| 00:00.33 | KalD|Work | ChrisRouse, brb. Are you using skinny or h323? |
| 00:00.56 | ChrisRouse | Kald: skinny |
| 00:01.03 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 00:01.04 | km- | tzanger: echocancel=32 had no effect |
| 00:01.20 | km- | is it possible I need it on the CO t1 |
| 00:01.25 | km- | as opposed to the pbx t1? |
| 00:01.42 | tzanger | km-: yeah try that -- set it to 'no' on the pbx T1 then |
| 00:01.58 | tzanger | I've found if you have multiple echo cans they can fight and end up not doing anything good at all |
| 00:02.37 | bjohnson | when I run safe_asterisk it shows in a coloured terminal and saves codes (I suspect ansi colour codes) into my logs .. making them harder to read. Is there a way to turn them off? |
| 00:02.58 | km- | still no effect |
| 00:03.05 | tzanger | ok try bumping it up to 64 |
| 00:03.07 | tzanger | and restart |
| 00:03.07 | km- | hehe, my wife is getting annoyed with me calling her |
| 00:03.11 | tzanger | :-) |
| 00:03.23 | km- | what would setting it to 128 do? |
| 00:03.28 | tzanger | that's part of the problem of deployments, you generally need a willing soul or a good far-end app |
| 00:03.30 | tessier_ | km-: Tell her she's lucky to have a phone in the kitchen at all |
| 00:03.34 | tzanger | you just increase the taps |
| 00:03.36 | tzanger | tessier_: HAHAHAHAHAHHAA |
| 00:03.49 | km- | tessier: Hahahah, that's wrong |
| 00:03.52 | tzanger | km-: lower # of taps means faster settling but not able to handle more delayed echo |
| 00:04.14 | tessier_ | I kill me. ;) |
| 00:04.37 | km- | tzanger: I'm going to try 128 and work my way down, because, if it's 128 and still echos, it means it's not going to get better through the echocancel setting, right? |
| 00:04.50 | mikegrb | bjohnson: just use cat to display the logs and they won't be hard to display, they will be colored |
| 00:04.52 | tzanger | no that's not necessarily right |
| 00:04.56 | km- | tzanger: ok |
| 00:04.57 | tzanger | and you can use 256 too but that's insane |
| 00:05.07 | tzanger | basicall you have to try each one and see how it's affect works |
| 00:05.11 | km- | the weird part is |
| 00:05.14 | mikegrb | bjohnson: or give less the option to preserve the ansi codes and it will display in color in less |
| 00:05.15 | km- | I tried calling my cell phone |
| 00:05.17 | km- | and there was no echo |
| 00:05.21 | tzanger | and then if that fails, try the same on the pbx side (turning it off on the telco side) |
| 00:05.31 | tzanger | km-: cell carriers all have VERY good echo cancellation |
| 00:06.10 | harryvv | bjohnson just to let you know for some reason it did authenticate on its own when I was googling the problem. It works now with just a minor codec bridge problem. |
| 00:06.26 | km- | I'll call my mom instead |
| 00:06.28 | km- | heh |
| 00:06.30 | tzanger | :-) |
| 00:06.37 | harryvv | km, no echo is a bit suprising. |
| 00:06.47 | tzanger | she'll be happy to hear from such a nice boy... at least until she realizes it's to test something |
| 00:06.51 | tzanger | instead of to talk to her |
| 00:07.26 | hcclNoodles | Tzanger I have just found out that British Telecom do not allow the "battery drop or battery reversal " apparantly a number of us in the UK have requested this to no avail. |
| 00:07.44 | tzanger | hcclNoodles: hmm |
| 00:08.03 | hcclNoodles | we just get the hang up tone |
| 00:08.04 | tzanger | so all british PBXes that use analogue trunks use inband detection? |
| 00:08.05 | tzanger | nasty |
| 00:08.18 | harryvv | btw, I will be looking for a more then your average UPS for my servers and need a external battery terminals on it. Would like to hook up my 125 amp/h battery to it. Anyone know. |
| 00:08.23 | tzanger | hcclNoodles: you can put a feature request or bounty in to get the british tone detected |
| 00:08.28 | tzanger | but I'd be surprised if it's not already there |
| 00:08.31 | *** part/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 00:08.33 | tzanger | you've set your zone to uk? |
| 00:08.43 | tzanger | i.e. the call progress tones all sound "normal" to you? |
| 00:08.49 | hcclNoodles | yes zone is uk |
| 00:08.59 | *** part/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 00:09.14 | hcclNoodles | there is a community of over 50 users on uk mailing lists all with the same issue |
| 00:09.24 | *** join/#asterisk sixTel (sixtel@sixTel.iax.cc) |
| 00:09.28 | harryvv | what issue |
| 00:09.30 | km- | tzanger: echocancel=yes has done it |
| 00:09.33 | km- | tzanger: echo all gone |
| 00:09.38 | bjohnson | mikegrb: I've gotten used to using editors for troubelshooting log file to make use of forward/backward action, selection highlighting, and text find |
| 00:09.39 | tzanger | so that's 128 then |
| 00:09.40 | rvhi | would latency affect echo cancellation? |
| 00:09.46 | tzanger | if you change echocancel to 128 it will be the same |
| 00:09.51 | tzanger | rvhi: yes |
| 00:09.59 | mikegrb | bjohnson: good idea |
| 00:10.07 | rvhi | i have 128, works fine within the city |
| 00:10.07 | tzanger | our PRI has echo only on certain calls |
| 00:10.25 | tzanger | it all has to do with the delay in the loop to the far end, where your voice is bouncing off their hybrid |
| 00:10.28 | rvhi | someone has a pap2 after 5k miles away, it is very bad |
| 00:10.31 | hcclNoodles | TDM400P not detecting UK hangup signal |
| 00:10.34 | mikegrb | bjohnson: try "TERM=vt100 safe_asterisk" or some such for startup |
| 00:10.40 | rvhi | choppy sound, not echo though |
| 00:10.41 | Grooby | grrrrr |
| 00:10.45 | tzanger | you have 5000 miles of copper? |
| 00:10.55 | harryvv | tzanger are you a end user supplier of voip |
| 00:10.59 | Grooby | codec_speex.so is having trouble loading libspeex.so.1 |
| 00:11.04 | Grooby | anyone having this problems before? |
| 00:11.04 | rvhi | pap2 on the internet |
| 00:11.08 | tzanger | just to local busineses and stuff |
| 00:11.18 | rvhi | i wish i had that much copper... :) |
| 00:11.21 | harryvv | okay what ups do you use and been happy with it |
| 00:11.27 | tzanger | rvhi: the echo cancellation on the pap2 should handle it |
| 00:11.35 | tzanger | I like APC upses |
| 00:11.56 | rvhi | when they call pstn number, pstn side hears choppy sound |
| 00:11.56 | tzanger | we have some rackmount 3kW ones I'm not keen on |
| 00:12.05 | rvhi | pap2 side is very clear |
| 00:12.07 | tzanger | they're nice enough but they don't turn back on after fully depleted |
| 00:12.17 | tzanger | rvhi: that means your packets are arriving in poor form |
| 00:12.21 | tzanger | high jitter |
| 00:12.22 | rvhi | i think that is echocancellation overdone |
| 00:12.24 | harryvv | do thay have a external 12 volt hookup ? I have a 125 amp hour battery thats on a constant 12 volt trickel charger and would like to attach it to a existing ups. |
| 00:12.34 | tzanger | rvhi: nope, not unless pap2 is very strange |
| 00:12.44 | tzanger | harryvv: some do, yes |
| 00:12.56 | rvhi | if the pstn side is digital phone, no problem |
| 00:12.59 | tzanger | harryvv: you have to be careful about two charging inputs |
| 00:13.06 | tzanger | rvhi: huh? |
| 00:13.25 | GodThor | is anyone install h323 on fedora3 (asterisk from CVS)? |
| 00:13.30 | harryvv | I know. thats for the battery right now. if its on a ups no sence in having the trickel charger on it. |
| 00:13.33 | rvhi | if pstn side uses a digital phone, e.g. a pbx system with pri connection |
| 00:13.40 | tzanger | rvhi: you --- [ internet, 5000 mile distance ] --- pap2 --- PSTN ?? |
| 00:13.44 | rvhi | but to analog home phone, is it choppy |
| 00:13.53 | harryvv | tzanger know which models might have one. |
| 00:13.54 | tzanger | how are you connecting a PAP2 to a PRI? |
| 00:14.02 | tzanger | harryvv: not offhand, but APC's site is (was) pretty good |
| 00:14.39 | harryvv | okay yea been looking at it |
| 00:14.39 | dstevens_ | Managed to compile Asterix on EPIA 5000 using Ubuntu without to much pain, make samples; make progdocs; having trouble command# sudo asterisk -vvvc returns "Illegal instruction" |
| 00:14.42 | rvhi | phone A -- pap2 --------------------- asterisk -- T1 PRI -- pstn |
| 00:14.43 | Himeko | tzanger you can set how much they recharge till they come back on |
| 00:14.55 | bjohnson | I keep getting: |
| 00:14.57 | bjohnson | Asterisk died with code 1. |
| 00:14.57 | bjohnson | Automatically restarting Asterisk. |
| 00:15.20 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 00:15.24 | rvhi | pstn ----- phone B |
| 00:15.42 | rvhi | if phone B is a home analog phone, or cell phone, sound quality is bad |
| 00:15.54 | tzanger | bjohnson: don't use safe_asterisk |
| 00:15.57 | km- | Feb 23 19:15:45 NOTICE[16876]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) |
| 00:15.58 | km- | <PROTECTED> |
| 00:15.59 | km- | hrmm |
| 00:15.59 | tzanger | let it die and see if you can get better stats |
| 00:16.05 | rvhi | if phone B is an office digital phone connecting a pbx, that's fine |
| 00:16.09 | tzanger | rvhi: oh |
| 00:16.17 | rvhi | pbx connects to pstn via pri |
| 00:16.19 | tzanger | rvhi: that is interesting |
| 00:16.27 | tzanger | rvhi: I don't have an answer to that one |
| 00:16.46 | tzanger | rvhi: test your theory -- disable echo cancellation altogether on the pap2 |
| 00:16.47 | rvhi | office phone is fine, because it is probably no digital/analog conversion, so no echo |
| 00:17.19 | tzanger | rvhi: it has nothing to do with digital/analog conversion, it has everything ot do with the hybrid circuit |
| 00:18.01 | rvhi | could choppy sound be overdone echo cancellation? |
| 00:18.08 | rvhi | i don't really get an echo in this case |
| 00:18.23 | tzanger | rvhi: i've not heard of that before, but disable the echo can altogether to try it |
| 00:18.48 | rvhi | i think i tried once to disable echo can |
| 00:18.50 | rvhi | it works |
| 00:19.11 | rvhi | but for users within 50 miles, i have to enable it |
| 00:20.22 | km- | <PROTECTED> |
| 00:20.22 | km- | Feb 23 19:20:03 NOTICE[17037]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) |
| 00:20.22 | km- | <PROTECTED> |
| 00:20.37 | km- | oh shitballs. |
| 00:20.43 | km- | I know what the problem is |
| 00:21.06 | buddah | pri full? |
| 00:21.12 | tzanger | km-: congestion |
| 00:21.20 | |Vulture| | anyone have a remote directory setup for the IP500? |
| 00:21.20 | tzanger | look up cause 3 in causes.h |
| 00:21.46 | rvhi | what is remote directory? |
| 00:21.51 | km- | tzanger: I had the wrong ip address on the remote side |
| 00:21.52 | km- | hehee |
| 00:21.57 | km- | I'm calling my buddy's office |
| 00:22.35 | |Vulture| | rvhi: a directory stored on a central web server |
| 00:23.24 | rvhi | i only know you can put it in directory.xml file and load it to the phone |
| 00:23.44 | rvhi | or you can use IP600 with a microbrowser to browse the web server |
| 00:23.48 | km- | hahaha |
| 00:23.49 | rvhi | but not on IP500 |
| 00:23.50 | km- | that is so rocking |
| 00:24.34 | tzanger | km-: put the last 3 patches from bug 2532 in |
| 00:24.35 | tzanger | I'm telling you |
| 00:24.40 | tzanger | you won't regret it |
| 00:24.40 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 00:24.47 | tzanger | so long as you're not using g729 |
| 00:25.19 | km- | tzanger: jitter buffer! |
| 00:25.23 | km- | tzanger: does it make it work better? |
| 00:25.35 | |Vulture| | ah I thought you could in the IP500 |
| 00:25.46 | |Vulture| | oh well |
| 00:25.54 | Poincare | damned, this is an addictive hobby |
| 00:26.10 | tzanger | km-: it does |
| 00:26.12 | tzanger | it's amazing |
| 00:26.24 | tzanger | stevekstevek has created one AMAZING new jitter buffer |
| 00:26.43 | km- | hehe |
| 00:27.21 | km- | I havent ever had experience with jitter |
| 00:27.34 | km- | hmm |
| 00:27.37 | km- | wait a sec |
| 00:27.48 | km- | what's the command to initiate a transfer? |
| 00:27.58 | km- | if you dont have a transfer button |
| 00:28.04 | Poincare | # |
| 00:28.10 | km- | I think I just figured out how to make transfer work on the legacy pbx |
| 00:30.16 | dstevens_ | <PROTECTED> |
| 00:30.20 | |Vulture| | Anyone know of a Firwall/Router for ~$500 that has an WAN port failover feature? |
| 00:30.48 | |Vulture| | dstevens_: post your error on pastebin.ca |
| 00:31.11 | km- | thats weird |
| 00:31.16 | km- | I hit pound |
| 00:31.20 | km- | and all I heard was ringing |
| 00:31.25 | km- | I couldn't transfer |
| 00:31.26 | Poincare | dstevens_: any indications with asterisk -vvvc? |
| 00:31.29 | Bentley | hi all, I think i once saw a thread about a special priority that gets executed b4 priority 1. Anyone know of such a thing? |
| 00:31.37 | km- | tzanger: I thought for sure that would have worked |
| 00:31.44 | ChrisRouse | Anyone else have experience with Cisco Call Manager and Asterisk? |
| 00:33.20 | dstevens_ | Poincare, If i run the command as -vvvc it return Illegal Error, If run with -vvv Then some text flows by is this what you mean. |
| 00:33.57 | Poincare | dstevens_: in that text that flows by might be an indication about what is going wrong... |
| 00:34.51 | |Vulture| | Anyone here have experiences with Megapath or Xspedius T1-Data? |
| 00:34.56 | *** join/#asterisk convey (~test@208-216-127-234.cust.gti.net) |
| 00:35.54 | convey | anyone have problems applying the broadvoice patch? |
| 00:35.59 | cbachman | dstevens_ what processor are you running on? |
| 00:36.03 | Grooby | convey, don't use the patch |
| 00:36.12 | Grooby | 1.0.5 should have the patch in it |
| 00:36.21 | convey | ok |
| 00:36.25 | convey | cool thanks |
| 00:36.30 | Grooby | you following the doc from bv? |
| 00:36.33 | Grooby | on asterisk howto? |
| 00:36.57 | convey | not me |
| 00:37.12 | convey | I was reading the broadvoice support pager |
| 00:37.23 | tzanger | km-: did you use T in the Dial() |
| 00:37.30 | km- | YES!!!! |
| 00:37.30 | Grooby | ok |
| 00:37.31 | km- | WOOT |
| 00:37.34 | km- | # transfer works |
| 00:37.38 | Grooby | hehehehe |
| 00:37.39 | km- | at least incoming direction |
| 00:37.44 | dstevens_ | processor model name VIA Samuel mobo EPIA 5000 |
| 00:37.48 | ChrisRouse | g |
| 00:37.57 | Grooby | so anyone got their speex codec to work?!?! |
| 00:38.02 | km- | tzanger: I can hit #499 to transfer the call from my nec phone to my x-lite phone when I dial in from my celly |
| 00:38.03 | dstevens_ | VIA Samuel 2 sorry |
| 00:38.16 | km- | tzanger: however, get this |
| 00:38.35 | |Vulture| | convey: tat info is outdated, check the wiki |
| 00:39.34 | tzanger | km-: well yeah, since * si seeing it |
| 00:39.36 | cbachman | dstevens_ your issue is the via chip. It's not completely compatible so you need to adjust a flag |
| 00:39.39 | tzanger | once it's in the NEC though I doubt you can get it out |
| 00:39.41 | km- | <PROTECTED> |
| 00:39.45 | km- | What is wrong with this dial line? |
| 00:39.57 | km- | When I dial |
| 00:40.07 | km- | I answer my cell (4849191400) but Asterisk doesnt detect the pickup |
| 00:40.17 | km- | if I remove the ,60,Ttr, Asterisk detects the pickup |
| 00:40.19 | tzanger | km-: it doesn't detect the pickup?? |
| 00:40.23 | km- | yeah |
| 00:40.23 | tzanger | first off, don't use 'r' |
| 00:40.26 | km- | its the weirdest thing |
| 00:40.42 | dstevens_ | cbachman, flag ahh how do i change my flag. |
| 00:40.45 | tzanger | and I wouldn't use 't' either, since someone could hit '#' and get access to your phone system |
| 00:40.49 | tzanger | at least the extensions part |
| 00:41.14 | km- | <PROTECTED> |
| 00:41.16 | km- | ok |
| 00:41.18 | km- | I'll try again now |
| 00:41.50 | *** join/#asterisk TheAx (~TheAx@d150-169-108.home.cgocable.net) |
| 00:42.25 | TheAx | Hi I just finished installing * from the latest CVS head and now it won't run anymore.. it givesd the following error |
| 00:42.26 | TheAx | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_cust_config_register |
| 00:42.41 | TheAx | any ideas y? |
| 00:42.46 | tzanger | TheAx: rm -rf /usr/lib/asterisk/modules and reinstall |
| 00:42.47 | TheAx | I am on a RH9 system |
| 00:42.58 | tzanger | you likely have old modules around that are confusing * when it tries to load |
| 00:43.05 | km- | weird |
| 00:43.15 | km- | it doesnt even work if I do it with the legacy pbx originating |
| 00:43.24 | km- | even if I do Tt |
| 00:43.28 | TheAx | OK.. thnaks tzanger.. |
| 00:43.29 | km- | but the answer is detected now |
| 00:43.33 | cbachman | dstevens_ google for i586 asterisk via You'll find that there are a number of references to this issue |
| 00:43.34 | km- | I guess the 'r' had something to do with it |
| 00:44.59 | cbachman | dstevens_ in particular: http://www.voip-info.org/tiki-index.php?page=Asterisk+Compile |
| 00:46.08 | terrapen | badass full moon out right now |
| 00:46.11 | terrapen | owwwwwwwwwwwwwwwwwwwllll |
| 00:46.30 | tzanger | yeah my kids are squirrely |
| 00:46.41 | dstevens_ | cbachman, Thanks for your help i will readup and recompile, and be back. |
| 00:46.44 | terrapen | my dog is lazy |
| 00:47.20 | terrapen | how do i dial up this dev conference? |
| 00:47.31 | terrapen | i see the /topic |
| 00:47.53 | terrapen | do i have to add an extension for it? |
| 00:48.27 | ChrisRouse | How do I associate an extension with an Agent? |
| 00:48.55 | km- | <PROTECTED> |
| 00:48.55 | km- | <PROTECTED> |
| 00:48.55 | km- | <PROTECTED> |
| 00:48.55 | km- | <PROTECTED> |
| 00:49.15 | km- | tzanger: so, I'm chatting on the phone, and, 60 seconds later, the phone disconnects |
| 00:49.27 | km- | tzanger: which leads me to realize -- Asterisk doesn't know when the call is connected. |
| 00:49.37 | TheAx | it is still the same... |
| 00:49.39 | ChrisRouse | Rather, when logging into Asterisk as an agent and dial an extension Asterisk tells me that the extension is not valid. Where is it getting that information? |
| 00:49.39 | TheAx | == Parsing '/etc/asterisk/musiconhold.conf': Found |
| 00:49.46 | TheAx | Warning, flexibel rate not heavily tested! |
| 00:50.15 | TheAx | [cdr_addon_mysql.so][root@PBX asterisk]# Junk at the beginning 49443303 |
| 00:50.28 | TheAx | and keeps giving "Warning, flexibel rate not heavily tested!" |
| 00:50.32 | km- | tzanger: is there some way I can tweak this thing a bit more so that it's a bit more intelligent about when the line's picked up? |
| 00:51.07 | TheAx | i removed the modules dir and recompiled asterisk, zaptel and libpri |
| 00:51.35 | TheAx | and it is still the same.. has anyone tried installing asterisk in the past hour or so? |
| 00:52.10 | TheAx | maybe something in the CVS head is mixed up?! Because I had a working version of it running and as soon as i downloaded the new CVS head and compiled it, this happend |
| 00:52.47 | |Vulture| | anyone here ever get a PRI with multiple LATA? |
| 00:52.55 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:53.20 | *** join/#asterisk goldenoldies (~goldenold@65.171.196.23) |
| 00:53.24 | goldenoldies | hi all |
| 00:53.40 | goldenoldies | anyone have any experience programming the speed dial buttons on the Cisco phones via TFTP? I cannot figure out what to modify in the XML File |
| 00:54.02 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 00:54.12 | shmaltz | anybody using polycom phones? |
| 00:54.15 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 00:54.22 | *** join/#asterisk JerJer (~JerJer@d9-46.rt-bras.che.centurytel.net) |
| 00:54.26 | buddah | yes |
| 00:54.31 | buddah | ip 500s |
| 00:54.35 | |Vulture| | yup |
| 00:55.14 | shmaltz | buddah, how do you have the lines configured? one sip or multiple sips? and did you disable call waiting? and if you did how? |
| 00:55.19 | JerJer | OT: anyone know how to port forward using iptables without knowing the WAN ip address? |
| 00:55.20 | buddah | one |
| 00:55.29 | buddah | and i never looked at the call waiting actually |
| 00:55.32 | buddah | wait |
| 00:55.33 | buddah | hmm |
| 00:55.37 | buddah | its active |
| 00:55.47 | shmaltz | |Vulture|, I believe xo offers this |
| 00:56.12 | tzanger | km-: what do you mean |
| 00:56.13 | buddah | you can probably turn it off via the phones .cfg |
| 00:56.16 | |Vulture| | shmaltz: thank you |
| 00:56.18 | shmaltz | buddah, whats the point of using one if the call doens't roll over? |
| 00:56.24 | tzanger | km-: yes it doesn't know it's connected |
| 00:56.33 | buddah | hell if i know, i just set em up for clients |
| 00:56.37 | shmaltz | |Vulture|, np, anytime |
| 00:56.41 | buddah | they give me a list, i make sure the stuff works |
| 00:56.59 | shmaltz | :) |
| 00:57.01 | buddah | but i think most of them dont do call waiting, they just go to voice mail |
| 00:57.03 | tzanger | unless you see "... has answered Zap/whatever" |
| 00:57.13 | shmaltz | buddah, how? |
| 00:57.19 | buddah | sec and ill show you |
| 00:58.09 | shido6 | boink |
| 00:58.26 | goldenoldies | someone must know how to program these goddamn Cisco phones remotely when running SIP or MGCP |
| 00:58.43 | goldenoldies | I am likely to send one to someone who can help me with this |
| 00:58.51 | terrapen | uhhh |
| 00:58.54 | terrapen | program them like how |
| 00:58.55 | km- | tzanger: is there a way to make asterisk more sensitive to whether the line is picked up? |
| 00:59.03 | km- | tzanger: is that that whole call supervision thing people have talked about |
| 00:59.17 | tzanger | km-: why is it not seeing the line picked up |
| 00:59.20 | terrapen | almost everything about them can/is set up from the TFTP files |
| 00:59.25 | km- | tzanger: no idea |
| 00:59.40 | km- | lemme try originating the call through sip |
| 00:59.42 | tzanger | km-: do you have to futz with the E&M timings |
| 00:59.46 | goldenoldies | The 7960 has 6 line/speed dial buttons... I want the phone to download via TFTP/FTP new button configs when the phones boot... I cannot find the option in the two .cnf files or the card.xml file. |
| 00:59.47 | tzanger | on teh telco side |
| 00:59.59 | tzanger | i.e. asterisk is looking for too long a wink or something |
| 01:00.01 | terrapen | golden, its in the config files |
| 01:00.07 | goldenoldies | which config file where |
| 01:00.10 | goldenoldies | I have the sample files |
| 01:00.12 | terrapen | hang on |
| 01:00.13 | km- | tzanger: I'm not sure |
| 01:00.16 | goldenoldies | thank you |
| 01:00.25 | km- | tzanger: it appears that originating the call from SIP doesn't change the fact that it cant find the connect |
| 01:00.31 | shmaltz | anybody know how to get the polycom to NOT pickup the second incoming call on the first line appearance, but on the second? (on the cisco, I disable the callwaiting, then I register all line appearances with the same sip account). |
| 01:00.33 | km- | tzanger: so maybe there does need to be some fiddling |
| 01:00.33 | tzanger | km-: you've eliminated all your other timing mods, right? |
| 01:00.39 | km- | tzanger: yes, I have |
| 01:00.41 | tzanger | km-: huh? |
| 01:00.48 | terrapen | SIPxxxxxxxxxxxxxxxx.cnf |
| 01:00.51 | tzanger | km-: ok |
| 01:00.53 | terrapen | where xxxx is the phones mac addy |
| 01:00.57 | km- | tzanger: I was wondering if the problem was coming from the legacy pbx of from the co line |
| 01:00.59 | tzanger | km-: try placing a call from the NEC out |
| 01:01.01 | terrapen | SIP00082194D85A.cnf |
| 01:01.01 | tzanger | does it work? |
| 01:01.03 | goldenoldies | k but what option do you specify in the file |
| 01:01.03 | km- | tzanger: the problem occurs in both places |
| 01:01.05 | terrapen | thats mine |
| 01:01.05 | tzanger | or does it cut off after 60s too |
| 01:01.13 | terrapen | golden, i swear its in the sample configs.... |
| 01:01.18 | km- | tzanger: the call succeeds but asterisk doesnt know it succeeds |
| 01:01.26 | km- | tzanger: all of them cut off after 60 seconds |
| 01:01.30 | tzanger | km-: ok so it looks like a more global problem |
| 01:01.30 | goldenoldies | I am staring at the sample configs man, I see MGCP gateway and everything, I do not see the button configs though |
| 01:01.33 | goldenoldies | nothing about the buttons |
| 01:01.34 | tzanger | generic wink settings are incorrect |
| 01:01.40 | terrapen | look for one that has lines like hits: |
| 01:01.41 | tzanger | km-: you can't ask the telco what their wink timings are can you? |
| 01:01.48 | terrapen | line2_name: "202" |
| 01:01.50 | km- | tzanger: I dunno if they'd know it |
| 01:01.53 | tzanger | km-: you can use zttool to see it too and try to estimate it |
| 01:01.57 | terrapen | line2_displayname: "Chris Snell x202" |
| 01:02.04 | terrapen | obviously with different values :P |
| 01:02.13 | goldenoldies | you the man, hopefully this works with mgcp! |
| 01:02.27 | terrapen | mgcp? |
| 01:02.29 | goldenoldies | message me your address, I'll send you a phone if this works, will know in 2 seconds |
| 01:02.33 | terrapen | this is sip, bruddah |
| 01:02.58 | TheAx | what does it mean when * says: Junk at the beginning 49443303 |
| 01:03.09 | TheAx | or "Warning, flexibel rate not heavily tested!" |
| 01:03.30 | TheAx | even when i compile the old version it still won't run |
| 01:03.38 | TheAx | and this was a working box |
| 01:03.48 | TheAx | all i did was download the new cvs head and compile it |
| 01:04.05 | tzanger | TheAx: that's the thing with CVS HEAD |
| 01:04.05 | km- | tzanger: get this, from the time the call is placed until the time I hang up my side, I never see the status on RxABCD go to '1' |
| 01:04.10 | tzanger | it changes all the time |
| 01:04.18 | km- | tzanger: there's a quick "wink" when I first pick up the line |
| 01:04.23 | tzanger | TheAx: before you get too far, make a copy of your /etc/asterisk and /etc/zaptel.conf |
| 01:04.31 | TheAx | whatever it has done, it is preventing me from even using my original version that i had running b4 |
| 01:04.31 | cbachman | jerjer: Some reading I did seemed to hint that you could leave off the -d and specify the interface alone? |
| 01:04.38 | km- | tzanger: but the status doesnt change from that point on, through call accepted, to hangup |
| 01:04.46 | tzanger | TheAx: I find that hard to believe |
| 01:04.49 | km- | tzanger: the hangup isn't detected either, I have to hang up myself |
| 01:04.59 | tzanger | TheAx: you didn't completely erase the old (new) asterisk then |
| 01:05.02 | TheAx | tzanger> me 2 |
| 01:05.05 | tzanger | or you played with your config files without backing them up |
| 01:05.08 | *** join/#asterisk SirPrize (~blah@host-84-9-105-17.bulldogdsl.com) |
| 01:05.26 | tzanger | km-: are you sure you're supposed to use e&M wink? |
| 01:05.37 | TheAx | i didn't earse the old (new) * , i just compiled the old one again |
| 01:05.56 | km- | tzanger: if I just set it to em, weird things happen |
| 01:06.02 | km- | tzanger: i'll try it again for the hell of it though |
| 01:07.22 | tzanger | hmm |
| 01:07.24 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 01:07.44 | SirPrize | is there a way to send SMS messages via SIP, in the UK? |
| 01:07.48 | km- | tzanger: switching to em causes no change on the CO T1 side |
| 01:07.59 | km- | tzanger: changing from em_w to em on the pbx side causes the pbx to act funky |
| 01:08.22 | terrapen | why not just send it by email? |
| 01:08.47 | terrapen | most celly providers have gateways |
| 01:08.47 | tzanger | km-: |
| 01:08.47 | tzanger | so your AB bits are 00 normally |
| 01:08.47 | tzanger | the telco places a call to you |
| 01:08.47 | SirPrize | mine charges money for that. :-S |
| 01:08.49 | tzanger | it sets 11 |
| 01:08.55 | km- | tzanger: yep |
| 01:08.56 | tzanger | you send back 010 |
| 01:09.02 | km- | uhm |
| 01:09.03 | km- | heh |
| 01:09.05 | km- | I dunno |
| 01:09.06 | SirPrize | :-) |
| 01:09.10 | tzanger | well watch it :-) |
| 01:09.12 | km- | is there a way to packet trace the signal? |
| 01:09.13 | tzanger | I'd imagine so |
| 01:09.15 | km- | it happen sso fast |
| 01:09.18 | tzanger | km-: not that I'm aware of |
| 01:09.20 | km- | lemme try it again |
| 01:09.22 | tzanger | you can uncomment some dbeugs |
| 01:09.36 | km- | ok |
| 01:09.40 | km- | I bring up zttool on the CO T1 |
| 01:09.47 | tzanger | km-: in zt_rbs_sethook |
| 01:09.50 | tzanger | in zaptel.c |
| 01:10.03 | km- | the first 11 lines are TxABCD 0, RxABCD 1, because I have data on those lines and I dont have them configured in the pbx |
| 01:10.10 | km- | so, the call would come out line 12 |
| 01:10.25 | km- | tx 1 rx goes 101 |
| 01:10.29 | km- | then stays 0 |
| 01:10.33 | TheAx | no |
| 01:10.41 | TheAx | oops |
| 01:10.42 | tzanger | either uncomment CONFIG_ZAPATA_DEBUG or make it so JUST that line (~1885) is printed |
| 01:10.43 | km- | from that point |
| 01:10.46 | tzanger | and recompile and reload the modules |
| 01:11.20 | TheAx | tzanger> should i delete the current none-working installation of * and trying compiling it again? |
| 01:11.32 | tzanger | TheAx: I don't know your exact situation |
| 01:11.50 | tzanger | TheAx: what exactly is it doing now that you've reverted? |
| 01:11.53 | km- | ok |
| 01:11.57 | km- | remaking zaptel module |
| 01:12.14 | TheAx | the same thing.. it won't run and will give those warning messages |
| 01:12.41 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:12.47 | tzanger | did you run make samples when you updated CVS HEAD too? (don't do it) |
| 01:12.50 | terrapen | this really is the best music on hold evar |
| 01:13.00 | km- | no |
| 01:13.02 | terrapen | especially if you have seen the movie |
| 01:13.03 | km- | I did not run make samples |
| 01:13.10 | tzanger | no not you km, theax |
| 01:13.13 | km- | brb bio break |
| 01:13.28 | TheAx | tzanger> no.. id didn't make the samples |
| 01:13.31 | tzanger | ok good |
| 01:13.40 | tzanger | pastebin the errors please? |
| 01:13.45 | TheAx | i also backed them up as u recommended |
| 01:14.54 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 01:15.00 | TheAx | tzanger> http://www.pastebin.com/245668 |
| 01:15.27 | TheAx | this is with my old backup compiled |
| 01:15.38 | TheAx | which gives the exact same eroor as the new cvs head |
| 01:16.11 | tzanger | TheAx: I don't think it has anything to do with mysql |
| 01:16.15 | tzanger | music on hold, maybe |
| 01:16.32 | |Vulture| | yea sounds like moh, try dissabling it |
| 01:16.37 | terrapen | i want to make a Music on Hold Favorites wiki page |
| 01:16.46 | tzanger | terrapen: :-) |
| 01:16.48 | terrapen | where people can post their favorite ideas for MoH |
| 01:17.00 | *** join/#asterisk ranliv (~ranliv@210.213.254.212) |
| 01:17.08 | km- | notifying reader data in block 0 |
| 01:17.09 | km- | hmm |
| 01:17.14 | km- | its spamming across the screen |
| 01:17.25 | TheAx | but moh was working 4 |
| 01:17.25 | km- | ok, time to only get the one we want |
| 01:17.27 | TheAx | b4 |
| 01:17.30 | tzanger | km-: that's why I said you might want to just make that one line print |
| 01:17.35 | tzanger | :-) |
| 01:17.52 | terrapen | tz: what do you use? |
| 01:18.01 | km- | remaking now :) |
| 01:19.09 | Nukemizer | Can I delay the prompt for "Meetme" ? when Ii call a conference room I get the first part of the intro chopped off.. |
| 01:19.25 | ranliv | hello guys! can anyone help me with my problem. I keep getting this error " WARNING[4293]: chan_iax2.c:2189 create_addr: No such host:" |
| 01:19.58 | ranliv | but from the linux console I can ping ng domain name |
| 01:20.19 | ranliv | also tried defining at /etc/hosts but still the same problem |
| 01:20.22 | Trionnis | Nukemizer: re-record it with a short pause at the beginning |
| 01:20.25 | Trionnis | ;) |
| 01:20.41 | tzanger | km-: |
| 01:20.43 | tzanger | http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00801123bb.shtml |
| 01:20.47 | Nukemizer | that works ! thanks |
| 01:20.49 | tzanger | E&M signaling is way too straightforward |
| 01:20.51 | tzanger | only two states |
| 01:21.04 | marc_c | how many mins can you on average term on a T1? |
| 01:21.10 | tzanger | one million |
| 01:21.20 | marc_c | no. thats full capacity! |
| 01:21.22 | tzanger | it's just shy of that (983000 or something) |
| 01:21.32 | Trionnis | welcome :) |
| 01:21.35 | tzanger | average term depends ENTIRELY on your usage patterns |
| 01:21.41 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 01:22.13 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 01:22.18 | km- | ok tzanger |
| 01:22.26 | TheAx | Damn.. |
| 01:22.38 | marc_c | ~200k? |
| 01:22.59 | TheAx | I don't know what the hell the new cvs head has done but * is refusing to load.. maybe I should delete the whole thing and reinstall... |
| 01:23.03 | km- | tzanger: I set bits to 15, state 2 in 64 signalling at first, then set 15, state 1 in 64, then set 15,state 1 in 64, then, there are no more bits set until I hang up the phone |
| 01:23.13 | km- | tzanger: the answer doesnt get detected, remote hangup doesnt get detected |
| 01:23.26 | TheAx | i even deleted my modukes dir and installed * again |
| 01:23.31 | tzanger | km-: well it's all 1111 that's why |
| 01:23.48 | tzanger | TheAx: try just noloading res_moh or whatever it's called |
| 01:24.07 | tzanger | you shouldn't be setting the bits to state 2 |
| 01:24.07 | km- | tzanger: ok, I'll accept that you know way more about this than I do -- any ideas how I fix that? |
| 01:24.18 | tzanger | offhand no, I am not sure why it's doing that |
| 01:25.30 | *** part/#asterisk TheAx (~TheAx@d150-169-108.home.cgocable.net) |
| 01:25.39 | km- | quite curious, huh |
| 01:25.50 | km- | damn telephony |
| 01:25.51 | km- | hehe |
| 01:26.43 | *** join/#asterisk TheAx (~TheAx@d150-169-108.home.cgocable.net) |
| 01:27.13 | tzanger | km-: stop asterisk |
| 01:27.17 | tzanger | dmesg -c |
| 01:27.20 | tzanger | and run ztcfg |
| 01:27.23 | tzanger | what's dmesg say |
| 01:27.26 | tzanger | anything interesting |
| 01:27.40 | TheAx | sorry how do i avoid * from loading res_moh? |
| 01:28.21 | km- | tzanger: nope, just that it's setting bits to 0 for channel te4/0/2/1-24 state 0 in 64 signalling |
| 01:28.32 | km- | tzanger: which I imagine means its idle? |
| 01:28.43 | tzanger | ok good |
| 01:28.54 | tzanger | now when you start asterisk and it stabilizes, what does it say |
| 01:29.54 | km- | same thing |
| 01:29.54 | km- | setting bits to 0 for channel te4/0/2/24 state 0 in 64 signalling |
| 01:29.54 | tzanger | ok good |
| 01:29.56 | tzanger | now pick up a line but don't dial anything |
| 01:30.13 | tzanger | (or rahter dial enough to get * to say "starting simple switch) |
| 01:30.16 | km- | on the CO side or PBX side? I cant pick up a line on the CO side without dialing |
| 01:30.29 | km- | unless I hack an extension that does Zap/g1 |
| 01:30.30 | tzanger | PBX side |
| 01:30.49 | km- | ok, pbx side says settings bits to 0 first |
| 01:30.51 | km- | then 15/1 |
| 01:30.52 | km- | then 0/0 |
| 01:30.59 | km- | hanging up sets 0/0 |
| 01:31.09 | tzanger | ok |
| 01:31.17 | km- | want me to hack that extension on CO side? |
| 01:31.18 | tzanger | now dial a # |
| 01:31.20 | tzanger | no |
| 01:31.23 | km- | ok |
| 01:31.38 | km- | no change |
| 01:31.57 | tzanger | hmm |
| 01:32.00 | km- | <PROTECTED> |
| 01:32.00 | km- | <PROTECTED> |
| 01:32.00 | km- | <PROTECTED> |
| 01:32.00 | km- | <PROTECTED> |
| 01:32.09 | tzanger | right |
| 01:32.42 | tzanger | but you should have seen the bits go from 0 to 15, then back to 0 after YOU hang up |
| 01:32.53 | Trionnis | lmao |
| 01:33.06 | Trionnis | my 4 year old just walked up here and said "hey dad....why do I have a crack in my butt?" |
| 01:33.11 | Trionnis | =| |
| 01:33.13 | km- | HAHAHA |
| 01:33.19 | tzanger | hahahaha |
| 01:33.26 | km- | tzanger: I just tried calling from my cellphoen to the asterisk box, and the states went |
| 01:33.27 | Trionnis | I'm still laughing so hard I can't think of an answer |
| 01:33.28 | Trionnis | lol |
| 01:33.28 | tzanger | my 5yo daughter asked me how to screw yesterday |
| 01:33.33 | Trionnis | =O |
| 01:33.37 | km- | 0/0 then 15/1 then 0/0 then 15/1 then 15/1 then 0/0 |
| 01:33.43 | tzanger | (we were putting together a fluorescent light and I told her to screw down the one piece of metal) |
| 01:33.48 | Trionnis | OOHHH |
| 01:33.50 | Trionnis | ... |
| 01:33.57 | km- | hahaha |
| 01:34.02 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp180870.sympatico.ca) |
| 01:34.15 | tzanger | km-: what is the number after the / |
| 01:34.15 | tzanger | hello jim |
| 01:34.18 | tzanger | long time |
| 01:34.20 | km- | the state |
| 01:34.22 | tzanger | I owe you an email or two |
| 01:34.23 | tzanger | ok |
| 01:34.29 | km- | the first number is the bits, second is the state |
| 01:34.30 | tzanger | so state 0 -> 1 -> 0 -> 1 -> 1 -> 0 |
| 01:34.43 | km- | right, on an incoming call from my cellphone to asterisk |
| 01:34.45 | JimVanM | tzanger: howdydoo |
| 01:34.49 | tzanger | km-: no no no no no |
| 01:34.53 | tzanger | well fuck it |
| 01:34.58 | tzanger | let's do tleco side first them |
| 01:34.59 | tzanger | er then |
| 01:35.10 | km- | I dont have a problem with the pbx side really |
| 01:35.11 | tzanger | so basically you see the wink |
| 01:35.16 | *** join/#asterisk jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:35.19 | km- | if I call from the pbx to the soft phone, it detects the answer |
| 01:35.21 | tzanger | km-: you just said you aren't able to sustain a call from either way |
| 01:35.33 | jayden | tzanger!!!! |
| 01:35.40 | km- | I can't sustain a call from either endpoint out to the CO |
| 01:35.42 | Trionnis | spa-2000 for 65USD including shipping a decent deal? |
| 01:35.48 | km- | I can sustain a call between the two endpoints on the system fine |
| 01:35.56 | Trionnis | refurb, btw |
| 01:36.02 | tzanger | km-: ah |
| 01:36.08 | tzanger | JimVanM: I'm pissed at you |
| 01:36.16 | tzanger | JimVanM: you didn't tell me I could light MWI from an ATA |
| 01:36.33 | tzanger | km-: hmm ok |
| 01:36.41 | JimVanM | tzanger: there's a whole lot I haven't told you, young grasshopper |
| 01:36.45 | km- | tzanger: ok, for the purposes of the rollout, I can at least remove the 60 second timeout and let it act like how it is |
| 01:36.46 | tzanger | JimVanM: hahaha |
| 01:36.54 | tzanger | km-: kind of |
| 01:37.11 | tzanger | km-: I am not going to be surprised if there are other problems, like the telco cutting you off since it didn't complete |
| 01:37.16 | tzanger | km-: but let's get back to work |
| 01:37.24 | tzanger | so you call * from the cell |
| 01:37.31 | tzanger | * is obviously winking |
| 01:37.43 | tzanger | and then obviously going offhook |
| 01:37.44 | JimVanM | tzanger: I have a box full of Norstar swag here, just waiting for a mad scientist to perform cruel experiments upon it |
| 01:37.46 | tzanger | and obviosusly going back |
| 01:38.10 | tzanger | JimVanM: yes I have to email you back and let you know that yes I want to get together and perform curel experiments on our MICS |
| 01:38.32 | tzanger | JimVanM: know anything about NEC Electra Elite 48s? |
| 01:38.46 | jayden | uhhhhh, you guys might wanna take this conversation private |
| 01:38.48 | jayden | :) |
| 01:38.55 | tzanger | km-: If you call your SIP phone from your cell phone does it work? |
| 01:39.08 | JimVanM | tzanger: no, but the thought of torturing Nortel gear gets me all weepy |
| 01:39.09 | km- | lemme try |
| 01:39.16 | tzanger | hahaha |
| 01:39.27 | JimVanM | jayden: don't worry, we'll post pictures! |
| 01:39.55 | km- | tzanger: yep, works fine, answer is detected |
| 01:39.59 | tzanger | ok |
| 01:40.08 | tzanger | now call your PBX from your SIP phone |
| 01:40.09 | km- | states for the call were 0,1,0,1,1,0 |
| 01:40.17 | km- | ok, just a sec |
| 01:40.22 | tzanger | km-: exactly, wink, offhook, onhook |
| 01:40.55 | km- | states were 2,1,1,0 |
| 01:41.11 | km- | lemme do it again |
| 01:41.14 | tzanger | state 2? |
| 01:41.16 | km- | just to be sure |
| 01:41.32 | tzanger | offhook, onhook, ring, kewl |
| 01:41.33 | km- | yes |
| 01:41.35 | tzanger | so yeah makes sense |
| 01:41.37 | km- | 2,1,1,0 |
| 01:41.47 | tzanger | what are teh bits for state 2 |
| 01:41.51 | km- | answer is detected, everything fine |
| 01:41.55 | km- | bits were 15 for state 2 |
| 01:41.57 | tzanger | oh okay |
| 01:42.01 | tzanger | but no wink |
| 01:42.01 | km- | and 15 for the state 1's |
| 01:42.08 | tzanger | interesting |
| 01:42.13 | tzanger | oh wait |
| 01:42.15 | tzanger | I'm dumb |
| 01:42.16 | km- | the state 0 was 0/0 |
| 01:42.23 | tzanger | * doesn't wink for outgoing, the receiver winks |
| 01:42.28 | tzanger | and we're not debugging rbs incoming |
| 01:42.32 | tzanger | ok |
| 01:42.38 | tzanger | so the problem is not with the co |
| 01:42.42 | tzanger | or with the pbx |
| 01:42.45 | tzanger | but rather with the CO and PBX |
| 01:42.58 | km- | you mean, the CO and asterisk? |
| 01:43.00 | tzanger | i.e. PSTN -> SIP = ok, and SIP -> PBX = ok |
| 01:43.05 | tzanger | no, I mean iwth the CO and PBX :-) |
| 01:43.11 | tzanger | can you call your cell from the SIP? |
| 01:43.18 | tzanger | and can you call your SIP from the PBX? |
| 01:43.22 | km- | I call the IVR, then dial the sip extension |
| 01:43.48 | km- | right |
| 01:43.52 | TheAx | this is wired now i get: |
| 01:43.54 | TheAx | <PROTECTED> |
| 01:43.55 | km- | PBX->SIP works, SIP->PBX works |
| 01:44.01 | km- | CO->* works |
| 01:44.11 | TheAx | this is with version 1 stable |
| 01:44.12 | km- | *->CO does not work I dont think, lemme reconfirm that |
| 01:44.17 | tzanger | TheAx: are you using mysql? |
| 01:44.28 | TheAx | i was, yes |
| 01:44.52 | tzanger | does /usr/lib/asterisk/modules/cdr_addon_mysql.so exist? |
| 01:45.16 | km- | tzanger: *->CO does not work |
| 01:45.29 | tzanger | km-: /me wonders if the CO is winking |
| 01:45.31 | TheAx | hrm.. no... |
| 01:45.31 | km- | tzanger: calling cell from SIP does not have any detections occuring |
| 01:45.32 | tzanger | it has to |
| 01:45.43 | tzanger | TheAx: you're trying ot load it and it doesn't exist. fix that. |
| 01:45.45 | *** join/#asterisk usam (~alx@203.156.48.176) |
| 01:45.55 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 01:45.58 | TheAx | let me install it again |
| 01:46.05 | tzanger | km-: what are the bits for SIP -> Cell |
| 01:46.08 | tzanger | er states rather |
| 01:46.13 | km- | just a sec |
| 01:46.20 | TheAx | tzanger.. thanks.. sorry i am juyst out of it today.. t his * problem was the least thing i needed |
| 01:46.20 | km- | clear the screen and try it again to be sure |
| 01:46.22 | TheAx | lol |
| 01:46.29 | tzanger | TheAx: no problem |
| 01:46.52 | km- | tzanger: 15/2 15/1 15/1 0/0 |
| 01:46.57 | km- | tzanger: the 0/0 comes when I hang up |
| 01:47.13 | *** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net) |
| 01:47.21 | usam | concerning x100p via zaptel, is there state that would provide me with "ringing" status? everytime i make call viz x100p, the asterisk always "Answered" the request... Am I right? |
| 01:47.30 | tzanger | km-: I think I see the problem |
| 01:47.41 | km- | if you tell me that it hates me |
| 01:47.42 | tzanger | km-: you're treating the telco as if it were a phone |
| 01:47.44 | km- | I'm going to cry |
| 01:47.49 | tzanger | you're ringing the telco |
| 01:47.53 | tzanger | which is wrong |
| 01:47.59 | km- | ok, how do I change that |
| 01:48.01 | km- | hehe |
| 01:48.05 | tzanger | how do you specify e&m but not fxo signalled? |
| 01:48.09 | tzanger | you want fxs signaling on the telco side |
| 01:48.15 | km- | lemme see the config samples |
| 01:48.39 | Inv_arp | quik q: i have inbound sip (BV) for my business, if iam on line and some1 else calls can they go thru my menu's in extension.conf also> |
| 01:48.43 | km- | tzanger: do I need sf_w instead of em_w? |
| 01:48.52 | tzanger | no I don't think so |
| 01:48.59 | tzanger | but it can't hurt to try |
| 01:49.08 | km- | what is sf_w anyway |
| 01:49.37 | tzanger | not what you want really |
| 01:49.47 | km- | yeah, asterisk dies if I set that |
| 01:50.07 | km- | do you think I need to set em and not em_w? |
| 01:50.09 | tzanger | try setting zaptel.conf to fxsks and zapata.conf to fxs_ks |
| 01:50.34 | tzanger | km-: no, the only difference between E&M and E&M_W is that winking lets the far end tell you when it's ready.. otherwise it's just a dumb delay |
| 01:51.10 | tzanger | km-: I'm just guessing here, btw, with this fxs/fxo stuff |
| 01:51.28 | tzanger | I mean ringing is the same as siezing the line |
| 01:51.35 | tzanger | but this is strange |
| 01:51.38 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 01:51.42 | tzanger | might get you to hack up the driver a bit more |
| 01:52.01 | km- | with fxsks, there's an immediate zap/12-1 answered sip/499, but, no disconnect |
| 01:52.11 | sivana | hehe |
| 01:52.25 | km- | and |
| 01:52.28 | km- | I cant receive any calls |
| 01:52.51 | tzanger | km-: it was just a guess |
| 01:53.25 | km- | yeah |
| 01:53.27 | km- | good try |
| 01:53.38 | ChrisRouse | Anyone know why Asterisk would tell me that an extension is not available for an auto login of an agent? |
| 01:53.41 | km- | I have to leave |
| 01:53.49 | km- | promised the wife I'd be out by 9 tonight |
| 01:54.33 | km- | I'm ok with leaving this one a quasi-mystery for the time being |
| 01:54.44 | km- | tzanger: want to hack on this some more friday night or possibly over the weekend? |
| 01:54.57 | tzanger | km-: you're in toronto? |
| 01:55.03 | km- | philadelphia |
| 01:55.11 | tzanger | ahh I was gonna say |
| 01:55.15 | tzanger | I'm going to torastricon on friday |
| 01:55.20 | sivana | hehe |
| 01:55.21 | tzanger | weekend might be better though |
| 01:55.28 | km- | sure |
| 01:55.37 | km- | I'll be idling here tomorrow and friday |
| 01:55.38 | tzanger | you might want to look in zt_rbsbits |
| 01:55.47 | tzanger | and add some debuggery there to see what's going on |
| 01:55.53 | km- | gotcha |
| 01:55.54 | tzanger | this is strange though |
| 01:55.57 | tzanger | it should be working |
| 01:56.04 | tzanger | you can place the call too |
| 01:56.07 | tzanger | which is what's baffling |
| 01:56.09 | km- | yeah, that's how it always goes with asterisk "THIS SHOULD BE WOKRING!!!" |
| 01:56.14 | tzanger | km-: :-) |
| 01:56.20 | km- | thats what i heard the whole time when I was getting hdlc working |
| 01:56.30 | km- | finally had to have citats patch the hdlc driver |
| 01:56.32 | km- | hehe |
| 01:56.40 | km- | okie doke, catch you guys tomorrow |
| 01:56.45 | sivana | later |
| 01:56.57 | km- | tzanger: thank you a million times over for all your help tonight, at least it's working well enough that nobody will know the difference tomoroow |
| 01:57.05 | tzanger | km-: :-) hopefully |
| 01:57.09 | tzanger | and you're welcome |
| 01:57.14 | km- | hehe |
| 01:57.16 | km- | ok, ttyl |
| 01:57.37 | tzanger | he's not seeing the network show answer |
| 01:57.38 | BrianR___ | tzanger: having fun with my norstar integration project - still no luck with tone detection based disconnect supervision though... |
| 01:57.39 | tzanger | which si what's weird |
| 01:58.01 | tzanger | BrianR___: I would have thought it would have been as simple as adding the frequencies to the callprogress code |
| 02:00.14 | *** join/#asterisk Beave (~beave@vistech.org) |
| 02:00.19 | Beave | hello all! |
| 02:00.20 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 02:00.25 | MrEntropy | yo |
| 02:00.33 | Inv_arp | http://www.jcreator.com/download.php?c=630ce14eb9efb1047438928bde43bbdb |
| 02:00.41 | Inv_arp | bah wrong chan |
| 02:00.54 | Grooby | tee hee hee |
| 02:01.25 | Beave | I have CLID dumping to a MySQL database, anyone have any suggestions on nice frontends to the CLID database? I thought about writting one, but no reason to re-invent the wheel... |
| 02:02.01 | BrianR___ | tzanger: I think I'm going to review (and modify if needed) the callprogress code. |
| 02:02.24 | Inv_arp | i have inbound sip (BV) for my business, if iam on phone and some1 else calls can they go thru my menu's in extension.conf also? |
| 02:02.25 | BrianR___ | Also, the norstar VMI has the ability to signal disconnect via DTMF. Rumore has it there's some code for that in CVS. |
| 02:02.45 | BrianR___ | Using the DTMF D tone would help avoid false disconnects in some cases too. |
| 02:03.12 | tzanger | BrianR___: yes I remember hearing about that in CVS |
| 02:03.20 | BrianR___ | Prolly less likely to get a "D" than a dialtone too.. |
| 02:03.23 | tzanger | I think it was * or # to disconnect but it was moved to D |
| 02:03.37 | BrianR___ | configured in features.conf, no less. |
| 02:04.01 | Graphikos | I have 2 IP phones... in a simple setup. Both can call each other, etc but you can't hear anything from one of the phones... any suggestions on how to trouble shoot that? |
| 02:04.14 | BrianR___ | tzanger: No no.. The guy thought his voicemail was sending a '*' but it was really senidng a 'D'. the disconnect= option in features.conf is aparently configurable.. |
| 02:04.24 | tzanger | ahh yes |
| 02:04.28 | tzanger | I remember now |
| 02:04.39 | *** join/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net) |
| 02:04.40 | MrEntropy | how does sipXpbx compare to asterisk in terms of features? |
| 02:04.55 | |Vulture| | doesnt :P |
| 02:04.58 | MrEntropy | haha |
| 02:05.05 | ManxPower | Well, it only supports SIP, for one thing |
| 02:05.16 | MrEntropy | what about hardware? |
| 02:05.23 | |Vulture| | yea IAX2 and Harware is where its at for * |
| 02:05.42 | |Vulture| | SIP is great for phone support.. but IAX is better for communications |
| 02:06.02 | dan2 | who do I get to see at von this year? |
| 02:06.44 | BrianR___ | http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf |
| 02:06.47 | BrianR___ | found it. |
| 02:07.06 | MrEntropy | i think it would be worth while coding a driver for asterisk to handle the audiocodes tp cards |
| 02:07.43 | BrianR___ | Now I gotta find the change in CVS and see if i can backport it to stable |
| 02:08.00 | BrianR___ | Is there a cvsweb anywhere for the asterisk source? |
| 02:08.39 | tzanger | BrianR___: yes |
| 02:08.42 | tzanger | www.asterisk.org |
| 02:09.05 | Graphikos | I have 2 IP phones... in a simple setup. Both can call each other, etc but you can't hear anything from one of the phones... any suggestions on how to trouble shoot that? |
| 02:10.08 | |Vulture| | Graphikos: is there a NAT or firewall inbetween them? |
| 02:10.14 | Graphikos | no |
| 02:10.29 | Graphikos | 192.168.0.7 & 8 ... |
| 02:10.45 | Graphikos | * on 5 |
| 02:10.50 | |Vulture| | the handset is plugged in right, correct? |
| 02:11.00 | Graphikos | heh... yes |
| 02:11.10 | |Vulture| | don't laugh I did it once :( |
| 02:11.11 | Graphikos | you can hear the other phone.... but nothing from it... |
| 02:11.19 | |Vulture| | strange |
| 02:11.38 | |Vulture| | try doing an echo test |
| 02:11.56 | Graphikos | heh, thats over my head right now.. I just bearly got this going... |
| 02:12.00 | |Vulture| | see if it is rx/tx to * |
| 02:12.03 | |Vulture| | ah 1s |
| 02:12.06 | |Vulture| | Ill pastebin it |
| 02:12.11 | Graphikos | thank you |
| 02:13.12 | |Vulture| | Graphikos: http://pastebin.ca/6357 put that in your extensions.conf |
| 02:13.25 | |Vulture| | then dial 111 from each phone and wait for the tone, then talk |
| 02:13.31 | *** join/#asterisk padf00t (~hq28@202.58.252.14) |
| 02:14.05 | padf00t | hi all |
| 02:14.23 | padf00t | i need to do some more understanding of the SWITCH statement in IAX |
| 02:14.30 | padf00t | where could i find a gud example |
| 02:14.32 | Grooby | w00t! got speex to work! |
| 02:14.33 | padf00t | or doc on that |
| 02:14.38 | Graphikos | works good from phone 1... let me try the bad phone.... across the hall |
| 02:14.54 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 02:15.13 | *** join/#asterisk DeepMahul (DeepMahul@83.132.224.59) |
| 02:15.48 | BrianR___ | tzanger: Couldn't find a cvsweb there, so i'm pulling down the whole damn source for some hacking.. |
| 02:15.56 | Graphikos | nope.. no echo repeat from that phone at all |
| 02:15.56 | BrianR___ | figured it'd come to this point anyway. |
| 02:15.59 | tzanger | oh I thought they hd one |
| 02:16.02 | trym | I have installed spandsp to have asterisk receive faxes. When a fax call is made to asterisk, asterisk starts whining about RFC3389. I also notice that the volume spandsp/asterisk is communicating with varies.. which is not normal for a fax session. Any suggestions? |
| 02:16.11 | BrianR___ | Maybe i wasn't looking hard enough. |
| 02:16.18 | trym | in other words.. not working |
| 02:16.18 | tzanger | ahh cvsup not cvsweb |
| 02:16.19 | tzanger | sorry |
| 02:16.29 | |Vulture| | Graphikos: very strange... try to plug the phone in somewhere else... do you know for a fact that this phone works? |
| 02:16.48 | BrianR___ | never played with cvsup before either.. Using it now to pull in all the stuff |
| 02:16.51 | Graphikos | its brand new... but no I don't know... |
| 02:16.57 | |Vulture| | what kinda phone? |
| 02:17.09 | Graphikos | Sipura SPA-841.. |
| 02:17.11 | Graphikos | they both are |
| 02:17.23 | |Vulture| | ah.. no exp. with those |
| 02:17.38 | Graphikos | swapping headsets just for fun |
| 02:17.43 | |Vulture| | k |
| 02:18.21 | Essobi | Anyone know why my sip trunks from my Cisco routers all land in default even thou I have a context set for them? |
| 02:18.32 | Essobi | I can't figure out why. :| |
| 02:18.38 | Essobi | I'm running -head. |
| 02:18.50 | Graphikos | nope.. no help |
| 02:19.24 | Graphikos | Do you think it might just be some sort of config problem? because its amazing I got this far.. |
| 02:19.53 | Graphikos | config files are pretty scrapped together... |
| 02:20.53 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:20.53 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:21.18 | |Vulture| | Graphikos: are your 2 entries in sip.conf the same? |
| 02:21.34 | Graphikos | yes... |
| 02:21.47 | *** part/#asterisk libpcp (libpcp@210.16.20.5) |
| 02:21.50 | |Vulture| | go into asterisk and type "sip show peers" |
| 02:21.57 | *** join/#asterisk nicolasg (~nicolasg@host-121.6.60.66-ta.adsl.netizen.com.ar) |
| 02:22.00 | |Vulture| | see if they look the same, except for the IPs |
| 02:22.23 | Graphikos | they do... |
| 02:23.24 | |Vulture| | Graphikos: duno what to say... they should work... you could use "sip debug" and see if you can find a problem |
| 02:23.25 | |Vulture| | but its prolly a little involved |
| 02:23.25 | Graphikos | ok... |
| 02:23.30 | Graphikos | thanks. |
| 02:23.32 | |Vulture| | np |
| 02:23.55 | Inv_arp | i have inbound sip (BV) for my business, if iam on phone and some1 else calls can they go thru my menu's in extension.conf also? |
| 02:24.36 | modulus_ | OMG |
| 02:24.42 | modulus_ | inv_arp BV? that's death wish. |
| 02:25.47 | Inv_arp | modulus_: ? works fine for me |
| 02:26.44 | Inv_arp | i hate the fact that they don use gsm tho |
| 02:27.40 | Inv_arp | modulus_: had bad experience with them? |
| 02:30.18 | modulus_ | inv_arp, i've only had good experiences with voip |
| 02:30.20 | BrianR___ | Hmm.. The example sup file for asterisk does checkout mode.. :( |
| 02:30.20 | modulus_ | NOT! |
| 02:30.30 | BrianR___ | Waiting again to get a real tree.. |
| 02:30.46 | bjohnson | Inv_arp: good question .. and you're in a position to find out. Does Broadvoice allow multiple concurrent incoming and/or outgoing calls and how do they charge for them? |
| 02:30.54 | *** part/#asterisk Damascene (Damascene@pcp0011401420pcs.ebrnsw01.nj.comcast.net) |
| 02:31.14 | Graphikos | I'd like to know that also... |
| 02:31.42 | shido6 | ..... |
| 02:31.52 | ManxPower | ~docs |
| 02:31.53 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 02:33.02 | *** join/#asterisk BoRiS (~boris@24.81.0.252) |
| 02:33.19 | bjohnson | Inv_arp: you may want to start by reading theier terms of service. And please note anything you find out on the wiki |
| 02:33.31 | shmaltz | anybody here that has a polycom phone? |
| 02:34.42 | *** join/#asterisk jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:36.16 | *** join/#asterisk lilneon (~tj_r3@200.108.28.153) |
| 02:36.23 | lilneon | hi and good evening guuys |
| 02:36.26 | Inv_arp | bjohnson: well when some1 else calls i get a signal to switch over b/c i have call waiting with them...but not sure if 2nd call can go thru my prompts |
| 02:36.27 | lilneon | and gyals |
| 02:36.47 | lilneon | hey guys anyone here know how to open up ports on a winxp machine? |
| 02:37.05 | Trionnis | later guys |
| 02:37.06 | Beirdo | with a crowbar |
| 02:37.38 | bjohnson | Inv_arp: I don't think so if call waiting is on. |
| 02:37.53 | Inv_arp | bjohnson: ahh k |
| 02:38.04 | Inv_arp | lilneon: u mean SP2 xp? |
| 02:38.17 | bjohnson | Inv_arp: can you turn off cw and try it? |
| 02:38.29 | Grooby | ugh |
| 02:38.33 | Grooby | speex is horrible |
| 02:38.47 | Inv_arp | bjohnson: ahh k will do |
| 02:38.56 | Inv_arp | Grooby: robotic? |
| 02:39.17 | Grooby | more like i can't make out what I was saying |
| 02:39.43 | Graphikos | Domo arigato, Mr. Roboto |
| 02:39.55 | lilneon | Inv_arp: yeah |
| 02:40.00 | Grooby | inv_arp, you have experience using speex? |
| 02:40.19 | Inv_arp | Grooby: nah just read bout it |
| 02:40.28 | lilneon | inv_arp: putting in exceptions but they not even getting applied.. cuz if i do a port scan the ports are still closed |
| 02:41.04 | Grooby | back to ilbc I go |
| 02:41.09 | Inv_arp | lilneon: closed means programs is not running doesnt mean firewalled |
| 02:41.22 | Grooby | also learned today that x-lite is much better than sjphone in terms of voice quality |
| 02:41.34 | lilneon | Inv_arp: but what if the programs are running? |
| 02:41.47 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:42.14 | Inv_arp | lilneon: check the ports they are listening on netstat -an (xp) netstat -tuanp (linux) |
| 02:42.26 | Inv_arp | linux version far better :) |
| 02:42.32 | bjohnson | Grooby: gsm seems to be more popular than ilbc around here .. but they are pretty close contenders (after ulaw as the favorite) |
| 02:43.01 | lilneon | Inv_arp: yeah doing that.. and the ports the program uses aren't there.. for windows.. but the program is running |
| 02:43.48 | Inv_arp | ohhh gsm :) |
| 02:44.18 | Inv_arp | lilneon: whats your setup and/or trying to do? |
| 02:45.30 | Grooby | i get better quality (IMO) with ilbc |
| 02:45.46 | Grooby | i guess i get more packet drops with my cable modem |
| 02:45.47 | Inv_arp | woah windoze has a nice proggy to map programs with ports ( ive been wanting that for years) |
| 02:46.31 | Inv_arp | fport |
| 02:50.32 | okieplaya | i have a IAXy s100 i have it all most setup can someone help me please |
| 02:51.13 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:51.13 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:51.16 | Inv_arp | okieplaya: if u ask a ques yea |
| 02:51.19 | lilneon | Inv_arp: i am trying to get my linux free tds to use mssql server 2000 on my windows box.. wrked before but not since installed sp2.. checked ms site.. they claimed to simply add the exception in the firewall for s\mssql |
| 02:51.27 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:51.46 | lilneon | Inv_arp: but that doesn't seem to wrk.. still getting a connection refused frm the linux side |
| 02:51.49 | *** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net) |
| 02:52.05 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:52.05 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:52.06 | okieplaya | ok i did every thing that pdf told me only thing i cant get goin IAXy with the following command: |
| 02:52.06 | okieplaya | ./iaxyprov <ip of the IAXy> <file> |
| 02:52.06 | okieplaya | i.e.: |
| 02:52.06 | okieplaya | ./iaxyprov 192.168.0.100 iaxy.conf.sample |
| 02:52.25 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 02:52.44 | Inv_arp | lilneon: a connection refused means program is not listening on specified port firewalls (including sp2) usually do not return refused |
| 02:54.04 | lilneon | Inv_arp: but mssql server is set to listen on port 1433 in windows.. it jsut doesn't show up in neither fport,nmap nor netstat |
| 02:54.12 | okieplaya | i never see this after saveing the iaxy.conf. If registration is successful, you will receive a notice on the command line |
| 02:54.37 | lilneon | Inv_arp: so.. i guess i should try reinstalling mssql server 2000 right? |
| 02:55.03 | Inv_arp | lilneon: on the 2000 server if you netstat -an is mssql listening on 1433? |
| 02:55.04 | okieplaya | any i deal what im doin wrong im useing ssh to setup that ok? |
| 02:55.25 | *** join/#asterisk sricard (sricard@HSE-Montreal-ppp133166.qc.sympatico.ca) |
| 02:55.52 | okieplaya | Inv_arp> any i deals |
| 02:56.47 | lilneon | Inv_arp: no it is not :S |
| 02:57.04 | lilneon | Inv_arp:any idea why not? and how i could get it to do so? |
| 02:57.36 | Inv_arp | lilneon: no idea since this is windoze ?(reboot)? |
| 02:57.39 | *** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net) |
| 02:59.02 | lilneon | Inv_arp: sigh... |
| 03:00.37 | syslod | Anyone working on QSIG name in here? |
| 03:01.07 | *** join/#asterisk NomadPCs (~service@65.113.210.103) |
| 03:01.43 | BrianR___ | Hmm.. It looks like DTMF disconnect is lumped in with a whole pile of other crap :( |
| 03:02.35 | NomadPCs | I'm looking to install asterisk on a single phone line for my small office - I'm very good with Linux, networking and PCs - looking for someone who can show me what I need to get for hardware. |
| 03:02.40 | dan2 | bkw_: am I going to get to see your at VoN? |
| 03:03.10 | *** join/#asterisk Graphikos (~Graphikos@71-32-6-49.spkn.qwest.net) |
| 03:03.11 | ariel_ | NomadPCs, simple start with asterisk@home add a tdm11b and your done. |
| 03:03.18 | NomadPCs | ty |
| 03:04.09 | *** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net) |
| 03:10.00 | okieplaya | Inv_arp ok i ask any help? |
| 03:11.13 | Inv_arp | okieplaya: hmm never used iaxy before... but usually anything iax isnt hard to setup |
| 03:11.49 | okieplaya | yea thats what would think |
| 03:12.00 | *** part/#asterisk lilneon (~tj_r3@200.108.28.153) |
| 03:13.54 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 03:17.49 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
| 03:18.24 | Ron-Na | I need some basic info, ... |
| 03:19.45 | Ron-Na | if I have a SIP phone somewhere, registered to my Asterisk box and I want to use NuFone as provider. I have registered NuFune with SIP, so that the media stream goes not through my server. Am I correct with that? |
| 03:20.13 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 03:20.28 | file | dan2: poke |
| 03:20.41 | Ron-Na | I found only info how setup Nufone with iax but not with sip, ... |
| 03:20.54 | BrianR___ | got it finally |
| 03:20.55 | BrianR___ | exten => s,1,Dial(local/in@fromvmi/n,10,H) |
| 03:21.03 | BrianR___ | makes DTMF disconnect supervision work |
| 03:21.25 | tzanger | BrianR___: nice! |
| 03:21.38 | file | cheater |
| 03:21.43 | file | but inventive. |
| 03:21.46 | BrianR___ | it's hardcoded to '*' in 1.0.5, but at least I can hack it to DTMF 'D' until the newer res_features is out. |
| 03:21.55 | znoG | Ron-Na: if you connect to your asterisk box, why not get asterisk to talk IAX to NuFone? |
| 03:22.05 | tzanger | BrianR___: ATAs can send 'D' on hangup? |
| 03:22.19 | BrianR___ | tzanger: They can send any DTMF tone, including A B C D |
| 03:22.24 | mishehu | I was looking at astcc for doing a prepaid solution in a shared office space I might be providing service to. Is there another prepaid solution out there already that doesn't require the dialer to dial a pin number for every call they make? |
| 03:22.28 | Ron-Na | because the sip phone is not at my site, it is somewhere ... |
| 03:22.28 | tzanger | BrianR___: how do you configure that ?? |
| 03:22.38 | file | mishehu: astcc can use callerid |
| 03:22.41 | BrianR___ | Oh. Sorry. not ATA's. VMI's.. |
| 03:22.46 | tzanger | ahh |
| 03:22.49 | BrianR___ | tzanger: The VMI is like a dual port ATA |
| 03:22.56 | Ron-Na | My thought was if I use iax, than the media stream must go through my server, ... |
| 03:22.57 | znoG | Ron-Na: and who does the phone connect to? |
| 03:22.58 | BrianR___ | tzanger: with two very important features... |
| 03:23.05 | tzanger | BrianR___: hmm |
| 03:23.06 | mishehu | file: oh it can? hmmm, I was reading the voip-info page about it, not the source. I'll have to look at teh source then. |
| 03:23.12 | Ron-Na | to my server |
| 03:23.14 | tzanger | I have a 0x8AM |
| 03:23.20 | BrianR___ | First, if you forward a call to a VMI's DN, it can play the original DN in DTMF. |
| 03:23.24 | tzanger | a few ATAs |
| 03:23.30 | tzanger | and a pair of two-port Flash modules |
| 03:23.49 | file | where's dan2... hrm |
| 03:23.49 | BrianR___ | second, it can play a tone on disconnect |
| 03:24.05 | BrianR___ | tzanger: ATA's are cheap on eBay. I got 3 of 'em for $150 |
| 03:24.09 | BrianR___ | That's 6 ports |
| 03:24.11 | znoG | Ron-Na: and you want the sip phone to connect to NuFone directly or via your asterisk server? |
| 03:24.14 | BrianR___ | err. s/ATA/VMI. |
| 03:24.15 | tzanger | ATAs or VMIs |
| 03:24.16 | tzanger | heh |
| 03:24.32 | BrianR___ | 3 VMI's gives you 6 ports.. And disconnect supervision. |
| 03:24.59 | tzanger | BrianR___: you connect them to * with FXO or FXS ports on *? |
| 03:25.03 | Ron-Na | znoG I want that the phone and NuFon is registered with me, but the phone is somewhere |
| 03:25.03 | tzanger | ATAs need FXO ports |
| 03:25.05 | BrianR___ | And the ability to tell what extension forwarded you a call... |
| 03:25.19 | znoG | Ron-Na: who cares? as long as it can connect to your asterisk box! |
| 03:25.20 | BrianR___ | tzanger: FXO ports on asterisk, just like an ATA or MOX8A. |
| 03:25.24 | tzanger | right |
| 03:25.39 | BrianR___ | I still haven't got callerid working... |
| 03:25.42 | Ron-Na | znoG my bandwidth does care !! |
| 03:25.49 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 03:25.58 | bjohnson_ | PTG123: you here? |
| 03:25.59 | tzanger | I can forward the calls to an 8xx extension on the PRI and see who called me that way |
| 03:26.04 | file | I have never gotten SIP reinvites working, that is - getting the audio to go direct |
| 03:26.06 | tzanger | but I can't light WMI yet |
| 03:26.14 | BrianR___ | tzanger: Yes.. If you have PRI none of this matters much. |
| 03:26.36 | tzanger | I can light MWI with an ATA or 0x8AM but that's dirty, especially since I have the PRI |
| 03:26.39 | znoG | Ron-Na: so, you want the phone to go STRAIGHT to NuFone, right? |
| 03:26.52 | BrianR___ | tzanger: But for doing like a Norstar 6x16 integraiton, the VMI is the only way. |
| 03:27.00 | BrianR___ | since the little norstars have no T1 sockets. |
| 03:27.03 | tzanger | BrianR___: right |
| 03:27.10 | Ron-Na | znoG, that what I was thinking should work, but billing via my server |
| 03:27.18 | tzanger | BrianR___: you don't happen to know how to light MWI from a public trunk PRI? |
| 03:27.19 | BrianR___ | And even if they did, for a system in thta price range you're not going to buy $1000 of T1 hardware... |
| 03:27.22 | znoG | Ron-Na: how you propose on doing that? |
| 03:27.26 | znoG | Ron-Na: this isn't radius |
| 03:27.27 | tzanger | Hell I'd settle for knowhing how to do it on a TIE trunk PRI |
| 03:27.31 | BrianR___ | tzanger: Does *1 work over DISA? |
| 03:27.34 | file | magic. |
| 03:27.42 | tzanger | BrianR___: never tried DISA... will tomorrow |
| 03:27.45 | tzanger | never thought of that |
| 03:27.53 | tzanger | BrianR___: damn man, you have all the ideas for Norstar |
| 03:28.12 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 03:28.12 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 03:28.14 | BrianR___ | doesn't work over DISA. Sorry. |
| 03:28.14 | bkw_ | yo |
| 03:28.15 | bkw_ | doughecka_, |
| 03:28.18 | bkw_ | is that you on /.. |
| 03:28.24 | syslod | tzanger: Isn't that a facility message and someones working on it? |
| 03:28.27 | BrianR___ | (I have a DID pointed at my norstar DISA) |
| 03:28.33 | tzanger | syslod: only if you can get SL1 |
| 03:28.33 | bkw_ | When Doug Heckman was installing a PC Pitstop program, he actually read the EULA. |
| 03:28.35 | BrianR___ | Using a single ATA port for lighting MWI on a norstar isn't so bad. |
| 03:28.37 | tzanger | syslod: and I can't get SL1 yet :-) |
| 03:28.37 | mishehu | file: get a room! |
| 03:28.47 | tzanger | BrianR___: no it's not but still |
| 03:28.48 | Ron-Na | znoG basic question: if I have a sip phone somewhere and call another sip phone somewhere, but both phones are registerd to my server, than the sip messages are coming to my server, but the media stream is going directly, am I right? |
| 03:29.07 | bjohnson_ | ~seen PTG123 |
| 03:29.09 | jbot | ptg123 is currently on #asterisk (1h 24m 30s) |
| 03:29.10 | BrianR___ | tzanger: Also needed for doing voice call and paging anyway, so... |
| 03:29.18 | syslod | tzanger: Does it work over QSIG. |
| 03:29.21 | mishehu | Ron-Na: I don't believe so. |
| 03:29.49 | tzanger | syslod: nope |
| 03:29.53 | tzanger | syslod: tried that already :-) |
| 03:30.00 | BrianR___ | QSIG? |
| 03:30.04 | tzanger | yeah |
| 03:30.09 | BrianR___ | What is QSIG? |
| 03:30.10 | tzanger | norstar MICS doesn't support qsig |
| 03:30.23 | tzanger | at least not without telling it it's not in north america, at which pint I bet a lot of other stuff breaks |
| 03:30.36 | tzanger | BrianR___: think of it as PBX interop signaling |
| 03:30.40 | BrianR___ | aah. |
| 03:31.08 | znoG | Ron-Na: i believe once the connection is made, your server doesn't know about the call anymore. At least that's how i thought it works. |
| 03:31.10 | BrianR___ | The norstar does support turning on MWI based on the MWI on an analog loop start trunk associated with a set. Not sure if it works with PRI. |
| 03:31.16 | syslod | QSIG is a euro standard really but many PBX support it. You can do DSS buttons, VM and all sorts of other stuff over ISDN. |
| 03:31.26 | tzanger | BrianR___: yes you can do that but only on very few defined sets |
| 03:31.57 | BrianR___ | Few defined sets? If you can do it for a target line and you have a target line assigned to every set... |
| 03:32.00 | tzanger | you can set a vm callback or something to that effect and I bet I can set MWI IEs but I think it's on a per-DID (and thus per-set) basis |
| 03:32.10 | tzanger | BrianR___: hmm |
| 03:32.35 | BrianR___ | The telco MWI passthru thing is done on a per-line basis. |
| 03:32.40 | Ron-Na | znoG: that what I read about too, ... How can I setup the NuFon with sip? Or does it not matter anyway if I use Iax ???? |
| 03:32.40 | tzanger | BrianR___: I would have thought that would make every set say "you've got msg" when the IE was sent |
| 03:32.56 | BrianR___ | tzanger: No idea... |
| 03:33.06 | tzanger | BrianR___: yes but you can only define like 4 or 5 (I think it was ridiculously low) Voicemail Ctrs |
| 03:33.12 | modulus_ | afk(smoke); |
| 03:33.18 | |Vulture| | with a PRI you can set your outbound CID... correct? |
| 03:33.32 | tzanger | |Vulture|: depends on the telco but generally yes |
| 03:33.34 | bkw_ | yes |
| 03:33.40 | syslod | You can also do on T1 CAS. |
| 03:33.48 | BrianR___ | tzanger: Yes. But the message center is the number that gets dialed when the user presses the "ANSWER MESSAGE" softkey |
| 03:33.59 | |Vulture| | oky, just trying to collect all my info... my first PRI ;) |
| 03:34.00 | tzanger | BrianR___: hmm |
| 03:34.09 | mishehu | hmmm, I'm seeing some docs on voip-info that point to asterisk branch 1.1... any estimate on when 1.1 will be stable? |
| 03:34.17 | BrianR___ | tzanger: So long as you don't care that the user has to enter both mailbox and password to check voicemail the message center limit is not a problem. |
| 03:34.33 | tzanger | BrianR___: I still think if you say line 187 is assigned ot all sets and then send a MWI on msg to the DID assigned to line 187 that all sets will show the message |
| 03:34.46 | tzanger | BrianR___: yeah I don't give a rat's ass about htat |
| 03:35.02 | BrianR___ | The real problem is whether or not MWI passthru is supported for target lines and if there's a way to indicate telco-voicemail message waiting over PRI. |
| 03:35.18 | BrianR___ | tzanger: Yes. Every set with line 187 assigned will light up as message waiting. At least that's how it works for POTS lines. |
| 03:35.30 | BrianR___ | How often do you put a target line on multiple sets though? |
| 03:35.35 | Ron-Na | How do I use exten => _91NXXNXXXXXX,2,Dial(IAX2/username@NuFone/${EXTEN:1} to sip ?? (Just replace IAX2 with SIP ???) |
| 03:35.43 | *** join/#asterisk VaHamish (~tgia@node-40243a81.dca.onnet.us.uu.net) |
| 03:35.45 | tzanger | BrianR___: well then you run into the limit that you can only define (I think) 30 DIDs |
| 03:35.51 | VaHamish | Wow??? |
| 03:35.56 | tzanger | so if you have 45 exetnsions there's 15 that you can't light up MWI on |
| 03:36.14 | BrianR___ | tzanger: The limit on a CICS is 160 or something. We have a DID for every set and we have over 100 sets. |
| 03:36.22 | VaHamish | I'm just going to jump in here... |
| 03:36.26 | tzanger | BrianR___: this is MICS |
| 03:36.36 | BrianR___ | err.. MICS.. We have a MICS... |
| 03:36.40 | tzanger | BrianR___: hmm |
| 03:36.42 | BrianR___ | Almostf ully expanded.. |
| 03:36.45 | VaHamish | I'm brand new to using Asterisk... and I need a bit of help getting my system configured.. |
| 03:36.48 | tzanger | I was sure I read in the manual that it was limited |
| 03:36.51 | tzanger | maybe just hte base then |
| 03:36.57 | VaHamish | Is there someone here who could help me out? |
| 03:37.07 | BrianR___ | I think it maxes out at something like 128 sets and 160 target lines. |
| 03:37.14 | tzanger | i.e. 8x32 is the default config I think |
| 03:37.18 | Essobi | Anyone lend me a hand debugging some sip stuff? I got a Cisco 5400 that always seems to land in [default] instead of my [peer] I have setup for it.. it completely ignores the context it's supposed to land in. |
| 03:37.21 | |Vulture| | VaHamish: voip-info will provide you with tutorials to getting started |
| 03:37.27 | tzanger | I'll have to try this tomorrow |
| 03:37.32 | modulus_ | back |
| 03:37.35 | VaHamish | I've been working through the tutorials for the last week or so.. |
| 03:37.37 | modulus_ | black |
| 03:37.39 | VaHamish | and I have the basics up and running, |
| 03:37.44 | BrianR___ | Yes.. The chassis comes with two 4 port analog line cards and 32 internal station ports |
| 03:37.52 | |Vulture| | VaHamish: then ask any specific Q and we will try to answer |
| 03:38.06 | tzanger | yeah and we have two additional modules, a 0x16 and a 8x0AM, or is it 16x0 and 0x8AM |
| 03:38.08 | BrianR___ | We have two service/fiber cards and two T1's in our cabinet. |
| 03:38.09 | VaHamish | but I have an TDM400 and am having problems getting it working. |
| 03:38.09 | tzanger | I think it's the latter |
| 03:38.22 | |Vulture| | VaHamish: what distro of linux? |
| 03:38.37 | VaHamish | I've got the drivers loaded, cause the lights are lit on the board, for the two modules. |
| 03:38.41 | |Vulture| | VaHamish: or better yet.. kernel |
| 03:38.59 | VaHamish | and the handset that's pluged in has battery, you can hear the tones when you press the keys. |
| 03:38.59 | BrianR___ | nostar is definantly the cheapest phone system around though. |
| 03:39.07 | tzanger | BrianR___: oh I remember what it was now |
| 03:39.12 | tzanger | I had assigned a DID to a set |
| 03:39.15 | tzanger | not a target line |
| 03:39.22 | |Vulture| | VaHamish: FXO or FXS modules? |
| 03:39.30 | VaHamish | I have one of each. |
| 03:39.31 | tzanger | i.e. 0000243 rings my set directly |
| 03:39.39 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
| 03:39.43 | tzanger | I also have 2914574 which is line 187 which is pvt to my set |
| 03:39.45 | |Vulture| | VaHamish: does it show them in ztcfg -vv ? |
| 03:40.17 | VaHamish | Channel 01: FXO Kewlstart |
| 03:40.25 | BrianR___ | We simply set the target line to APPR&RING on the person's DN - that's how we do all of the direct inward dialing. |
| 03:40.25 | VaHamish | Channel 04: FXS Kewlstart |
| 03:40.40 | tzanger | BrianR___: yeah that makes more sense |
| 03:41.14 | BrianR___ | making private lines all over the place causes you to run into limits on the norstar. |
| 03:41.17 | |Vulture| | VaHamish: seems like it works to me.. whats the problem? |
| 03:41.25 | BrianR___ | every line in our CICS is set to type public |
| 03:41.33 | VaHamish | well there's no dial tone on the handset.. |
| 03:41.40 | tzanger | BrianR___: *nods* |
| 03:41.59 | tzanger | I'll have to see if I can light MWI via PRI IE |
| 03:42.04 | |Vulture| | VaHamish: Ive never used FXS modules, so I cant tell you :( |
| 03:42.11 | VaHamish | I have sip configured, I can call from one sip phone on one computer to another.. |
| 03:42.12 | BrianR___ | tzanger: If you make that go, I'd be interested to hear :) |
| 03:42.26 | tzanger | BrianR___: will definitely keep you in the loop |
| 03:42.42 | BrianR___ | I should sell preconfigured asterisk/norstar voicemails on eBay :) |
| 03:42.45 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:42.49 | tzanger | :-) |
| 03:42.57 | tzanger | don't steal my business plan :) |
| 03:43.00 | BrianR___ | Heh heh |
| 03:43.25 | BrianR___ | Actually, there's a comapny which makes a four port PCI card that speaks the Norstar station protocol... |
| 03:43.31 | tzanger | BrianR___: yeah |
| 03:43.34 | tzanger | dialogic |
| 03:43.38 | VaHamish | I tell you it's pretty darn frustrating.. |
| 03:43.38 | tzanger | they're expensive though |
| 03:43.49 | BrianR___ | Since each Norstar station port has 2 bearer channels, you can run eight IVR calls that way |
| 03:43.54 | tzanger | from what I've learned the station ports are all ISDN BRI anyway |
| 03:44.22 | BrianR___ | I know there's seperate settings and a bunch of DN's reserved for ISDN BRI ports on my CICS system. |
| 03:44.30 | tzanger | yup |
| 03:44.48 | Graphikos | is there any way to flush all registered peers? |
| 03:44.51 | BrianR___ | I don't think I can use an ISDN TA on an ordinary station module port though. |
| 03:45.02 | tzanger | BrianR___: no, it's a fucked up protocl |
| 03:45.15 | tzanger | similar to how SL1 is like qsig but not |
| 03:45.16 | *** join/#asterisk Chuji (Chuji@pcp09930971pcs.tulipgrove.tn.nash.comcast.net) |
| 03:45.26 | BrianR___ | If the norstar wasn't so close to obsolescence I'd go through the reverse engineering effort. |
| 03:45.29 | tzanger | I plugged in an optical receiver to a fiber expansion card |
| 03:45.38 | tzanger | the norstar was sending AIS |
| 03:45.42 | BrianR___ | The nortel station module<->KSU protocol would be an interesting place to look at too. |
| 03:45.43 | BrianR___ | AIS? |
| 03:45.50 | tzanger | so I'm fairly confident that the modules all talk over SL1 |
| 03:46.01 | tzanger | red alarm (all 1s) |
| 03:46.10 | tzanger | so with some hacking I can make an optical monitor |
| 03:46.20 | tzanger | and see both sides of the conversation between cabinets |
| 03:46.24 | BrianR___ | The norstar fiber optics are weird. |
| 03:46.27 | tzanger | plug it into a T100P and see |
| 03:46.29 | tzanger | heh |
| 03:46.31 | BrianR___ | It's just a regular LED. |
| 03:46.31 | tzanger | BrianR___: howso? |
| 03:46.33 | tzanger | yes |
| 03:46.37 | tzanger | it's standard HP transcievers |
| 03:46.43 | BrianR___ | Can't go more than a few feet. |
| 03:46.44 | tzanger | we use them for our medium voltage gate firing circuitry |
| 03:47.05 | BrianR___ | had to get a MCK mod extender to run a module to the 1st floor. Conduit was too crowded to run a 25 pair... |
| 03:47.38 | BrianR___ | (The mod extender converts the norstar station module fiber port to single mode fiber and back again) |
| 03:47.42 | BrianR___ | s/single/multi |
| 03:47.43 | okieplaya | context=yourcontext what is yourcontext? |
| 03:47.46 | tzanger | *nod* |
| 03:47.49 | okieplaya | whats that mean |
| 03:48.15 | BrianR___ | I thought the T100P was POTS fxo only.. |
| 03:48.23 | tzanger | BrianR___: that's X100P |
| 03:48.25 | tzanger | T100P is T1 |
| 03:48.28 | BrianR___ | Oh yes.. |
| 03:48.52 | BrianR___ | I bougth generic X100P's for $10/ea on eBay.. I'm wondering if they just suck and that's why I can't make callerid go.. |
| 03:48.55 | tzanger | dammit |
| 03:48.58 | tzanger | it's quarter to 11 |
| 03:49.02 | tzanger | I wanted to be in bed a half hour ago |
| 03:49.05 | tzanger | thanks a lot BrianR___ :-) |
| 03:49.06 | BrianR___ | I'm going to hit up k-mart or something and buy a $10 cidco caller id receiver |
| 03:49.10 | BrianR___ | heh.. Let's talk again tomorrow. |
| 03:49.15 | tzanger | BrianR___: absolutely |
| 03:49.19 | tzanger | 'night |
| 03:49.25 | BrianR___ | ttyl. |
| 03:54.15 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 03:54.22 | Essobi | Baaah. |
| 03:54.44 | Essobi | I'm mad as hell. |
| 03:55.26 | *** join/#asterisk _daver_ (~daver@ns1.tmok.com) |
| 03:58.05 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 03:58.49 | VaHamish | Why is that Essobi? |
| 03:59.19 | loud | broadvoice, i bet. |
| 04:01.34 | Graphikos | you guys keep making me have second thoughts about BV... |
| 04:02.05 | loud | Graphikos, i bought an account last night, i feel guilty already. |
| 04:02.10 | Graphikos | ha ha.. |
| 04:02.18 | Graphikos | well I've been quite happy with BV.. |
| 04:02.27 | loud | bah, i dont know .. guess i have to test it more .. |
| 04:02.28 | Graphikos | but I haven't done anything major with it |
| 04:02.30 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 04:02.36 | _daver_ | i'm using LiveVoip here |
| 04:02.38 | loud | i cant be quite happy only with g711. |
| 04:02.41 | _daver_ | I had voicepulse for a while |
| 04:02.42 | Graphikos | never experienced anything else... |
| 04:02.56 | loud | see, livevoip is another issue, they do have good service. |
| 04:03.11 | _daver_ | livevoip has terrible customer service response times |
| 04:03.16 | VaHamish | Does anyone have any experience with the TDM400p? |
| 04:03.17 | _daver_ | they take days to respond to emails |
| 04:03.25 | loud | you have to call, not just email |
| 04:04.04 | _daver_ | also, when i first signed up, my DID was getting constant busies - they finally fixed that. |
| 04:04.28 | pulu | i call around the world more than 40 hours a month to my wife using broadvoice for US$10... sometimes it's a bit flaky but I don't know anyone else that can do it cheaper |
| 04:04.37 | okieplaya | does any one have good place for VOIP PRI like to get 30 DID over VOIP |
| 04:04.41 | loud | watch this: Got SIP response 500 "Internal Server Error" back from 147.135.12.128 |
| 04:04.45 | loud | gues who's ip is that. |
| 04:05.57 | ariel_ | VaHamish, ask the question someone here might be able to help. |
| 04:06.09 | ariel_ | okieplaya, where are you located? |
| 04:06.18 | pulu | okieplaya: if you post more details to the asterisk-biz list i'm sure lots of people will write you back |
| 04:06.43 | VaHamish | I'm trying to configure my TDM400p, i've got the drivers loaded, I've got ztcfg showing the two modules, but I can't get a dial tone on the phone I pluged into the board. |
| 04:07.26 | ariel_ | VaHamish, did you plug the phone into the correct plug. Is one fxo and otherone fxs? |
| 04:07.55 | VaHamish | Yes, I have battery on the handset, I can press a number and hear the dtmf. |
| 04:08.14 | VaHamish | Yes, one is fxo, the other is fxs, and only one gives the handset battery. |
| 04:08.17 | |Vulture| | Anyone here ever use Megapath or Xspedius T1-Data service? |
| 04:09.28 | VaHamish | Ok first question, ztcfg shows the FXO in channel 1, should the line in /etc/zaptel.conf for that channel be fxoks=1? |
| 04:09.51 | JunK-Y | its a FXO card? |
| 04:10.24 | VaHamish | ztcfg reports it as: Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 04:10.48 | *** part/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 04:10.52 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 04:11.05 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
| 04:11.26 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 04:11.35 | carlosh | hello, anyone has had success compiling gaim-sip under fedora 3 ? many thanks! |
| 04:12.28 | carlosh | or , what any suggestions for a multifunctional soft client for fedora 3 ? |
| 04:12.36 | ariel_ | VaHamish, the first port most of the time is the one you plug the phone too 2nd phone line. Now do you remember what color the modules are. |
| 04:12.38 | Grooby | installed x-lite on my imac g5. I am really impressed by it..no audio feedback from mic input |
| 04:12.41 | Grooby | very very cool |
| 04:12.58 | VaHamish | first port is the one closes to the motherboard? |
| 04:13.05 | ariel_ | hello JunK-Y |
| 04:13.08 | VaHamish | closest? |
| 04:13.15 | carlosh | i heard x-lite was already available for linux, but could not find it.. |
| 04:13.16 | ariel_ | no |
| 04:13.17 | JunK-Y | lo ariel_ |
| 04:13.28 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) |
| 04:13.34 | JunK-Y | how are ya today? |
| 04:13.35 | ariel_ | first port is what I would call top of card. |
| 04:13.52 | ariel_ | JunK-Y, just fine. |
| 04:14.20 | VaHamish | Ok, so there are four ports if MB is the MotherBoard, they go [MB 4 3 2 1] right? If so the phone is plugged into port 1. |
| 04:14.49 | Silik0n | damn hotel room with a nice highspeed wired connection |
| 04:15.13 | carlosh | I installed linphone first, along with its pre-requisites... it was ok.. but not really there yet.. so I tried gaim-sip, I haven't been able to successfully compile it... |
| 04:15.27 | VaHamish | and ztcfg -vv reports that as an FXO. |
| 04:15.37 | okieplaya | <ariel_> sorry im in muskogee oklahaoma |
| 04:16.17 | okieplaya | <pulu> where do i post this |
| 04:17.53 | ariel_ | okieplaya, wow I don't know any one that has pri service there. But you can check with X/O and qwest for service? |
| 04:17.57 | carlosh | could anyone suggest or tell what softphone are you using on linux ? thanks. |
| 04:18.21 | Sedorox | kphone? |
| 04:18.30 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 04:18.33 | okieplaya | yea i have seen it on on myphonecompany.com but they dont call back |
| 04:18.45 | ariel_ | VaHamish, is asterisk running? |
| 04:18.58 | VaHamish | Yes. |
| 04:19.00 | *** join/#asterisk BoRiS (~boris@24.81.0.252) |
| 04:19.02 | okieplaya | o really qwest has voip pri |
| 04:19.05 | ariel_ | okieplaya, get a reseller don't go direct. |
| 04:19.27 | okieplaya | reseller like? |
| 04:19.40 | okieplaya | i have read every thing i get my hands on |
| 04:19.41 | ariel_ | here we are paying for a pri with x/o 499 per month and 3.20 per group of 20 did's. |
| 04:19.56 | okieplaya | wow |
| 04:20.02 | okieplaya | man thats great |
| 04:20.08 | okieplaya | from> |
| 04:20.10 | okieplaya | ? |
| 04:20.28 | marc32344 | ariel-- location? |
| 04:20.30 | ariel_ | I am in Miami, Florida |
| 04:20.35 | carlosh | ariel .. where is that ? |
| 04:20.38 | carlosh | ok |
| 04:20.48 | |Vulture| | ariel_: can you get multiple local areas on that PRI? |
| 04:20.48 | marc32344 | ariel--contract? |
| 04:20.53 | okieplaya | can u talk on all 20 at the same time |
| 04:21.04 | |Vulture| | okieplaya: shhh |
| 04:21.09 | |Vulture| | lol |
| 04:21.12 | *** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com) |
| 04:21.17 | |Vulture| | its a PRI |
| 04:21.22 | ariel_ | |Vulture|, they used to but since B/S did a letter we now have to pay for the 3 county. |
| 04:21.24 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 04:21.30 | ariel_ | pri yes |
| 04:21.44 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:21.52 | |Vulture| | B/S is crap we play just about that for 5 POTs in miami |
| 04:21.54 | jsmith | Can anyone tell me what the Progress() application is supposed to do? (Yes, I'm documenting it as part of the Asterisk Documentation Project) |
| 04:22.06 | okieplaya | shhh? most phone companys have pri but its a group of 20 and only talk 2 at a time |
| 04:22.17 | ariel_ | X/O is offering a 6 pots rest data for 399. |
| 04:22.27 | ariel_ | and the 6 pots is pri |
| 04:22.39 | |Vulture| | Id rather just get a full pri for $100 more |
| 04:22.40 | VaHamish | Yeah, go it working.. |
| 04:22.43 | Sedorox | jsmith: ummm... my guess... announces the progress in the queue.. or.. the progress of a call... |
| 04:22.44 | VaHamish | hot damn.. |
| 04:23.02 | ariel_ | VaHamish, which port was it? |
| 04:23.04 | Sedorox | I wish I could get a decent pri around here... :/ |
| 04:23.26 | VaHamish | my problem was in zapata.conf |
| 04:23.52 | okieplaya | why u say that <|Vulture|> |
| 04:23.53 | VaHamish | I had the two channels configured both with fxs_ks. One should be fxs_ks, the other fxo_ks |
| 04:24.00 | marc32344 | ne1 knows what hardware packet8 uses? |
| 04:24.02 | jsmith | Sedorox: Unfotunately, guesses won't work here. I need to know, so that I can document it properly. |
| 04:24.09 | Sedorox | yea |
| 04:24.27 | |Vulture| | okieplaya: I was just messing around |
| 04:24.31 | okieplaya | o |
| 04:25.03 | |Vulture| | na I am new to PRIs any info I can get is helpful :) |
| 04:25.12 | okieplaya | anyone reselling voip in the room |
| 04:25.18 | ariel_ | okieplaya, sounds like isdn |
| 04:25.40 | okieplaya | yea it is |
| 04:25.48 | okieplaya | there 50$ |
| 04:25.52 | okieplaya | sucks ass |
| 04:26.03 | okieplaya | small town |
| 04:26.22 | ariel_ | okieplaya, not bad price for isdn. it's a going away here. |
| 04:27.11 | mishehu | has anybody used the Citel Handset Gateway for legacy digital pbx phones? |
| 04:28.29 | mishehu | I'm looking for some information and user experiences with the device |
| 04:29.13 | carlosh | KPHONE does not seem to have a password field in its configuration page, so it fails to register... :( |
| 04:30.06 | ariel_ | carlosh, add wine and use xlite |
| 04:30.44 | mishehu | add wine, and get drunk |
| 04:31.02 | carlosh | ariel_ : does xlite work ok with asterisk ? |
| 04:31.13 | Sedorox | yes |
| 04:31.14 | ariel_ | yes |
| 04:31.18 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 04:32.27 | znoG | no |
| 04:32.39 | znoG | .. just felt like being different :) |
| 04:35.56 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 04:35.58 | Sedorox | lol |
| 04:39.16 | carlosh | Serodox / ariel_ thanks, I just hoped i could find gaim-sip rpm.... it'd be kool to have IM as well as SIP (if not IAX2).. |
| 04:39.47 | carlosh | gaim-sim is compliant with lindows / linspire.. only |
| 04:39.55 | _Vile | oh you lamo's where's the conf people? |
| 04:40.12 | _Vile | wrong chan |
| 04:40.12 | Sedorox | hmmmm |
| 04:40.15 | Sedorox | lol |
| 04:40.56 | marc32344 | are the small voip providers getting ANY signups? |
| 04:41.05 | _Vile | im sure |
| 04:41.17 | _Vile | but they're not getting good LD rates |
| 04:41.27 | _Vile | too small of volumes |
| 04:41.32 | ariel_ | some small ones are actually doing well. |
| 04:41.52 | Sedorox | if I knew of some small ones I would probably get stuff from them |
| 04:41.53 | _Vile | but i'm sure some of them are doing OK |
| 04:41.54 | marc32344 | dont see many residents trust their phone line to a no-name providers. |
| 04:42.15 | _Vile | most residents go with local voip providers if they exist in their area |
| 04:42.24 | _Vile | otherwise they go with the names they hear on TV |
| 04:42.36 | _Vile | or stick with their current ILEC, CLEC etc |
| 04:42.53 | _Vile | dont ask why |
| 04:43.05 | marc32344 | why? |
| 04:43.38 | bkw_ | kllkj |
| 04:43.44 | Sedorox | my mom doesn't wanna switch :(:( she had a friend on vonage and when she was talking to her apparnetly they were having some problems.. but I'm not sure if their connection at the time was the best for it anyway |
| 04:43.54 | _Vile | because 1) they like local service, everyone likes keeping money local 2) they've heard of the VoIP provider on TV or 3) the local ILEC/CLEC is the only choice |
| 04:44.16 | _Vile | and really in that instance the CLEC will provide voip while the ilec will not |
| 04:44.31 | _Vile | so it's just a matter of whether you have any clecs in your area |
| 04:44.41 | _Vile | and clecs are going away |
| 04:44.49 | _Vile | since the 1996 reversal crap is happening |
| 04:44.56 | marc32344 | I dont even think that even local competition is possible. NO brand. |
| 04:44.58 | ariel_ | The biggest problem with vonage and others at the start were problems with the like of tivo it really gave them a bad name. |
| 04:45.03 | _Vile | under the nigger bush administration |
| 04:45.21 | _Vile | and I use that term as vulgar as I can use it |
| 04:45.22 | _Vile | . |
| 04:45.38 | Sedorox | how did tivo give vonage a bad name? |
| 04:45.59 | ariel_ | they would not work. |
| 04:46.13 | Sedorox | oh together? |
| 04:46.18 | marc32344 | how can a small provider go up against the big names without any brand recognition? |
| 04:46.29 | marc32344 | offering lower prices will not do |
| 04:46.31 | _Vile | marc, I have brand and I'm not competing with the ILEC atm in terms of VoIP |
| 04:46.37 | _Vile | and selling $20 less |
| 04:46.37 | ariel_ | tivo plugged into there line would not get updates. |
| 04:46.40 | _Vile | per month |
| 04:46.45 | Sedorox | ooooooo ok.... |
| 04:46.52 | Sedorox | interesting |
| 04:47.01 | _Vile | create a brand and market it, easy |
| 04:47.02 | ariel_ | marc32344, service will. |
| 04:47.10 | Sedorox | someone in my gall here on campus has a tivo.. was gnna try and hack it :-p |
| 04:47.23 | marc32344 | create brand?? huge cost!! |
| 04:47.23 | Sedorox | hall* |
| 04:47.43 | ariel_ | In fact it's still a problem. But now tivo has an internet option so it's going to be less of a problem. |
| 04:47.43 | _Vile | dude you're smoking crack, creating a brand is registering a name and getting word of mouth |
| 04:47.52 | _Vile | advertising on the radio maybe |
| 04:47.57 | _Vile | or local TV maybe |
| 04:48.15 | _Vile | work deals with local TV and radio stations, cut them deals on phone or internet service |
| 04:48.16 | Sedorox | ariel_: yea... |
| 04:48.18 | ariel_ | you hit the customers you have for basic service. then you ask for reference then you flood the small area with ads. |
| 04:48.23 | _Vile | get yourself recognized and call it a day |
| 04:48.31 | _Vile | word of mouth is best |
| 04:48.36 | marc32344 | most small voip have sat idle. really. |
| 04:48.51 | Sedorox | anyone know a few good small providers? |
| 04:49.02 | ariel_ | www.race.com is one. |
| 04:49.20 | _Vile | i only deal w/ 100k minute + providers |
| 04:49.26 | _Vile | as my customers |
| 04:49.31 | _Vile | so I can't comment much here |
| 04:49.38 | _Vile | aside from my past experience |
| 04:49.53 | Sedorox | hmmm |
| 04:50.12 | ariel_ | _Vile, so your a voip provider yourself? |
| 04:50.24 | Sedorox | I'm looking for the company I'm working with right now.. we just got asterisk up for between us all and stuff and its great.. but still need outside access... so.. |
| 04:50.33 | _Vile | only to big customers, but yes.. I stem from a background as being a small guy though |
| 04:50.35 | Sedorox | but the catch is its a canadian company |
| 04:50.55 | marc32344 | you need big name to go into residential market. |
| 04:50.56 | Beirdo | why is that a catch? |
| 04:50.59 | _Vile | look at some canadian voip providers, I think protus can help you |
| 04:51.23 | Sedorox | I haven't been any to see a lot of voip providers that do CND did's |
| 04:51.40 | Beirdo | ah |
| 04:51.45 | Sedorox | and I haven't seen a lot of CND voip providers |
| 04:51.46 | Beirdo | that is a bit of an issue, yes |
| 04:51.50 | _Vile | there's a couple that can, I don't do cdn, I only do 48 |
| 04:51.58 | Sedorox | some.. dunno where they are... do have some dids.. like I think BV... but... |
| 04:51.58 | _Vile | ask protus |
| 04:52.02 | Beirdo | talk to bjohnson, he's working on a list |
| 04:52.02 | Sedorox | yea... |
| 04:52.09 | _Vile | I think their flat rate is 1.75 |
| 04:52.12 | Sedorox | protus? |
| 04:52.18 | marc32344 | now the big players are lower their fees... margins are going down fast |
| 04:52.31 | _Vile | marc, no diff than the standard ld market |
| 04:53.47 | marc32344 | unless you have lots of bandwidth, going up against them with a single T1 will be hard |
| 04:54.12 | _Vile | I have enough bandwidth for what I do |
| 04:54.20 | ariel_ | you can't go it with a t1 |
| 04:54.32 | _Vile | and I deal in DS-3s and not T-1s |
| 04:54.52 | _Vile | though, annoying T-1 customers seem to enjoy bothering me |
| 04:55.22 | marc32344 | the upfront cost is rising |
| 04:55.26 | _Vile | tnt is cool |
| 04:55.33 | _Vile | good sip termination boxes |
| 04:55.50 | ariel_ | _Vile, there programming sucks it's worst then the cisco's cli |
| 04:56.15 | ariel_ | But when you get then configure there up and running and you can forget them almost. |
| 04:56.34 | _Vile | i disagree, depending on the platform, cisco's programming is more intuitive |
| 04:56.51 | _Vile | ariel, which is the way it should be :) |
| 04:57.17 | marc32344 | so the local cable company gives it at $15/month. How much lower will you have to offer to overcome the brand awareness? |
| 04:57.22 | marc32344 | $5? |
| 04:57.38 | _Vile | im looking at the outdated and completely unsupported VCO series when cisco bought summa4 as a LCR solution atm |
| 04:57.50 | _Vile | vco80 atm |
| 04:58.39 | _Vile | $15/mo? |
| 04:58.43 | _Vile | you're kidding me |
| 04:58.48 | _Vile | that should be illegal |
| 04:58.59 | _Vile | i swear |
| 04:59.07 | _Vile | the utility companies are taking over again |
| 04:59.18 | ariel_ | well 19.95 mark is looking like the braking point don't think it will get lower any time soon. |
| 04:59.30 | _Vile | it's $35/mo here |
| 04:59.41 | ariel_ | 35 for cable? |
| 04:59.49 | _Vile | for cable internet |
| 04:59.52 | ariel_ | I was thinking 19.95 for voip service. |
| 05:00.02 | marc32344 | no, voip serv |
| 05:00.11 | ariel_ | here cable is around 49 per month for internet access. |
| 05:00.12 | loud | ok im less angry with bv .. works kinda OK right now .. |
| 05:00.18 | _Vile | the local cable company is looking at going into VoIP now too |
| 05:00.25 | _Vile | which should be illegal as well |
| 05:00.43 | _Vile | regulation is needed, but won't happen under this administration |
| 05:00.47 | ariel_ | same here got a mailer from them. |
| 05:00.59 | ariel_ | no please no more regulations |
| 05:01.08 | Beirdo | why should it be illegal? |
| 05:01.08 | _Vile | you have to have regulation |
| 05:01.08 | Sedorox | guess we're gonna start to see cable modems from the likes of Motorola (and cisco will have to come out with a new one) that has fxs ports ini t... |
| 05:01.17 | _Vile | otherwise you kill local competition |
| 05:01.25 | _Vile | and send money to other states |
| 05:01.33 | Beirdo | the cable companies have a vast IP network, why shouldn't they be allowed to do VOIP |
| 05:01.38 | _Vile | which drains local economy |
| 05:01.40 | _Vile | etc etc etc |
| 05:01.59 | _Vile | read up |
| 05:01.59 | ariel_ | but that is why there are many different vies. |
| 05:02.14 | ariel_ | ./vies/views |
| 05:02.15 | marc32344 | atleast two big providers lowered their rates in the past 3months |
| 05:02.31 | Beirdo | who cares where the money ends up, it's all going to the rich anyways |
| 05:02.31 | _Vile | if you let local cable companies compete in the voice market |
| 05:02.38 | _Vile | then you now have two ILECs to deal with |
| 05:02.39 | moonwick | yeah, let's regulate voip so we can make it more expensive and protect the little guys. |
| 05:02.40 | ariel_ | what I don't want to see is the big mergers of the bells coming up. |
| 05:02.42 | moonwick | pfft. |
| 05:02.43 | _Vile | who will only fix prices |
| 05:03.04 | _Vile | the local ilecs are buying out the IXCs |
| 05:03.11 | _Vile | since they get their LD license back |
| 05:03.17 | Sedorox | everyone is buying out someone |
| 05:03.20 | _Vile | and can now compete |
| 05:03.28 | Beirdo | who cares? |
| 05:03.37 | ariel_ | voip service, sat service, wireless service it's all going to merge in the end. |
| 05:03.41 | _Vile | anyone who does local service cares |
| 05:03.45 | _Vile | or should care |
| 05:03.49 | Sedorox | ariel_: sad.. but true... |
| 05:04.07 | Beirdo | if they can't compete, they die. such is life |
| 05:04.26 | marc32344 | the T1 providers win. |
| 05:04.30 | _Vile | uhm, read up on the 1996 Telecommunications Act and see why CLECs were put into place. |
| 05:04.34 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 05:04.39 | _Vile | and then come back and talk here |
| 05:04.42 | _Vile | until then stfu |
| 05:04.46 | Mike | someone knows whats the last wisip firmware version? |
| 05:04.52 | Beirdo | I don't care why it was put into place, it's irrelevant anyways |
| 05:05.10 | _Vile | it's very relevant |
| 05:05.23 | Beirdo | regulation has not made the US telecom market any cheaper for the consumer, just more confusing |
| 05:05.46 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 05:05.46 | _Vile | I can offer a data T-1 for $99, the ILEC is charging you $500+ |
| 05:05.46 | ariel_ | actuall I think it's made it more expensive |
| 05:05.49 | _Vile | tell me about local competition. |
| 05:05.57 | Beirdo | and I will not STFU. Maybe you should just chill out. |
| 05:06.01 | Sedorox | _Vile: where you at? :-p |
| 05:06.06 | _Vile | central oregon |
| 05:06.11 | Sedorox | damn |
| 05:06.15 | _Vile | rural |
| 05:06.26 | _Vile | zone 2 T1s are no more than $100 more. |
| 05:06.37 | Sedorox | hmm |
| 05:06.43 | Beirdo | there's no reason that companies shouldn't be allowed to be in the marketplace just to save the little guys, that's absurd |
| 05:06.51 | _Vile | they classed us at zone 1 for whatever reason |
| 05:07.09 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 05:07.15 | _Vile | Beirdo, you should really do your history. |
| 05:07.26 | Beirdo | those who can't compete die out |
| 05:08.07 | ariel_ | _Vile, how far do you have to be from the big city to get zone 2? |
| 05:08.11 | _Vile | Beirdo, when the ilec owns the facilities that you purchase and decides to raise rates on you, and basically tells you what your selling point is, and then sells lower, that's illegal. |
| 05:08.20 | _Vile | it's all about facilities |
| 05:08.35 | _Vile | if I could own the copper pairs I have going into the businesses I serve |
| 05:08.46 | _Vile | then we'd be discussing a different matter. |
| 05:09.00 | marc32344 | how much initial is needed to go into local res voip? |
| 05:09.03 | _Vile | ariel, 30-40 minutes. |
| 05:09.03 | Beirdo | how is that related in any way to cable companies offering VOIP? |
| 05:09.13 | _Vile | or 10-20 miles or so |
| 05:09.37 | _Vile | Beirdo, because cable companies have local facilities into most people's homes |
| 05:09.50 | Beirdo | and you already said you are offering T1 at 1/5 of the price of the ILEC, you are already competing |
| 05:09.57 | *** join/#asterisk mooboi (~selfsck@silenceisdefeat.org) [NETSPLIT VICTIM] |
| 05:10.07 | Beirdo | so? |
| 05:10.10 | _Vile | only at their grace, lately they seem to be able to raise the price on everything |
| 05:10.11 | Beirdo | so do most ISPs |
| 05:10.18 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
| 05:10.21 | _Vile | read up man |
| 05:10.33 | Beirdo | VOIP is a broadband data transport issue |
| 05:10.40 | _Vile | FCC regulations are reversing the 1996 act of splitting the bell |
| 05:10.43 | Beirdo | it doesn't fit into the telco paradigm |
| 05:10.54 | Beirdo | so what? |
| 05:11.03 | Beirdo | in Canada we NEVER split our Bell |
| 05:11.10 | _Vile | sad |
| 05:11.16 | Sedorox | yea |
| 05:11.18 | Sedorox | and you got telus |
| 05:11.23 | Sedorox | which chargs out the arse |
| 05:11.29 | Beirdo | and we have good service at comparable rates to the US, and it is a heck of a lot confusing |
| 05:11.43 | _Vile | I give up |
| 05:11.48 | _Vile | your service sucks |
| 05:11.53 | Sedorox | lol |
| 05:11.53 | _Vile | you've got great health care |
| 05:12.00 | shido6 | what is ffdefault.cfg? |
| 05:12.04 | _Vile | and we don't |
| 05:12.06 | shido6 | anyone have an example |
| 05:12.07 | _Vile | let's call it even |
| 05:12.13 | Sedorox | :-p |
| 05:13.01 | Beirdo | let's not :) |
| 05:13.09 | _Vile | ok |
| 05:13.14 | _Vile | then I'm on the high side |
| 05:13.17 | _Vile | and your service still sucks |
| 05:13.29 | _Vile | bell canada |
| 05:13.47 | Sedorox | lol |
| 05:13.53 | _Vile | that's like asking someone for a boot |
| 05:14.06 | Beirdo | VOIP has not much to do with old-style telcos |
| 05:14.22 | _Vile | doesn't matter, your country still sucks |
| 05:14.29 | Beirdo | otherwise I'd never be allowed to run my own VOIP service to bypass Bell |
| 05:14.36 | Sedorox | oi.... |
| 05:14.39 | Beirdo | you are saying that THAT should be regulated |
| 05:14.48 | Beirdo | and I say phooey on that |
| 05:15.05 | _Vile | I'm saying that if you're a US company, then regulation is very important to being your own VoIP provider |
| 05:15.12 | _Vile | and very important to you as being a CLEC |
| 05:15.24 | _Vile | who cares about canada, I know nothing about them other than that they talk funny |
| 05:15.36 | |Vulture| | lol |
| 05:15.41 | Beirdo | spoken like a true ignorant Yank :) |
| 05:15.44 | *** part/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 05:15.44 | Sedorox | lol |
| 05:15.50 | _Vile | southern boy here |
| 05:16.12 | Beirdo | I'll continue to call you a Yank while you continue to act like one :) |
| 05:16.46 | ariel_ | _Vile, southern boy....hum he is out west. |
| 05:17.04 | _Vile | I partnered with a guy in Canada about 6 years ago doing jsworld.com, pcgaming.com, netpedia.com, he in turned fucked me over, so, I'll continue to disrespect you people until I guess that's been made right |
| 05:17.06 | ariel_ | well it's my bed time. I have work to do in the morning. See you all later. |
| 05:17.08 | Beirdo | as a consumer, disallowing companies from providing us service is NEVER in our best interest. |
| 05:17.13 | _Vile | I'm currently out west, yes |
| 05:17.29 | |Vulture| | night ariel_ |
| 05:17.29 | Beirdo | Oh, blame an entire country for one scumbag? |
| 05:17.42 | _Vile | Beirdo, yeah it's called discrimination. |
| 05:17.50 | |Vulture| | lol |
| 05:17.55 | Beirdo | OK, in that case maybe I should disrespect all of the US due to Bill Gates :) |
| 05:18.04 | Beirdo | hardly fair reasoning |
| 05:18.20 | Sedorox | oi |
| 05:18.38 | _Vile | actually, Bill Gates is worse than the bell's. |
| 05:18.47 | _Vile | hurts more companies than the big ilecs |
| 05:18.57 | Beirdo | no kidding |
| 05:19.21 | _Vile | I'd assume get rid of him, hell if we did that, I'd quit my job and enjoy holidays for the next 30 years |
| 05:19.29 | Sedorox | lol |
| 05:19.30 | _Vile | but it won't happen |
| 05:19.41 | Beirdo | and whether you like it or not, ultimately, it's what hurts the consumers that's the most important, not what hurts companies trying to break into the market |
| 05:19.50 | Beirdo | as such, some regulation is necessary |
| 05:20.01 | Beirdo | but be careful not to wish for too much |
| 05:20.18 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 05:20.24 | _Vile | when I'm pricing phone lines at $25 w/ no FCC Subscriber Line Charge, charging the customer no more than $28/mo and the ilec is charging them $38/mo |
| 05:20.27 | _Vile | who benefits? |
| 05:20.28 | bkw_ | ibook*CLI> show version |
| 05:20.28 | bkw_ | Asterisk CVS-HEAD-02/23/05-23:18:54 built by brian@ibook.local on a Power Macintosh running Darwin |
| 05:20.34 | Sedorox | I just don't understand why BW prived are so hugh... |
| 05:20.39 | Sedorox | damn |
| 05:20.40 | Beirdo | the ILECs are screwing the customers, granted |
| 05:20.55 | _Vile | and who's giving them the power to do so? |
| 05:21.00 | Beirdo | and that's always been the case |
| 05:21.07 | _Vile | and when they raise their rates, then what? |
| 05:21.11 | _Vile | regulation. |
| 05:21.12 | Beirdo | the FCC and the customers allow it |
| 05:21.13 | _Vile | needed. |
| 05:21.27 | Sedorox | yea.. |
| 05:21.30 | _Vile | customers don't... customers turn to the competition. |
| 05:21.38 | Beirdo | that's the way it works with capitalism. supply and demand |
| 05:21.42 | _Vile | if you eliminate the competition then we're back to monopolyland. |
| 05:21.51 | Beirdo | if the consumers will pay it, it happens |
| 05:22.01 | Beirdo | and we do pay it, but grudgingly |
| 05:22.06 | _Vile | only because they want phone service, though |
| 05:22.16 | Beirdo | yes |
| 05:22.27 | Beirdo | you already have regulation to allow for CLECs |
| 05:22.36 | Beirdo | as do we |
| 05:22.38 | _Vile | my point is, it's reversing |
| 05:23.06 | Beirdo | well, maybe you guys shoulda voted for a different administration :) |
| 05:23.26 | _Vile | not my fault |
| 05:23.31 | Sedorox | the problem isn't clecs and ilecs.. as such... its them getting into VOIP.. Which is a data service.. voip and since they already have a telephone backend.. and a name.. they'll have a monopoly very fast again |
| 05:23.32 | _Vile | everyone I know voted against that fat fuck |
| 05:23.48 | |Vulture| | he is fat? |
| 05:23.51 | Sedorox | lol |
| 05:23.54 | _Vile | colin powells' son is running the FCC |
| 05:23.54 | Beirdo | Sedorox: not likely. |
| 05:24.02 | _Vile | how did that happen |
| 05:24.03 | Sedorox | we didn't vote him tho |
| 05:24.09 | _Vile | and no, we didn't |
| 05:24.13 | _Vile | he was appointed |
| 05:24.13 | Sedorox | at least he's stepping down... |
| 05:24.16 | _Vile | by that dumbass |
| 05:24.16 | Sedorox | yea... |
| 05:24.18 | _Vile | and yes he is |
| 05:24.20 | Sedorox | hehe |
| 05:24.31 | Beirdo | while people can offer it cheaper, they will not have a monopoly |
| 05:24.38 | |Vulture| | anyone ever use Xspedius or Megapath? |
| 05:24.43 | _Vile | Beirdo, they raised our price by $3 |
| 05:24.43 | marc32344 | the small player has really no chance. |
| 05:24.47 | _Vile | got rid of UNE-P |
| 05:24.51 | _Vile | per line |
| 05:24.59 | _Vile | they will raise it another $3 in the next year |
| 05:25.03 | _Vile | $6 more per line |
| 05:25.16 | Beirdo | what kind of line? |
| 05:25.19 | _Vile | pots |
| 05:25.20 | Sedorox | Beirdo: true.. but the thing is.. they have the name.. so more people will goto them.. and they will charge more for other providers to has access to the PSTN.. |
| 05:25.23 | |Vulture| | yea our communications taxes are crap |
| 05:25.34 | _Vile | from $16 to $21.. |
| 05:25.37 | _Vile | and when we sell at $25 |
| 05:25.39 | Beirdo | well, that's a failure of FCC |
| 05:25.44 | Beirdo | but still |
| 05:25.48 | _Vile | it makes profit go to $4/line |
| 05:26.18 | Beirdo | it's also called inflation |
| 05:26.20 | |Vulture| | _Vile: will you be passing on the increase to new customers? |
| 05:26.49 | _Vile | Vulture, yep, anyone who's not contracted |
| 05:26.49 | _Vile | which hurts sales |
| 05:26.52 | |Vulture| | what does the ILEC charge? |
| 05:27.09 | _Vile | depends on the service, but not off by much |
| 05:27.20 | _Vile | $35 vs $25 I think |
| 05:27.27 | |Vulture| | we get raped by Bellsouth down here in Florida |
| 05:27.35 | _Vile | but they also charge that fcc subscriber line charge |
| 05:27.37 | _Vile | which is all profit |
| 05:27.46 | _Vile | so deduct $6.50 from that |
| 05:28.04 | _Vile | I used to get raped by Bellsouth in TN |
| 05:28.38 | Beirdo | the whole point is... VOIP negates the need for POTS |
| 05:28.57 | |Vulture| | I use Bellsouth in FL, Verizon in CA and I forgot the ilec in St. Louis, MO |
| 05:29.05 | Beirdo | if you are trying to make a profit doing POTS, good luck. |
| 05:29.33 | _Vile | true, but offer reliable 911 service and negotiate calea, then let me know how your service is. |
| 05:29.35 | |Vulture| | voip has the tendancy to be sketchy |
| 05:29.47 | mishehu | ugh. |
| 05:29.53 | mishehu | speex pisses me off |
| 05:29.59 | Beirdo | calea is a stupid idea, your government's on crack |
| 05:30.00 | |Vulture| | mishehu: why is that? |
| 05:30.13 | _Vile | Beirdo, i've never disagreed with that. |
| 05:30.15 | Inv_arp | when u guys say POTS line u mean the phone numbers themselves? |
| 05:30.21 | shido6 | no |
| 05:30.23 | _Vile | inv, no, copper |
| 05:30.25 | |Vulture| | DIDs are #s |
| 05:30.27 | mishehu | it works perfectly fine when the originating point is IAX2, but if the originating point is SIP g711ulaw, audio gets all chopped and desynced |
| 05:30.27 | shido6 | plain old telephone service |
| 05:30.29 | |Vulture| | POTS are regular lines |
| 05:30.31 | shido6 | co lines |
| 05:30.34 | shido6 | analog lines |
| 05:30.38 | Inv_arp | ahh k |
| 05:30.42 | Inv_arp | continue |
| 05:30.50 | shido6 | Inv_arp keep listening :) |
| 05:30.54 | shido6 | been doin it for 3 yrs |
| 05:30.58 | VaHamish | Ok, I've got another Zaptel question.. |
| 05:31.13 | mishehu | |Vulture|: you used speex at all? |
| 05:31.14 | Sedorox | lol |
| 05:31.17 | Beirdo | calea sucks ass. :) |
| 05:31.24 | |Vulture| | I have 7 offices on * right now, and I am about to bring a T1 PRI and a T1-Data to my main office to terminate inbound calls and send them to each office |
| 05:31.41 | Sedorox | hmmm |
| 05:31.55 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 05:32.12 | Inv_arp | |Vulture|: got any examples of your setup on wiki blogs etc....? |
| 05:32.28 | VaHamish | does Zap/1 refer to the module in channel 1, and so then the module in channel 4, would be Zap/4? |
| 05:32.29 | MrEntropy | is a 200 response still considered an INVITE? otherwise what method is it? |
| 05:32.42 | |Vulture| | then send LD over VPC/NuFone and trying to get extended local areas on my PRI |
| 05:32.46 | file | a 200 is a sip reply |
| 05:32.48 | |Vulture| | Inv_arp: no |
| 05:32.49 | Sedorox | |Vulture|: here's a question.. I'm doing something similar but with just three * boxes... do you use switch lines.. or have everything hard coded? |
| 05:32.58 | file | 200 OK, it has a status code... 200 |
| 05:32.59 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 05:33.11 | _Vile | going to bed, later |
| 05:33.13 | MrEntropy | file: so it doesn't come under any sip methods? |
| 05:33.20 | Sedorox | night _Vile |
| 05:33.23 | Beirdo | there's only so much profit to be made in the telco world. to get it cheap enough to compete, you need to be doing something higher density than one line per pair of copper |
| 05:33.24 | file | sip methods are used in sip requests |
| 05:33.30 | |Vulture| | Sedorox: right now I have TDMs in each office bringing in POTS... but they are so expensive that 6 POTS pretty much equals a PRI in some areas |
| 05:33.36 | file | ie: INVITE, ACK, BYE, INFO, REFER, CANCEL |
| 05:33.45 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 05:33.53 | MrEntropy | i see, ok, thank you |
| 05:34.05 | Sedorox | |Vulture|: well yea.. but wasn't sure which you did.. because I did have switch but I had to switch one back to sip instead of iax |
| 05:34.06 | Sedorox | :/ |
| 05:34.12 | file | sleeeeeeep I must go to sleeeeeeeeep |
| 05:34.14 | marc32344 | anyone knows what equip packet8 runs? |
| 05:34.20 | Sedorox | ./file & |
| 05:34.20 | Sedorox | :-p |
| 05:34.56 | Inv_arp | |Vulture|: gsm from office to office? |
| 05:35.02 | |Vulture| | Sedorox: I have noticed over the past 5 months a lot of VoIP providers have gotten much more stable, almost to the point I don't worry about it |
| 05:35.08 | |Vulture| | ilibc |
| 05:35.12 | Beirdo | _Vile: you just need a better regulator :) CRTC is actually good for some things |
| 05:35.23 | *** join/#asterisk nwhit (~chatzilla@65.107.59.67.ptr.us.xo.net) |
| 05:35.25 | |Vulture| | Id like to do 729 though |
| 05:35.38 | |Vulture| | if I do the PRI I will prolly do 729 |
| 05:35.55 | Sedorox | |Vulture|: no.. between the boxes.. to dial extentions across.. do you do direct... or switch? |
| 05:36.04 | |Vulture| | direct |
| 05:36.05 | *** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com) |
| 05:36.08 | Sedorox | ok |
| 05:36.15 | Inv_arp | im confused isnt a PRI 24 lines? |
| 05:36.16 | godsmoke | is this old news?: http://www.linuxdevices.com/news/NS9180984123.html |
| 05:36.18 | nwhit | Hey all... i am having trouble with the current cvs release of asterisk... the voicemail system won't accept any of the passwords... it reports in the log that they are invalid, but it puts a B in front of it any ideas? |
| 05:36.25 | Sedorox | Inv_arp: I believe so |
| 05:36.26 | |Vulture| | Inv_arp: 23B+1D |
| 05:36.30 | Sedorox | hehe |
| 05:36.42 | |Vulture| | 23 voice and a data channel |
| 05:36.53 | Inv_arp | so what does it mater on codec that is used for PRI |
| 05:37.03 | Inv_arp | if all u can have is 24 |
| 05:37.20 | |Vulture| | Inv_arp: I am talking about sending these lines to the other offices through the main office |
| 05:37.40 | |Vulture| | Inv_arp: for example the test case will be a large office in Jacksonville, FL with a small office in Daytona, FL |
| 05:37.46 | Sedorox | we're (the company I work with) looking into getting a office in Canada (where the main people are and the servers and stuff) but I'm not sure if we wanna get a T1 and split it.. two T1's one data and one voice... keep like 3 pots and cable for internet.. or what.. |
| 05:38.08 | |Vulture| | Inv_arp: when someone dials Daytona, FL it will go into Jacksonville through the PRI and route over the internet to Daytona and ring into their system |
| 05:38.17 | Inv_arp | ahhh |
| 05:38.39 | |Vulture| | and when someone picks up a line to make an outbound call in Daytona, it goes over the inet to jacksonville, then dials out on the PRI |
| 05:38.49 | |Vulture| | basically making a mini VoIP provider |
| 05:38.58 | Inv_arp | nice |
| 05:39.12 | Inv_arp | all with * and cisco i bet |
| 05:39.14 | |Vulture| | however NuFone offers the best LD rates I've ever seen |
| 05:39.27 | nwhit | any ideas on my voicemail problem? |
| 05:39.28 | |Vulture| | * and Polycom, I use to use Cisco, but I prefer Polycoms |
| 05:39.36 | Sedorox | hehe |
| 05:39.48 | Inv_arp | i use voipjet for longdistance |
| 05:39.58 | Inv_arp | and BV for incoming |
| 05:40.12 | |Vulture| | Im surprised you don't use BV for LD |
| 05:40.14 | Sedorox | nwhit: I remember seeing you saying something like that before... or at least someone else..dunno what ended up with it |
| 05:40.27 | Inv_arp | dont call outbound that much |
| 05:40.36 | nwhit | sedorox: probably me... are you using the latest cvs? |
| 05:40.49 | |Vulture| | Inv_arp: ah you have a local BV account? |
| 05:40.55 | Inv_arp | heh everyone is shocked how i can chge my callerid on fly lol |
| 05:41.00 | Inv_arp | |Vulture|: yes |
| 05:41.05 | Sedorox | here's a question.. what do you guys recommened for about 3-5 calls in and out... solid.. what kind of BW/internet connection would you recommend? |
| 05:41.09 | PyroSteve | hey guys |
| 05:41.13 | Inv_arp | BYOD 5.9h + 1.50 a month |
| 05:41.14 | |Vulture| | Inv_arp: yea CID passing is great |
| 05:41.16 | PyroSteve | i have a small problem |
| 05:41.20 | Sedorox | nwhit: I tried it.. but I had problems with my sip phone.. so I moved back to 1.0.5 |
| 05:41.29 | PyroSteve | i have a call file that is being created |
| 05:41.35 | |Vulture| | Sedorox: thats my exact setup in 5 offices |
| 05:41.46 | Sedorox | what do you use? T1's? |
| 05:42.01 | PyroSteve | to dial out to the pstn via my sip <-> pstn provider |
| 05:42.05 | |Vulture| | Sedorox: I use 4 POTS and then overflow into VoicePulse |
| 05:42.08 | nwhit | sedorox: thanks |
| 05:42.16 | mishehu | ugh. |
| 05:42.20 | mishehu | fricking speex |
| 05:42.21 | Inv_arp | but its hard comming from Linux admin/php to voip phones so my phreaking grammer isnt to good |
| 05:42.23 | MrEntropy | UAS is the system recieving the call and UAC is the client who made the call, is that correct? |
| 05:42.27 | Sedorox | nwhit: for what? lol |
| 05:42.36 | |Vulture| | Sedorox: you might be better getting a Fractional T1, 6voice, rest data, bring it into a T100P and have inet and voice for the office |
| 05:42.45 | Sedorox | yea... |
| 05:42.50 | PyroSteve | and to put the called number in the a context and extension that reads dtmf |
| 05:43.04 | |Vulture| | Inv_arp: don't worry Ive only been at it for 7 or so months and I am still learning |
| 05:43.12 | PyroSteve | it seems like everything works, but asterisk wont detect the dtmf |
| 05:43.16 | nwhit | i'm going to try moving it back... it always seems to give me a fit when I do, though |
| 05:43.17 | Sedorox | because we have staff in the US and CND... so we're look at this as a solution to help.. but just not sure on the BW needs if we have it terminating in one point and to have the other point dial out, w/o there being jitter and stuff |
| 05:43.34 | Sedorox | nwhit: oh... try installing to a different directory |
| 05:43.39 | Sedorox | and copying your conf's over |
| 05:43.41 | mishehu | anybody else out here using speex successfully? |
| 05:43.54 | Sedorox | Nein here |
| 05:44.06 | |Vulture| | Sedorox: do you have a lot of inter-office dialing, like the office in CDN calling the office in US? |
| 05:44.42 | Sedorox | not right now... I mean we do conferences and stuff over it about once a month.. but we're looking to eventually have the tech support calls go anywhere and such |
| 05:44.53 | Sedorox | so right now.. no.. eventually.. yes |
| 05:45.18 | Sedorox | and inter-office is the most common right now.. since we only one have pots out in the US and one in cnd |
| 05:45.21 | |Vulture| | Sedorox: prolly cheapest to use a service like Nufone on outbound calls |
| 05:45.30 | nwhit | sedorox: yeah.. i ran into a library issue I think before |
| 05:45.32 | mishehu | damn, linphone causes asterisk to crash |
| 05:45.36 | Sedorox | well yea.. but I'm looking at the pipe I need to the internet to handle this... |
| 05:45.40 | Sedorox | nwhit: oo ok |
| 05:45.50 | Grooby | wazaaaaaa |
| 05:45.53 | |Vulture| | Sedorox: how many calls at once? |
| 05:45.58 | PyroSteve | asterisk wont read dtmf tones from a call that was initially made by a call file |
| 05:46.07 | Sedorox | maybe 3-5 on a good day... |
| 05:46.32 | _Vile | Bierdo, I have 3 lawsuits pending against Qwest atm, I would definitely say I need a better regulator |
| 05:46.33 | Sedorox | I've been thinking what you said.. frac. T1 |
| 05:46.43 | _Vile | but I have to go to bed |
| 05:46.49 | Sedorox | night _Vile |
| 05:46.58 | |Vulture| | Sedorox: yea frac t1, ilbc iax2 to voicepulse/nufone... your set |
| 05:47.02 | _Vile | g'night |
| 05:47.08 | Sedorox | yea |
| 05:47.10 | |Vulture| | got 6 lines on inbound and PLEANTY of bandwidth |
| 05:47.16 | Sedorox | do you think isdn would be good too? |
| 05:47.16 | VaHamish | Hey, question from the simpleton here.. ;-) |
| 05:47.20 | Sedorox | hehe |
| 05:47.27 | Sedorox | VaHamish: ok :-p |
| 05:48.03 | VaHamish | I've got my asterisk system set up with a X400p |
| 05:48.07 | Sedorox | the main point will probably come in, in CND.. so I'm not sure which would be cheaper. Frac T1 or ISDN... |
| 05:48.13 | Sedorox | ok |
| 05:48.17 | VaHamish | I think that's the card, the one that holds 4 modules. |
| 05:48.24 | VaHamish | I've got an FXO and an FXS.. |
| 05:48.33 | |Vulture| | Sedorox: for ilbc you only need like 150k of bandwidth for 6 calls |
| 05:48.40 | VaHamish | and it's configured correctly, I've got a hand set pluged into port 1, and it rings and I can dial with it. |
| 05:48.42 | Sedorox | kb or kB? |
| 05:48.48 | VaHamish | But now I want to bridge to an outgoing line. |
| 05:48.51 | |Vulture| | kilobytes |
| 05:48.54 | Sedorox | kk |
| 05:49.00 | VaHamish | so I've plugged a line from the wall into port 4. |
| 05:49.01 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 05:49.15 | |Vulture| | so like a DSL connection would work |
| 05:49.23 | VaHamish | I'm guessing I refer to that port as Zap/4, correct? |
| 05:49.38 | Sedorox | to dial out of.. yes |
| 05:49.52 | VaHamish | Is there a special wire I need for this, a cross over? |
| 05:49.53 | Sedorox | |Vulture|: is that total? so like 75k in and out? |
| 05:49.59 | |Vulture| | Sedorox: I got it... Frac T1, then split it into half 384k/6lines |
| 05:50.02 | VaHamish | cause I don't get a dial tone, when I dial out. |
| 05:50.10 | VaHamish | There is a dial tone on the line. |
| 05:50.21 | |Vulture| | Sedorox: http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 check out the case study |
| 05:50.24 | VaHamish | oh, you know maybe the cord I have in there is, a cross over.. |
| 05:50.25 | |Vulture| | its pretty accurate |
| 05:50.51 | Sedorox | VaHamish: if your using your phone in the FXS port to dial out of the FXO port.. * will provide the dialtone to the phone |
| 05:50.53 | Inv_arp | whats a frac t1 go for (south bellsouth mia) these days? |
| 05:50.57 | Sedorox | |Vulture|: thanks |
| 05:50.59 | |Vulture| | just make sure the pipe for VoIP isn't the same pipe for Inet in the office... it can get scary |
| 05:51.08 | Sedorox | trying to get different ideas and plans to see which is the best |
| 05:51.16 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:51.18 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 05:51.18 | |Vulture| | Inv_arp: bellsouth... they are expensive still |
| 05:51.34 | Sedorox | yea. I was thinking about splitting a T1 for that.. but I guess maybe cable for inet and a frac t1 for voice |
| 05:51.43 | Inv_arp | fract t1 == frame relay? |
| 05:51.44 | Sedorox | well.. voip anyway |
| 05:51.51 | Sedorox | Inv_arp: good question... |
| 05:51.56 | neopher | anyone know where the sccp phone firmware is located in cisco callmanager |
| 05:51.56 | |Vulture| | Inv_arp: around $800 for a full T1-Data and like $750 for a full PRI |
| 05:52.24 | |Vulture| | Inv_arp: and in my experience Fract T1s aren't much cheaper than full T1s |
| 05:52.37 | Sedorox | so it might be better to get a full :-p |
| 05:53.08 | *** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net) |
| 05:53.12 | Sedorox | still waiting on the one dude to call Shaw/Telus to see what the going rate for a T1 up there is |
| 05:53.22 | |Vulture| | Sedorox: yea I am not sure what kinda rates you get but ariel_ in miami was talking about a deal for $500 for 6 PRI voice channels and the rest data... thats a nice ass deal |
| 05:53.23 | Inv_arp | but i can get sdsl atabout same speed for $150 a month , T1 more stable i guess |
| 05:53.48 | |Vulture| | Inv_arp: a lot of times a SDSL connection is just a partial T1 |
| 05:53.59 | Sedorox | hehe |
| 05:54.07 | Inv_arp | ahhh |
| 05:54.20 | Sedorox | but thats voice in the US.. hehe :-p |
| 05:54.26 | Sedorox | allwell.. least I know of more options now |
| 05:54.44 | Sedorox | I have a chart here of the prices for a ISDN PRI but.. eh... |
| 05:54.50 | Sedorox | stil have to look it over.. kinda confusing |
| 05:55.01 | Sedorox | or seems REALLYY expensive |
| 05:55.08 | |Vulture| | Sedorox: is this in CDN? |
| 05:55.17 | Sedorox | yea |
| 05:55.18 | tuxinator_linux | Sedorox, I am may be able to help you figure it out |
| 05:55.26 | Sedorox | tuxinator_linux: mmm ookk |
| 05:55.48 | |Vulture| | they were just talking about this like 30min ago, how CDN has rape you in the ass fixed prices |
| 05:55.50 | Sedorox | |Vulture|: has helped a lot.. just getting different ideas right now to present... |
| 05:56.11 | Sedorox | hehe |
| 05:56.12 | Sedorox | yea |
| 05:56.20 | Sedorox | tuxinator_linux: I don't think I can do anything DCC... |
| 05:56.37 | *** join/#asterisk andrew` (~andrew@adsl-67-119-25-11.dsl.snfc21.pacbell.net) |
| 05:56.47 | Sedorox | wait.. should be able to.. public ipv6.. hmm.. anyway... |
| 05:57.11 | |Vulture| | whereas I see quotes from clecs being cheap (not always as reliable) and ilec being sometimes close to double the clecs |
| 05:57.22 | Sedorox | wow.. |
| 05:57.24 | tuxinator_linux | Sedorox: I am not sure how else to do a private chat |
| 05:57.29 | Sedorox | PM |
| 05:58.08 | Sedorox | I would like to have us be all VOIP.. and get termination from a voip provider.. |
| 05:58.16 | Inv_arp | tuxinator_linux: eg.. /msg person what are you wearing? :) |
| 05:58.37 | Sedorox | but I hear it would be cheaper if I got a data-only pipe... so I'm thinking maybe actually have it terminate in the states... with a data-only pipe.. |
| 05:58.42 | tuxinator_linux | thanks Inv |
| 05:58.43 | Sedorox | but with CND did's and such... |
| 05:58.49 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 05:59.46 | Beirdo | Sedorox: where in Canada are you looking for? |
| 05:59.58 | Sedorox | Umm.. Creston BC area... |
| 06:00.06 | Sedorox | I think... |
| 06:00.15 | Sedorox | hehe |
| 06:00.30 | bkw_ | w00t |
| 06:01.03 | shido6 | bkw_ u use skinny? |
| 06:01.05 | shido6 | 7920 |
| 06:01.08 | shido6 | 7902 |
| 06:01.08 | shido6 | ? |
| 06:01.18 | Beirdo | so that's 604? |
| 06:01.32 | Sedorox | area code? |
| 06:01.33 | Sedorox | 250... |
| 06:01.38 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 06:01.38 | Beirdo | 250? |
| 06:01.41 | Beirdo | interesting |
| 06:01.51 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-85-148.sentechsa.net) |
| 06:01.54 | Beirdo | 604 is the old one, I guess they split it? |
| 06:01.57 | bkw_ | shido6, no |
| 06:01.59 | Sedorox | dunno |
| 06:02.06 | bkw_ | i wish I was skinny |
| 06:02.07 | Sedorox | I know the land-line and cell is that |
| 06:02.19 | Sedorox | and its listed on most sites for international calling to canada... |
| 06:03.07 | neopher | bkw: do you happen to know where to look in CCM for fireware files, I have a 30 VIP that i need to upgrade to work with * |
| 06:03.08 | Beirdo | that might be tough to get VOIP terminated DID for |
| 06:03.20 | Beirdo | you may need to roll your own |
| 06:03.26 | Sedorox | hmmm |
| 06:03.33 | Beirdo | Vancouver, maybe |
| 06:03.42 | Sedorox | yea.. its close to there.. |
| 06:03.46 | Beirdo | Creston's kinda in the middle of nowhere |
| 06:03.51 | Sedorox | hehe yup :-p |
| 06:03.58 | Beirdo | Creston's a LONG way from Vancouver |
| 06:04.01 | Sedorox | I'm not sure if we're really worried about local calling |
| 06:04.06 | Beirdo | looks to be closer to Calgary :) |
| 06:04.16 | Sedorox | argh.. thats what I shoulda said. calgary.. |
| 06:04.17 | Sedorox | sorry |
| 06:04.19 | bkw_ | ok it makes and takes calls |
| 06:04.21 | bkw_ | thats a plus |
| 06:04.35 | Sedorox | I think we're mainly look for a Toll-free CND line.. but eh |
| 06:04.41 | Beirdo | Freeworldtel is based in Edmonton. |
| 06:05.17 | Sedorox | hmmm |
| 06:05.28 | Grooby | asterisk on ibook |
| 06:05.30 | Grooby | niiiicceeeee |
| 06:05.43 | Beirdo | direct.freeworldtel.com |
| 06:05.56 | Sedorox | more or less it comes down to us being a hosting provider and want to have say upto 10 incoming and say 10 outgoing (more realisticly.. 5 each way MAX) |
| 06:06.01 | Beirdo | no idea as to quality, etc |
| 06:06.09 | Sedorox | but with CND and US dids.. mainly toll free I would assume |
| 06:06.52 | Beirdo | bjohnson: you in? |
| 06:07.14 | Sedorox | Beirdo: looks like a good site.. |
| 06:07.23 | Beirdo | they list toll-free |
| 06:07.31 | Beirdo | dunno if they suck or not though :) |
| 06:07.45 | Sedorox | yea |
| 06:07.47 | Sedorox | well we'll find out |
| 06:07.50 | Sedorox | thanks for the link :) |
| 06:07.54 | Beirdo | but worth researching. |
| 06:07.54 | marc32344 | how many res dids can you load on a T1? average |
| 06:07.57 | Beirdo | no problem |
| 06:07.58 | Sedorox | brb |
| 06:08.10 | Sedorox | dids.. howevery many you want I would assume |
| 06:08.28 | marc32344 | no. without running into busy signals] |
| 06:08.30 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 06:08.57 | VaHamish | thanks a lot folks, I made real progress tonight... |
| 06:09.12 | Beirdo | marc32344: doing voice to the telco? 23 normal voice channels |
| 06:09.13 | VaHamish | I'm very psyched. |
| 06:09.17 | VaHamish | g'night |
| 06:09.48 | marc32344 | beirdo-- how many local customers/t1? |
| 06:10.12 | Sedorox | hehe |
| 06:10.19 | Beirdo | on a T1, you can use 23 voice calls at a time |
| 06:10.30 | Beirdo | assuming you are doing voice. |
| 06:10.36 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 06:10.51 | marc32344 | yes. but how many clients can you support, ievonage type |
| 06:11.00 | Beirdo | huh? |
| 06:11.03 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:11.11 | Nukemizer | I am tryin got get Zap/1-1 (the extension) to dial out using Zap/3-1 ( CO line 1) and Zap/2-1 to use zap/4-1 I can not seem get the right combination Does anyone have an example I could review |
| 06:11.26 | Nukemizer | trying to get |
| 06:11.49 | Beirdo | marc32344: are you running it in data mode, or voice mode? |
| 06:11.53 | marc32344 | voice |
| 06:12.04 | Beirdo | then 23 calls simultaneous max |
| 06:12.08 | Sedorox | hmmmmmm |
| 06:12.18 | Beirdo | how many more times do I have to say it? |
| 06:12.22 | marc32344 | how many mins can you do on a T1? max capacity is 1M. |
| 06:12.31 | |Vulture| | okay I am back now |
| 06:12.31 | Beirdo | mins? |
| 06:12.33 | Beirdo | huh? |
| 06:12.35 | |Vulture| | just testing some stuff :) |
| 06:12.36 | marc32344 | minutes |
| 06:12.48 | Beirdo | what are you talking about? |
| 06:13.15 | |Vulture| | on a T1 you can use 24 lines unless it is a PRI |
| 06:13.15 | |Vulture| | then it is 23 |
| 06:13.21 | marc32344 | how many calls minutes can you do/month on a T1 PRI? |
| 06:13.30 | Beirdo | |Vulture|: sorry, you are right there :) |
| 06:13.39 | Beirdo | how many minutes are there in a month? |
| 06:13.48 | tuxinator_linux | It's unmeeterd |
| 06:13.51 | tuxinator_linux | usually |
| 06:13.53 | shido6 | its highly unlikely you are going to FILL an entire T 24/7 |
| 06:13.54 | |Vulture| | marc32344: depends on your plan |
| 06:14.00 | |Vulture| | marc32344: most do a local area, then LD |
| 06:14.04 | Beirdo | you'd have to look at how the T1 is billed, but usually, it's unmetered last I heard |
| 06:14.16 | |Vulture| | unmetered LD? |
| 06:14.30 | Beirdo | no no |
| 06:14.38 | Sedorox | hehe, it isn't too bad... |
| 06:14.40 | Beirdo | I meant unmetered for local |
| 06:14.41 | Beirdo | <PROTECTED> |
| 06:14.43 | |Vulture| | haha yea I was like where the hell do you guys buy your PRIs |
| 06:14.44 | marc32344 | beirdo-- how about busy lines? |
| 06:14.49 | tuxinator_linux | Vulture, sorry came in the middle of that one |
| 06:14.59 | Beirdo | what do you mean busy lines? |
| 06:15.10 | marc32344 | 25 concurrent calls |
| 06:15.20 | |Vulture| | marc32344: you mean what happens if you fill 24 lines and someone calls in? |
| 06:15.25 | marc32344 | yes |
| 06:15.30 | Beirdo | you can't peg up a call if the thing's fully used |
| 06:15.58 | |Vulture| | marc32344: depends on you provider |
| 06:16.05 | |Vulture| | some will play "all circuits are busy" other you will get congestion |
| 06:16.14 | |Vulture| | I prefer "all circuits" |
| 06:16.25 | marc32344 | how many clients will you run on a single T1 PRI? |
| 06:16.36 | |Vulture| | me? |
| 06:16.48 | Beirdo | you asked the same damn thing last night did you not? :) |
| 06:17.02 | marc32344 | no answer |
| 06:17.13 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
| 06:17.16 | marc32344 | like this. |
| 06:17.20 | |Vulture| | lol I was so zoned out last night... damn flu season |
| 06:17.29 | Beirdo | you would have to determine your call patterns, etc |
| 06:17.29 | Sedorox | lol |
| 06:17.52 | Beirdo | once you've done that, you can figure out how many T1s you need to support the calls |
| 06:17.59 | Beirdo | just like you were told the other night |
| 06:18.01 | |Vulture| | marc32344: I will be running 7 offices off a PRI if I can get it done right |
| 06:18.07 | *** part/#asterisk Ahewes (~rsb@adsl-69-107-39-45.dsl.pltn13.pacbell.net) |
| 06:18.50 | |Vulture| | but they are small offices, ~5 lines each, and in different time zones, so peek hours change to lighten load |
| 06:19.34 | marc32344 | 200? |
| 06:20.07 | Beirdo | WTF? |
| 06:20.17 | Beirdo | are you going to listen any time soon? |
| 06:20.39 | Beirdo | figure out the call patterns, and from that determine how much you can put on the T1 |
| 06:20.40 | *** join/#asterisk Fanguin (~Fanguin@p50818948.dip0.t-ipconnect.de) |
| 06:21.04 | Essobi | Anyone using a Cisco 2XXX-5XXX router terminating into an * box? |
| 06:21.06 | Beirdo | if you have 200 callers who all want to be on the phone for hours at a time, it ain't gonna work |
| 06:21.14 | shido6 | Essobi |
| 06:21.15 | shido6 | YES |
| 06:21.26 | shido6 | I just helped set one up again |
| 06:21.27 | Essobi | I can't seem to get asterisk to happily accept my 5400 as a peer. |
| 06:21.29 | Beirdo | if they do very little calling, short calls, etc, then maybe |
| 06:21.31 | marc32344 | there is a number. it averages out |
| 06:21.39 | Beirdo | right |
| 06:21.41 | Essobi | I can call out, but my sip.conf entry doesn't seem to be right. |
| 06:21.44 | shido6 | if you can wait until the morning I can get the guy to assist |
| 06:21.44 | Sedorox | I have a 2501 but it just sits here.... |
| 06:22.03 | Essobi | Nah, I was just hoping to get a sip.conf example entry |
| 06:22.03 | Beirdo | so go from the average and worst case call load.... and determine how many T1s you need to support it |
| 06:22.54 | Essobi | IT's wierd.. I can call my 5400 from * and it talks just fine.. |
| 06:23.13 | Essobi | my 5400 drops a coll on my * box and it says it's from an unauthed peer |
| 06:23.23 | |Vulture| | brb |
| 06:25.36 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 06:25.36 | *** mode/#asterisk [+o twisted] by ChanServ |
| 06:26.36 | Essobi | wierd thing is, it'll land in the [default] if I turn on guest mode. |
| 06:26.42 | Essobi | but it won't land on the peer.. |
| 06:27.04 | Essobi | Hey twisted.. you got an example peer entry for a cisco sip router? |
| 06:29.40 | shido6 | http://www.voip-info.org/wiki-Asterisk+cisco+FXO |
| 06:29.48 | shido6 | thats what I used |
| 06:29.54 | Essobi | I've got it setup exactly like that one. |
| 06:30.05 | Essobi | and it's landing in [default] only when guest it on |
| 06:30.07 | Essobi | is |
| 06:31.31 | Essobi | chan_sip.c:8017 handle_request: Failed to authenticate user |
| 06:31.41 | shido6 | err K |
| 06:31.44 | shido6 | u cant register |
| 06:31.45 | Essobi | heh |
| 06:31.51 | Essobi | umm |
| 06:31.53 | Essobi | it's a peer |
| 06:31.56 | shido6 | right |
| 06:32.05 | |Vulture| | k back |
| 06:32.08 | Essobi | I thought peers didn't register |
| 06:32.14 | shido6 | right |
| 06:32.15 | shido6 | they dont |
| 06:32.17 | Essobi | :| |
| 06:32.27 | |Vulture| | friend |
| 06:32.27 | Essobi | So wtf is it trying to register for? |
| 06:32.37 | shido6 | something is wrong in your setup :) |
| 06:32.40 | Essobi | :P |
| 06:32.41 | |Vulture| | must have register turned on |
| 06:33.10 | shido6 | bleh |
| 06:33.11 | shido6 | going to bed |
| 06:33.16 | |Vulture| | night shido6 |
| 06:34.43 | Essobi | |Vulture| I don't. |
| 06:34.50 | Essobi | that's what's wierd |
| 06:34.55 | Essobi | this shit worked with 1.0 |
| 06:35.08 | Essobi | -head broxored it. :) |
| 06:35.11 | |Vulture| | oh you changed nothing.. just version? |
| 06:35.14 | |Vulture| | 1.0.5? |
| 06:35.30 | Essobi | yup |
| 06:35.32 | Essobi | Feb 24 01:36:47 NOTICE[18715]: chan_sip.c:8017 handle_request: Failed to authenticate user <sip:1231\ |
| 06:35.32 | Essobi | 231234@my.ip.5400.ip> |
| 06:35.39 | Essobi | It's pissing me off. Heh. |
| 06:36.40 | nwhit | YEAH!!! it works! |
| 06:36.43 | Essobi | http://www.pastebin.com/245731 <-- My current sip.conf entry for the peer. |
| 06:37.22 | nwhit | ouch |
| 06:37.31 | Essobi | Mehe. |
| 06:37.51 | Essobi | Quit trying to brighten up my gloomy situation. ;) |
| 06:37.54 | nwhit | i've been dealing with a problem with the voicemail system for days... |
| 06:38.12 | nwhit | and it works now... I can go home and get some sleep |
| 06:38.19 | Essobi | ahh |
| 06:38.20 | Essobi | :) |
| 06:38.26 | Essobi | Those simple times are nice. |
| 06:40.00 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.219.246) |
| 06:40.40 | Essobi | Pssh, I really don't want all my inbound sip peers to land in one context and have to send them to the appropriate places. That's ghetto. |
| 06:41.05 | Sedorox | feels like my laptop is going to fry itself :( |
| 06:41.42 | Essobi | Quit sit it on the cisco dust and singe it. |
| 06:41.46 | Essobi | quick even. |
| 06:42.19 | Sedorox | HDD's running ar 140F |
| 06:42.23 | Sedorox | thats not good... |
| 06:43.42 | Essobi | wow |
| 06:43.44 | Essobi | yea |
| 06:43.49 | Essobi | clean the fan out |
| 06:43.57 | Essobi | and shut off seti |
| 06:44.26 | Sedorox | lol |
| 06:44.31 | Sedorox | its my laptop :( |
| 06:44.44 | Sedorox | the HDD's are in the front.. and for some reason don't have any vent. |
| 06:44.52 | Sedorox | and stupid me got the 7200rpm drives and not the 54000 |
| 06:44.53 | Sedorox | 5400 |
| 06:45.15 | Sedorox | and I had a natural KB sitting on top so I could use that.. stopping airflow over the palm rest.. yea.. bad idea |
| 06:45.23 | |Vulture| | ...sleeping pill time :) |
| 06:45.50 | Sedorox | lol |
| 06:45.52 | Sedorox | night :-p |
| 06:45.59 | |Vulture| | takes a bit to kick in |
| 06:46.14 | |Vulture| | damn flu messed me up and i got to take these to sleep |
| 06:46.18 | Sedorox | I should probably head in :-/ |
| 06:46.21 | Sedorox | hmm |
| 06:46.43 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 06:46.47 | marc32344 | is the govarion cards identical to digium? |
| 06:46.54 | Essobi | FFs this is stupid. |
| 06:46.59 | shaZwaz | hi ppl |
| 06:47.30 | Sedorox | looks like GSM will be my best bet for iax2 trunking |
| 06:47.33 | Essobi | Sedorox yea bad idea |
| 06:47.48 | Sedorox | just wish I didn't get the 'machine gun affect' when calling across it with the phones I have |
| 06:47.56 | Essobi | nice |
| 06:48.17 | Essobi | w a a a t t t a a a r r r e y y y yoooo u u u u do dooo o o o oing |
| 06:48.20 | Essobi | just enunciate |
| 06:48.22 | |Vulture| | Sedorox: why not ilbc? |
| 06:48.23 | Essobi | you'll be fine |
| 06:48.25 | Essobi | :) |
| 06:48.34 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 06:48.35 | Sedorox | lol |
| 06:48.44 | |Vulture| | <-- ilbc man |
| 06:48.49 | Sedorox | actually you can't hear anything besides lots o clicking |
| 06:48.50 | Sedorox | of* |
| 06:48.55 | Sedorox | I noticed :-p |
| 06:49.00 | Sedorox | I haven't tried it yet.. I should |
| 06:49.01 | Essobi | damn |
| 06:49.12 | Sedorox | had the problem before.. codec stuff... |
| 06:49.27 | Essobi | Piece of shite cost more then my car. |
| 06:49.28 | Essobi | GRRR. |
| 06:49.31 | |Vulture| | hahaha |
| 06:49.37 | Sedorox | lol |
| 06:49.52 | Sedorox | been doing that lately |
| 06:50.04 | Sedorox | "yea.. I would like that... wait. I can buy a used car for that.. nvm" |
| 06:50.14 | Essobi | WTF is * tryng to register my 5400 as a sip friend instead of a peer. |
| 06:50.24 | Essobi | GRRR |
| 06:50.25 | Sedorox | check your sip.conf? |
| 06:50.31 | Essobi | Ayup. |
| 06:50.46 | Essobi | <Essobi> http://www.pastebin.com/245731 <-- My current sip.conf entry for the peer. |
| 06:51.09 | Essobi | I don't see anything wrong. I've tweeked and tweeked on it to no avail. |
| 06:51.16 | Essobi | I can dial out the peer all day long. |
| 06:51.19 | Sedorox | do you see it listed in 'sip show peers'? |
| 06:51.23 | Essobi | Ayup. |
| 06:51.26 | Essobi | :| |
| 06:51.38 | Sedorox | so which way doesn't work? * -> 5400 or 5400 -> * |
| 06:52.01 | Essobi | I even get the chan_sip.c:8017 handle_request: Failed to authenticate user during sip debug peer as5400-2 |
| 06:52.10 | Essobi | * -> 5400 works great |
| 06:52.20 | Essobi | 5400 -> * gives the above |
| 06:52.32 | Sedorox | is that all your seeing? |
| 06:52.36 | Essobi | only way I can get it to work is to use the allow guest thing |
| 06:52.54 | Essobi | then land all of the 5X00s into a default context |
| 06:52.58 | Essobi | which sucks |
| 06:53.10 | Sedorox | sounds like a config issue on the 5400 |
| 06:53.26 | Essobi | I copied the FXO settings above. |
| 06:53.27 | Essobi | :| |
| 06:53.28 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 06:53.36 | Essobi | from voip-info |
| 06:53.52 | Essobi | pisser is it WAS working.. and -head broxored it. |
| 06:53.54 | Essobi | GRRR. |
| 06:54.03 | Sedorox | try adding username=<something> |
| 06:54.10 | Sedorox | and using that username in the register from the 5400 |
| 06:54.26 | Sedorox | -head broke a lot of sip stuff for me |
| 06:54.30 | Sedorox | hence why I went back to 1.0.5 |
| 06:54.41 | Essobi | No shit? :| |
| 06:54.44 | shaZwaz | back to 1.0.5 ? |
| 06:54.50 | Sedorox | from cvs-head |
| 06:54.55 | shaZwaz | sounds like back to future |
| 06:54.57 | Sedorox | we.. whatever the latest is I pulled... |
| 06:54.59 | Sedorox | lol |
| 06:55.07 | Essobi | I'm thinking about it.. I really want to use the new spandsp thou. |
| 06:55.19 | Sedorox | dunno what to tell ya |
| 06:55.20 | Essobi | and I heard it would only build on -head. |
| 06:55.27 | Essobi | *SIGH* |
| 06:55.47 | shaZwaz | spandsp :-/ |
| 06:56.08 | Essobi | Well damnit. |
| 06:56.17 | Essobi | someone fix -head |
| 06:56.27 | Sedorox | lol |
| 06:58.07 | Sedorox | how does this sound: |
| 06:58.27 | Sedorox | for a US50/Canada Toll Free Number: $2.49/month and $.03/min |
| 06:58.42 | |Vulture| | nice |
| 06:58.48 | Sedorox | and either... |
| 06:59.08 | |Vulture| | thats a great deal, but I think nufone does offer it for .02/min but more /month |
| 06:59.58 | Sedorox | $2.49/month + $.02/min for US did.., $4.50/month+$0.02/min for CND did, (Unlimited incoming) |
| 07:00.03 | viLeR | Somebody have a Iptables script that works with Ata Behind Nat ? |
| 07:00.15 | Sedorox | or $10.49/month for unlimited |
| 07:00.20 | Sedorox | with unlimited incoming |
| 07:00.28 | Inv_arp | viLeR: a scipt to do what? f |
| 07:00.38 | Sedorox | on the above planes.. unlimited inbound channels |
| 07:00.46 | Sedorox | not minutes |
| 07:01.10 | Sedorox | This is FreeWorldTel someone pointed me to |
| 07:01.14 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:01.16 | Sedorox | doesn't seem too bad |
| 07:01.17 | pashah | hi! |
| 07:01.27 | pashah | Sedorox: url? |
| 07:02.05 | Sedorox | http://direct.freeworldtel.com/ |
| 07:02.25 | Sedorox | apparently thats for IP solution providers and stuff.. but I think I could use it for the biz I'm doing.. dunnon |
| 07:02.43 | viLeR | Inv_arp: for rtp stream |
| 07:02.46 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 07:02.58 | marc32344 | what is the difference between govarion and digium cards? |
| 07:02.59 | Beirdo | I'm sure unlimited really isn't unlimited |
| 07:03.33 | Sedorox | Unlimited Plan DIDs: |
| 07:03.33 | Sedorox | Unlimited Rate: $10.49 per month with unlimited incoming minutes. |
| 07:03.43 | |Vulture| | Sedorox: nice prices |
| 07:03.51 | Sedorox | yea... |
| 07:03.54 | Beirdo | ahhh, incoming |
| 07:03.56 | Beirdo | duh |
| 07:04.21 | Sedorox | yea.. the other plans.. like per-minute ones |
| 07:04.27 | Sedorox | had unlimited channels |
| 07:04.29 | Sedorox | not minutes |
| 07:04.30 | Sedorox | :-p |
| 07:06.38 | *** part/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 07:09.26 | Beirdo | Sedorox: where are the plans shown? |
| 07:12.25 | *** join/#asterisk scythelx (~wow@pc-24-151-59-224.newt1.ct.charter.com) |
| 07:12.31 | scythelx | hello all, does iaxtel work still? |
| 07:13.01 | RestLessGemini | well yes! i guess |
| 07:13.16 | scythelx | hmm strange |
| 07:13.22 | elric | decent IAX hard phones are hard to find though |
| 07:13.32 | scythelx | well |
| 07:13.34 | scythelx | Executing Dial("SIP/bethos.sales-724f", "IAX2/kknott:biff2baff@iaxtel.com/17004286161@iaxtel") in new stack |
| 07:13.54 | Beirdo | way to give everyone your password |
| 07:13.56 | Sedorox | Beirdo: I signed up for a account... its free to sign up |
| 07:13.59 | RestLessGemini | lol |
| 07:14.01 | Sedorox | lol |
| 07:14.10 | scythelx | i dont care |
| 07:14.12 | scythelx | im just testing it |
| 07:14.12 | Sedorox | sounds like me.... |
| 07:14.15 | Beirdo | and it shows the plans after you login? |
| 07:14.16 | scythelx | Auto fallthrough, channel 'SIP/bethos.sales-c31b' status is 'CONGESTION' |
| 07:14.26 | Sedorox | # Quotes: * |
| 07:14.26 | Sedorox | * Jul 11 23:20:14 <worth> Linux kernel: Free, # |
| 07:14.26 | Sedorox | # GNOME Desktop: Free, X-chat: Free, Posting a * |
| 07:14.26 | Sedorox | * screenshot with your IRC password in it so everyone # |
| 07:14.26 | Sedorox | # can see it: Priceless * |
| 07:14.26 | RestLessGemini | yeah he can signup for another.. its FREE |
| 07:14.27 | Sedorox | * (Yes, he's referring to Me) # |
| 07:14.45 | Sedorox | Beirdo: when you go to the did's page, it tell you under them the price |
| 07:15.08 | Beirdo | ahhh, after you sign up |
| 07:15.10 | Sedorox | then you can add funds or what not to get the dids.. |
| 07:15.10 | Sedorox | yea |
| 07:15.24 | Sedorox | and its iax too :-p |
| 07:15.25 | scythelx | would ser somehow mess it up? its sip phone -> ser -> asterisk -> iaxtel |
| 07:15.28 | Beirdo | do they list the prices for 800 in tehre? |
| 07:15.34 | Sedorox | yea |
| 07:15.34 | Sedorox | here |
| 07:15.54 | Sedorox | Random Toll-Free DID: |
| 07:15.54 | Sedorox | Random Toll-Free Rates: |
| 07:15.54 | Sedorox | US48: $1.50/per month, and 2.2?/min. |
| 07:15.54 | Sedorox | US50/Canada: $2.49/month, and 3?/min. |
| 07:15.55 | Beirdo | in particular, US50/CDN toll free :) |
| 07:16.02 | Beirdo | wow |
| 07:16.07 | Beirdo | that's pretty good |
| 07:16.09 | Sedorox | For vanity toll-free numbers and 1-900 services, please contact support for details. |
| 07:16.12 | Sedorox | hehe |
| 07:16.27 | Sedorox | yea.. I forget who told me about it.. I'll look above... |
| 07:16.31 | Beirdo | I could care less what toll free number I'd get |
| 07:16.33 | Sedorox | but thats the one thing with me is CND dids.. |
| 07:16.36 | Sedorox | yea.. same here |
| 07:16.37 | Beirdo | I did :) |
| 07:16.39 | Beirdo | hehe |
| 07:16.41 | Sedorox | oh |
| 07:16.42 | Sedorox | lol |
| 07:16.43 | Sedorox | :-p |
| 07:16.44 | Sedorox | thought so |
| 07:16.49 | Beirdo | but I didn't know the toll-free rates |
| 07:17.09 | Sedorox | yea.. if you signup (it is free) it tells you under the did selection |
| 07:17.16 | Sedorox | and they have a pretty good selection too |
| 07:17.52 | Beirdo | is that US$ or CDN$? (not that there's much difference these days) |
| 07:18.01 | Sedorox | US I'm sure |
| 07:18.10 | Beirdo | either way |
| 07:18.10 | Sedorox | and apparently their server is in TX |
| 07:18.11 | Beirdo | nice |
| 07:18.27 | Sedorox | because when you select for the iax registration.. it has to pick th server.. only one is in TX |
| 07:18.47 | Beirdo | hehe |
| 07:18.52 | Sedorox | but.. can't do iax register till funds are in your account :-p |
| 07:19.01 | Beirdo | :) |
| 07:19.36 | Beirdo | so once I get paid... I'll be setting up nufone and them, likely |
| 07:19.56 | Sedorox | hehe |
| 07:20.02 | Sedorox | I'm gonna talk to the owner about this |
| 07:20.05 | Sedorox | see what we can do... |
| 07:20.15 | Beirdo | owner of? |
| 07:20.19 | Sedorox | do some calcs from the page |Vulture| sent me on our bandwitdh costs |
| 07:20.24 | Sedorox | the company I work with |
| 07:20.28 | Sedorox | thats doing all this |
| 07:20.28 | Sedorox | hehe |
| 07:20.29 | Beirdo | ahhh |
| 07:20.55 | Beirdo | for me it'd just be for me :) |
| 07:20.55 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
| 07:21.06 | Sedorox | yea |
| 07:21.12 | Sedorox | I would love to talk my mom into it for at home |
| 07:21.17 | Sedorox | get a small asterisk box setup |
| 07:21.26 | Sedorox | and get a fxs card for the cordless's |
| 07:21.35 | Sedorox | errr... TDM400 |
| 07:21.40 | Beirdo | yeah |
| 07:21.45 | Beirdo | TDM22P |
| 07:21.48 | Sedorox | w/e |
| 07:21.48 | Beirdo | B |
| 07:21.48 | Sedorox | :-p |
| 07:21.50 | Beirdo | dammit |
| 07:21.51 | Beirdo | :) |
| 07:22.03 | Sedorox | or maybe a two port sipura |
| 07:22.05 | Sedorox | but anyway |
| 07:22.05 | Beirdo | 2 FXO, 2 FXS |
| 07:22.13 | Beirdo | and then a sipura or two :) |
| 07:22.15 | Beirdo | heh |
| 07:22.16 | Sedorox | well I would only need fxs's at that poiint.. but yea |
| 07:22.27 | Beirdo | and a WiSIP would be so sweet |
| 07:22.38 | Sedorox | have to do calc first on BW.. since we only have cable |
| 07:22.40 | Sedorox | hehe, yes it would |
| 07:22.46 | Beirdo | although they likely suck rocks |
| 07:22.53 | Sedorox | why's that? |
| 07:23.06 | Beirdo | I dunno, cheaply made or something |
| 07:23.12 | Beirdo | I'd have to try one :) |
| 07:23.18 | Sedorox | lol |
| 07:24.10 | Sedorox | I would like to get signed up now and try this out.. but eh... I don't have the money.. and I'm not doing anything w/o the OK |
| 07:24.10 | Sedorox | :-p |
| 07:24.48 | Sedorox | we pay I think $8/gig a month for BW... so... |
| 07:24.52 | Beirdo | well, I'm in the unenviable position of not having been paid for 7 weeks |
| 07:24.54 | Beirdo | grrr |
| 07:25.02 | Sedorox | :(:(:( |
| 07:25.09 | Sedorox | I don't have a job.. so.. it sucks. big time |
| 07:25.17 | Beirdo | once I get paid, I'll be spending |
| 07:25.18 | Beirdo | :) |
| 07:25.21 | Sedorox | lol |
| 07:25.32 | Sedorox | once I start getting paid again (waiting to hear back from a few places) |
| 07:25.34 | Sedorox | I'll be saving |
| 07:25.36 | Beirdo | I put $10 into voipjet for the meantime |
| 07:25.41 | Sedorox | think my car is starting to go.. so.. |
| 07:25.43 | Sedorox | hehe |
| 07:26.03 | Beirdo | needed something for now |
| 07:26.08 | Sedorox | yup |
| 07:26.12 | Sedorox | well I got my cellphone |
| 07:26.20 | Beirdo | and although their service is somewhat crappy at times, it's cheeeep |
| 07:26.37 | Sedorox | I really wish they made one of those jacks that you can plug your cellphone into and it rings a housephone for the i730.. would hook it into asterisk |
| 07:26.39 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 07:26.40 | Sedorox | hehe |
| 07:26.51 | Beirdo | hehe |
| 07:26.51 | Sedorox | wb |
| 07:27.09 | Beirdo | I have been half considering pricing a cellular fixed station |
| 07:27.10 | Beirdo | hehe |
| 07:27.12 | Sedorox | they have it for like every model.. BUT the nextel ones... |
| 07:27.16 | Sedorox | lol |
| 07:27.29 | Sedorox | of the motorola line anyway |
| 07:27.45 | Beirdo | I think my mike phone is an i730 too |
| 07:27.59 | Sedorox | Canada.. probably |
| 07:28.08 | Beirdo | yes |
| 07:28.17 | Beirdo | same thing as Neztel |
| 07:28.28 | Sedorox | is it silver with a color screen, where the display it at the top, near the ant, not at the bottom? |
| 07:28.35 | Beirdo | yep |
| 07:28.43 | Beirdo | company phone |
| 07:28.45 | Sedorox | yea.. I know.. was trying to talk my friend into getting it hwen I got nextel.. because you can use Direct Connect accross |
| 07:28.50 | Sedorox | yea.. i730 |
| 07:28.53 | Beirdo | great for 2-way :) |
| 07:28.54 | Sedorox | mine's black.. but same phone |
| 07:28.57 | Sedorox | I love it... |
| 07:29.12 | Sedorox | I got it because of my g/f having netel (parents family plan) and my church... |
| 07:29.27 | Sedorox | I use like 100-200mins a month on the DC |
| 07:29.27 | Beirdo | ahh |
| 07:29.49 | *** join/#asterisk msupino (~msupino@gateway.sd.com) |
| 07:30.11 | Sedorox | don't have too much reception problems either.. 'cept for here in the dorm. so I got one of those car magnet ant's and stick it out my window. and all is good |
| 07:30.20 | Beirdo | hehe |
| 07:30.21 | *** join/#asterisk Graphikos (~Graphikos@71-32-6-49.spkn.qwest.net) |
| 07:30.30 | Beirdo | it works pretty much everywhere here |
| 07:30.38 | Graphikos | is GSM a crappy codec or what? |
| 07:30.39 | Sedorox | the concrete walls don't help.. when the main station is on the other side of the building I'm on... |
| 07:30.46 | Beirdo | except in the elevator in the condo building at home |
| 07:30.46 | Sedorox | I don't think so Graphikos |
| 07:30.50 | Sedorox | well yea... |
| 07:30.56 | Graphikos | I didn't think so either.. |
| 07:31.01 | Beirdo | works great in the elevator at work |
| 07:31.09 | Graphikos | haveing quality issues between locations... |
| 07:32.15 | Sedorox | Graphikos: could be other things like BW too.... try a different codec and see if you have the same problem |
| 07:32.45 | Graphikos | hard to think its BW... and we did try a ulaw codec... *siigh* |
| 07:33.33 | Sedorox | don't use ulaw |
| 07:33.38 | Sedorox | thats high BW consuming :-;p |
| 07:33.47 | Graphikos | well it actually sounded better than GSM |
| 07:34.05 | Sedorox | wel yea.. it will... |
| 07:34.17 | Graphikos | its probably our method... |
| 07:34.19 | Sedorox | try ILBC... |
| 07:34.20 | Sedorox | ulaw... |
| 07:34.22 | Sedorox | er |
| 07:34.23 | Sedorox | alaw |
| 07:34.35 | Graphikos | doing this thru a VPN.. which we should probably throw in another PBX |
| 07:34.42 | Sedorox | <PROTECTED> |
| 07:36.00 | Graphikos | oh well.. figgure it out eventually I guess... |
| 07:36.48 | Sedorox | just try different ones and see what works best for you.. thats my suggestion anyway |
| 07:36.58 | Sedorox | I'm sure someone else will say otherwise |
| 07:37.40 | Graphikos | thanks. ;) |
| 07:38.31 | Sedorox | yup |
| 07:48.53 | Sedorox | turning in |
| 07:48.53 | Sedorox | night |
| 07:48.59 | Graphikos | nite |
| 07:55.36 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
| 07:55.43 | kamran | hi |
| 07:57.13 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 07:57.52 | kamran | any one know how to call one application from other application |
| 07:58.33 | kamran | <PROTECTED> |
| 07:58.33 | kamran | > DialApp = pbx_findapp("Dial"); |
| 07:58.33 | kamran | > |
| 07:58.33 | kamran | > int ret=0; |
| 07:58.33 | kamran | > if (app) |
| 07:58.34 | kamran | > { |
| 07:58.35 | kamran | > ret = pbx_exec(chan, DialApp, "SIP/2000", 1); |
| 07:58.37 | kamran | > } |
| 07:58.39 | kamran | > else |
| 07:58.41 | kamran | > { |
| 07:58.43 | kamran | > } |
| 08:04.12 | kamran | hello developers |
| 08:05.18 | libpcp | hi guys |
| 08:08.55 | libpcp | if im installing TE405P on my asterisk, do i need a working ISDN line to my ISDN carrier before I can test the zaptel and zapata.conf? |
| 08:09.23 | kamran | is there any developer on this list |
| 08:09.33 | RestLessGemini | libpcp: No. i dont think so |
| 08:09.51 | RestLessGemini | just make zaptel and zapata.conf entries and you are done |
| 08:12.11 | libpcp | RestLessGemini: eventhough i dont have a real line of ISDN? |
| 08:12.49 | libpcp | i tried to configure zaptel and zapate.conf but when i tried to start my asterisk i got an error: Feb 24 16:19:02 WARNING[8663]: Ignoring switchtype |
| 08:13.00 | libpcp | Feb 24 16:19:02 ERROR[8663]: Unknown signalling method 'pri_cpe' |
| 08:14.10 | RestLessGemini | well yeah .. actually zaptel and zapata.conf only care for TE405P, i've never installed one but its default behaviour .. if i am not mistaking |
| 08:15.15 | libpcp | RestLessGemini: so i really need to have a working line of ISDN before I could use the config of zapata and zaptel right? |
| 08:16.01 | RestLessGemini | well take a look here http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+config+zapata.conf&diff=12 |
| 08:17.50 | TheEmperor | hello, can anyone tell me how to get callerid show to me on my softphone when i am being called? :) |
| 08:18.07 | *** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 08:20.13 | RestLessGemini | libpcp : also check switchtypes |
| 08:20.16 | RestLessGemini | -; Switchtype: Only used for PRI. |
| 08:20.16 | RestLessGemini | -; |
| 08:20.16 | RestLessGemini | -; national: National ISDN 2 (default) |
| 08:20.16 | RestLessGemini | -; dms100: Nortel DMS100 |
| 08:20.16 | RestLessGemini | -; 4ess: AT&T 4ESS |
| 08:20.17 | RestLessGemini | -; 5ess: Lucent 5ESS |
| 08:20.19 | RestLessGemini | -; euroisdn: EuroISDN |
| 08:20.21 | RestLessGemini | -; ni1: Old National ISDN 1 |
| 08:20.35 | RestLessGemini | and |
| 08:20.37 | RestLessGemini | -; PRI Dialplan: Only RARELY used for PRI. |
| 08:20.38 | RestLessGemini | -; |
| 08:20.38 | RestLessGemini | -; unknown: Unknown |
| 08:20.38 | RestLessGemini | -; private: Private ISDN |
| 08:20.38 | RestLessGemini | -; local: Local ISDN |
| 08:20.38 | RestLessGemini | -; national: National ISDN |
| 08:20.40 | RestLessGemini | -; international: International ISDN |
| 08:20.42 | RestLessGemini | -; |
| 08:20.44 | RestLessGemini | -;pridialplan=national |
| 08:21.09 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 08:22.10 | libpcp | yeah i have that settings |
| 08:22.18 | libpcp | im using euroisdn |
| 08:24.30 | RestLessGemini | signalling=pri_net |
| 08:24.44 | RestLessGemini | brb |
| 08:25.36 | RestLessGemini | you are also see this doc http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x842.html |
| 08:25.37 | RestLessGemini | brb |
| 08:40.48 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:42.31 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:42.40 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:51.18 | RoyK | SIP/1000052-fad1<ZOMBIE> (queue-ks s 1 ) Up Queue briiz-customerservice|t|||0 |
| 08:51.19 | RoyK | wtf? |
| 08:51.27 | shaZwaz | hi RoyK |
| 08:51.59 | RoyK | hi |
| 08:52.40 | *** join/#asterisk JerJer (~JerJer@dsl-106-170.che.centurytel.net) |
| 08:53.00 | JerJer | is it just me or is slashdot's site all screwy? |
| 08:53.02 | *** join/#asterisk gr0mit (~gr0mit@router1.txrx.org.uk) |
| 08:54.08 | *** join/#asterisk bsenicar (~bsenicar@BSN-77-186-131.dsl.siol.net) |
| 08:54.47 | RoyK | JerJer: fsckedup from here |
| 08:55.05 | RoyK | possibly slashdotted |
| 08:55.35 | *** join/#asterisk e3eli3h (~Chris@static-np1-5.cytanet.com.cy) |
| 09:03.29 | *** join/#asterisk e3eli3h (~Chris@static-np1-5.cytanet.com.cy) |
| 09:05.47 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 09:07.57 | *** join/#asterisk amer (~aaa@203.99.60.27) |
| 09:08.20 | amer | ok here is what I have |
| 09:08.54 | amer | SIP proxy --------- Asterisk------------ SoftSwitch-------TDM |
| 09:10.28 | amer | When I get a call from SIP proxy I can see that the from field in SIP header is correct but when asterisk fwds the call to the SSW it changes the from field and outs "asterisk" instead of the actual user who made the call |
| 09:10.37 | amer | how can I fix this? |
| 09:12.08 | amer | SIP proxy from: "1408XXXXXXX" <sip"1408XXXXXXX@x.x.x.x>;tag=bla bla bla |
| 09:12.08 | RoyK | SetCallerID? |
| 09:12.09 | RoyK | :P |
| 09:12.20 | amer | that doesn't work |
| 09:13.07 | amer | Asterisk from: ""1408XXXXXXX" <sip"asterisk@x.x.x.a>;tag bla bla |
| 09:14.03 | amer | my softswitch gets confused and sets the callerID to asterisk and here callerName is not supported so I get 00000 on Landlines and cell phones |
| 09:14.39 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 09:15.40 | amer | have I made myself clear |
| 09:16.11 | RoyK | I don't know, sorry |
| 09:16.19 | amer | np |
| 09:20.14 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:20.59 | shaZwaz | anyone tried the latest cvs head ? |
| 09:25.10 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 09:26.33 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 09:26.44 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:34.11 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 09:35.16 | *** join/#asterisk LenzX (~lenz-ml@213-92-107-83.f5.ngi.it) |
| 09:36.58 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
| 09:37.11 | pashah | shaZwaz: somebody had problems today with it |
| 09:39.37 | LenzX | hello, I need somebody who can call me via IAX or FWD - I changed the setup and I am not sure everything still works :-) |
| 09:42.03 | xpasha | anybody could say why I have this? -> |
| 09:42.03 | xpasha | <PROTECTED> |
| 09:42.04 | xpasha | Jan 26 07:31:39 WARNING[16579]: chan_zap.c:9608 setup_zap: Ignoring faxdetection |
| 09:42.04 | xpasha | Jan 26 07:31:39 ERROR[16579]: chan_zap.c:9429 setup_zap: Unknown signalling method 'pri_cpe' |
| 09:42.16 | xpasha | libpri is compiled and installed |
| 09:44.02 | shaZwaz | pashah: what sort of ? |
| 09:46.53 | pashah | shaZwaz: do not remember |
| 09:46.56 | Zeeek | LenzX you can call yourself on FWD or use the web phone booth have it call you |
| 09:47.30 | amer | SIP proxy --------- Asterisk------------ SoftSwitch-------TDM |
| 09:47.33 | amer | my softswitch gets confused and sets the callerID to asterisk and here callerName is not supported so I get 00000 on Landlines and cell phones |
| 09:47.33 | shaZwaz | Zeeek there is one Call me serivce too |
| 09:47.39 | amer | When I get a call from SIP proxy I can see that the from field in SIP header is correct but when asterisk fwds the call to the SSW it changes the from field and outs "asterisk" instead of the actual user who made the call |
| 09:48.01 | amer | SIP proxy from: "1408XXXXXXX" <sip"1408XXXXXXX@x.x.x.x>;tag=bla bla bla |
| 09:48.04 | amer | Asterisk from: ""1408XXXXXXX" <sip"asterisk@x.x.x.a>;tag bla bla |
| 09:48.39 | shaZwaz | pashah: wanna know if it is related to jitter buffer |
| 09:48.46 | shaZwaz | anyway thanks |
| 09:48.56 | Zeeek | that's the "phone booth" |
| 09:49.19 | LenzX | Zeex: where do I find the phone booth? |
| 09:49.27 | Zeeek | http://www.freeworlddialup.com/content/view/sitemap/2 |
| 09:49.27 | Zeeek | http://www.freeworlddialup.com/support/configuration_guide |
| 09:49.27 | Zeeek | http://www.freeworlddialup.com/advanced/iax |
| 09:49.27 | Zeeek | http://www.freeworlddialup.com/support/forum |
| 09:49.27 | Zeeek | http://www.freeworlddialup.com/advanced/service_numbers |
| 09:49.34 | Zeeek | check that stuff |
| 09:50.30 | LenzX | Zeex: I already checked, but I did not see such a service. I tried the 55555 but there's nobody on. |
| 09:51.52 | LenzX | anyway I tried to call me back and it worked. Sorry |
| 09:51.58 | Zeeek | what did you check - the answer is on the first page up there |
| 09:52.43 | LenzX | oh yes, I found it. never logged on :-) |
| 09:55.11 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 10:00.03 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:03.20 | Zeeek | seek and ye shall run out of disk space |
| 10:11.21 | *** join/#asterisk nicox (~nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 10:11.25 | nicox | Hello |
| 10:11.44 | Zeeek | good morning |
| 10:11.45 | nicox | Do anyone knows something about Asterisk_Realtime? |
| 10:11.56 | nicox | good morning |
| 10:13.21 | RestLessGemini | Shoot your Question nicox... I am sure more then 60% ppl sitting here knows about it |
| 10:13.40 | RoyK | nicox: see the wiki |
| 10:13.42 | Zeeek | I don't |
| 10:13.59 | Zeeek | ~lart RoyK |
| 10:14.00 | RoyK | ~lart Zeeek |
| 10:14.22 | Zeeek | that's the worst lart I've ever seen, Outlook Express ! |
| 10:14.30 | RoyK | :) |
| 10:14.44 | Zeeek | go ahead and shoot me! |
| 10:14.49 | RoyK | ~lart Zeeek |
| 10:15.02 | Zeeek | anything but OE or Outlook |
| 10:15.21 | RoyK | netcat |
| 10:15.22 | Zeeek | I spent two hours converting an Outlook message file to be importaed by Thunderbird |
| 10:15.31 | nicox | okay, i have an asterisk server running with ast_data with one simple problem: Sip over nat! there is no chance to do it, that says wiki and all people, is this problem solved in asterisk_realtime? |
| 10:15.44 | RoyK | nat=yes |
| 10:15.45 | Zeeek | in fact, you can't import or convert an Outlook message file unless you have.... Outlook! |
| 10:16.33 | Zeeek | no good, the file is encrypted!!! |
| 10:16.45 | RoyK | Zeeek: in fact, the easiest way to convert from outlook is an external imap account :P |
| 10:17.00 | Zeeek | nicox : http://willypick.mindsay.com/?entry=10 |
| 10:17.05 | nicox | <PROTECTED> |
| 10:17.18 | Zeeek | yes but this was a file sent by an attorney via email att. |
| 10:17.40 | RoyK | ~time |
| 10:17.40 | jbot | extra, extra, read all about it, time is 1 dimensional, or everlasting |
| 10:17.48 | RoyK | ~time cet |
| 10:17.49 | Zeeek | asking the atty for a different format, even if he knew how, would cost at least $25-$50 |
| 10:18.00 | Zeeek | ~date |
| 10:18.01 | jbot | Thu Feb 24 10:18:01 2005 |
| 10:18.08 | Zeeek | nyuk, nyuk |
| 10:18.30 | RoyK | ~date cet |
| 10:18.40 | Zeeek | ~yermutha cet |
| 10:18.47 | RoyK | :) |
| 10:20.05 | Zeeek | apparently she's not |
| 10:20.23 | Zeeek | seems to be GMT |
| 10:20.31 | Zeeek | ~EST |
| 10:20.32 | jbot | i guess est is Eastern Standard Time, but if you're outside of Indiana you probably need to use EST5EDT instead, or US/Eastern, or (use the tzconfig program to do this) |
| 10:22.33 | amer | ~who |
| 10:27.02 | RoyK | ~fsck Zeeek |
| 10:27.04 | jbot | e2fsck /dev/Zeeek : warning! filesystem contains dickheads! |
| 10:27.04 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 10:27.11 | RoyK | lol |
| 10:28.13 | Zeeek | errrrr.... wait a minute |
| 10:28.34 | Zeeek | just a fscking minute |
| 11:02.50 | Zeeek | no wonder he weighs so much |
| 11:03.24 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 11:11.30 | *** join/#asterisk Inv_arp (junya@adsl-8-231-123.mia.bellsouth.net) |
| 11:14.38 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
| 11:23.38 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:34.00 | tzafrir | can't mozilla/win32 import from outlook? (not that I really care) |
| 11:37.45 | libpcp | i would like to ask what is the correct pinning of cable for TE410P ? |
| 11:41.43 | Zeeek | yes as long as you have outlook |
| 11:41.56 | Zeeek | fucking Outlook can't be read without Outlook |
| 11:42.23 | Zeeek | So if anyoine ever needs to read one of these .pst files to recover mail, they're fscked! |
| 11:42.32 | Zeeek | Good to know |
| 11:42.42 | Zeeek | even if you don't care :) |
| 11:44.39 | Zeeek | are nufone tollfree DID free for Canadians too? (866) I'm guessing not from what I read about CA |
| 11:46.30 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 11:47.29 | *** join/#asterisk tom-win (~tom-w@sar.biz) |
| 11:50.22 | Damin | It sucks getting up early.. |
| 11:54.05 | file | Damin: no sleep for you! |
| 11:58.05 | *** part/#asterisk pashah (~pashah@relay.patentica.com) |
| 12:03.42 | djin | Zeeek, there are pst -> mbox converters. |
| 12:05.47 | libpcp | if im going to add an TE410P E1 card to my existing asterisk box, what driver do i need to use? |
| 12:06.10 | djin | zaptel |
| 12:06.37 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 12:06.41 | djin | do you already have digium hardware in your machine, or ztdummy? |
| 12:07.11 | djin | then just 'modprobe wct4xxp' |
| 12:11.41 | libpcp | if i do make install on zaptel directory, do i need to recompile the asterisk source again? |
| 12:12.13 | libpcp | djin: the TE410P is already plugged-in to my asterisk machine |
| 12:12.24 | Zeeek | djin - yes but they REQUIRE Outlook to work - I downloaded them all |
| 12:12.24 | libpcp | djin: do i need the libpri? |
| 12:12.38 | Zeeek | I've never seen a situation like that before |
| 12:12.51 | Zeeek | the converters use MAPI interface |
| 12:15.30 | MrEntropy | hehehe...did slashdot get 0wn3d or something? |
| 12:23.33 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 12:27.35 | RestLessGemini | libpcp: yes you need to recompile asterisk source again |
| 12:29.16 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 12:30.44 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 12:30.58 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 12:31.26 | heka | Hello, is there any callback implementation with asterisk |
| 12:31.27 | heka | ? |
| 12:31.48 | RoyK | it's possible via the manager interface |
| 12:32.46 | heka | RoyK: was your last message for me? |
| 12:33.02 | RoyK | heka: yep |
| 12:33.25 | heka | RoyK: is there any documentation on how to do that ? |
| 12:33.33 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 12:33.45 | Delvar | heka: yes asterisk can do callback, look on voip-info.org im sure thers a lot of info |
| 12:33.50 | RoyK | heka: see the wiki. there's something there |
| 12:34.36 | heka | I did a search but I didnt get any interesting |
| 12:37.03 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 12:38.11 | JunK-Y | http://slashdot.org/ is odd today :( |
| 12:43.20 | *** join/#asterisk e3eli3h (~Chris@static-np1-5.cytanet.com.cy) |
| 12:44.18 | *** join/#asterisk stypjan (hidden-use@sdtn-9.netactive.co.za) |
| 12:44.20 | stypjan | hi all |
| 12:44.35 | stypjan | sorry to barge in, but I got a quick question re: extensions. |
| 12:44.48 | stypjan | How can I send a certain CLI to an upstream server? |
| 12:45.14 | stypjan | ie, anything that I dial with 09 at the front get's sent to server x.y.z with caller ID 09231515125(example) |
| 12:47.47 | djin | djin: do i need the libpri? -> yes |
| 12:49.45 | *** join/#asterisk sd-tux (user2267@emasq.stusta.mhn.de) |
| 12:52.23 | JunK-Y | makes everything going to the same server, and with _09X. goes to ur 2nd server. |
| 12:55.45 | stypjan | yeah, i got that bit |
| 12:55.45 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 12:55.45 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 12:55.55 | stypjan | but I want to know, if I can "fool" the other server into thinking that i'm calling from 5551234 |
| 12:56.19 | Zeeek | setcalleridnum() |
| 12:57.03 | stypjan | eh? |
| 12:57.27 | Zeeek | try this from the CLI: show applications |
| 12:58.15 | stypjan | aah, Zeeek: you talking about this? |
| 12:58.15 | stypjan | <PROTECTED> |
| 12:58.35 | Zeeek | stypjan - as I said, look at the list of applications |
| 12:58.50 | Zeeek | then read the ones that talk about what interests you |
| 13:00.24 | stypjan | okay, this is all new to me, just when i thought my n00bness was waring off |
| 13:00.39 | stypjan | but eh, where do I use this application? I got this far |
| 13:00.50 | stypjan | asterisk*CLI> show application SetCallerID |
| 13:00.50 | stypjan | asterisk*CLI> |
| 13:00.50 | stypjan | <PROTECTED> |
| 13:00.50 | stypjan | [Synopsis]: |
| 13:00.50 | stypjan | Set CallerID |
| 13:00.51 | stypjan | [Description]: |
| 13:00.53 | stypjan | <PROTECTED> |
| 13:00.53 | Zeeek | in that case check off the items you've already read: |
| 13:00.55 | stypjan | value. Sets ANI as well if a flag is used. Always returns 0 |
| 13:00.56 | Zeeek | Starter tutorial: |
| 13:00.56 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 13:00.56 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 13:00.56 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 13:00.56 | Zeeek | THE reference of the moment: |
| 13:00.57 | Zeeek | http://www.asteriskdocs.org |
| 13:01.07 | stypjan | lol |
| 13:01.27 | Zeeek | if you have read any two of the above you would know |
| 13:02.00 | stypjan | mmm |
| 13:02.19 | Zeeek | the read of pride would be more useful |
| 13:02.55 | stypjan | i'm busy walking towards the read of pride ;) |
| 13:03.01 | stypjan | thanks for the kick in the right direction |
| 13:03.06 | Zeeek | heh |
| 13:03.15 | Zeeek | the kick on IRC is pretty blodless :) |
| 13:03.46 | stypjan | i like my ass bloodless, so irc is better |
| 13:03.50 | RoyK | ~lart Zeeek |
| 13:04.02 | Zeeek | don't start that agian! |
| 13:04.09 | RoyK | :) |
| 13:04.28 | Zeeek | count to ten in norwegian or something |
| 13:07.39 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 13:07.46 | fishboy1669 | hi there |
| 13:07.53 | Zeeek | fishy! |
| 13:08.12 | djin | Zeeek, getting back @ outlook conversions. |
| 13:08.15 | fishboy1669 | anyone know how to remove the message "asterisk" when dialing in from a restricted no cli zap channel? |
| 13:08.20 | fishboy1669 | hi zeek hows things |
| 13:08.21 | fishboy1669 | ? |
| 13:08.23 | djin | Did you ever check Outport (http://outport.sourceforge.net/)? |
| 13:08.43 | fishboy1669 | i want to change it to say "withheald number" |
| 13:08.55 | fishboy1669 | its on a zap channel |
| 13:08.58 | Zeeek | djin I tried 'em all |
| 13:09.10 | *** join/#asterisk GodThor (~ninja@212.110.95.139) |
| 13:09.22 | Zeeek | fish I think you can do that in sip.conf |
| 13:09.27 | djin | Zeeek, really, really all? :) |
| 13:09.31 | GodThor | what this mean: wrapendpoint.cxx:915: error: 'class H323AudioCodec' has no member named 'IsDescendant' ???? |
| 13:09.34 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 13:09.35 | *** join/#asterisk Whisk (~whisk@whisk.gotadsl.co.uk) |
| 13:09.36 | fishboy1669 | any idea how? |
| 13:09.40 | Zeeek | i think so - I googled for hours and hours and hours |
| 13:10.15 | Zeeek | but the fact is, I stopped because I had Outlook at the office. I just did it there, direct import to Thunderbird and zipped that to my son who needed it |
| 13:10.25 | GodThor | when i build asterisk-oh |
| 13:10.43 | Zeeek | fish I think there is a line to add in sip.conf - I *think* |
| 13:12.07 | Zeeek | what if you do a setcalleridname when there is none? That's what I did |
| 13:12.52 | fishboy1669 | cheers zeek |
| 13:13.12 | GodThor | anyone? |
| 13:13.13 | heka | anybody know any good howto for making a callback system with asterisk? |
| 13:14.15 | Zeeek | you get it fishboy1669 ? |
| 13:16.59 | *** join/#asterisk RGi_ (~rgi@gw-a.adcom.stord.as) |
| 13:18.01 | fishboy1669 | not yet |
| 13:18.16 | fishboy1669 | im reading up on http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 13:18.28 | fishboy1669 | usecallerid etc |
| 13:18.30 | Zeeek | you do an If( callerid == '') |
| 13:18.35 | Zeeek | except that isn't the syntax |
| 13:18.46 | Zeeek | and if it's blank you do a setcalleridname |
| 13:18.48 | fishboy1669 | but thats in zapata.conf |
| 13:18.59 | fishboy1669 | aha i see where your comming from |
| 13:19.01 | Zeeek | that would be in extensions |
| 13:19.21 | Zeeek | I thought there was an assignment possible to replace "Asterisk" though |
| 13:19.26 | Zeeek | and there may be |
| 13:19.33 | Zeeek | or change it in the source |
| 13:19.47 | fishboy1669 | but would callerid = '' be right as what would get passed would probably be callerid = 'asterisk' ??? |
| 13:19.47 | Zeeek | you can change UserAgent for example in sip.conf |
| 13:20.13 | Zeeek | no not if I understand you, which I am not sure to be doing |
| 13:20.41 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 13:20.42 | fishboy1669 | what u say makes sence |
| 13:20.53 | Zeeek | wait I'll get the lines |
| 13:20.54 | RGi_ | Hi.. I have a litle strange problem with Silence supression and music on hold... when I put someone on hold and Asterisk plays music on hold to them I have the problem with silence suppression that asterisk wont send packets out to the client that is lisening to the MOH. but.. I only get that problem when I use MOH.. not with regular speach and voice conversations.. any ideas ? |
| 13:21.13 | fishboy1669 | i have an incoming call from pstn on a x100p zap channel but the cli is withheald so the cli on the phone is "asterisk" |
| 13:21.41 | fishboy1669 | RGi check your codecs |
| 13:21.56 | fishboy1669 | i think u have to use certain codecs for moh |
| 13:22.22 | fishboy1669 | or have the g729 lics for the codec conversion |
| 13:22.33 | Zeeek | fishboy this is with number but it would work with name as well: |
| 13:22.35 | Zeeek | exten => s,3,GotoIf($[X${CALLERIDNUM} != X]?5) |
| 13:22.35 | Zeeek | exten => s,4,SetCIDNum(${CALLERIDNUM}) |
| 13:22.57 | RGi_ | fishboy1669 :ahh !! thanx ! I`l look into that ! :D |
| 13:23.49 | Zeeek | wait that doesn't look right |
| 13:24.12 | Zeeek | here it is |
| 13:24.13 | Zeeek | exten => s,4,GotoIf($[X${CALLERIDNUM} != X]?s,6) |
| 13:24.13 | Zeeek | exten => s,5,SetCIDNum(0000) |
| 13:24.44 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:25.05 | fishboy1669 | cheers zeek |
| 13:25.43 | didz_ | anyone knows whats happening for a "m" DTMF ??? --------- > DTMF digit: m on Zap/1-1 |
| 13:27.23 | fishboy1669 | zeeek heres is someone else suggestion edit chan_sip.c in channels dir and change Asterisk and asterisk with whatever you want .. and then recompile it .. it will solve your problem |
| 13:27.49 | Zeeek | exten => s,4,GotoIf($[X${CALLERIDNUM} != X]?s,6)why would I wanna do that? |
| 13:27.58 | Zeeek | damn paste |
| 13:28.12 | fishboy1669 | ? |
| 13:28.24 | fishboy1669 | what is worng with it? |
| 13:28.50 | Zeeek | why would I want to screw around modifying source? I'd have to change it again when I upgrade (if I ever do) |
| 13:29.04 | fishboy1669 | true |
| 13:29.16 | Zeeek | anyway, you asked... there is one way |
| 13:29.25 | fishboy1669 | im thinking using your plan first as i dont wanna bring the cust pbx down |
| 13:29.32 | fishboy1669 | its only just been in stalled |
| 13:29.40 | fishboy1669 | yours is safer method |
| 13:29.46 | Zeeek | that would tend to sap confidence |
| 13:29.58 | fishboy1669 | exactly |
| 13:30.34 | fishboy1669 | sorry if i offended u pasting the other method up just thought u may be interested in it i wasnt slating your effort in giving me a solution |
| 13:31.11 | fishboy1669 | hope u r cool |
| 13:31.32 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 13:33.09 | fishboy1669 | zeeek u still speaking to me? |
| 13:34.53 | fishboy1669 | zeeeeeeeeeeeeeeeeeeeeeeeekkkkkkkkkkkkkkkkk |
| 13:35.06 | fishboy1669 | im worried i have offended u :( |
| 13:35.13 | Zeeek | fish no the "damn paste" was ME I pasted the stupid thing by accident |
| 13:35.24 | Zeeek | no other than the fishy smell |
| 13:35.41 | fishboy1669 | oh so what is the correct method then |
| 13:35.42 | Zeeek | I am not offened |
| 13:35.54 | Zeeek | or even effended |
| 13:36.16 | Zeeek | correct methond of what? I pasted by accident and was frustrated |
| 13:36.23 | *** join/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo) |
| 13:36.43 | GMsoft | hi |
| 13:36.44 | fishboy1669 | ?so which is the right code from the stuff u pasted |
| 13:37.03 | GMsoft | does external serial isdn works with chan_modem ? |
| 13:37.44 | ManxPower | GMsoft, Generally no. |
| 13:38.45 | GMsoft | any doc about this somewhere ? like supported models and co ? |
| 13:39.33 | ManxPower | GMsoft, no. chan_modem is pretty much unsupported and undocumented. Buy a cheap ISDN card if you are not in the USA/CA |
| 13:40.13 | GMsoft | ok. tnx |
| 13:40.16 | ManxPower | Rumor is that the chan_modem* was written over spring break by a bunch of geeks with too much beer and two few girls and has not been touched since. |
| 13:40.29 | GMsoft | heh |
| 13:41.00 | ManxPower | GMsoft, You can assume you can't do anything with aterisk with a modem on a serial port. |
| 13:41.27 | e3eli3h | FXO won't answer incoming calls and I'm suffering from the "Ring/Off-hook in starnge state 6 on channel n" syndrome. Anyone out there who can help me? |
| 13:41.28 | GMsoft | ok. that was my guess but Ihad some hope :) |
| 13:41.37 | Grooby | so chan_modem was written by those 2 girls while the geeks were drinking beer? |
| 13:41.52 | ManxPower | e3eli3h, callprogress=no busydetect=no in /etc/asterisk/zapata.conf |
| 13:42.04 | e3eli3h | they are there |
| 13:42.27 | ManxPower | e3eli3h, analog FXO? |
| 13:42.36 | e3eli3h | yes. TDM13B. |
| 13:42.56 | ManxPower | e3eli3h, the only other thing I can think of is that you have Ringmaster/Distinctive Ring on the line. |
| 13:43.36 | e3eli3h | i can get it to ring an internal grandstream phone with my extensions file, but when i pick up the handset it just keeps ringing |
| 13:43.49 | ManxPower | Ah. You are a newbie. |
| 13:43.52 | ManxPower | ~docs |
| 13:43.53 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:44.13 | e3eli3h | been through all of them. i'm into my second week or reading docs. |
| 13:44.21 | e3eli3h | extensively! |
| 13:45.36 | e3eli3h | second week of reading docs that is |
| 13:45.57 | GMsoft | mhh besides the fact that most pstn (yes pstn this time :) modems doesn't provide full duplex voice, would such support be easily codable/fixable in chan_modem or whatever ? |
| 13:46.33 | ManxPower | GMsoft, even the ones that DO support full duplex, the APIs are secret. |
| 13:46.54 | ManxPower | They are designed for voicemail, which doesn't care if you have 100ms latency. |
| 13:47.00 | *** join/#asterisk bugsmoke (~mayday@c-24-15-165-107.client.comcast.net) |
| 13:47.01 | ManxPower | ..er...1000ms |
| 13:47.14 | GMsoft | yeah. my next question would have been about the documentation of all this stuff :) |
| 13:47.17 | GMsoft | ah k |
| 13:47.17 | GMsoft | too bad |
| 13:47.50 | Zeeek | 100ms is GOOD |
| 13:47.56 | GMsoft | I think I'll buy one 3fxs+1fxo tdm400 :) |
| 13:48.04 | Zeeek | I WISH I had 100ms |
| 13:48.20 | GMsoft | :) |
| 13:48.25 | ManxPower | Zeeek, 100ms latency between the PC and a local FXO/FXS device sucks. |
| 13:48.41 | Zeeek | I'm talking about 100ms to the caller or ISP |
| 13:48.49 | ManxPower | Zaptel has like 5ms latency |
| 13:48.52 | Zeeek | 90% of my provers are a little over 100 |
| 13:49.01 | GMsoft | ITU-T recommanded latency for a call is 150 ms |
| 13:49.02 | bjohnson | for anyone who cares, I think I've got safe_asterisk not using those crappy colours now |
| 13:49.04 | Zeeek | ah I missed the zaptel part :) |
| 13:49.08 | ManxPower | Zeeek, That's why we don't use ITSPs much |
| 13:50.35 | ManxPower | Ugh. I have to spend all day doing a new Asterisk install |
| 13:54.19 | Zeeek | isn't that what you love best in this life? |
| 13:55.56 | ManxPower | no. |
| 13:56.03 | ManxPower | that would be money |
| 13:56.26 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 13:56.31 | pashah | hello |
| 13:57.06 | Poincare | ManxPower: just money? |
| 13:57.26 | tzanger | ManxPower: so far so good? |
| 13:58.17 | fishboy1669 | catch u later guys got to change lans now |
| 13:58.18 | fishboy1669 | bye |
| 13:59.33 | Whisk | hi - i'm getting reproduceable seg faults with latest cvs - is the procedure for providing debug traces etc documented anywhere? |
| 14:00.13 | *** part/#asterisk GodThor (~ninja@212.110.95.139) |
| 14:00.34 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 14:00.34 | Zeeek | Manx if you loved money that much, you would have chosen another path |
| 14:00.41 | GMsoft | Whisk: ulimit -c unlimited |
| 14:00.47 | GMsoft | that will give you a core dump |
| 14:00.54 | GMsoft | which the dev can analyze |
| 14:01.12 | *** join/#asterisk sabre (~urfos@69.149.209.83) |
| 14:01.47 | Whisk | ok - sorry to be dumb, but you run that command before starting asterisk? |
| 14:02.14 | GMsoft | yep |
| 14:02.32 | GMsoft | and preferably run something like |
| 14:02.38 | GMsoft | asterisk -vvvvvvvvvv |
| 14:02.55 | GMsoft | instead of starting it with the init scripts |
| 14:03.34 | goatmilk | has anyone compiled asterisk with gcc4? |
| 14:04.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 14:04.16 | puzzled | hi |
| 14:04.30 | Moc | hi |
| 14:05.17 | mooboi | heya |
| 14:05.59 | Grooby | anyone running the CVS HEAD? |
| 14:06.05 | Grooby | how's the jitter buffer working out? |
| 14:06.30 | mooboi | iiiiii dont know ;0 |
| 14:09.58 | *** join/#asterisk oej (~oej@ua-213-115-215-100.cust.bredbandsbolaget.se) |
| 14:11.31 | grailink | is there an alternative to goto.. i.e. something like a procedure call that can return to where the stack originally was before the call? |
| 14:11.43 | grailink | like call(x) |
| 14:12.04 | grailink | in extensions.conf that is... |
| 14:12.58 | *** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar) |
| 14:13.54 | Whisk | thx GMsoft |
| 14:15.12 | GMsoft | Whisk: np |
| 14:15.56 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 14:16.50 | oej | ~seen anthm |
| 14:16.58 | jbot | anthm <~anthmct@CPE-69-76-83-52.wi.rr.com> was last seen on IRC in channel #asterisk, 1d 16h 2m 59s ago, saying: 'yacto'. |
| 14:17.20 | dsmouse | grailink: I think dial might be able to do that |
| 14:17.28 | dsmouse | *think* |
| 14:17.54 | grailink | k.. i'll give it a try |
| 14:17.57 | grailink | thx |
| 14:18.06 | *** join/#asterisk ScarletCrusader (~GMMiller@wsip-66-210-74-254.mc.at.cox.net) |
| 14:18.25 | dsmouse | grailink: also look up forking on voip-info |
| 14:18.35 | dsmouse | grailink: they have some tricks with dial there |
| 14:24.17 | grailink | i *wish* they made these conf files like a procedural script (java/php/etc)... i'd be done by now :) |
| 14:24.54 | mishehu | mornings suck. |
| 14:25.24 | grailink | anyone here use diax? |
| 14:26.42 | *** join/#asterisk TheEmperor (TheEmperor@218.111.48.18) |
| 14:28.52 | *** join/#asterisk nix000 (~nixman@66.11.190.225) |
| 14:29.12 | nix000 | anyone expert in tdm technologies here like t1/e1 ? |
| 14:29.52 | TheEmperor | hello |
| 14:30.32 | TheEmperor | can i ask if kernel 2.6 is ok to use with *? |
| 14:30.37 | Essobi | it's fine |
| 14:30.44 | grailink | works perfectly.. better actually |
| 14:31.01 | TheEmperor | graillink: how does it work better? |
| 14:31.28 | grailink | meetme needs a timer and if you don't have zaptel hardware it needs you to have a special usb controller |
| 14:31.36 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:31.36 | grailink | with 2.6 there is a high-res timer built into the kernel |
| 14:31.52 | TheEmperor | i c |
| 14:31.54 | TheEmperor | interesting |
| 14:32.14 | TheEmperor | i was thinking of using debian with kernel 2.6, would that be ok |
| 14:32.22 | grailink | i had to upgrade a * box from to 2.6 last night to get meetme to work right |
| 14:32.38 | Grooby | so with 2.6, i don't need ztdummy? |
| 14:32.41 | grailink | i'd think any 2.6 kernel would be ok. i'm a suse fan but its all the same |
| 14:32.52 | grailink | u need ztdummy but it uses the kernel instead of the usb timer |
| 14:32.53 | TheEmperor | sweet! |
| 14:33.27 | Grooby | ahhh i c i c |
| 14:33.35 | grailink | u running x and all that or just a server box? |
| 14:34.06 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 14:34.24 | nix000 | grailink: what is meetme ? i cant seem to find a wesite about it. |
| 14:34.35 | grailink | meetme is the * conf capability |
| 14:34.36 | Darwin35 | its in the wiki pages |
| 14:35.01 | grailink | its really nice |
| 14:35.01 | Darwin35 | ~wiki |
| 14:35.31 | Darwin35 | ~wiki meetme |
| 14:35.40 | TheEmperor | can anyone help? i've already made a .call file and moved it to the outgoing directory how do i run it? |
| 14:35.49 | Hmmhesays | voodoo |
| 14:36.02 | Darwin35 | who killed wiki |
| 14:36.32 | grailink | voip-info.org is up |
| 14:36.38 | grailink | for me that is... |
| 14:36.45 | Hmmhesays | will smith killed wiki |
| 14:36.48 | grailink | it could be because i'm special |
| 14:36.51 | grailink | ;) |
| 14:37.18 | *** join/#asterisk g00dy (~g00dy@69-17-136-9.kingkom.com) |
| 14:37.23 | Hmmhesays | I do my callbacks with an agi |
| 14:37.24 | Darwin35 | job is having issues |
| 14:37.30 | Darwin35 | the wiki pages are up |
| 14:37.36 | Darwin35 | jbot |
| 14:37.41 | tzafrir | TheEmperor, your callfile is probably not readable by Asterisk |
| 14:37.54 | tzafrir | Is asterisk run as non-root? |
| 14:39.00 | tzafrir | basically moving the call file to the outgoing directory should run it |
| 14:39.07 | grailink | i'm no totally sure but i don't see why it would need root |
| 14:39.08 | g00dy | what hardware do you guys recomend for a newbie asterisk tester? |
| 14:39.18 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:39.44 | tzafrir | grailink, root is not needed. But if you're not root there is the option that you cannot read the call file |
| 14:39.52 | nix000 | anyone used asterisk snmp support ? |
| 14:40.05 | TheEmperor | tzafrir: i am running as root |
| 14:40.36 | TheEmperor | tzafrir: i tried a few times but it doesn't work... |
| 14:40.43 | TheEmperor | tzafrir: i called it 1.call |
| 14:41.23 | grailink | tzafrir: i run as root too.. but i've never tried to run it as anything else either. i'm sure there's a way to do it. |
| 14:41.49 | g00dy | any comments on the : LINYSYS PAP2? |
| 14:42.09 | tzafrir | TheEmperor, does the file remain in the outgoing directory? |
| 14:42.35 | TheEmperor | tzafrir: no, it's gone when i move it |
| 14:42.47 | tzafrir | TheEmperor, so it is consumed by asterisk |
| 14:43.08 | TheEmperor | tzafrir: yeah, but nothing happens |
| 14:43.15 | tzafrir | Now go to the CLI, set verbose = 3; and try to figure out the errors |
| 14:43.22 | TheEmperor | tzafrir: maybe i wrote it wrong |
| 14:43.33 | bjohnson | on a SPA 2000 .. is FXS Port Output Gain the volume to the handset or would that be the FXS Port Input Gain setting? |
| 14:44.22 | tzafrir | grailink, -U is the way to run as non-root, BTW |
| 14:44.36 | Hmmhesays | bjohnson: i bet it says in the manual |
| 14:44.59 | TheEmperor | tzafrir: so what should i do now? |
| 14:45.33 | bjohnson | grailink: the superdial macro takes a procedure approach and returns to where it was called |
| 14:46.39 | bjohnson | Hmmhesays: have you read the manual? It doesn't explain much |
| 14:47.39 | TheEmperor | tzafrir: maybe i can show you what i wrote in the .call file? it's not long |
| 14:48.06 | tzafrir | 2 or three lines can be pasted here. |
| 14:48.06 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 14:48.10 | Hmmhesays | bjohnson: no |
| 14:48.16 | Hmmhesays | but i'm in the RTFM kinda mood |
| 14:48.30 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
| 14:48.34 | BrianR___ | YAY! DTMF based disconnect supervision works! |
| 14:48.41 | bjohnson | Hmmhesays: example: FXS Port Input Gain is defined as Input Gain in dB. Valid values are 6.0 to infinity. Up to 3 decimal places |
| 14:48.43 | Chotaire | http://support.microsoft.com/default.aspx?scid=KB;en-us;q276304 |
| 14:48.54 | TheEmperor | tzafrir: Channel : Zap/4/921416980, MaxRetries:4,RetryTime:60,WaitTime:30 |
| 14:48.56 | tzafrir | TheEmperor, but did you check the trace in the CLI ? |
| 14:49.07 | bjohnson | that still doesn't tell me if it is the gain to (or from) the handset |
| 14:49.19 | TheEmperor | tzafrir: how do i check trace in CLI? |
| 14:49.28 | tzafrir | asterisk -r |
| 14:49.36 | tzafrir | set verbose 3 |
| 14:49.36 | Hmmhesays | receive gain is generally considered gain for incoming to the port |
| 14:49.41 | tzafrir | set debug 1 |
| 14:49.58 | tzafrir | (or use some other values for 'verbose' and 'debug') |
| 14:50.09 | TheEmperor | when i type in set debug 1, it says no such command |
| 14:50.21 | tzafrir | Anyway, from which channel is this and to which extension? |
| 14:50.24 | bjohnson | Hmmhesays: incoming from * or incoming from the analog phone? |
| 14:50.34 | TheEmperor | is my .call file correct? i basically want it to dial a number |
| 14:50.40 | TheEmperor | from channel 4 |
| 14:50.41 | kamran | hi all any one using http://www.voip-info.org/wiki-CallingCard+Applications |
| 14:50.46 | Hmmhesays | from the analog side |
| 14:50.56 | TheEmperor | dialling 921416980 |
| 14:51.03 | tzafrir | That would be: Channel: Zap/4 |
| 14:51.10 | TheEmperor | yup, got that |
| 14:51.20 | tzafrir | to which extension should it dial? |
| 14:51.22 | Hmmhesays | rx gain = the people transmitting to asterisk, tx gain = asterisk transmitting to the end user |
| 14:51.26 | Hmmhesays | probably |
| 14:51.32 | *** join/#asterisk bladex (~bladex@bonk.personal.engin.umich.edu) |
| 14:51.33 | TheEmperor | i want it to dial an actual phone number |
| 14:51.42 | TheEmperor | or must it dial an extension? |
| 14:52.17 | Hmmhesays | you could always just turn one waaaaaaaay up |
| 14:52.22 | Hmmhesays | then you'd be able to tell quickly |
| 14:52.52 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:52.59 | tzafrir | TheEmperor, asterisk knows about extensions (from certain contexts). |
| 14:53.30 | tzafrir | Asterisk has no idea you happen to consider some of those (minus some prefix) as phone numbers |
| 14:53.36 | TheEmperor | tzafrir: so maybe i mistake was making it dial an actual number? |
| 14:53.50 | TheEmperor | it dials and extension and then in the exten file i specify where that goes? |
| 14:53.55 | tzafrir | Something like: |
| 14:54.03 | tzafrir | Extension: 921416980 |
| 14:54.19 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
| 14:54.35 | TheEmperor | so instead of Channel, put in Extension: 1234 ? |
| 14:54.38 | TheEmperor | would that work? |
| 14:54.41 | TheEmperor | i can try that |
| 14:57.36 | bladex | howdy, howdy .... anybody using linux, what linux distro are you running on? |
| 14:58.30 | *** join/#asterisk Alejandriax26 (~nurbina23@proxy.more.cl) |
| 14:58.35 | bladex | oops |
| 14:58.41 | Alejandriax26 | somebody from chile? |
| 14:58.47 | bladex | not a war, i'm just wondering what people are using |
| 14:58.56 | Hmmhesays | anyone using linux IN HERE?!?!?! nawwwwww |
| 14:59.30 | Hmmhesays | I like abacus linux... simple easy to use and requires no electricity |
| 14:59.46 | bjohnson | here's a different issue I have that I think may just be something I'm missing. I'm using the authbyCID macro from the wiki and it authenticates and dials the internal extension that the caller inputs, but the caller doesn't hear ringing .. they hear a repeated beep (kinda souns like a busy tone) |
| 14:59.52 | bladex | don't the beads get expensive making it hard to scale? |
| 15:00.26 | Hmmhesays | good for small to medium sized installations.... the real problem is hiring the super smart monkeys to run the calculations |
| 15:00.39 | tzafrir | I've seen people use (alphbetical order) Debian, Fedora, Gentoo, little Mandrake, RHEL clones (mostly CentOS), slackware, and very little SuSE |
| 15:01.06 | bjohnson | wow .. alphabetical order even |
| 15:01.15 | jsolares | Unbuntu! |
| 15:01.23 | Hmmhesays | lol |
| 15:01.25 | bladex | i am only asking because i use crux which is Very lean, but i ran into a problem with devfs versus udev |
| 15:01.27 | GMsoft | tzafrir: gentoo even have support for asterisk on parisc cpu :) |
| 15:01.42 | Hmmhesays | i use debian |
| 15:01.49 | Hmmhesays | because apt is my friend |
| 15:01.55 | jsolares | well ubuntu is kinda like a debian clone |
| 15:01.59 | jsolares | apt is good |
| 15:02.04 | Hmmhesays | yeah knoppix too |
| 15:02.04 | djin | apt works in redhat as well. |
| 15:02.06 | TheEmperor | tzafrir:pbx_spool.c:194 apply_outgoing: At least one of app or extension must be specifies, along with tech and dest in file /var/spool/asterisk/outgoing/1.call |
| 15:02.13 | Hmmhesays | yeah but you gotta install it |
| 15:02.28 | jsolares | well knoppix is really debian with custom debs on top, while ubuntu i think is only using apt |
| 15:02.31 | Hmmhesays | and it's a pain in the ass if you run into an apt dependancy problem |
| 15:02.35 | jsolares | i think i think that |
| 15:02.37 | bjohnson | true .. yum is the default system |
| 15:02.49 | tzafrir | For reference on supported Debian platforms: http://packages.debian.org/testing/comm/asterisk |
| 15:02.53 | TheEmperor | tzafrir:pbx_spool.c:304 scan_service: Invalid file contents deleting. Any idea? |
| 15:03.17 | Hmmhesays | Ubuntu is a Linux distribution that starts with the breadth of Debian |
| 15:03.47 | Hmmhesays | I dislike the name enough to keep me from using it |
| 15:04.10 | jsolares | tzafrir: i think my next asterisk box is going to be debian |
| 15:04.18 | Essobi | anyone have any ideas why when my AS5400 dials my * box, * never matches it to the peer entry? |
| 15:04.20 | tzafrir | Knoppix is really a mix of Debians with some custom debs on to[ |
| 15:04.40 | jsolares | as my brother the debian nut says, if it's not on debian repository it doesnt exists :X |
| 15:04.52 | Essobi | maha |
| 15:04.58 | kamran | hi all any one using http://www.voip-info.org/wiki-CallingCard+Applications |
| 15:05.03 | Hmmhesays | i like the fact that you can download the entire repository weekly |
| 15:05.08 | Hmmhesays | on cd's |
| 15:05.15 | Essobi | That's scarey. |
| 15:05.27 | Hmmhesays | iso 1-15 is regenerated weekly |
| 15:05.40 | tzafrir | Hmmhesays, why would you like to do that? The whole point is that you don't need to. |
| 15:05.44 | Essobi | "1-15" is exactly what scares me. |
| 15:05.52 | Essobi | It's 15 fricking disks. |
| 15:06.02 | Hmmhesays | they have dvd distro's too |
| 15:06.08 | Essobi | ^o_O^ |
| 15:06.20 | Essobi | Baaah, humbug. |
| 15:06.22 | tzafrir | Essobi, which is why you never load everything (unless you want to set up a mirror |
| 15:06.23 | tzafrir | ) |
| 15:06.29 | jsolares | i know someone that found it useful to burn all 15 cd's |
| 15:06.53 | Essobi | tza I run debian. :) I DONT NEED NO STEEEENKIN ISOS! |
| 15:07.03 | Hmmhesays | tzafrir: just to have it... most likely after the nuclear holocaust and I can't get connected to a debian repository to install more packages |
| 15:07.26 | Essobi | Heh.. We have a reposotory like.. 20 minutes from wher eI'm sitting. |
| 15:07.35 | Alejandriax26 | Who know if there are a distributor of digium in chile? Thanks. But I need to buy the Wildcard TE110P. |
| 15:07.48 | Essobi | I'll just drop on the local net, and be like.. DEWD.. BURN ME SOME SHIZZLE |
| 15:08.18 | Hmmhesays | lol, I would ignore that statement like you weren't even there |
| 15:08.19 | Essobi | I should run a local apt-mirror thou. |
| 15:08.58 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 15:09.10 | Essobi | Hmmhesays That was sarcasm, you'know. |
| 15:09.12 | TheEmperor | tzafrir: Any idea? |
| 15:09.26 | *** join/#asterisk NirS_UK (~root@81.27.72.23) |
| 15:09.30 | NirS_UK | hello all |
| 15:09.32 | Essobi | I SMOTE THEEE! |
| 15:09.32 | NirS_UK | anybody home ? |
| 15:09.36 | Essobi | Hell na. |
| 15:09.43 | Essobi | Only the cockroaches. |
| 15:09.48 | NirS_UK | any |
| 15:09.59 | Essobi | Now drop the cereal and shut off the light on the way out. |
| 15:10.11 | NirS_UK | any one has an idea why would a TE410P card flicker like crazy on all leds after loading the ZAPTEL module ? |
| 15:10.31 | Essobi | Umm.. Bad voltage? Motherboard? And you called digium? |
| 15:10.40 | NirS_UK | hadn't called digium yet |
| 15:10.43 | Essobi | Dude. |
| 15:10.44 | NirS_UK | motherboard ? |
| 15:10.52 | Essobi | Use their support. |
| 15:10.53 | NirS_UK | just installed th |
| 15:10.54 | |Vulture| | Essobi: damn man did you sleep? |
| 15:11.00 | NirS_UK | their support isn't available at this time |
| 15:11.01 | Essobi | Mahaha. No. |
| 15:11.20 | NirS_UK | the module loads ok, and recognizes the card no problem |
| 15:11.34 | Essobi | Ehh? Digium ain't open yet? |
| 15:11.36 | NirS_UK | but the possibility of a power issue is a possibility |
| 15:11.41 | |Vulture| | still trying to register as a peer? |
| 15:11.45 | |Vulture| | I mean friend |
| 15:11.46 | NirS_UK | well, it sh |
| 15:11.47 | Essobi | Shewww. I'm going to open up an eastern seaboard support center. |
| 15:11.50 | NirS_UK | it should be by now |
| 15:11.52 | NirS_UK | good idea |
| 15:11.57 | Essobi | :) |
| 15:12.26 | Essobi | |Vulture| I've been using a peer in the config but the 5400 looks like it's trying to register as a friend.. |
| 15:12.31 | Essobi | Why I don't know.. |
| 15:13.06 | Essobi | Funny thing is.. none of the docs says it supports anything other then a peer config (the 5400 that is..) |
| 15:13.28 | |Vulture| | Essobi: did you try setting it as a friend and see what happens? |
| 15:14.37 | tzafrir | TheEmperor, any trace? |
| 15:15.15 | Essobi | Oh, I down graded to 1.05 too |
| 15:15.42 | Essobi | that didn't help.. I was sure I had inbound working.. I'm thinking I had one of my did's in default or default included one of the contexts it was in. |
| 15:15.44 | Essobi | :| |
| 15:16.32 | Essobi | Same dealio. |
| 15:16.32 | Essobi | Looking for 8665289720 in default |
| 15:16.35 | Essobi | :| |
| 15:16.40 | tzafrir | TheEmperor, each field should be in a separate line. And you should indeed have an Extension: field |
| 15:16.41 | Essobi | It won't match on the peer |
| 15:16.43 | Essobi | or friend |
| 15:18.46 | Essobi | From: <sip:ORIGINATINGANI@my.as.5400.ip>;tag=7D65ED88-8C3 <-- Is the header supposed to look like that? |
| 15:19.21 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 15:19.44 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 15:19.47 | jero | hi |
| 15:19.57 | Grooby | hi |
| 15:20.49 | Essobi | Okay.. This is screwed. |
| 15:21.12 | Essobi | I added the ANI of the inbound user as the username.. and it landed in the context it was supposed to. |
| 15:21.15 | Essobi | WTF is up with that? |
| 15:21.35 | bjohnson | here's a different issue I have that I think may just be something I'm missing. I'm using the authbyCID macro from the wiki and it authenticates and dials the internal extension that the caller inputs, but the caller doesn't hear ringing .. they hear a repeated beep (kinda souns like a busy tone) |
| 15:22.06 | *** part/#asterisk e3eli3h (~Chris@static-np1-5.cytanet.com.cy) |
| 15:22.09 | BrianR___ | Is there any way to make fxo cards answer before the second ring? I don't care if I lose caller id... |
| 15:22.15 | *** join/#asterisk jero (~sflphone@199.243.85.90) |
| 15:22.19 | jero | hi |
| 15:22.26 | bjohnson | why would that be? It does actually ring the phones. And a person can pickup the ringing phones and have a conversation like normal |
| 15:23.42 | Essobi | Umm. |
| 15:23.52 | Essobi | sounds like your ringback is simulated. |
| 15:23.56 | Essobi | Zapata? |
| 15:23.59 | TheEmperor | tzafrir: did that.. |
| 15:24.09 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
| 15:24.12 | Essobi | Maybe you have the wrong set of notification tones turned up. |
| 15:24.42 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-62.z13-95-67.customer.algx.net) |
| 15:26.27 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:26.27 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:26.28 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:26.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:27.17 | Essobi | *SIGH* |
| 15:27.51 | Essobi | Anyone have an idea why I can only get my as5400 to register as a sip friend with the ANI being the username? |
| 15:28.33 | [ro]nic3try | is possible that contact to differ of from in a sip message ? |
| 15:28.37 | bjohnson | Essobi: you talking to me? My devices are Sipra |
| 15:28.37 | *** join/#asterisk ast_freak (~yircme@hades-out.universalsystems.net) |
| 15:28.41 | bjohnson | Sipura |
| 15:29.11 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc00i.dialup.mindspring.com) |
| 15:29.59 | [ro]nic3try | pls ? |
| 15:33.45 | ast_freak | Can anyone help me with debugging some X-lite sip phones? I had set qualify=yes which had worked fine for months, but now they are lagging big time. I set qualify=2000 since the lagged phones seemed to have ~1600 ms lag time. But some of them are still lagging at 2 secs! I did a sip debug peer on one, and it seems to be retransmitting an OPTIONS command like this: |
| 15:33.56 | ast_freak | <PROTECTED> |
| 15:33.57 | ast_freak | Retransmitting #4 (no NAT): |
| 15:33.59 | ast_freak | OPTIONS sip:5004@10.251.87.237:5060 SIP/2.0 |
| 15:34.00 | ast_freak | Via: SIP/2.0/UDP 10.251.86.242:5060;branch=z9hG4bK26cc9c4e |
| 15:34.02 | ast_freak | From: "asterisk" <sip:asterisk@10.251.86.242>;tag=as06acfd3d |
| 15:34.03 | ast_freak | To: <sip:5004@10.251.87.237:5060> |
| 15:34.05 | ast_freak | Contact: <sip:asterisk@10.251.86.242> |
| 15:34.06 | ast_freak | Call-ID: 46a215b841183d8e09c0ac443e6d1f10@10.251.86.242 |
| 15:34.08 | ast_freak | CSeq: 102 OPTIONS |
| 15:34.10 | ast_freak | User-Agent: Asterisk PBX |
| 15:34.11 | ast_freak | Date: Thu, 24 Feb 2005 15:29:47 GMT |
| 15:34.12 | ast_freak | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER |
| 15:34.14 | ast_freak | Content-Length: 0 |
| 15:34.15 | ast_freak | Can anyone help me? |
| 15:34.27 | Beirdo | gah |
| 15:34.34 | Beirdo | ~pastebin |
| 15:34.35 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 15:34.57 | ast_freak | Sorry, I didn't know that this counted as a flood. |
| 15:35.08 | heison | do we have any dundi participant here? |
| 15:35.47 | *** join/#asterisk km- (user@brdgw1.rttx.com) |
| 15:37.11 | Beirdo | muhahah |
| 15:37.19 | Beirdo | ~nickometer ast_freak |
| 15:37.19 | jbot | 'ast_freak' is 14.000% lame, beirdo |
| 15:37.24 | *** join/#asterisk lorion (~test@63.115.106.66) |
| 15:37.24 | Beirdo | fun stuff |
| 15:37.36 | Delvar | ~nickometer Delvar |
| 15:37.36 | jbot | 'Delvar' is 0.000% lame, delvar |
| 15:37.46 | Beirdo | ~nickometer [ro]nic3try |
| 15:37.46 | jbot | '[ro]nic3try' is 32.000% lame, beirdo |
| 15:37.55 | Beirdo | oooh, we have a winner |
| 15:38.02 | heison | ~nickometer jbot |
| 15:38.02 | jbot | 'jbot' is 0.000% lame, heison |
| 15:38.11 | D31V4rrr | ~nickometer D31V4rrr |
| 15:38.11 | jbot | 'D31V4rrr' is 99.550% lame, d31v4rrr |
| 15:38.15 | Beirdo | heheh |
| 15:38.15 | D31V4rrr | rar! |
| 15:38.17 | djin | wow |
| 15:38.21 | `Sauron | Yawn. |
| 15:38.27 | `Sauron | Y'all really need a life. |
| 15:38.30 | `Sauron | Really. |
| 15:38.40 | Beirdo | `Sauron: your point being? |
| 15:38.41 | [ro]nic3try | ~nicometer Beirdo |
| 15:38.45 | D31V4rrr | ~nickometer `Sauron |
| 15:38.45 | jbot | '`Sauron' is 14.000% lame, d31v4rrr |
| 15:38.48 | D31V4rrr | hehe |
| 15:39.10 | [ro]nic3try | ~nickometer Beirdo |
| 15:39.10 | jbot | 'Beirdo' is 0.000% lame, [ro]nic3try |
| 15:39.15 | [ro]nic3try | upz |
| 15:39.31 | `Sauron | Beirdo: A big boy like you should've figured that out already, no? |
| 15:39.32 | ast_freak | Can anyone help me with the sip problem? |
| 15:40.35 | Beirdo | `Sauron: I know I need a life. :) |
| 15:42.28 | lorion | I am getting a 1 way audio path on my phones that are mat'd behind a firewall. Anyone know if I need to open allow any special ports? |
| 15:43.30 | dsmouse | LoRez: sip, iax, ? |
| 15:43.46 | LoRez | dsmouse: fix your tab completion :) |
| 15:43.51 | dsmouse | damnit |
| 15:47.57 | Nugget | heh |
| 15:48.15 | dsmouse | terracon and terrapen get me all the time too |
| 15:48.27 | Essobi | Anyone running a cisco ASXXXX as a sip peer into to an * box? |
| 15:48.53 | kamran | hello |
| 15:49.09 | kamran | any one know how to set LD_LIBRARY_PATH |
| 15:49.28 | kamran | i am getting error while gcc |
| 15:49.50 | kamran | gcc -shared -Xlinker -x -o app_prepaid_auth_pin.so app_prepaid_auth_pin.o -lmysqlclient |
| 15:49.50 | kamran | /usr/bin/ld: cannot find -lmysqlclient |
| 15:49.50 | kamran | co |
| 15:50.00 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 15:50.09 | *** join/#asterisk adjacent_ (~scott@office.bftwave.com) |
| 15:50.19 | kamran | any one have any idea |
| 15:50.39 | ScarletCrusader | anyone Use Nabisco Routers in their $hop? |
| 15:50.56 | Grooby | lorion, port 10000-20000 for RTP? |
| 15:51.08 | Whisk | hmm |
| 15:51.09 | |Vulture| | anyone using speex on voicepulse? |
| 15:51.10 | Nugget | this router is packed by weight, not volume. Traffic may have settled during transmission. |
| 15:51.16 | Whisk | my asterisk seems to be grinding to a halt |
| 15:51.18 | Essobi | kamran You don't have the mysql client libraries installed. |
| 15:51.31 | Essobi | Nugget Hah. |
| 15:51.39 | Whisk | took 2 mins to do an agentcallbacklogin and now it's not responding at all :( |
| 15:51.53 | kamran | Essobi: i have mysql libraries at /usr/lib/mysql |
| 15:51.53 | Essobi | Whisk asterisk -r? |
| 15:51.54 | Hmmhesays | hmm my asterisk has suddenly stopped sending voicemail--> email |
| 15:52.01 | lorion | Grooby: Here is my config - Asterisk -> Internet -> PIX525 -> Phone |
| 15:52.08 | Essobi | kamran Have you re-ran ldconfig since you installed them? |
| 15:52.09 | Whisk | console doesn't show anything |
| 15:52.10 | |Vulture| | Hmmhesays: check maillog |
| 15:52.17 | ast_freak | Can anyone help me with my sip problems? |
| 15:52.22 | Essobi | if so, then /usr/lib/mysql ain't in your /etc/ld.conf |
| 15:52.27 | Grooby | lorion, so where's the router/nat sit? |
| 15:52.35 | Essobi | Whisk top -S? |
| 15:52.40 | kamran | ok |
| 15:52.46 | ScarletCrusader | hold the cup closer to your mouth |
| 15:52.51 | Essobi | mahaha |
| 15:53.05 | Essobi | paint it mauve.. it's got more ram. :) |
| 15:53.10 | Whisk | asterisk isn't using any cpu |
| 15:53.20 | Essobi | then what is? |
| 15:53.21 | Essobi | :) |
| 15:53.22 | Whisk | box is on about 10% |
| 15:53.23 | Whisk | nothing |
| 15:53.45 | Essobi | something is sucking.. whip out your handy dandy admin skills and look. |
| 15:53.55 | |Vulture| | Anyone here using Speex? |
| 15:54.22 | Whisk | it's ok when i first start *, but goes crappy after a while |
| 15:54.26 | Grooby | vulture, I tried |
| 15:54.36 | Grooby | it sucked bad |
| 15:54.40 | Essobi | |Vulture| Hey, get this.. my AS5400 works fine if I use the inbound caller ID as the username= in the sip config with a friend type. |
| 15:54.41 | Essobi | :| |
| 15:54.43 | kamran | Essobi: how to set this |
| 15:55.08 | Essobi | emacs /etc/ld.so.conf |
| 15:55.12 | Trionnis | erf |
| 15:55.16 | |Vulture| | Essobi: oh wow.. thats just strange |
| 15:55.20 | |Vulture| | Grooby: what do you use? |
| 15:55.20 | Essobi | add the directory and rerun ldconfig |
| 15:55.25 | Trionnis | anyone seen the 403 Forbidden problem with xlite? |
| 15:55.29 | Essobi | Yea.. I'm groveling in #cisco now for help. |
| 15:55.29 | Grooby | i use speex with x-lite |
| 15:55.32 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 15:55.40 | Trionnis | (yes, I'm screwing with a softphone, ridicule me) |
| 15:55.42 | Trionnis | ;) |
| 15:55.43 | |Vulture| | Grooby: Ive been using ilbc and it isn't bad |
| 15:55.47 | HitTop | Hi, anyone using polycom phone here? |
| 15:55.49 | Grooby | yeah |
| 15:55.52 | Grooby | i use ilbc |
| 15:55.53 | Grooby | and i like it |
| 15:55.55 | |Vulture| | HitTop: I do |
| 15:55.59 | lorion | Grooby: the router is on the phone side |
| 15:56.02 | Grooby | ilbc when I am outside, and ulaw when I am in the house |
| 15:56.27 | Grooby | lorion, is nat=yes for that extension's config? |
| 15:56.27 | kamran | Essobi: it is already there |
| 15:56.43 | lorion | Grooby: I have included nat=yes |
| 15:56.48 | Grooby | hmmmm |
| 15:56.57 | Essobi | now is that the server libs or the client libs? |
| 15:57.02 | Essobi | Install both. |
| 15:57.08 | Grooby | what kinda phone? |
| 15:57.16 | lorion | Grooby: The asterisk box is not behind a FW, the phones are. |
| 15:57.29 | lorion | Grooby: I am using X-lite |
| 15:57.30 | Essobi | lorion Ewww. |
| 15:57.36 | Grooby | that's interesting |
| 15:57.54 | lorion | Grooby: the phones login fine and I can leave voicemail, I just don't hear anything |
| 15:58.03 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 15:58.09 | Hmmhesays | is there anyway I can see if voicemail is trying trying to send and email out? |
| 15:58.32 | ast_freak | Hmmhesays, mailq |
| 15:58.43 | lorion | grooby: when I type "sip show peers" I see the phones. |
| 15:58.43 | Grooby | Lorion, can you open port on your FW for UDP outgoing on 10000 to 20000? |
| 15:58.56 | ast_freak | Hmmhesays, better yet, # tail -f /var/log/maillog |
| 15:59.00 | lorion | Grooby: sure |
| 15:59.11 | dsmouse | lorion: try sip show channels |
| 15:59.12 | lorion | Grooby: brb |
| 15:59.12 | Grooby | i know at one of my client site, I had to open up the outgoing ports for RTP in order for my x-lite to work |
| 15:59.27 | Hmmhesays | heh, **frozen** that doesn't look good |
| 15:59.45 | lorion | dsmouse: 0 active sip channels |
| 16:00.02 | dsmouse | lorion: oh, right cause you arn't on the phone atm |
| 16:00.08 | kamran | i have checked /etc/ld.so.conf it has /usr/lib/mysql |
| 16:00.09 | lorion | lol |
| 16:00.22 | kamran | and here i have all libraries |
| 16:01.30 | km- | dammit, I hate it when I send people e-mail and they're like "oh, I deleted that months ago, can you send it to me again?" |
| 16:01.34 | lorion | Grooby: allow outside -> inside UDP 10000 - 20000, correct? |
| 16:01.46 | km- | its hard searching through all that friggin e-mail |
| 16:01.51 | Grooby | lorion, is it NATTED? |
| 16:02.01 | Grooby | I was thinking inside => out UDP 10000-20000 |
| 16:02.02 | lorion | grooby: yes |
| 16:02.25 | Grooby | if it's outside -> in, then you have to configure which internal IP they go to |
| 16:02.44 | lorion | grooby: true |
| 16:03.11 | Grooby | also when you dial to VM, do sip show channels |
| 16:03.15 | Grooby | and see if that channel shows up |
| 16:03.58 | mishehu | has anybody used a Citel Link Handset Gateway for (legacy) PBX phones before? I'm looking for info and user responses to the device... |
| 16:04.50 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 16:06.06 | lorion | Grooby: that rule stop intenet traffic |
| 16:06.11 | lorion | Grooby: brb |
| 16:06.40 | Grooby | hmmmf...guess that's a little different from the firebox |
| 16:07.45 | *** join/#asterisk sricard (sricard@HSE-Montreal-ppp133166.qc.sympatico.ca) |
| 16:08.06 | |Vulture| | oh man.. you guys will get a kick out of this |
| 16:08.12 | Essobi | Hmm. |
| 16:08.18 | |Vulture| | I just created an extension for a one "Lovely Butts" |
| 16:08.18 | Trionnis | trying to register xlite as a sip extension, sip debug shows that I keep getting 403 Forbidden |
| 16:08.22 | Trionnis | help? |
| 16:08.23 | |Vulture| | that is someones REAL name |
| 16:08.30 | Essobi | Haha. |
| 16:08.41 | Grooby | lol |
| 16:08.43 | |Vulture| | I had to confirm that this was not a joke |
| 16:08.45 | Essobi | I knew a guy nameds Bob Head. |
| 16:08.57 | Trionnis | Anita Dick |
| 16:09.07 | Essobi | and I went to school with Anita Douche |
| 16:09.07 | Trionnis | was a woman my mom worked with for many years |
| 16:09.12 | |Vulture| | Essobi: we have a couple of funny ones, "Mike Googe" "Dick Super" and now "Lovely Butts" |
| 16:09.12 | Trionnis | laf |
| 16:09.16 | shmaltz | anybody here dealt with VoipSupply? |
| 16:09.20 | |Vulture| | Essobi: lol |
| 16:09.33 | |Vulture| | shmaltz: yes they are good |
| 16:09.33 | Essobi | Her dad was a teach.. Bob Douche. |
| 16:09.37 | Essobi | teacher |
| 16:09.55 | Essobi | oh and Peter Richard Johnson was in my gym class. :) |
| 16:10.03 | Essobi | That was classic.. he looked like Beavis. |
| 16:10.04 | |Vulture| | lol |
| 16:10.15 | Grooby | how you guys like the IP500 phones? |
| 16:10.17 | |Vulture| | sounds like a porn star name |
| 16:10.22 | Essobi | Heh. |
| 16:10.22 | |Vulture| | love them! |
| 16:10.40 | shmaltz | |Vulture|, thanks, I also think they are good, but I'm having an accounting issue with them, so I was wondering if it's just a mistake (which at the moment I'm treating it as such) or have there been other mistakes like this. |
| 16:11.14 | *** part/#asterisk bladex (~bladex@bonk.personal.engin.umich.edu) |
| 16:11.15 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 16:11.21 | fishboy1669 | hi guys |
| 16:11.22 | |Vulture| | shmaltz: never had any problem with them, is it a big issue? |
| 16:11.53 | Trionnis | so can someone give me a couple hints here? |
| 16:11.57 | Trionnis | pretty please? |
| 16:11.58 | Trionnis | :) |
| 16:12.07 | shmaltz | |Vulture|, it's some extra charge on my MasterCard, and the bookeeper seems not know what I'm talking about |
| 16:12.21 | Trionnis | suprisingly enough, this is the first time I've bothered with a softphone |
| 16:12.42 | Trionnis | and the google, it does nothing! |
| 16:13.19 | shmaltz | Trionnis, take out the password (blank) for know, add it in once it works |
| 16:13.47 | Essobi | My * box is sending SIP Status: 407 Proxy Authentication Required |
| 16:13.49 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-217-180.dsl.scarlet.be) |
| 16:13.55 | Essobi | back to my as5400 |
| 16:13.56 | Essobi | :| |
| 16:13.58 | Essobi | FFS |
| 16:14.09 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 16:14.13 | Trionnis | k |
| 16:14.16 | Trionnis | I'll give it a try |
| 16:15.14 | Grooby | so how are you guys encrypting the sip passwords? |
| 16:15.41 | sricard | anybody haa a SPA-841 phone, i hear a small annoying rumbling noise in the handset and Sipura is not answering my emails |
| 16:16.00 | sricard | it's brand new |
| 16:16.08 | shmaltz | sricard, who you talking on the other end with the spa841? |
| 16:16.43 | shmaltz | sricard, I mean what hardware? |
| 16:17.15 | Grooby | sricard, i'll be getting that phone tomorrow |
| 16:17.49 | shmaltz | Grooby, why do u want them encrypted? |
| 16:18.00 | sricard | asterisk |
| 16:18.14 | sricard | shmaltz: asterisk |
| 16:18.17 | Grooby | so other people don't create sip connection to my asterisk |
| 16:18.33 | shmaltz | sricard, thanks :( what hardware I asked |
| 16:18.34 | Essobi | then use IP restrictions. |
| 16:18.48 | sricard | shmaltz: if i leave a voice mail or do a recording, i can hear it back |
| 16:18.55 | Essobi | I thought all sip passwords flybythewire unencrypted. |
| 16:19.08 | shmaltz | Grooby, you mean if sniffed? |
| 16:19.15 | Trionnis | er |
| 16:19.17 | Trionnis | Feb 24 10:19:03 NOTICE[27131]: chan_sip.c:4874 register_verify: Peer 'brooks' is trying to register, but not configured as host=dynamic |
| 16:19.22 | |Vulture| | only IAX2 can be encrypted from my understanding |
| 16:19.36 | Trionnis | it's not dynamic, it's through ipsec |
| 16:19.41 | |Vulture| | see Trionnis is using IP restrictions |
| 16:19.46 | shmaltz | Trionnis, you got the solution in the error. do host => dynamic |
| 16:19.47 | Hmmhesays | farking sales people |
| 16:19.51 | Trionnis | but it's not |
| 16:19.52 | Trionnis | lol |
| 16:20.19 | |Vulture| | Trionnis: how is your setup working via VPN? |
| 16:20.20 | sricard | shmaltz: asterisk is running on Gentoo (P3 1000 512mb) going through a netgear 100mb/s switch. Is this the info you are looking for? |
| 16:20.27 | Hmmhesays | if I hear "if this box works, this newer revision should work" one more time I"M going to kill someone |
| 16:20.54 | bjohnson | Essobi: there is a guy named Jack Imhof that works for the Ontario Ministry of Agriculture |
| 16:21.06 | |Vulture| | lol |
| 16:21.07 | sricard | anybody from Sipura here? |
| 16:21.21 | Trionnis | |Vulture|: openswan from my m0n0wall to the server |
| 16:21.22 | shmaltz | sricard, nope I meant the phone of the other person. I think b4 you come to conclusions that you should first try it with another phone, and not rely on the VoiceMail |
| 16:21.39 | Trionnis | I think it just connected tho |
| 16:21.40 | Trionnis | :) |
| 16:21.45 | Trionnis | lemme test it |
| 16:22.16 | shmaltz | Trionnis:):):):) |
| 16:22.23 | |Vulture| | Anyone use FreeWorldTel Direct? |
| 16:22.47 | *** join/#asterisk marc_c (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 16:22.50 | sricard | shmaltz: i tried via the PSTN going through an FXO on my TDM400 and it is the same, also tried via an inter pbx link via IAX2 and it is the same |
| 16:23.22 | sricard | shmaltz: there is no dought that it is the phone if it's what you want to get at... |
| 16:23.30 | shmaltz | have you tried it with another * box? sricard. |
| 16:23.52 | sricard | shmaltz: at another location, same |
| 16:24.10 | shmaltz | I hear. I have a SPA 841, but I didn't test it yet. |
| 16:24.13 | Grooby | yeah |
| 16:24.14 | Grooby | if it's sniffed |
| 16:24.59 | shmaltz | Grooby, I don't think that a VOIP account is wort so much that it should be sniffed out, sniffing is just too much work |
| 16:25.28 | |Vulture| | plus you would see it on CDR |
| 16:25.36 | marc_c | whats the difference between govarion and digium cards? |
| 16:25.42 | shmaltz | gtg will be back soon. |
| 16:25.56 | sricard | shmaltz: you planning to test it soon? I sent 2 emaisl in a row to Sipura: 1-SPA-3000 problem; 2-SPA-841, they responded on the first email right away and never responded on the second, emailed again a few days ago and no response yet |
| 16:26.02 | *** join/#asterisk RGi_- (RGi@computer-36-dmz.rgi.as) |
| 16:26.48 | sricard | shmaltz: it seems they are not interested in responding, may be they are aware of the problem and have no resolution for it :-( |
| 16:26.52 | Grooby | ok |
| 16:26.53 | RGi_- | is it in the sip.config file I can set what codec I use ? |
| 16:27.26 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 16:29.54 | Grooby | that's for sip connections |
| 16:30.05 | Grooby | which codecs are allow for sip connections i mean |
| 16:30.09 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 16:30.11 | Mother_ | hi all |
| 16:30.13 | Essobi | Well.. if anyone cares to look.. the tethereal dumps, sip debug peer and sip.conf entries I'm using are at http://spider.teledvance.com/sip-debug.txt |
| 16:30.21 | Mother_ | anyone use a 7960 through a double NAT? |
| 16:30.48 | Mother_ | I have a problem that no matter how I configure the RTP port ranges, it always picks a port outside the range and thus fails |
| 16:31.15 | mishehu | SIP and double nat are bad. |
| 16:31.43 | bjohnson | sricard: what was the spa 3k issue? |
| 16:31.48 | Mother_ | yeah I know, but this is a single phone inside a NAT talking to the * on the other NAT, so in theory, if I map the UDP port range on both routers to the fixed IPs it should work |
| 16:32.08 | bjohnson | mishehu: few people have it working |
| 16:32.24 | Grooby | Mother, that's how my setup is |
| 16:32.35 | Mother_ | i.e. in rtp.conf I limit the range 10000 to 10020, the same is configured on the 7960, and ports 10000->10020 are routed to the respective IPs inside the NATs |
| 16:32.42 | bjohnson | mishehu: zeeek got it to work by specifing the remote LAN ip address and the remote LAN subnet mask |
| 16:32.43 | *** part/#asterisk oej (~oej@ua-213-115-215-100.cust.bredbandsbolaget.se) |
| 16:32.50 | aminorex | SIP is sheer madness |
| 16:32.55 | bjohnson | I got it to work by using FWD as a go-between |
| 16:32.56 | |Vulture| | by double NAT you mean SIP phone out of 1 NAT and into a NAT with a * behind it? |
| 16:33.01 | bjohnson | yes |
| 16:33.01 | *** join/#asterisk JerJer (~JerJer@dsl-106-170.che.centurytel.net) |
| 16:33.19 | |Vulture| | I use WRT54g routers and have no problems |
| 16:33.22 | bjohnson | actually in my case .. * behind nat .. remote sip phone behind 2 nat routers |
| 16:33.23 | Mother_ | |Vulture|: yes |
| 16:33.36 | JerJer | hell I have Asterisk running on my WRT54gs |
| 16:33.38 | JerJer | as my home PBX |
| 16:33.40 | Grooby | Mother, I have the same setup as vulture and have no problem |
| 16:33.48 | |Vulture| | JerJer: they rock! :) |
| 16:33.51 | Mother_ | 7960 <- NAT -> DSL <-> NAT <-> * |
| 16:33.54 | aminorex | that would solve the SIP problem |
| 16:34.03 | JerJer | maybe more like Jam |
| 16:34.03 | Grooby | i didn't have to bother with port forward on the phone side tho |
| 16:34.03 | |Vulture| | Mother_: wow that looks like my setup |
| 16:34.06 | Grooby | just the asterisk side |
| 16:34.10 | bjohnson | mainly because it isn't double nat then |
| 16:34.18 | |Vulture| | JerJer: you use the Sveasoft firmware? |
| 16:34.19 | JerJer | they would rock if it was a x86 processor |
| 16:34.24 | cbachman | JerJer, I was looking at doing the same thing under OpenWrt on a Motorola wr850g (similar to a wrt54g) |
| 16:34.26 | JerJer | no FPU is kinda suckky |
| 16:34.34 | JerJer | OpenWRT |
| 16:34.42 | |Vulture| | ah |
| 16:35.17 | Grooby | jerjer, how's * on wrt? |
| 16:35.25 | Mother_ | then in the SIP debbuging, why do I get this: Peer audio RTP is at port 80.32.X.X:27466 |
| 16:35.33 | JerJer | usable for the average home user |
| 16:35.40 | JerJer | i've had 4 calls up at once |
| 16:35.51 | sivana | ~seen sixtel |
| 16:35.53 | jbot | sixtel <sixtel@sixTel.iax.cc> was last seen on IRC in channel #asterisk, 49d 11h 17m 8s ago, saying: 'no such host, not in sip.conf right'. |
| 16:35.53 | JerJer | transcoding ulaw to gsm |
| 16:35.58 | Grooby | nice ok |
| 16:37.04 | Mother_ | in sip.conf I also have host=dynamic, because if I specify the remote IP in host it cannot register |
| 16:37.09 | Grooby | Mother: 80.32.x.x is your phone IP? |
| 16:37.13 | Mother_ | yes |
| 16:37.28 | Grooby | and nat=yes for the phone right? |
| 16:37.37 | Mother_ | yep, in sip.conf nat=yes |
| 16:37.47 | mishehu | anybody using speex and can help give some insight? I've been trying speex 1.0.4, and speex works with asterisk when the originating extension is iax, even if it's not speex to start with (in this case, it's an iaxy, g711ulaw). speex always works in this case. However, if the originating extension is using SIP g711ulaw, transcoded to speex, audio gets all chopped and 80% lost... |
| 16:37.54 | Grooby | (for that extension or under [general]?) |
| 16:38.00 | Mother_ | for that extension |
| 16:38.31 | Grooby | wacky |
| 16:38.54 | Mother_ | type=peer, host=dynamic, reinvite=no, canreinvite=no, nat=yes |
| 16:39.03 | Grooby | mishehu, i tried speex 1.0.4 yesterday and was having the same problem |
| 16:39.14 | mishehu | Grooby: have you tried speex 1.1.6 ? |
| 16:39.18 | JerJer | Mother_: reinvite=no is not an option |
| 16:39.25 | Grooby | the beta? no |
| 16:39.29 | JerJer | and why don't you want them to re-invite anways? |
| 16:39.39 | bjohnson | the ability to hack the WRT54gs has really made it popular. I wonder if router manufacturers have learned anything from that? |
| 16:39.56 | Grooby | isn't it canreinvite=no? |
| 16:40.14 | Mother_ | well I had both, this was taken from an example configuration |
| 16:40.15 | mishehu | I replaced speex 1.0.4 with 1.1.6 for test, and even though asterisk is dynamically linked to speex libraries, it seems that I might need to rebuild asterisk against 1.1.6, as asterisk crashes whenever speex is negoatiate then. |
| 16:40.22 | Mother_ | should I remove the reinvite? |
| 16:40.24 | Trionnis | bjohnson: I doubt it |
| 16:40.24 | JerJer | Mother_: then the sample is wrong |
| 16:40.26 | JerJer | look at the source |
| 16:40.29 | Mother_ | ok |
| 16:40.34 | Grooby | mishehu, did you ldconfig? |
| 16:40.42 | Grooby | i had to do that else my * crashes when tryiing to load the so |
| 16:40.43 | mishehu | Grooby: of course. |
| 16:40.59 | mishehu | ldconfig, and restarted asterisk. |
| 16:41.02 | Grooby | mishehu, I pretty much gave up on it..hehehehe |
| 16:41.12 | Grooby | ilbc works great for me so I am sticking w/ that |
| 16:41.14 | Essobi | Hey JerJer.. Can you look at something real quick at tell me if I'm missing something painfully obvious? http://spider.teledvance.com/sip-debug.txt |
| 16:41.45 | yasha | Is it possible to have a follow me try to reach 1 * ext first, then try Cell Phone, then if no answer come back to 2nd * ext and try the 2nd cell phone and then if no answer go to * VM box? |
| 16:41.50 | mishehu | Grooby: it's a shame. my hardware is good enough that transcoding from pretty much any codec to speex is only 28 to 30 ms delay, and speex at gsm bandwidth sounds like g711ulaw... |
| 16:42.03 | shido6 | peers dont have contexts Essobi |
| 16:42.04 | Grooby | really? |
| 16:42.15 | Grooby | maybe another try later down the road |
| 16:42.25 | Essobi | shido6 What? Inbound peers don't HAVE a context? |
| 16:42.34 | tzanger | Essobi: an inbound peer is a user |
| 16:42.37 | shido6 | what? |
| 16:42.44 | shido6 | inbound peers? you mean users... ? |
| 16:42.49 | |Vulture| | I just made a call with ilbc... wow great quality |
| 16:42.52 | Essobi | No shit? |
| 16:42.54 | shido6 | and users dont need hosts |
| 16:43.01 | Mother_ | is there anywhere else where RTP port range is restricted other than rtp.conf? |
| 16:43.05 | mishehu | Grooby: yeah, I think gsm sounds kind of crappy... I really couldn't tell the diff between speex and g711ulaw at gsm bandwidth... |
| 16:43.05 | Essobi | Baah. |
| 16:43.22 | Essobi | So.. What, have a user with no name and password just an IP set? |
| 16:43.25 | ionix | Hey, anyone has ANY IDEA on how I can fill in a name when I have a phone number ? Trying to find a way to access the RBOC database. |
| 16:43.36 | |Vulture| | do you guys use "trunk=yes" in your iax.conf? |
| 16:43.50 | Mother_ | |Vulture|: I tried, but it refused to work |
| 16:44.09 | Mother_ | was trying to trunk four PSTN over IAX to another * |
| 16:44.11 | Grooby | Mother, the nat on the phone side is port forwarding you said? |
| 16:44.19 | Mother_ | Grooby: yes |
| 16:44.40 | Mother_ | but what really puzzles me is that the RTP port it picks is outside the range limits |
| 16:44.42 | Grooby | is there a reason for that? if the phone's iniating the connection and nat=yes in sip.conf |
| 16:45.00 | Grooby | the router should be smart enough to keep that connection open |
| 16:45.13 | Mother_ | Grooby: so the phone will start the RTP port towards the * first then? |
| 16:45.20 | Grooby | yeah |
| 16:45.23 | Mother_ | OK |
| 16:45.33 | JerJer | shido6: users don't ~need~ hosts, but ~can~ use hosts to authenticate via IP |
| 16:45.45 | Mother_ | still can't figure out why it picks 27000something when it's limited to 10000-10020 |
| 16:45.52 | shido6 | yes. |
| 16:46.03 | Grooby | i am guessing 10000-10020 is on the * side |
| 16:46.07 | Grooby | the phone side can be anything |
| 16:46.15 | Grooby | might be something in the phone configuration |
| 16:46.36 | |Vulture| | you can set RTP ports in * config |
| 16:46.46 | Grooby | but that's RTP port for * right? |
| 16:46.52 | |Vulture| | correct |
| 16:46.52 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 16:46.55 | Grooby | not the phone |
| 16:47.04 | |Vulture| | RTP for the phone usually isn't configurable |
| 16:47.05 | Grooby | Mother, did you add canreinvite=no? |
| 16:47.15 | Mother_ | |Vulture|: I am doing so, in rtp.conf I have limited 10000-10020, and the same on the phone, and in the sip debug output I see it's picking 27470, 27472, etc |
| 16:47.18 | mishehu | |Vulture|: I use trunk=yes in iax.conf |
| 16:47.27 | Grooby | maybe asterisk is reinviting and send SIP connection from your provider to phone |
| 16:47.30 | |Vulture| | mishehu: notice any difference? |
| 16:47.30 | Essobi | So I shouldn't use a single friend entry? I need a peer and a user for each router? |
| 16:47.44 | Mother_ | Grooby: yes, it was like that, let me check again |
| 16:47.52 | mishehu | |Vulture|: I don't do enough calling to notice. (normally only one channel open) |
| 16:48.18 | yasha | GUYS: Is it possible to have a follow me try to reach 1 * ext first, then try Cell Phone, then if no answer come back to 2nd * ext and try the 2nd cell phone and then if no answer go to * VM box? |
| 16:48.40 | mishehu | yasha: you can do whatever you like |
| 16:48.55 | mishehu | from the extensions conf at least |
| 16:49.07 | yasha | So it is possible to bring the call back from cell phone back to *? |
| 16:49.14 | JerJer | yasha: sure |
| 16:49.32 | Essobi | I guess I'm missing something blindingly important. |
| 16:49.41 | JerJer | type=friend is evil |
| 16:49.47 | JerJer | and it WILL bite you, eventually |
| 16:49.54 | Essobi | How the hell do you get an as5400 server to allow a call to an * box? |
| 16:50.13 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 16:50.16 | JerJer | Essobi: SIP |
| 16:50.18 | yasha | Can someone please give me an example of "follow me" that would asnwer * ext, then call cell phone and then if no answer, back to an * VM? |
| 16:51.06 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 16:51.08 | heison | essobi: http://lists.digium.com/pipermail/asterisk-users/2004-February/036180.html |
| 16:52.41 | Essobi | so... My dial-out (peer) works fine.. |
| 16:52.50 | eipi | i have problem authenticanting SIP from database (sip_friends). ANyone have working and want help me? |
| 16:54.53 | yasha | Guys, anyone? |
| 16:55.33 | Essobi | If I'm reading this right... |
| 16:56.02 | Essobi | the sip.conf page on voip-info says.. the From header recieved is matched to the type=user entry.. |
| 16:56.32 | Essobi | My as5400 is sending From: <sip:THEANIOFTHECALLER@192.168.0.1>;tag=7DBDED74-59E |
| 16:56.56 | *** join/#asterisk demon|werk (~demonrage@dsl017-022-045.chi1.dsl.speakeasy.net) |
| 16:56.56 | Essobi | So I need to have a seperate user for every fricking inbound ANI? That's wack. |
| 16:57.42 | Essobi | Oh.. it should just match on the host= shouldn't it? |
| 16:57.48 | shido6 | u can |
| 16:58.37 | JerJer | there should be a way to set an actual username |
| 16:58.42 | JerJer | and secret |
| 17:00.18 | Essobi | I havn't found any documentation confirming that on cisco.com |
| 17:00.27 | Essobi | I assumed as much myself. |
| 17:00.49 | Essobi | Cause it'd be nice to have different DID's from one box land in different contexts. |
| 17:01.03 | *** join/#asterisk jalsot (~tamas@195.56.44.83) |
| 17:01.17 | *** join/#asterisk DoCatwork (~doc@pD951C53C.dip.t-dialin.net) |
| 17:01.19 | bjohnson | yasha: wiki |
| 17:01.34 | bjohnson | basically dial with a timeout set |
| 17:01.39 | DoCatwork | hello peoples |
| 17:01.43 | bjohnson | check the superdial macro on the wiki |
| 17:02.00 | yasha | bjohnson: Thanks buddy... |
| 17:02.07 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:02.30 | DoCatwork | can someone say me if asterisk can route incoming isdncalls to annother isdncard?? |
| 17:03.33 | roamer323 | anyone familiar with the DOCSIS MTAs and the cable digital phones that are being rolled out all over by TimeWarner, Shaw in Canada, etc? |
| 17:03.44 | DoCatwork | i mean can it simulate an isdnbox an send incoming calls to intern numbers like 12 15 etc |
| 17:05.40 | ScarletCrusader | Excuse me, does anyone have experience with the Budge Tone 100 Phones from the Test Kit? i'm just trying to get the phone to talk to the PBX system |
| 17:06.27 | DoCatwork | you must add a user for the phone and login with this logindates to your asterisk box |
| 17:06.33 | ScarletCrusader | The PBX is Operational and I've installed the sample scripts |
| 17:06.43 | shido6 | u have the pword to the phone ScarletCrusader? |
| 17:06.49 | ScarletCrusader | yes |
| 17:07.05 | ScarletCrusader | shido6: yes |
| 17:07.15 | ScarletCrusader | DoCatwork: THanks |
| 17:09.46 | DoCatwork | i come back later if some more peoples are there |
| 17:10.30 | km- | Asterisk rocks. |
| 17:10.39 | km- | just in case no one else realized it. |
| 17:10.42 | Grooby | lol |
| 17:10.45 | ScarletCrusader | lol |
| 17:11.35 | phreak | I know this isn't the spot-on subject for this channel, but since people here _might_ have had experience with such things I'll try: What do you call the equipment you have on your feet to climb up in telephone/power-poles? Directly translated from swedish it's 'pole shoes', but it doesn't seem to be the right term in english. |
| 17:11.58 | outtolunc | pole climbers <G> |
| 17:12.07 | trig | climbing spikes? |
| 17:12.12 | trig | http://www.esscodist.com/shopsite_sc/store/html/page6.html |
| 17:13.09 | phreak | trig: No, but similiar. The things I mean is very much more reliable, you can rest on them. But since you have it around the pole, no branches/whatever can be in the way. |
| 17:13.28 | *** join/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net) |
| 17:13.55 | outtolunc | http://www.buckinghammfg.com/linemen/pcpc.html |
| 17:14.02 | km- | they also have kind of like a belt strap that they use to climb as well |
| 17:14.03 | outtolunc | they are called pole climbers |
| 17:14.03 | phreak | outtolunc: From what I found at first on google it doesnt seem to be the right thing too, but thanks you too for some pointers, some site might have them both :) |
| 17:14.16 | *** join/#asterisk Tili (~Tili@202-133-65-121-dialup.sat.net.pk) |
| 17:14.23 | Mother_ | Grooby: in the Cisco, where did you configure the RTP port range? |
| 17:14.32 | Mother_ | in SIPDefault.cnf? |
| 17:14.45 | Grooby | lol |
| 17:14.45 | *** join/#asterisk Jackthe (~jesse@thewhitehouse.adsl.utwente.nl) |
| 17:14.46 | Grooby | i have no clue |
| 17:14.48 | outtolunc | that link you did is for gaff guards |
| 17:14.51 | Grooby | never had cisco phones |
| 17:14.59 | eipi | i have problem authenticanting SIP friends from database (sip_friends). ANyone have working and want help me? |
| 17:15.17 | km- | Buckingham Pole Climbers, also known as pole spikes, leg irons and pole hooks, are widely used in the CATV, Telecommunication and Electrical industries. Pole Climbers have relatively short gaffs |
| 17:16.31 | *** join/#asterisk Ad-Hoc (~ad-hoc@62.1.246.83) |
| 17:16.50 | *** part/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net) |
| 17:16.57 | phreak | I guess I'm looking for a too specifik thing, and the word is more general. So I'll check these words out, thanks y'all. |
| 17:17.00 | greg_work | is that still used? around here everyone seems to use boom trucks |
| 17:17.33 | phreak | http://www.linjedon.se/images/stolpsko.jpg <-- There you have exactly what I'm looking for. |
| 17:18.21 | tzanger | phreak: wtf is that? |
| 17:18.35 | tzanger | ahh pole climbers |
| 17:19.05 | Essobi | Man.. This is aggravating the crap out of me. |
| 17:19.24 | Essobi | I can't get a sip type=user that'll let my as5400 land on it, no matter what I do. |
| 17:20.11 | tzanger | outtolunc: aha |
| 17:20.50 | outtolunc | i honestly have never seen a pair like that |
| 17:21.36 | bjohnson | roamer323: what's the question? |
| 17:22.01 | *** join/#asterisk Shaneful (~sharper@d154-20-37-11.bchsia.telus.net) |
| 17:22.46 | Essobi | I'm getting a 404 not found error from * to my as5400. |
| 17:25.03 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
| 17:25.32 | *** join/#asterisk Signuts (~signuts@209.172.11.54) |
| 17:26.24 | Signuts | Hey all, i've got a probably simple question about the VoiceMail() app. When a user presses * it throw's me into the 'a' extension, and i'm going to exec VoiceMailMain(u${EXTEN}@mycontext), but ${EXTEN} doesn't seem to be set, as it still asks for the mailbox # |
| 17:26.57 | Signuts | I am probably using the wrong variable, but can't seem to locate the proper one to use |
| 17:31.14 | *** join/#asterisk zapa (~zapa@201.135.161.28) |
| 17:31.38 | km- | tzanger: is therre something special I need to do to make asterisk keep the ANI? |
| 17:31.52 | km- | tzanger: I'm noticing that I'm not getting caller ID info on any calls coming into the system |
| 17:32.16 | km- | watching gastman, it says "<unknown>" on the caller id |
| 17:32.19 | *** join/#asterisk LarsAC (~chatzilla@pD9501019.dip0.t-ipconnect.de) |
| 17:32.30 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:32.52 | LarsAC | how can I receive calls from two sip providers? |
| 17:33.04 | LarsAC | it seems, sip.conf allows only one in the general section |
| 17:33.17 | km- | huh? You can register as many as you'd like |
| 17:33.25 | km- | my box at home had 6 seperate providers |
| 17:33.54 | LarsAC | so you simply use several register lines ? |
| 17:33.54 | km- | right |
| 17:34.01 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 17:34.05 | km- | define multiple friend entries and then use multiple register lines |
| 17:34.35 | LarsAC | I dont have a fried entry at all for now |
| 17:34.45 | bjohnson | type=users for incoming calls |
| 17:34.47 | *** join/#asterisk EC-ASP (~alfredo@Intelideas-Avanzia.Mesena.MAD.ES.INTELIDEAS.NET) |
| 17:34.49 | EC-ASP | Hi |
| 17:35.04 | EC-ASP | I'm coming in search of clue |
| 17:35.29 | bjohnson | 42 |
| 17:35.30 | EC-ASP | I'm running a Debian sarge with the latest drivers |
| 17:35.33 | EC-ASP | for a TE110 |
| 17:35.37 | lorion | I need a recommendation for a free SIP Softphone to test with, not X-Lite. |
| 17:35.40 | EC-ASP | it has been working well |
| 17:35.41 | bjohnson | uh sorry .. jumped straight to the answer |
| 17:35.47 | LarsAC | so the register stuff should rather go to individual sections rather than to general ? |
| 17:35.48 | EC-ASP | :) |
| 17:35.52 | km- | lorion: whats the problem with x-lite? |
| 17:36.01 | km- | larsac: no, the register lines show up under general |
| 17:36.12 | EC-ASP | problem, after upgrading kernel, module no longer loads |
| 17:36.17 | km- | larsac: you just need to define user entries for the lines you register |
| 17:36.20 | EC-ASP | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 17:36.23 | km- | ec-asp: recompile the drivers |
| 17:36.27 | lorion | km-: i am getting one way audio and a message that reade Maximum retries exceeded on call. |
| 17:36.27 | EC-ASP | I did |
| 17:36.36 | km- | lorion: are you using nat? |
| 17:36.37 | LarsAC | km-: to dial out or to be called ? |
| 17:36.41 | EC-ASP | checked out the latest drivers as well |
| 17:36.43 | EC-ASP | out of cvs |
| 17:36.52 | lorion | km-: yes |
| 17:37.03 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 17:37.05 | km- | ec-asp: I'm running debian sarge on a 2.6.10 system and zaptel and wct4xxp load fine |
| 17:37.09 | lorion | km-: I created a static nat for testing.. still no luck |
| 17:37.29 | EC-ASP | Well, actually they load but ztcfg complains and Asterisk doesn't start |
| 17:37.30 | LarsAC | ec-asp: 2.6.9-smp works fine too |
| 17:37.35 | Mother_ | hah |
| 17:37.39 | km- | lorion: whenever i've had problems with x-lite, nat has been to blame. I use x-lite here at the office without NAT and it works fine. |
| 17:37.48 | Mother_ | if I map the entire port range the Cisco pretends to use, it works |
| 17:37.55 | km- | ec-asp: does the console show "Found a wildcard" type of message? |
| 17:38.10 | eipi | i have problem authenticanting SIP from database (sip_friends). ANyone have working and want help me? |
| 17:38.21 | EC-ASP | km-, console only says something when I load zaptel |
| 17:38.21 | lorion | km-: I am positive it is a nat issue. |
| 17:38.31 | EC-ASP | but wcte11xp is silent |
| 17:38.37 | km- | lorion: have you thought about trying an iax soft phone as opposed to a sip? |
| 17:38.47 | km- | iax is less nat-annoyed |
| 17:39.12 | LarsAC | km-: is there only one context in which I can handle incoming calls ? |
| 17:39.14 | lorion | km-: no I am pretty new to this. |
| 17:39.42 | km- | larsac: there's one context that sip calls dump into, but you can define the extension that the calls come in from |
| 17:39.44 | lorion | km-: I am assuming an iax phone would be configured in the IAX.conf file |
| 17:39.48 | *** join/#asterisk Jearil (~Jearil@216-224-56-213.client.dsl.net) |
| 17:39.57 | LarsAC | km-: okay, starting to get it |
| 17:40.13 | km- | if you say register=> yourname:yourpass@sip.provider.com/1000, all incoming calls will go to 1000@sipcontext |
| 17:40.17 | km- | or something to that effect |
| 17:40.26 | km- | the /1000 part is what I'm trying to illustrate |
| 17:40.59 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 17:40.59 | *** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
| 17:41.03 | km- | I had a context called [voipincoming] where I defined multiple extensions and then sent calls out from that extension |
| 17:41.08 | km- | larsac: yeah. |
| 17:41.10 | lorion | BRB Lunch... |
| 17:41.15 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 17:41.39 | EC-ASP | I'm a bit annoyed at this issue, it's not the first time |
| 17:41.53 | loud | 2.6.10 problems ? |
| 17:41.56 | EC-ASP | kind of frustrating, an error that doesn't give a clue as to what's wrong |
| 17:41.57 | EC-ASP | yep |
| 17:42.17 | EC-ASP | 2.6.9 fine, 2.6.10 no way |
| 17:42.23 | loud | yep, happened to me too |
| 17:42.32 | loud | fc3 ? |
| 17:42.43 | EC-ASP | sorry, fc 3? |
| 17:42.48 | loud | fedora core 3 ? |
| 17:42.56 | EC-ASP | nope, Debian 3.1 (sarge) |
| 17:43.00 | km- | I dunno what the issue is with you guys, I'm using 2.6.10 with my te405p |
| 17:43.08 | EC-ASP | stock kernel? |
| 17:43.09 | km- | detects the card, spans all come up just fine |
| 17:43.12 | loud | i assume you have edited the udev stuff already |
| 17:43.12 | km- | yep, stock kernel |
| 17:43.14 | zapa | hi all,can anyone help me, i am having this error, i don´t know wha coul be wrong chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 17:43.14 | EC-ASP | maybe a library function? |
| 17:43.25 | marc_c | can one T1 PRI be shared among more than one box? |
| 17:43.25 | EC-ASP | km-, can you lsmod for us? |
| 17:43.46 | km- | Module Size Used by |
| 17:43.46 | km- | wct4xxp 71232 0 |
| 17:43.46 | km- | zaptel 225188 59 wct4xxp |
| 17:43.46 | km- | pbx01:/etc/asterisk# |
| 17:43.58 | *** join/#asterisk lyroy (~info@modemcable007.224-203-24.mc.videotron.ca) |
| 17:44.10 | EC-ASP | So, not crc or anything |
| 17:44.15 | EC-ASP | Hmmmm... |
| 17:44.28 | km- | I have those compiled into the kernel |
| 17:44.34 | EC-ASP | ah |
| 17:44.46 | km- | pbx01:/etc/asterisk# zcat /proc/config.gz |grep -i crc |
| 17:44.46 | km- | CONFIG_CRYPTO_CRC32C=y |
| 17:44.46 | km- | CONFIG_CRC_CCITT=y |
| 17:44.46 | km- | CONFIG_CRC32=y |
| 17:44.46 | km- | CONFIG_LIBCRC32C=y |
| 17:44.56 | loud | oh wait |
| 17:44.58 | loud | zaptel 191236 55 wcfxo,wctdm,wct1xxp |
| 17:45.03 | loud | 2.6.10-1.741_FC3smp |
| 17:45.05 | loud | it does work. |
| 17:45.16 | km- | bbiafm |
| 17:45.28 | outtolunc | 2.6.10-1.14_FC2 works also |
| 17:45.28 | eipi | i have problem authenticanting SIP from database (sip_friends). ANyone have working and want to help me? (i configured extconfig.conf, created tables, have unixodbc working). Voicemail, voicemessages and extensions are working from db |
| 17:46.18 | *** join/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 17:46.41 | lyroy | Someone know what is the phone number for DIDnumbers please. |
| 17:46.52 | *** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com) |
| 17:47.03 | loud | whiskey tango foxtrot ? |
| 17:47.46 | *** join/#asterisk ChrisRouse (~crouse@67.131.247.187) |
| 17:47.47 | eipi | tango yes... foxtrot maybe... whiskey on the rocks please |
| 17:48.21 | Beirdo | eipi: better not be scotch on the rocks, that's heresy |
| 17:48.23 | loud | :D |
| 17:48.36 | alt_phil | Hey all. Anyone out there have issues with getting a Digium Wildcard TE110P T1 card syncing to the telco as your clock source? I've set my timersource to 1 for the span, yet the card still reports its sync source as internally synced and we're getting tons of frame slips. |
| 17:48.38 | Signuts | anyone familiar with asterisk VoiceMail() , When someone presses '#' it forwards me to the 'a' extension, but I seem to be losing the # dialed, (I don't want the user to have to enter their mailbox #) |
| 17:48.54 | marc_c | can a T1 PRI line be shared among more than one server? |
| 17:49.13 | loud | marc_c, if you do iax trunks, maybe |
| 17:49.29 | ChrisRouse | Signuts: Maybe I am not understanding the question. Is your problem that you want the person to not have to dial in their extension? |
| 17:49.40 | pcm | alt_phill: digium tech support |
| 17:49.48 | pcm | alt_phill: call :) |
| 17:49.49 | ChrisRouse | Signuts: Like having Asterisk know what exten you are dialing from? |
| 17:49.51 | xeet2 | alt_phil: if it can't sync, it will try internally clocked |
| 17:50.12 | xeet2 | alt_phil: make sure you are set to the right framing and linecode, that can make it not sync up |
| 17:50.20 | lyroy | Does someone know what is the phone of Bell Canada, I need get info for DID numbers and I can get that number on the website? |
| 17:50.43 | Signuts | ChrisRouse, basically I call voicemail() and if the users presses * it takes them to voicemailMain() in the 'a' extension. asterisk asks for the mailbox number, when i'm already calling voicemailmain like, VoiceMailMain(${EXTEN}@vmcontext) |
| 17:51.07 | ChrisRouse | Signuts: Let me check for you. |
| 17:51.08 | Signuts | ${EXTEN} expands to 'a' I think, it is not infact their phone number (which is also their mailbox #) |
| 17:51.25 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 17:51.34 | Signuts | ChrisRouse, it'd be niec if I could do a variable dump on a call. it'd be nice to see what's set =) |
| 17:51.59 | ChrisRouse | Signuts: You can look up the NoOp() command |
| 17:52.32 | Signuts | I did find that, but it's a pain to make up variables names that mighthelp. I checked NoOP(${ARG1}) and NoOp(${ARG2}) with no avail |
| 17:54.18 | ChrisRouse | Signuts: Do you know what ${EXTEN} is giving you back? |
| 17:54.38 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:55.18 | Signuts | ${EXTEN} seems to be printing 'a' |
| 17:55.50 | ChrisRouse | Signuts: What about {$CALLERID}? |
| 17:56.23 | ChrisRouse | Signuts: It seems that when you transfer to the "a" extension that it is overriding the ${EXTEN} variable. |
| 17:56.45 | Signuts | yeah, i'll check CALLERID, but i'm not necessarily calling from my extension, so it'd fill in a bad mailbox # |
| 17:57.50 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 17:57.55 | ChrisRouse | Signuts: Then how would you stop the user from having to type their extension? |
| 17:58.10 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
| 17:58.11 | *** join/#asterisk empire667 (~user1@h71032.upc-h.chello.nl) |
| 17:58.12 | Signuts | ChrisRouse, ${CALLERID} fills in properly with the # i'm coming from. |
| 17:58.23 | ChrisRouse | Signuts: Unless I misunderstood |
| 17:58.26 | Signuts | ChrisRouse, they variable i'm trying to get is the phone number dialed |
| 17:58.45 | |Vulture| | Signuts: then you want {$EXTEN} |
| 17:58.48 | |Vulture| | urg |
| 17:58.54 | |Vulture| | ${EXTEN} |
| 17:59.18 | *** join/#asterisk [cc]smart (~smart@62.65.149.158) |
| 17:59.25 | Hmmhesays | are sipura's spa2000's reliable? |
| 17:59.38 | alt_phil | I'd say so. I've got 20 of 'em at work |
| 17:59.39 | Signuts | they dial their voicemail box, say it's 1234, in which I call VoiceMail(1234), but in VoiceMail app, if they press *, it jumps to 'a' extension, in the a extension, ${EXTEN} is set to 'a', when in fact it should be the number dialed. I'm running asterisk 1.0.5 |
| 17:59.42 | |Vulture| | Hmmhesays: for what? |
| 17:59.43 | loud | absolutely. |
| 17:59.49 | alt_phil | the spa2000's aren't bad at all |
| 17:59.52 | |Vulture| | for phones they are great |
| 17:59.57 | Hmmhesays | want to put them as off site extensions for home users |
| 18:00.00 | |Vulture| | for fax they are pretty good |
| 18:00.06 | Hmmhesays | to their main office pbx |
| 18:00.07 | |Vulture| | but then again fax is so so |
| 18:00.20 | |Vulture| | Hmmhesays: yea good choice |
| 18:00.28 | Hmmhesays | that's what I needed to know |
| 18:00.31 | Sedorox | Signuts: for voicemail.. use # instead of * to pass the extention |
| 18:00.34 | alt_phil | I dunno about fax on the 2000's. We had tons of problems with them and ended up going with a 4 port TDM card for our faxes |
| 18:00.39 | ChrisRouse | |Vulture| Does ${EXTEN} get replaced when the script transfers to an extension? |
| 18:00.49 | Hmmhesays | these users wont' be doing fax, so we're golden |
| 18:00.59 | ChrisRouse | Signuts: Why do you have them hitting *? or is that what voicemail does? |
| 18:01.12 | Juggie | it says on the wiki that voicemail is limited to 99 messages per inbox, does anyone know if that is still the case? |
| 18:01.20 | |Vulture| | ChrisRouse: not sure, you might need to store it as a global variable if your passing it between contexts |
| 18:01.22 | Signuts | it's built into the VoiceMail() app, '*' is my only option |
| 18:01.41 | alt_phil | yes Juggie, it is. I have to run an autodelete script to keep my messages down. |
| 18:01.44 | *** join/#asterisk deRost (~deRost@054.209-89-66-0.interbaun.com) |
| 18:01.58 | ChrisRouse | |Vulture| Ohh variables. did not know that you could create your own. Must investigate. |
| 18:02.00 | Sedorox | hmmm |
| 18:02.02 | Hmmhesays | now I just have to make sure 2x TDM400P's with 4 fxo modules a piece will work fine in a 2.8ghz machine |
| 18:02.30 | |Vulture| | Hmmhesays: why not get a T100P and get a frac t1 instead of 8 POTS? |
| 18:02.32 | Signuts | So I could call setVar before I exec' voicemail(), perhaps that'll do it |
| 18:02.38 | alt_phil | I've only got one, in a 2.8 machine, it runs good. So can't really help ya there. |
| 18:02.45 | ChrisRouse | Signuts: The documentation for VoicemailMain tells me that * is for help. |
| 18:03.00 | Hmmhesays | |Vulture| because they have 8 pots lines, and that's pretty much written in stone at this location |
| 18:03.16 | ChrisRouse | Signuts: You should already be in the voicemail box at this point. |
| 18:03.16 | Hmmhesays | this is in the middle of bumfark north dakota |
| 18:03.22 | Juggie | alt_phil, do you know if the 99 limit exists in CVS head still? |
| 18:03.26 | |Vulture| | ah, well I know 1 TDM400 with 4 FXO works great on my Dell SC420 2.8s |
| 18:03.50 | |Vulture| | Hmmhesays: and 2.8 is overkill if your not using any compression for a system like that |
| 18:04.05 | Hmmhesays | yeah a 2.8ghz machine should be able to handle 8 calls.... voice compression for the 5-6 offsite extensions |
| 18:04.08 | alt_phil | Juggie: I know it does in CVS-v1-0-02 |
| 18:04.09 | Signuts | ChrisRouse, I may be confusing us, there are two applications here. pressing star in VoiceMail() takes me to the 'a' extension, at which point I call VoiceMailMain() |
| 18:04.11 | |Vulture| | but I've always liked overkill vs too little machine |
| 18:04.27 | Hmmhesays | yes... i don't need to be hearing back from these people after i've done this install |
| 18:04.30 | |Vulture| | Hmmhesays: yes, you will be fine |
| 18:05.02 | ChrisRouse | Signuts: Ok, so * is your extension definition. So if you are not calling from your own extension then how do you get the voicemail box. |
| 18:05.30 | |Vulture| | Hmmhesays: I have a dual xeon 2.8 for 46 lines with 7 * boxes interconnected with ilibc between and SIP to phones, the tests show it as overkill |
| 18:05.40 | Hmmhesays | nice |
| 18:06.16 | |Vulture| | $1400 after tax and S&H... I thought it was cheap as hell... 2U from dell |
| 18:06.25 | Sedorox | hmm |
| 18:06.31 | Sedorox | what model? |
| 18:06.34 | |Vulture| | 2850 |
| 18:06.40 | Sedorox | hmmm |
| 18:07.04 | |Vulture| | 1GB ram and 73GIG UW320 drive, this is a test model |
| 18:07.14 | EC-ASP | arg, I'm stuck |
| 18:07.20 | |Vulture| | the production one will be just around 2k with Raid 1 and redundant PSUs |
| 18:07.20 | EC-ASP | now 2.6.9 doesn't work also |
| 18:07.26 | *** part/#asterisk LenzX (~lenz-ml@213-92-107-83.f5.ngi.it) |
| 18:07.40 | EC-ASP | why can't they get these drivers straight? :( |
| 18:07.59 | EC-ASP | I have never seen those problems with any other hardware |
| 18:08.11 | EC-ASP | (end of rant) |
| 18:08.20 | outtolunc | make sure you uname -r actually equals your *current* sourcecode |
| 18:08.32 | EC-ASP | verified |
| 18:08.40 | EC-ASP | uname -r is 2.6.10, source is 2.6.10 |
| 18:08.41 | outtolunc | and the error you get US? |
| 18:08.47 | |Vulture| | 2.6.9-1.667 works fine here |
| 18:08.54 | outtolunc | er is? |
| 18:08.55 | EC-ASP | s5:~# modprobe wcte11xp |
| 18:08.56 | EC-ASP | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 18:08.56 | EC-ASP | FATAL: Error running install command for wcte11xp |
| 18:09.07 | |Vulture| | lspci -vv |
| 18:09.09 | outtolunc | lsmod |
| 18:09.31 | EC-ASP | 0000:00:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 18:09.31 | EC-ASP | <PROTECTED> |
| 18:09.31 | EC-ASP | <PROTECTED> |
| 18:09.31 | EC-ASP | <PROTECTED> |
| 18:09.33 | EC-ASP | (etc, etc) |
| 18:09.35 | KalD|Work | EC-ASP, did you check /proc/pci to make sure the kernel sees the card? |
| 18:09.43 | *** join/#asterisk santiago (~santiago@63.245.86.121) |
| 18:09.48 | EC-ASP | s5:~# lsmod |
| 18:09.49 | EC-ASP | Module Size Used by |
| 18:09.49 | EC-ASP | wcte11xp 21696 0 |
| 18:09.49 | EC-ASP | zaptel 216836 1 wcte11xp |
| 18:09.49 | EC-ASP | crc32 2976 0 |
| 18:09.49 | EC-ASP | crc32c 800 0 |
| 18:09.51 | EC-ASP | libcrc32c 1536 1 crc32c |
| 18:09.54 | bkw_ | EC-ASP, SMACK |
| 18:09.55 | EC-ASP | capability 2888 0 |
| 18:09.57 | EC-ASP | commoncap 3360 1 capability |
| 18:09.57 | Juggie | STOP PASTEING |
| 18:09.57 | bkw_ | use a paste bin |
| 18:09.58 | |Vulture| | no more |
| 18:09.59 | Beirdo | ~pastebin |
| 18:10.00 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 18:10.00 | *** kick/#asterisk [EC-ASP!~bkw_@bkw.developer.and.friend.of.asterisk] by bkw_ (bkw_) |
| 18:10.08 | |Vulture| | lol |
| 18:10.13 | Beirdo | ~nickometer EC-ASP |
| 18:10.13 | jbot | 'EC-ASP' is 97.970% lame, beirdo |
| 18:10.17 | |Vulture| | damn he was about to paste all that? |
| 18:10.20 | bkw_ | yes |
| 18:10.27 | |Vulture| | ~nickometer |Vulture| |
| 18:10.27 | jbot | '|Vulture|' is 26.000% lame, |vulture| |
| 18:10.28 | Juggie | ~nickometer Juggie |
| 18:10.28 | jbot | 'Juggie' is 0.000% lame, juggie |
| 18:10.28 | bkw_ | I took care of it my childeren |
| 18:10.34 | Juggie | score. |
| 18:10.36 | |Vulture| | hahah Juggie rox! |
| 18:10.39 | bkw_ | ~nickometer bkw_ |
| 18:10.39 | jbot | 'bkw_' is 0.000% lame, bkw_ |
| 18:10.40 | *** join/#asterisk EC-ASP (~alfredo@Intelideas-Avanzia.Mesena.MAD.ES.INTELIDEAS.NET) |
| 18:10.42 | outtolunc | when you were gonna compile did you modprobe -r the wct and the zaptel before reloading? |
| 18:10.44 | EC-ASP | sorry |
| 18:10.48 | KalD|Work | ~nickometer KalD|Work |
| 18:10.48 | jbot | 'KalD|Work' is 61.000% lame, kald|work |
| 18:10.52 | KalD|Work | ouch |
| 18:10.57 | KalD|Work | that's over half |
| 18:11.00 | |Vulture| | EC-ASP: try pastebin.ca next time |
| 18:11.09 | bkw_ | NEXT!!! |
| 18:11.19 | |Vulture| | :P |
| 18:11.28 | Juggie | bkw, i'm doing up asterisk against a RFP at work, the wiki says voicemail is restricted to 99 messages per box, is this still the case? |
| 18:11.29 | [Outcast] | bkw_ : rofl |
| 18:11.53 | EC-ASP | outtolunc, I did |
| 18:11.55 | EC-ASP | even rebooted |
| 18:12.00 | |Vulture| | Juggie: I can confirm thats the case in 1.0.5 |
| 18:12.01 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:12.09 | Juggie | seems like an odd limit, since messages are stored as msg#### |
| 18:12.15 | BrianR___ | Juggie: I think it's 99 per folder.. |
| 18:12.16 | EC-ASP | Oddly, this box has been working well |
| 18:12.18 | EC-ASP | with 2.6.9 |
| 18:12.22 | outtolunc | are you doing make clean; make linux26; make install? |
| 18:12.28 | EC-ASP | outtolunc, yes |
| 18:12.35 | outtolunc | are you reading the output for errrors |
| 18:12.40 | EC-ASP | I think I've done it like 5 or 6 times today :) |
| 18:12.42 | EC-ASP | yes |
| 18:12.42 | |Vulture| | BrianR___: that would make sense I just know INBOX will max at 99 |
| 18:12.53 | EC-ASP | no errors apart from two innocent-looking compiler warnings |
| 18:13.01 | outtolunc | well there has to be an error somewhere, if NOT, then move the card slot |
| 18:13.04 | BrianR___ | I wonder if that's a configurable limit somewhere... |
| 18:13.05 | Sedorox | make linux26? |
| 18:13.19 | |Vulture| | BrianR___: not in any conf, prolly in the actual source |
| 18:13.40 | BrianR___ | It certainly makes sense to have some limit lower than 10k messages to prevent wierd problems from filling your VM system.. |
| 18:13.41 | |Vulture| | I remember someone talking about doing a patch for it but duno how that turned out |
| 18:13.57 | |Vulture| | BrianR___: yea thats how I found out the limit was 99 :P |
| 18:14.21 | Juggie | very odd limit |
| 18:14.32 | EC-ASP | outtolunc, but it has worked right in the same slot where it is now |
| 18:14.34 | Juggie | espicially since the files are msg#### |
| 18:14.40 | Juggie | so it already has room for 9999 there. |
| 18:14.48 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:14.56 | BrianR___ | 99 is a sensible default for the inbox.. I'd argue for an even lower default as most users don't get many messages... |
| 18:14.58 | outtolunc | ec: so, the bus could have reset |
| 18:15.12 | BrianR___ | would be nice to have it configurable though. |
| 18:15.12 | outtolunc | you are 'trying' to get an error at this stage |
| 18:15.13 | |Vulture| | yea 99 msgs is crazy to be in an inbox |
| 18:15.15 | tzanger | BrianR___: ok looking at putting DISA in here |
| 18:15.17 | Juggie | BrianR___, thats true, but i have to present the limits of the system |
| 18:15.20 | Beirdo | BrianR___: as long as it can be overridden, 99 should be fine :) |
| 18:15.29 | Juggie | i would rather say there are no limits, and you can set max to whatever u want |
| 18:15.33 | BrianR___ | tzanger: Oh. I already tested DISA for MWI - doesn't work. |
| 18:15.35 | Juggie | is there a setting for max in voicemail.conf? |
| 18:15.40 | |Vulture| | no |
| 18:15.42 | Beirdo | but really. who leaves 99 in there anyways |
| 18:15.48 | tzanger | BrianR___: dammit |
| 18:15.49 | tzanger | heh |
| 18:15.54 | tzanger | I just assigned a DISA DN to 0000123 |
| 18:15.55 | Juggie | Beirdo, thats not the point. |
| 18:15.55 | EC-ASP | outtolunc, now resetting... |
| 18:15.58 | |Vulture| | Beirdo: people who don't check the mail and let it fill up |
| 18:16.06 | BrianR___ | tzanger: I got 4 ports from the norstar connecting to asterisk over VMI's though. And working disconnect supervision now. |
| 18:16.07 | KalD|Work | EC-ASP, have you tried changing the base location in the Makefile to point to where /proc/pci thinks your card is at? |
| 18:16.15 | Beirdo | Juggie: I know, you need to know what the limit actually is. :) |
| 18:16.24 | BrianR___ | tzanger: The VMI's even hunt and vice-versa. |
| 18:16.26 | |Vulture| | Juggie just likes to be able to tell them they can have "Unlimited" |
| 18:16.29 | eipi | i have problem authenticanting SIP from database (sip_friends). ANyone have working and want to help me? (i configured extconfig.conf, created tables, have unixodbc working). Voicemail, voicemessages and extensions are working from db |
| 18:16.44 | *** join/#asterisk point (~point@office.rtcomm-yug.ru) |
| 18:16.59 | EC-ASP | KalD|Work, I'll do now, because last reboot hasn't changed anything |
| 18:17.05 | BrianR___ | tzanger: I will try to do a writeup on this setup soon. It's probably the only way to cheaply integrate asterisk with smaller norstars like the 3x8 and the 6x16 and so on. |
| 18:17.13 | tzanger | BrianR___: using Dial(,,H) |
| 18:17.19 | tzanger | BrianR___: yeah |
| 18:17.20 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 18:17.35 | Juggie | on a side note however MWI works perfect from voicemail |
| 18:17.40 | Juggie | on both my cisco and mitel sip phones. |
| 18:18.02 | |Vulture| | Polycom too :) |
| 18:18.14 | Sedorox | and BT100's |
| 18:18.28 | BrianR___ | tzanger: Yep. FOr incoming calls it's Dial(local/in@fromvmi/n,10,H) The '/n' is very important too - it keeps the local channel from getting optimized out of the route. |
| 18:18.31 | BrianR___ | tzanger: s/route/path |
| 18:18.36 | BrianR___ | tzanger: For outbound calls, use 'h' |
| 18:18.38 | Juggie | well theres no limit of 99 in the file structure |
| 18:18.46 | Juggie | i'll have to check out the source. |
| 18:19.06 | tzanger | BrianR___: interesting, I didn't know there was a '/n' part |
| 18:19.06 | BrianR___ | tzanger: In 1.0.5, I changed the disconnect tone from '*' to 'D' - it's configurable in the CVS version from features.conf though. |
| 18:19.43 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 18:20.01 | jayden | hey tzanger... wassup |
| 18:20.04 | BrianR___ | tzanger: '/n' means "no native transfer" or something to that effect. It keeps that Dial from getting optimized away so it's always there listening for the disconnect DTMF tone. Otherwise you'd have to make sure the 'D' is in every dial statement in your dialplan - and even then you might wind up with trouble on things like meetme. |
| 18:20.22 | tzanger | BrianR___: I understand |
| 18:20.43 | BrianR___ | tzanger: But the disconnect supervision is flawless now for both inbound and outbound calls |
| 18:21.20 | BrianR___ | tzanger: I also found out that the caller-id on the VMI boxes I have here is junk. Bought a standalone pots phone with callerid and it still doesn't work. |
| 18:21.42 | EC-ASP | KalD|Work, baseaddr is memory region or I/O ports? |
| 18:21.46 | *** join/#asterisk Ad-Hoc (~ad-hoc@62.1.246.83) |
| 18:21.47 | KalD|Work | i/o |
| 18:22.35 | BrianR___ | is there a devel conf call coming up? |
| 18:22.38 | tzanger | BrianR___: how do I flash on a norstar set? |
| 18:22.45 | |Vulture| | at 2 est right? |
| 18:23.04 | BrianR___ | tzanger: feature 71. Not sure if it will flash non-trunk pots lines though. |
| 18:23.13 | Mother_ | in sip who sets the rtp port, asterisk or the client? |
| 18:23.32 | EC-ASP | KalD|Work, no luck... Same message |
| 18:23.47 | KalD|Work | EC-ASP, you did a make clean etc |
| 18:23.50 | Juggie | looking at app_voicemail.c i dont see any restriction for 99 yet... anyways i'm only about 1/2 way. |
| 18:23.58 | EC-ASP | nope, I'll compile from clean now |
| 18:24.25 | Juggie | the only code i read so far which could affect was the count messages function |
| 18:24.35 | Juggie | and it was written withno limit. |
| 18:24.55 | tzanger | hmm |
| 18:25.01 | tzanger | I must have something wrong with DISA |
| 18:25.08 | |Vulture| | Juggie: the limit is actually 100 |
| 18:25.15 | Juggie | #define MAXMSG 100 |
| 18:25.16 | |Vulture| | msg0000.txt |
| 18:25.18 | tzanger | I can't even hit another extension from it |
| 18:25.19 | Juggie | aha |
| 18:25.30 | tzanger | I get stutter dial tone (MICS intenral) |
| 18:25.36 | tzanger | but then I can't hit anything |
| 18:25.39 | |Vulture| | Juggie: :) |
| 18:26.05 | Juggie | i wonder what happens once asterisk reaches msg9999.txt |
| 18:26.19 | |Vulture| | Juggie: that would be easy to movie into voicemail.conf |
| 18:26.19 | BrianR___ | mailbox full? |
| 18:26.22 | Juggie | i dont think it fills in the gaps of messages you delete. |
| 18:26.36 | Juggie | well, if there are only like 50 messages... |
| 18:26.40 | Juggie | let me test something. |
| 18:26.55 | |Vulture| | brb food |
| 18:27.01 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 18:27.55 | tzanger | yup I just have too much security on my DISA |
| 18:28.19 | EC-ASP | KalD|Work, no joy; same error |
| 18:28.32 | tzanger | PRI-A (line1-23) has access, remote package 01, whee |
| 18:28.42 | KalD|Work | EC-ASP, hmmm... what does dmesg report? |
| 18:28.58 | *** join/#asterisk kongnamool (~sexton@astound-64-85-253-249.ca.astound.net) |
| 18:29.15 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:29.33 | EC-ASP | not much actually |
| 18:29.41 | EC-ASP | no more than what I get interactively |
| 18:29.43 | Juggie | ok, so i left 3 msgs, and then deleted # 2 |
| 18:29.49 | Juggie | which is msg0001.* |
| 18:29.56 | Juggie | lets see how it handels the numbering |
| 18:30.00 | EC-ASP | zaptel loads and unloads |
| 18:30.01 | EC-ASP | Zapata Telephony Interface Registered on major 196 |
| 18:30.04 | EC-ASP | and |
| 18:30.09 | EC-ASP | Zapata Telephony Interface Unloaded |
| 18:30.32 | EC-ASP | but ztcfg -vv says, well, that dreaded no such device... |
| 18:30.47 | *** join/#asterisk phantam (~root@63.210.60.199) |
| 18:30.53 | Juggie | hmm... it worked find as it moved all played messages to old |
| 18:31.10 | tzanger | BrianR___: ok how do I set up DISA properly |
| 18:31.13 | Juggie | so the file system should be no restriction on the inbox |
| 18:32.21 | sivana | ~seen sixtel |
| 18:32.23 | jbot | sixtel <sixtel@sixTel.iax.cc> was last seen on IRC in channel #asterisk, 49d 13h 13m 38s ago, saying: 'no such host, not in sip.conf right'. |
| 18:32.49 | phantam | hey guys |
| 18:32.51 | phantam | quick question |
| 18:33.12 | [Outcast] | shoot |
| 18:33.13 | phantam | does anyone know how to change the location where the cdr/csv is saved |
| 18:33.56 | EC-ASP | nah |
| 18:33.59 | EC-ASP | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 18:34.04 | EC-ASP | frustrating |
| 18:34.13 | *** join/#asterisk {Sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 18:34.25 | {Sean} | anyone done an application where it allows the called to choose their own MOH? |
| 18:34.31 | {Sean} | caller rather |
| 18:35.15 | visik7 | I want * on my mobile phone :) |
| 18:35.27 | visik7 | is there a port on symbian :) |
| 18:35.32 | phantam | :( |
| 18:36.28 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 18:36.32 | [Outcast] | phantam: tring to find an aswer one sec |
| 18:36.57 | phantam | thx |
| 18:37.03 | BrianR___ | tzanger: Just set the DISA DN to the suffix of an unused DID on your PRI. |
| 18:37.17 | tzanger | yes |
| 18:37.18 | tzanger | I did htat |
| 18:37.34 | tzanger | and when I dial it I get a stutter dialtone for a second or two and hten the MICS' internal dialtone |
| 18:37.44 | tzanger | but I can't dial a 3-digit exten or an extenral number |
| 18:37.57 | Juggie | i think i'll have a crack at making vm limit configurable, anyone up for testing it when i'm done? |
| 18:38.47 | phantam | im waiting on someone in the world to fix h323 |
| 18:38.50 | phantam | lol |
| 18:39.24 | eipi | anyone is workign with odbc? |
| 18:40.11 | phantam | not odbc but eventually gonna try to figure out how to save all my cvs's in a mysql |
| 18:40.24 | phantam | on a remote server but havent yet figured out how to accomplish it |
| 18:40.25 | phantam | lol |
| 18:40.34 | greg_work | Juggie: vm limit ? how many messages per box you mean? |
| 18:40.38 | eipi | ok |
| 18:41.34 | ChrisRouse | Anyone have experience with Cisco Call Manager Integration? |
| 18:42.56 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 18:43.55 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 18:44.15 | BrianR___ | tzanger: The stutter dialtone wants a COS password, I think. Once you enter that you get an internal dialtone with access level controlled by the COS password. |
| 18:45.31 | ChrisRouse | Maybe a question that is more specific |
| 18:46.00 | phantam | hmm |
| 18:46.08 | phantam | i have a feeling [Outcast] isnt having much luck |
| 18:46.09 | phantam | lol |
| 18:46.19 | ChrisRouse | How do I get Asterisk to reconize an extesion when I have integrated the system with Call Manager |
| 18:46.19 | *** join/#asterisk LarsAC (~chatzilla@pD9501C02.dip0.t-ipconnect.de) |
| 18:46.23 | phantam | anyone else know |
| 18:47.15 | *** join/#asterisk CMike (~a_mike@c-dc4171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 18:47.23 | ChrisRouse | Sorry really new to all of this. |
| 18:48.39 | tzanger | BrianR___: ahhh |
| 18:49.43 | *** join/#asterisk lyroy_ (~lyroy@picachou.csaffluents.qc.ca) |
| 18:50.08 | lyroy_ | Does someone knoe if the Linksys (Vonage) can work with an asterisk server or if it is lockeed |
| 18:50.32 | greg_work | ChrisRouse: asterisk is all about contexts. to dial an extension, it has to have an exten=> line in or included in the current context |
| 18:50.48 | JerJer | lyroy_ yes....they are not locked, just pre-configured |
| 18:51.35 | ChrisRouse | greg_work: I understand that. I am able to dial from my phone into the agent application. I can even enter my password for the agent I defined. However, I am unable to specify the extension that I am calling from so that AgentCallbackLogin can call me back. |
| 18:51.47 | lyroy_ | so do I need to put another fireware or use the one that is aleready there? |
| 18:52.17 | ChrisRouse | I have gotten the Agent system to work if I sit in the Queue but we are looking for Asterisk to call the Agent when there is a client in the queue for that Agent. |
| 18:52.43 | Sedorox | ChrisRouse: look into AgentCallBack |
| 18:53.07 | Sedorox | you'll notice, especially if you look on the console |
| 18:53.30 | ChrisRouse | Sedorox: I have and every time I type in an extension it tells me that the extension is not valid |
| 18:53.32 | Sedorox | that it uses the CID to set the callback number |
| 18:53.50 | Sedorox | hmmm |
| 18:53.53 | *** part/#asterisk phantam (~root@63.210.60.199) |
| 18:53.58 | ChrisRouse | Sedorox: Let me double check |
| 18:54.35 | Sedorox | like for example.. my phone is on extention 2001, and thats what the phone is setup with.. so when I login to the agent.. it see's it as 2001, and uses that as the call back number |
| 18:55.12 | ChrisRouse | Sedorox: Do you mean AgentCallbackLogin? That is what I am attempting to use. |
| 18:55.20 | Sedorox | yea.. sorry |
| 18:55.34 | ChrisRouse | Sedorox: The system prompts me for a new extension. |
| 18:55.52 | bjohnson | lyroy: jerjer loves to say that. the simple fact is that if you have a voip provider supplied pap2 (not a PAP2-NA), unless you have access to Sipura development tools, you will be unable to use it with anything else |
| 18:55.58 | ChrisRouse | Sedorox: So when I type my extension the system tells me that it is not a valid extension for that agent. |
| 18:56.11 | Sedorox | I don't have mine setup like that... |
| 18:56.12 | Sedorox | sorry |
| 18:57.02 | Beirdo | my PAP2 is almost due for some speed-holes |
| 18:57.13 | ChrisRouse | Sedorox: Do you have your system set up as Asterisk is your call manager? |
| 18:57.18 | Beirdo | if I weren't so busy working on other stuff it'd be there already :) |
| 18:57.21 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-230-248.dsl.scarlet.be) |
| 18:57.37 | bjohnson | speed-holes? |
| 18:57.43 | Beirdo | yeah |
| 18:57.47 | Beirdo | like on the simpsons |
| 18:57.51 | bjohnson | what is a speed-hole? |
| 18:58.02 | ChrisRouse | Sedorox: I am also getting Feb 24 10:56:48 WARNING[27144]: chan_agent.c:1282 __login_exec: Extension '1902' is not valid for automatic login of agent '1001' |
| 18:58.02 | Beirdo | when homer took the pickaxes to his car to give it speed holes |
| 18:58.10 | jsolares | rofl Beirdo |
| 18:58.10 | bjohnson | odd |
| 18:58.13 | Grooby | lol |
| 18:58.27 | Grooby | i was thinking in terms of a hole on the speedometer |
| 18:58.30 | bjohnson | missed that one I guess |
| 18:58.32 | Beirdo | Feb 24 13:41:26 NOTICE[1060]: chan_iax2.c:5673 socket_read: Peer 'voipjet' is now TOO LAGGED (2045 ms)! |
| 18:58.38 | jsolares | it was very funny |
| 18:58.39 | Beirdo | argh |
| 18:58.40 | Grooby | "but officer, I don't have 60mph, I only have 80mph and up" |
| 18:58.46 | Sedorox | I'm not sure... sorry |
| 18:58.54 | Sedorox | I followed the guide on voip-supply |
| 18:58.56 | Beirdo | it was a good episode |
| 18:59.00 | bjohnson | Beirdo: 42 ms for me |
| 18:59.15 | ChrisRouse | Sedorox: I will look into it. Maybe it can help me too. Thank you VERY much for the help. |
| 18:59.16 | Beirdo | it does this every so often |
| 18:59.22 | Sedorox | yup |
| 18:59.25 | Sedorox | sory I couldn't help more |
| 18:59.30 | ChrisRouse | Sedorox: I do appriciate it. At least it is a different direction. |
| 18:59.31 | Beirdo | but my connection to FWD's IAX doesn't slip |
| 18:59.46 | bjohnson | 30 ms for me |
| 18:59.47 | EC-ASP | Guys, I have this server that suddenly doesn't work with a TE110p with which it was perfectly happy before |
| 18:59.50 | *** join/#asterisk darkskiez (~mhb@host-84-9-91-127.bulldogdsl.com) |
| 18:59.50 | EC-ASP | and at the same time |
| 18:59.51 | Beirdo | I swear, voipjet has craptacular connectivity or soemthing |
| 18:59.53 | bjohnson | iaxtel I gave up on though |
| 18:59.56 | EC-ASP | bind 9 doesn't work |
| 19:00.00 | EC-ASP | does that ring a bell? |
| 19:00.02 | |Vulture| | so is the dev conf starting now? |
| 19:00.11 | |Vulture| | anyone know what the discussion is going to be? |
| 19:00.12 | Beirdo | it says 54ms right now |
| 19:00.33 | bjohnson | |Vulture|: yeah .. lets go eavesdrop |
| 19:00.43 | ChrisRouse | Sedorox: What is the URL for that site? |
| 19:00.49 | Sedorox | ~docs |
| 19:00.50 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:01.02 | Sedorox | second to the last |
| 19:01.05 | tzanger | BrianR___: you're right, set a cos password and get another dialtone I can dial out on now but can't do anything else... I can't call an internal extension or anything |
| 19:01.07 | bjohnson | or .. let's bash Beirdo on HIS meetme |
| 19:01.18 | bjohnson | wuaa haaa haa |
| 19:01.27 | Beirdo | hmm? |
| 19:01.36 | Beirdo | I don't have meetme setup yet :) |
| 19:01.54 | bjohnson | IAX2/guest@rakdanit.shavedgoats.net/3100 |
| 19:02.00 | tzanger | shavedgoats hahaha |
| 19:02.16 | jsolares | hehe |
| 19:02.22 | *** join/#asterisk mbranca_home (~matteo@host-84-222-6-8.cust-adsl.tiscali.it) |
| 19:02.30 | Beirdo | that's mishehu's setup :) |
| 19:03.06 | Grooby | lol |
| 19:03.36 | BrianR___ | tzanger: Now that's odd. Make sure the remote access profile is not buggered. |
| 19:03.50 | tzanger | yeah it could be, heh |
| 19:03.57 | |Vulture| | bjohnson: I am listening now |
| 19:04.14 | |Vulture| | talking about codec mismatches with reinvite |
| 19:04.30 | Grooby | what are you guys listening to? |
| 19:04.32 | *** join/#asterisk jdb1968 (~jdb1968@S010600045af29653.cg.shawcable.net) |
| 19:04.39 | tzanger | BrianR___: linepool access is PRI-A only (all I have), remote page=n, |
| 19:04.40 | tzanger | that's it |
| 19:04.42 | Grooby | how to shave goats? |
| 19:05.01 | BuckRogers | wow |
| 19:05.09 | jsolares | Grooby: i think the thing on the topic |
| 19:05.29 | BuckRogers | there was a time in this room when VoIP was discused |
| 19:05.40 | jsolares | you lie! |
| 19:05.53 | BuckRogers | tzanger, knows what im talking about |
| 19:05.57 | tzanger | I do? |
| 19:06.01 | tzanger | heh |
| 19:06.09 | |Vulture| | no, we always just talked about moose penis |
| 19:06.25 | BuckRogers | ok thats just plain old inapporparate |
| 19:06.25 | Grooby | stupid trillian...can't view topic |
| 19:06.26 | Grooby | sigh |
| 19:06.39 | Beirdo | so don't use stupid trillian :) |
| 19:06.51 | Grooby | i refuse! |
| 19:06.54 | BrianR___ | tzanger: That _should_ work... I don't think I had to configure anything else for extension dialling here. |
| 19:06.55 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 19:07.07 | Grooby | like how I still use rotary phone here at home |
| 19:07.21 | Grooby | shinny red phone labeled "bat phone" |
| 19:07.38 | tzanger | hmmfdsa |
| 19:07.40 | tzanger | odd |
| 19:07.41 | |Vulture| | lol with a plexiglass buble around it? |
| 19:07.53 | Grooby | how you know?!?! |
| 19:07.53 | *** join/#asterisk zno (~zeno@ip-160-79-174-102.autorev.intellispace.net) |
| 19:07.54 | Grooby | lol |
| 19:07.57 | |Vulture| | ;) |
| 19:08.26 | Grooby | the "batstrisk" |
| 19:08.39 | *** join/#asterisk |Barcode (~uid@h-68-165-204-41.chcgilgm.covad.net) |
| 19:08.44 | |Vulture| | roary that generates tone dialing? |
| 19:08.54 | zno | when I park someone the parked call extension is not read back. I see in the log that it started playing back 7 - 0 and then it gets cutoff |
| 19:09.07 | Grooby | sure...hehehehe |
| 19:09.09 | zno | I just updated cvs 5 minutes ago |
| 19:09.58 | zno | however, when I directly dial 700 (parking ext) I get the parked extension read back to me |
| 19:10.15 | zno | maybe it's my phone? |
| 19:11.12 | Grooby | time for some mandarine chicken from wendy's |
| 19:11.48 | tzanger | who's talking right now |
| 19:11.53 | tzanger | mark or brian? |
| 19:12.19 | tzanger | that's paul |
| 19:12.58 | yasha | in Asterisk@home configs, which sections starts the dialplan? |
| 19:13.35 | tzanger | he said "eeks" |
| 19:13.37 | tzanger | that's mark |
| 19:14.09 | Hmmhesays | if I want to add a tdm400p to a machine do I have to do anything besides modprobe zaptel wctdm ? |
| 19:14.18 | Hmmhesays | I mean, add a second one |
| 19:14.21 | tzanger | Hmmhesays: nope |
| 19:14.33 | Hmmhesays | perfect |
| 19:14.53 | km- | doesnt the tdm400p require either the wcfxo or wcfxs drivers? |
| 19:15.00 | km- | or has that changed since I last used it |
| 19:15.02 | Hmmhesays | the new driver is wctdm |
| 19:15.10 | km- | gotcha |
| 19:15.10 | Grooby | in order to listen to this conference, I need an iax client huh? |
| 19:15.10 | Hmmhesays | as of 2/5/05 |
| 19:15.20 | km- | oh, conference |
| 19:15.24 | km- | lemme jump on |
| 19:15.41 | Beirdo | there we go. ilbc bridged from work |
| 19:15.44 | tzanger | :-) |
| 19:15.57 | |Vulture| | ilbc is my new fav codec |
| 19:16.05 | Sedorox | lol |
| 19:16.07 | bjohnson | Grooby: yes |
| 19:16.10 | tzanger | |Vulture|: I want to lvoe it |
| 19:16.10 | Sedorox | y? |
| 19:16.13 | bjohnson | Grooby: like asterisk |
| 19:16.25 | |Vulture| | tzanger: whats holding you back |
| 19:16.41 | Grooby | gasp |
| 19:16.42 | Grooby | you can? |
| 19:16.58 | bjohnson | exten => _7,1,Dial(IAX2/guest@66.250.68.194/996) |
| 19:17.02 | tzanger | |Vulture|: it sounds like ass |
| 19:17.04 | tzanger | compared to gsm |
| 19:17.08 | tzanger | at least to everyone here at the office |
| 19:17.11 | km- | is this muted? |
| 19:17.13 | Beirdo | sounds fine to me |
| 19:17.16 | tzanger | km *1 toggles mute |
| 19:17.19 | Grooby | ahhh |
| 19:17.20 | |Vulture| | sounds better than gsm to me |
| 19:17.31 | |Vulture| | much |
| 19:17.32 | km- | am I mute by default? |
| 19:17.38 | bjohnson | any way to tell how many people in the dev conference |
| 19:17.39 | |Vulture| | thats strange |
| 19:17.45 | tzanger | |Vulture|: about 50% of the people I talk to say ilbc rocks over gsm, and the other 50% say the opposite. |
| 19:17.52 | EC-ASP | it seems that the cvs version of the zaptel driver guesses ok the OS version |
| 19:17.56 | tzanger | |Vulture|: and then invariably I get a few people saying ulaw's the only true way |
| 19:18.03 | EC-ASP | no joy, though, as I still can't get the 110p to work |
| 19:18.23 | |Vulture| | tzanger: ulaw is great, but not out over the net... too much bandwidth |
| 19:18.29 | xkev | queues.conf: timeout=, is that supposed to be per member, or per round of call attempting? |
| 19:18.30 | bjohnson | tzanger: from what I've read .. gsm wins out over ilbc by a slight margin |
| 19:18.48 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 19:18.49 | |Vulture| | yea gsm wins because it uses less bandwidth |
| 19:18.51 | bjohnson | I've been using ulaw over the net but have been thinking of using gsm |
| 19:19.00 | |Vulture| | but I think ilbc sounds better... personally |
| 19:19.02 | km- | I'm using ilbc |
| 19:19.08 | km- | hehe |
| 19:19.17 | km- | ooh, is that jitter? |
| 19:19.19 | tzanger | |Vulture|: every time I change the codec from gsm to ilbc here I get complaints about audio quality |
| 19:19.20 | km- | heh |
| 19:19.22 | tzanger | this is 30+ people |
| 19:19.30 | bjohnson | I should connect to the conference with gsm .. just need to figure out how |
| 19:19.46 | tzanger | speaker identify yourslef |
| 19:20.11 | |Vulture| | Im on the conf at ilbc sounds great to me |
| 19:20.16 | yasha | In Asterisk@home configs, which section (like: [default]) and in what file (like: extension) starts the dialplan? |
| 19:20.23 | km- | I've been on it for about 4 minutes and the chopping is getting bad |
| 19:20.30 | km- | I'm wondering if it's jitter |
| 19:20.32 | |Vulture| | tzanger: is it possible because ilbc uses more bandwidth that your hitting your limit and getting choppy calls? |
| 19:20.32 | tzanger | km-: sounds fine to me |
| 19:20.36 | tzanger | |Vulture|: no |
| 19:20.40 | tzanger | I'm nowhere near my limit |
| 19:20.42 | tzanger | it's not choppy |
| 19:20.43 | |Vulture| | oky |
| 19:20.44 | km- | it gets better for a sec, then gets worse again |
| 19:20.45 | tzanger | it's "gravelly" |
| 19:20.47 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:20.50 | bjohnson | how do I change my codec to the dev conf? |
| 19:20.55 | km- | sounds like a warble almost |
| 19:20.55 | Grooby | bjohnson, doesn't work |
| 19:21.02 | bjohnson | logs say I'm using ulaw now |
| 19:21.06 | |Vulture| | tzanger: what VoIP provider? |
| 19:21.07 | tzanger | bjohnson: create a peer and disable=all,enable=fave_codec |
| 19:21.07 | bjohnson | Grooby: it is what I use |
| 19:21.10 | Hmmhesays | heh, this is pretty interesting |
| 19:21.11 | tzanger | |Vulture|: me |
| 19:21.19 | tzanger | *--*--PRI |
| 19:21.20 | |Vulture| | tzanger: ah :) |
| 19:21.27 | Grooby | i got "everyone is busy/congested" |
| 19:21.28 | bjohnson | tzanger: what peer would it use? |
| 19:21.29 | greg_work | hm, so i guess app_voicemail can't send to a TAP service .... |
| 19:21.33 | tzanger | bjohnson: create one |
| 19:21.46 | |Vulture| | tzanger: who do you use as a provider? I am trying to set that up right now for my offices |
| 19:21.57 | *** join/#asterisk e3eli3h (~e3eli3h@83.168.2.150) |
| 19:21.57 | |Vulture| | tzanger: looking at XO cause of the local calling areas |
| 19:21.57 | km- | tzanger: think my T1 oddities are worth bringing up on the conference or should we get some more data about what's happening before we raise it as a bug? |
| 19:21.59 | tzanger | |Vulture|: me |
| 19:22.00 | tzanger | :-) |
| 19:22.07 | km- | we use XO here |
| 19:22.08 | tzanger | I my connection to the internet is through Ikano |
| 19:22.15 | *** join/#asterisk florz (nobody@odnb-d9baa542.pool.mediaWays.net) |
| 19:22.34 | |Vulture| | km-: do you guys use multiple LATAs? |
| 19:22.36 | PyroSteve | guys, i need help with my call file |
| 19:22.40 | PyroSteve | my call file works |
| 19:22.52 | km- | uhm, dont think so, |
| 19:22.56 | *** join/#asterisk Inv_arp (junya@adsl-8-230-20.mia.bellsouth.net) |
| 19:22.58 | PyroSteve | but I can navigate throught my server |
| 19:23.03 | km- | we only have numbers through our one local CO |
| 19:23.28 | PyroSteve | my call file is supposed to call my cellphone, and then stick the call into a context that run the DISA command |
| 19:23.30 | |Vulture| | km-: oky, yea I am trying to get a central office with like 2 PRIs and then all the sub offices connect to it to dial out and rx calls |
| 19:23.37 | ScarletCrusader | Excuse me, dont anyone know which configuration file or program I would enter user auth for SIP phones? |
| 19:23.49 | PyroSteve | after that my dtmf tones wont work so cant dial an numbers or passwords |
| 19:24.04 | Grooby | i am retarded |
| 19:24.06 | Sedorox | bbl |
| 19:24.08 | Grooby | i enter the wrong ip |
| 19:24.12 | |Vulture| | lol |
| 19:24.22 | km- | ooh, I just looked up at the cisco router |
| 19:24.26 | Beirdo | heheh |
| 19:24.26 | km- | I'm getting lights |
| 19:24.33 | Inv_arp | ScarletCrusader: sip.conf u mean? |
| 19:24.38 | Grooby | btw bjohnson, i am using bt headset w/ x-lite..works great..sjphone sucks |
| 19:24.51 | |Vulture| | km-: is that bad? |
| 19:25.09 | km- | orange lights |
| 19:25.14 | km- | bad flashing lights |
| 19:25.31 | |Vulture| | collisions? |
| 19:25.58 | ScarletCrusader | Inv_arp: i've been looking in that file but i dont see an example to actually enter auth information. BTW i've been known to have Idiots Bliendness |
| 19:26.42 | *** part/#asterisk point (~point@office.rtcomm-yug.ru) |
| 19:26.47 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 19:27.44 | bjohnson | damn .. I can connect with a staright dial command .. but can't get using a iax.conf entry to work |
| 19:27.57 | bjohnson | Grooby: I use the exact sting I pasted |
| 19:28.22 | Grooby | bjohnson, I type the wrong IP |
| 19:28.26 | km- | oh |
| 19:28.36 | km- | the orange flashy was ethernet I think |
| 19:28.50 | km- | for a second I thought it was T1 errors |
| 19:30.37 | EC-ASP | I think I got it |
| 19:30.50 | EC-ASP | in my haste, some usually needed options in networking were not selected |
| 19:31.00 | EC-ASP | Now compiling... |
| 19:31.13 | km- | oops |
| 19:31.14 | *** part/#asterisk didz_ (didz_@200.218.192.52) |
| 19:31.26 | km- | time to apologize to asterisk |
| 19:31.27 | km- | :) |
| 19:31.32 | EC-ASP | hehe... I guess so |
| 19:31.34 | Hmmhesays | why is that? |
| 19:31.36 | EC-ASP | sorry, asterisk |
| 19:31.45 | EC-ASP | cause I ranted about loosy drivers |
| 19:31.52 | km- | he was complaining that the drivers sucked earlier :) |
| 19:31.56 | Hmmhesays | lol |
| 19:32.01 | Meznev | Is anyone using the zaptel driver in freebsd with digium cards? |
| 19:32.05 | EC-ASP | though I'd like to say that the error message could be a bit more informative |
| 19:33.08 | km- | ec-asp: the problem may be that what happened was too generic to say specifically. What is it supposed to say? "No such device, meaning, you may not have the card inserted, you may have the networking drivers not compiled into your kernel, your kernel may be malfunctioning, there may be a full solar eclipse, etc" |
| 19:33.32 | km- | although I 100% sympathize with what you're saying |
| 19:33.35 | EC-ASP | km-, agree on that - maybe a README that lists the kernel requirements? |
| 19:33.40 | znoG | is anyone using Freshtel successfully? |
| 19:33.44 | km- | THAT is a fantastic idea |
| 19:34.09 | lyroy | Does someone what is the PAP2 configuation password for Vonage please? |
| 19:34.21 | km- | there is a README file but it doesnt list what goodies you need |
| 19:36.34 | Grooby | so who has voice in this conference? |
| 19:36.52 | PyroSteve | my dtmf tones coming from the pstn are ingored when a call is made from a call file |
| 19:36.58 | Hmmhesays | no idea |
| 19:37.14 | ennuyeux7 | is the expression in execif the same format as that for gotoif ? |
| 19:37.24 | Delvar | anyone know how to use execif? i cant seem to get it to work, and yes iv read the show application :) |
| 19:37.49 | Hmmhesays | does everyone have voice? |
| 19:38.03 | Beirdo | lyroy: nobody will be telling you on here even if they do know, that's not something you share in a public forum |
| 19:38.07 | Grooby | nope |
| 19:38.12 | Grooby | i have no idea who's talking |
| 19:38.16 | Hmmhesays | me neither |
| 19:38.20 | Grooby | just something about cell phones being dropped in toilet |
| 19:38.24 | Delvar | i try execif([${VAR}=STRING]|SetVar|${VAR2}=FLIP) |
| 19:38.26 | Hmmhesays | that sound like shit |
| 19:38.35 | Delvar | but doesnt actualy do anyhting |
| 19:40.23 | Hmmhesays | good lord I hate repeating myself |
| 19:41.44 | *** part/#asterisk Fanguin (~Fanguin@p50818948.dip0.t-ipconnect.de) |
| 19:42.03 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 19:42.14 | guugmember | hello guys, who has played with Varion cards |
| 19:43.00 | bjohnson | damn .. put in user instead of username |
| 19:44.14 | bjohnson | still getting ulaw though |
| 19:44.15 | JerJer | guugmember: they are junk |
| 19:44.19 | JerJer | buy from Digium, support Asterisk |
| 19:45.10 | bjohnson | damn .. No authority found |
| 19:45.30 | modulus_ | hi jerjer |
| 19:45.32 | modulus_ | hi bjohnson |
| 19:45.33 | JerJer | add some authority then |
| 19:45.40 | JerJer | [bob] |
| 19:45.42 | JerJer | type=user |
| 19:45.45 | JerJer | context=hoe |
| 19:45.47 | bjohnson | for the devcon |
| 19:45.49 | km- | buy digium, digium r0x |
| 19:46.05 | bjohnson | <PROTECTED> |
| 19:46.08 | bjohnson | ? |
| 19:46.21 | JerJer | then that box is fux0red |
| 19:46.23 | JerJer | not yours |
| 19:46.33 | Hmmhesays | are you trying to get ot the conference? |
| 19:46.34 | Hmmhesays | *to |
| 19:46.36 | Hmmhesays | lol |
| 19:46.43 | bjohnson | yes |
| 19:47.03 | Hmmhesays | exten => dial,1,IAX2/guest@66.250.68.194/996 |
| 19:47.16 | JerJer | dial ?! |
| 19:47.18 | Hmmhesays | exten => 123456,1,IAX2/guest@66.250.68.194/996 |
| 19:47.20 | bjohnson | I can connect with ulaw with exten => _8,1,Dial(IAX2/guest@66.250.68.194/996) but I'm trying to force gsm |
| 19:47.22 | Hmmhesays | heh |
| 19:47.23 | JerJer | getting closer |
| 19:47.33 | Hmmhesays | lol, fark |
| 19:47.38 | JerJer | _8 ?! |
| 19:47.41 | bjohnson | err |
| 19:47.42 | bjohnson | 8 |
| 19:47.44 | bjohnson | same thing |
| 19:47.47 | modulus_ | jbot 50 shekel? |
| 19:47.51 | ManxPower | Apparently everyone's BRAIN is at LUNCH. |
| 19:47.54 | JerJer | how about 996,1,Dial,IAX2/guest@66.250.68.194/996 |
| 19:48.06 | Hmmhesays | my irc is farked |
| 19:48.07 | Hmmhesays | brb |
| 19:48.14 | *** part/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 19:48.51 | *** join/#asterisk kippi (~one@cpc1-hatf3-6-0-cust233.lutn.cable.ntl.com) |
| 19:48.51 | bjohnson | JerJer: but how to force gsm? I'm trying to make a iax.conf entry but keep getting no authority found |
| 19:48.55 | kippi | hi |
| 19:49.12 | znoG | anyone have firefly? |
| 19:49.14 | kippi | i have exten => 6696,1,Dial(sip/6696,20) |
| 19:49.15 | km- | manx: hey dude! |
| 19:49.15 | kippi | exten => 6696,3,Voicemail(u6696) |
| 19:49.15 | kippi | exten => 6696,4,Voicemail(b6696) |
| 19:49.15 | kippi | exten => 6666,102,Hangup |
| 19:49.20 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 19:49.33 | km- | manx: my dial problem was fixed by using CVS and the "emdigitwait" variable |
| 19:49.47 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 19:49.52 | Hmmhesays | heh much better |
| 19:49.59 | kippi | when i ring my extension and no one picks it up it dosn't go to the voicemail, how to I do this? |
| 19:50.04 | km- | manx: apparently others have had problems with dialing times on e&m wink systems |
| 19:50.05 | JerJer | bjohnson: are they allowing GSM? |
| 19:50.12 | JerJer | make a type=peer out of it |
| 19:50.17 | bjohnson | good question |
| 19:50.21 | bjohnson | I have a type-peer |
| 19:50.25 | bjohnson | type=peer |
| 19:50.34 | bjohnson | and get same error with allow=ulaw |
| 19:50.35 | JerJer | 996,1,Dial,IAX2/guest@devcon/996 |
| 19:51.39 | JerJer | how about /j #asterisk-dev ? |
| 19:52.57 | bjohnson | ? |
| 19:53.18 | bjohnson | ?? pastebin |
| 19:53.24 | bjohnson | ~pastebin |
| 19:53.25 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:53.25 | sivana | ManxPower: you there? |
| 19:54.36 | guugmember | JerJer, i will buy in the budget of my company for research doesnt allow me |
| 19:55.02 | guugmember | JerJer, if * runs ok they will give me more budget |
| 19:55.03 | bjohnson | http://pastebin.ca/6373 gives me chan_iax2.c:5510 socket_read: Call rejected by 66.250.68.194: No authority found |
| 19:55.10 | guugmember | JerJer, do you know what I mean? |
| 19:55.38 | Beirdo | bjohnson... |
| 19:55.49 | bjohnson | btw .. what IS difference between _8 and 8 as an extension? Wouldn't they perform the same? |
| 19:56.02 | Beirdo | exten => 7,1,Dial(IAX/guest@66.250.68.194/996) |
| 19:56.10 | bjohnson | Beirdo: yes I know |
| 19:56.14 | Beirdo | no need to set up a peer :) |
| 19:56.14 | |Vulture| | yea I just connected to the conf in both gsm and ilbc... ilbc sounds MUCH better |
| 19:56.22 | bjohnson | Beirdo: I'm trying to force a codec |
| 19:56.26 | Beirdo | ahh |
| 19:56.26 | Grooby | <---- sticking with ulaw |
| 19:56.27 | Grooby | lol |
| 19:56.31 | Hmmhesays | interesting conference |
| 19:56.36 | guugmember | any other Varion user? |
| 19:56.39 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 19:56.42 | kippi | hi, i have http://pastebin.ca/6374 but afer about 15 secs it just hangs up and dosn't go to vouc |
| 19:56.46 | bjohnson | Beirdo: I'm defaulting to ulaw .. I want to try gsm |
| 19:56.57 | kippi | voicemail, can anyone help? |
| 19:57.02 | Delvar | im off |
| 19:57.03 | Delvar | nn all |
| 19:57.12 | guugmember | km-, have you worked with varion? |
| 19:57.13 | Hmmhesays | 2 voicemail priorities? |
| 19:57.22 | Hmmhesays | how do you make it to the second one? |
| 19:57.34 | guugmember | km-, my problem is budget I am in the research stage of asterisk |
| 19:57.52 | |Vulture| | guugmember: whats the budget? |
| 19:58.31 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 19:58.39 | Hmmhesays | $5? |
| 19:58.47 | |Vulture| | lol |
| 19:59.12 | Mneumonic | anyone know an easy way to play the hold music 3 seconds before transferring the call to an extension? |
| 19:59.53 | guugmember | |Vulture|, US$2000, for the server, the phones and a card |
| 20:00.03 | |Vulture| | how many phones, and what kinda line? |
| 20:00.23 | guugmember | |Vulture|, have already bought them, grandstream 102 |
| 20:00.26 | guugmember | 3 phones |
| 20:00.32 | guugmember | |Vulture|, 1 IAXy |
| 20:00.39 | guugmember | |Vulture|, 1 US$900 server |
| 20:00.58 | guugmember | |Vulture|, a TDM04B for 4 fxo |
| 20:00.59 | shmaltz | anybody ever explored SprintPCSs' RadioLink feautre? it uses SIP |
| 20:01.00 | |Vulture| | jesus.. $900 server for a 3 phone test? |
| 20:01.29 | guugmember | |Vulture|, a $900 because we are looking to make it the * server of my company |
| 20:01.36 | |Vulture| | ah oky |
| 20:01.47 | guugmember | |Vulture|, so I cant ask for another US$1500 just for a test |
| 20:01.58 | |Vulture| | why do you need more $$? |
| 20:02.07 | |Vulture| | you have everything to test |
| 20:02.11 | guugmember | |Vulture|, have you experienced with varion cards? |
| 20:02.18 | |Vulture| | guugmember: no sorry |
| 20:02.20 | guugmember | |Vulture|, but E1 |
| 20:02.43 | guugmember | any one can give me professional consulting with Varion? |
| 20:03.06 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 20:04.06 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
| 20:04.44 | sivana | who's got a good call forwarding macro? |
| 20:05.16 | km- | heh |
| 20:05.22 | km- | we put out the $1500 for the TE405P |
| 20:05.22 | Hmmhesays | define 'good' |
| 20:05.25 | km- | very happy we did |
| 20:05.40 | km- | I'm learning something very interesting -- we always have at least 2 calls going on at once at our office |
| 20:05.46 | km- | I never realized we had that much call volume |
| 20:05.59 | Hmmhesays | 2 calls at once? |
| 20:06.10 | Hmmhesays | someone's got a dishwasher going there |
| 20:06.23 | km- | what? :) |
| 20:06.27 | jaiger | Hmmhesays, I think someone's hosing down his car |
| 20:06.38 | Hmmhesays | lol |
| 20:06.54 | km- | the humor is lost on me, someone explain to me wtf they're talkin about? :) |
| 20:06.56 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:07.16 | Hmmhesays | something about mark and his odd patches |
| 20:07.24 | km- | ahhh |
| 20:07.30 | bjohnson | sivana: the superdial macro |
| 20:07.30 | km- | conference call still going on I guess |
| 20:07.42 | |Vulture| | km-: there is a text chat going on too |
| 20:07.54 | km- | Anyone have any idea why I'm not getting caller ID on my T1? |
| 20:07.55 | Hmmhesays | yeah, i'm really glad these guys are smarter than me, i might be able to learn something, lol |
| 20:08.04 | km- | I thought all T1's had ANI, as in, it wasnt an option |
| 20:08.23 | agave-txlink | heh |
| 20:08.27 | Beirdo | interesting conference |
| 20:08.36 | Hmmhesays | i wish you could see who was talking though |
| 20:08.38 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 20:09.01 | `Sauron | Hmmhesaid. |
| 20:09.44 | bjohnson | ahhh .. why can't I connect??? |
| 20:10.15 | Beirdo | because you are trying to do it in a funky way :) |
| 20:10.21 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 20:10.22 | topping | in the text conference, is this like an anonymous IAX connection? |
| 20:10.28 | Hmmhesays | just put an extension in your extensions.conf |
| 20:10.29 | topping | erm, voice conference |
| 20:10.33 | lorion | anyone have problems diaing out via broadvoice? |
| 20:10.34 | Beirdo | it bridged me no problem using ilbc |
| 20:10.37 | bjohnson | Beirdo: so how do I force gsm |
| 20:10.44 | bjohnson | Beirdo: how? |
| 20:10.46 | Beirdo | I don't know that you can |
| 20:10.55 | kippi | how can i find out what the default number is for my voicemail? |
| 20:10.55 | Beirdo | I'm calling from Firefly in IAX mode |
| 20:11.10 | Hmmhesays | why are you forcing gsm? just put an extension in your dialplan and allow ilbc |
| 20:11.33 | bjohnson | ahh .. I'd like to figure out how to do other codecs in *. I think my config looks right .. just doesn't work |
| 20:11.47 | sivana | bjohnson: where would I find it? |
| 20:11.57 | Beirdo | voipjet's such a bitch to me lately |
| 20:13.27 | bjohnson | sivana: wiki |
| 20:13.27 | topping | is there a phone client like firefly that will do IAX that works on Linux? |
| 20:13.35 | bjohnson | topping: iaxcomm |
| 20:13.45 | bjohnson | I don't know if it's like firefly |
| 20:14.00 | topping | that's okay, just a client... thanks! :-) |
| 20:14.21 | Grooby | beirdo, how you feel about firefly |
| 20:14.22 | Grooby | ?? |
| 20:14.31 | sivana | bjohnson: is that the one ManxPower made? |
| 20:14.33 | Grooby | i tried it once..never really like the interface.... |
| 20:14.54 | Beirdo | I hate the interface, but it works |
| 20:14.59 | Grooby | ok |
| 20:15.17 | Grooby | hehe...pretty much how I feel.... |
| 20:16.12 | bjohnson | sivana: don't know |
| 20:16.23 | bjohnson | sivana: one what? |
| 20:16.50 | KalD|Work | not I =( |
| 20:17.37 | *** join/#asterisk Alric (~nbowyer@64.6.45.2) |
| 20:18.57 | shmaltz | anybody ever explored SprintPCSs' RadioLink feautre? it uses SIP |
| 20:19.00 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
| 20:21.23 | yasha | Weird Problem: I have 2 EXT. One will ring and NOT go into VM (eventially call will timeout/hang up), the othes goes into VM when not answered like it should. Any ideas what it might be? |
| 20:21.51 | Inv_arp | hmmm ilbc or gsm .. both sound same for me for dev/conf ilbc seems like it demands less bandwidth tho |
| 20:21.51 | Grooby | xit |
| 20:24.50 | Beirdo | If only I could put the voices with the nicks :) |
| 20:24.57 | terrapen | hey |
| 20:25.00 | terrapen | how do i get on the conf. call |
| 20:25.06 | terrapen | or is it over? |
| 20:25.20 | bjohnson | according to my iax debug my peer is trying to connect to number 996 in context 996 .. where does it get that context from? |
| 20:25.20 | Beirdo | read the topic. |
| 20:25.24 | Beirdo | it's still going |
| 20:25.26 | yasha | guys, does anyone have an idea??? |
| 20:25.29 | Juggie | has anyone done anything with the cisco 7960 in a language other then french? |
| 20:25.35 | terrapen | i don't know hwo to dial that, beirdo |
| 20:25.40 | Juggie | er, other then english |
| 20:25.40 | terrapen | do i just assign it an extension |
| 20:25.41 | Juggie | i ment |
| 20:25.43 | terrapen | and dial the extension? |
| 20:25.45 | Beirdo | yes |
| 20:25.46 | terrapen | ok |
| 20:25.58 | Beirdo | just assign an extension to dial to it. |
| 20:26.04 | bjohnson | exten => 7,1,Dial(IAX2/guest@66.250.68.194/996) |
| 20:26.06 | Beirdo | then make sure to mute yourself :) |
| 20:27.14 | Shaneful | I have an FXO card in asterisk. have setup the dial plan like I did with my PRI card. but when asterisk dials an outside number it dials g1/EXTEN |
| 20:27.17 | terrapen | word |
| 20:27.18 | terrapen | im on |
| 20:27.30 | terrapen | how is this speaking? |
| 20:27.38 | terrapen | eww |
| 20:27.40 | terrapen | err |
| 20:27.41 | terrapen | who |
| 20:28.14 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 20:28.35 | terrapen | are we all muted by default? |
| 20:28.43 | terrapen | i put my phone on mute anyway |
| 20:28.46 | Inv_arp | terrapen: i would assume so |
| 20:28.46 | Beirdo | I would expect so |
| 20:28.56 | jsolares | it does say, mutted when you dial in |
| 20:28.57 | Beirdo | I manually muted on my end too |
| 20:29.09 | terrapen | so i guess they unmute the core devs |
| 20:29.12 | Inv_arp | some idiot woulda screamed on it by now |
| 20:29.31 | bjohnson | why would my peer try to call the 996 context? |
| 20:29.41 | Beirdo | I dunno |
| 20:29.58 | Inv_arp | bjohnson: still try to use gsm? |
| 20:29.59 | Beirdo | I'm not gonna fight trying to make a peer definition when this works fine |
| 20:30.08 | yasha | Weird Problem, please help: I have 2 EXT. One will ring and NOT go into VM (eventially call will timeout/hang up), the othes goes into VM when not answered like it should. Any ideas what it might be? |
| 20:30.13 | terrapen | what is that video game noise in the background |
| 20:30.19 | Inv_arp | terracon: yea i get it too |
| 20:30.26 | Inv_arp | terracon: what codec u using? |
| 20:30.30 | terrapen | ulaw |
| 20:30.38 | terrapen | maaybe it means ppl are joining |
| 20:30.39 | terrapen | or leaving |
| 20:30.43 | Inv_arp | hmm im ilbc |
| 20:30.47 | bjohnson | a haaa exten => 7,1,Dial(IAX2/asteriskdevcon/996@) |
| 20:30.50 | Inv_arp | yea prob |
| 20:30.59 | terrapen | ; Asterisk Conference call |
| 20:30.59 | terrapen | exten => 7070,1,Wait(1) |
| 20:30.59 | terrapen | exten => 7070,2,Dial(IAX2/guest@66.250.68.194/996) |
| 20:30.59 | terrapen | exten => 7070,3,Hangup |
| 20:31.01 | terrapen | :P |
| 20:31.02 | |Barcode | I'm using ilbc and I hear the sound too. |
| 20:31.33 | bjohnson | I had the beep in ulaw too |
| 20:31.38 | bjohnson | got gsm working |
| 20:31.56 | ariel_ | just wanted to let everyone know I was at the show in Miami today and the Digium booth where Mark was over run with people. |
| 20:32.06 | *** join/#asterisk nani707 (~nene@nat-149.sjc1.globix.net) |
| 20:32.15 | nani707 | Hello everybody, |
| 20:32.20 | Inv_arp | ariel_: ahh man i forgot... where in miami? |
| 20:32.22 | ariel_ | There were so many people trying to see him that there was a traffic jam. |
| 20:32.34 | ariel_ | Inv_arp, are you not here. |
| 20:32.39 | |Barcode | How often do these conferences take place.. Should I keep this extension in my extensions.conf? |
| 20:32.44 | ariel_ | The are open Friday too. |
| 20:32.51 | bjohnson | http://pastebin.ca/6375 here's how it works as a peer to control codec |
| 20:32.52 | Inv_arp | ariel_: in miami yes show no |
| 20:33.11 | nani707 | I tried to connect DVG-1402S voip adaptor to Asterisk, iam getting sip 403 forbidden error, is there a quickfix |
| 20:33.19 | Inv_arp | ariel_: where can i get more info addr etc... |
| 20:33.21 | bjohnson | no beep in gsm |
| 20:33.43 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 20:33.44 | bjohnson | nani707: what is DVG-1402S ? |
| 20:33.45 | *** join/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 20:33.46 | terrapen | who is this speaking? |
| 20:33.53 | ManxPower | I would NEVER write a SOFTPHONE! |
| 20:33.54 | visik7 | what is gfp ? |
| 20:34.01 | nani707 | it is a 2port FXS with inbuilt router |
| 20:34.08 | ariel_ | Inv_arp, about addr??? |
| 20:34.17 | nani707 | it is a 2port FXS with inbuilt router DVG-1402S |
| 20:34.26 | nani707 | same one used for Lingo |
| 20:34.30 | bjohnson | nani707: good. first find out what protocol it uses |
| 20:34.31 | cbachman | |Barcode, they are trying for every thursday |
| 20:34.35 | nani707 | however i could unlock with known password on net |
| 20:34.38 | Inv_arp | ariel_: address website? heard about it but totally forgot |
| 20:34.50 | ariel_ | ok just a sec. |
| 20:34.53 | nani707 | supports SIP2.0 Johnson |
| 20:35.07 | mkhan | Hello.. I am first time here . would anybody help me to make something understand |
| 20:35.07 | |Barcode | cbachman: Thanks. Now I have something to keep me awake on Thursdays.. lol |
| 20:35.26 | bjohnson | if you have a working * then just define it in sip.conf and extensions.conf |
| 20:35.52 | ariel_ | Inv_arp, www.itexpo.com |
| 20:36.01 | Inv_arp | woah total 6.6 kbytes using ilbc nice thruput for conf |
| 20:36.02 | bjohnson | oops .. got the beep |
| 20:36.08 | Inv_arp | me 2 |
| 20:36.13 | nani707 | i have working asterisk and defined it in the sip.conf [1001] |
| 20:36.13 | nani707 | type=friend |
| 20:36.13 | nani707 | username=1001 |
| 20:36.13 | nani707 | secret=1001 |
| 20:36.13 | nani707 | host=dynamic |
| 20:36.13 | nani707 | context=partner |
| 20:36.15 | nani707 | defaultip=192.168.10.201 |
| 20:36.17 | nani707 | canreinvite=no |
| 20:36.25 | nani707 | dtmfmode=rfc2833 |
| 20:36.25 | nani707 | if i use X-lite it works |
| 20:36.33 | bjohnson | Inv_arp: where do you get the bitrate? |
| 20:36.42 | nani707 | if i use it with Dlink VOIP adaptor i get SIP 403 |
| 20:36.46 | nani707 | forbidden error |
| 20:36.47 | Inv_arp | bjohnson: is us iptraf on my * box |
| 20:36.52 | Inv_arp | err use |
| 20:36.58 | bjohnson | oh |
| 20:38.05 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 20:38.12 | nani707 | did anybody receive SIP 403 before |
| 20:38.17 | nani707 | any fix for that |
| 20:38.35 | Katty | gosh, 300 people :) |
| 20:38.50 | nani707 | the digest authentication is not working i beleive, |
| 20:38.57 | bjohnson | nani707: that looks ok |
| 20:39.06 | nani707 | but is there a way to force basic authenticaiton? |
| 20:39.10 | *** join/#asterisk GDRA (~1054@209.51.68.120) |
| 20:39.14 | bjohnson | nani707: if you can connect as that with xlite .. then the problem must be the hardware |
| 20:39.20 | Inv_arp | Incoming rates: 3.2 kbytes/sec 35.0 packets/sec |
| 20:39.24 | Inv_arp | Outgoing rates: 3.3 kbytes/sec 34.8 packets/sec |
| 20:39.38 | nani707 | the same hardware i used with FWD and it worked johnson |
| 20:39.49 | Inv_arp | hmm getting echo now |
| 20:40.03 | GDRA | Anyone have a minute for what I think should be a simple solution to a problem? |
| 20:40.04 | jsolares | woa calls on nufone for the 23'rd were free, upgrading software rules |
| 20:40.41 | bjohnson | nani707: perhaps I spoke woringly .. it isn't the hardware, it's the hardware config |
| 20:40.55 | GDRA | I've considered that Katty |
| 20:41.08 | Katty | you didn't consider it enough! *grin* |
| 20:41.14 | nani707 | config i have double checked, and sure iam using right parameters |
| 20:41.15 | Katty | what's your problem? |
| 20:41.24 | GDRA | Well, then I considered what I would do when I lost my job... |
| 20:41.47 | Katty | could get messy that way... |
| 20:41.48 | GDRA | Basically I want to have 1 asterisk server connect to another asterisk server directly, no voip provider between them |
| 20:42.13 | nani707 | I guess it is IAX trunking GDRA i did not implement that yet though |
| 20:42.56 | GDRA | Would it be possible to just use SIP? |
| 20:43.24 | bjohnson | GDRA: you could use SIP .. but most people use iax for that |
| 20:43.26 | *** part/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 20:43.29 | nani707 | other way might be just to define SIP extensions on both sides and goup them , |
| 20:43.31 | km- | hmm, my x-lite transfer button is greyed out |
| 20:43.42 | km- | do I need to set a special option in sip.conf to allow transfers? |
| 20:43.51 | bjohnson | km-: no |
| 20:43.51 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 20:44.09 | Inv_arp | GDRA: IAX2 all the way no nat probs, trunking is avail less overhead etc.. etc.. |
| 20:44.19 | nani707 | Johnson can you suggest me a good asterisk prepaid application, |
| 20:44.24 | bjohnson | km-: three ways to transfer I think. Each are documented on wiki. None need sip.conf config |
| 20:44.29 | bjohnson | nani707: no |
| 20:44.39 | km- | bjohnson: just wondering why the transfer button is greyed out |
| 20:44.47 | bjohnson | do you mean a voip provider? |
| 20:44.53 | bjohnson | km-: no idea here |
| 20:45.04 | bjohnson | I don't use xlite |
| 20:45.17 | km- | what do you use? |
| 20:45.25 | km- | I'm thinking of using iaxcomm |
| 20:47.43 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:47.57 | buddah | does * support faxing at all? |
| 20:48.14 | *** part/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 20:48.15 | denon | buddah: sorta, google for asterisk rxfax txfax |
| 20:48.21 | buddah | k |
| 20:48.44 | terrapen | anybody listening to this discussion on the conf call? |
| 20:48.47 | ScarletCrusader | can some one give me an example of a sip registration format ir [username]:[password]@[ip-address or dns] not sure if this is correct |
| 20:48.48 | KalD|Work | yeah |
| 20:48.50 | km- | denon: hey! |
| 20:48.54 | KalD|Work | terrapen, I am |
| 20:48.55 | km- | denon: ltns |
| 20:48.57 | terrapen | i'm trying to understand why they want to pull all that stuff out of the care |
| 20:48.58 | Beirdo | no, nobody's listening... :) |
| 20:49.00 | terrapen | err core |
| 20:49.07 | ScarletCrusader | thanks |
| 20:49.10 | denon | hey km |
| 20:49.14 | ScarletCrusader | i'll keep hacking at it |
| 20:49.15 | Beirdo | just a pile of us |
| 20:49.21 | *** join/#asterisk Capouch (501@12.176.248.4) |
| 20:49.21 | terrapen | and move it to, essentially, plug-ins |
| 20:49.24 | denon | how ya been doin? |
| 20:49.28 | terrapen | not doubting it |
| 20:49.32 | terrapen | just trying to understand |
| 20:49.38 | km- | denon: hehe, cant complain, we got Asterisk here at work now... Trying to work out all the kinks |
| 20:49.43 | denon | terrapen: for a number of reasons .. its nice to have a streamlined core, if you want to make an appliance, for example, that just does a few things .. |
| 20:49.45 | terrapen | can someone fill me in? |
| 20:49.46 | km- | denon: do you know much about T1's? |
| 20:50.03 | denon | terrapen: its also less code to break .. which would impact the core reliability of the pbx |
| 20:50.09 | terrapen | ok. |
| 20:50.15 | Beirdo | denon: which one of the speakers are you? :) |
| 20:50.29 | terrapen | but wouldn't the overall progress of asterisk slow down? |
| 20:50.34 | denon | hah, I dont do much speaking on * |
| 20:50.43 | Beirdo | ah |
| 20:50.47 | *** join/#asterisk Ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 20:50.48 | terrapen | because you are depending on implementations of commonly used apps to be done in each laanguage |
| 20:50.52 | terrapen | ie perl, ruby, etc |
| 20:50.59 | denon | terrapen: on the contrary .. with things as modules, devs can go off on their own tangents, without code having to be really robustly tested in the core |
| 20:51.13 | Inv_arp | its a good idea |
| 20:51.19 | terrapen | denon, exactly....but then everybody is working for themselves as opposed to a common goal |
| 20:51.30 | dan2 | sweet |
| 20:51.39 | dan2 | my panasonic cordless sip phone will arrive shortly |
| 20:51.40 | denon | terrapen: well, not really -- everyone works on their respective modules, and you have standards in place for them to plug in .. |
| 20:51.40 | terrapen | you'll have 20 implementations of Dial(), etc |
| 20:51.41 | Juggie | has anyone done more then english on a cisco 7960? |
| 20:51.48 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
| 20:52.11 | denon | terrapen: if you have lots of people working on the core, there's more chance of their efforts overlapping, and conflicting things getting changed |
| 20:52.23 | Beirdo | and things breaking :) |
| 20:52.27 | denon | think of it like kernel development vs applications |
| 20:52.30 | terrapen | i don't doubt about the power of it |
| 20:52.31 | Inv_arp | dan2: panasonic cordless sip how much? |
| 20:52.38 | dan2 | Inv_arp: free |
| 20:52.45 | Beirdo | I like the direction you guys are thinking of taking |
| 20:52.47 | terrapen | but it seems that the core team would have to implement the common, popular applications in each language first |
| 20:52.49 | denon | you dont want Sun developing Sun Office into the linux kernel, do you? |
| 20:52.49 | jsolares | terrapen: but they wont get into the cvs |
| 20:52.52 | terrapen | so that they get done correctlyu |
| 20:52.54 | Inv_arp | dan2: perf price |
| 20:52.56 | dan2 | Inv_arp: (I work for a voip company (broadvoice)) |
| 20:53.05 | terrapen | denon, of course not |
| 20:53.07 | lorion | has anyone seen an error "Maximum retries exceeded on call"? |
| 20:53.09 | Inv_arp | dan2: heh k |
| 20:53.14 | denon | terrapen: I dont think I follow .. each language. . |
| 20:53.25 | terrapen | but you want to make sure that it works right and works the same as another word processing package |
| 20:53.31 | terrapen | maybe i'm confused about this |
| 20:53.33 | jsolares | you dont have to write them for each language terrapen, just the api |
| 20:53.38 | bjohnson | km-: I use hardware these days |
| 20:53.44 | Beirdo | terrapen: you'd only need to make an app in *one* language |
| 20:53.52 | terrapen | they are talking about moving much of the functionality to 'external', non-core code |
| 20:53.59 | terrapen | that could be implemented in various languges |
| 20:54.00 | Beirdo | hopefully, people tend to use the same languages |
| 20:54.00 | terrapen | correct? |
| 20:54.12 | denon | right .. most likely just moving the existing C into modules .. |
| 20:54.19 | Katty | well, I suppose I'll jump right in |
| 20:54.35 | jsolares | i dont think people are going to go and recode everything in every language out there |
| 20:54.40 | Katty | i'll be setting up asterisk for the Very First Time and haven't the slightest clue about it. |
| 20:54.42 | denon | no, there's no need |
| 20:54.59 | Katty | I suppose my first question will be the prefered OS |
| 20:55.03 | denon | Katty: linux |
| 20:55.05 | bjohnson | Katty: linux |
| 20:55.06 | Katty | does it depend upon hardware |
| 20:55.08 | Katty | obviously |
| 20:55.10 | bjohnson | nope |
| 20:55.10 | Katty | but which one |
| 20:55.14 | denon | Katty: thats a distribution :) |
| 20:55.17 | Inv_arp | Katty: any |
| 20:55.18 | Katty | i'm currently using debian |
| 20:55.22 | jsolares | that's fine |
| 20:55.24 | bjohnson | debian seems to be a winner |
| 20:55.24 | Inv_arp | Katty: me 2 |
| 20:55.28 | denon | Katty: everyone likes everyhting .. debian is good though |
| 20:55.29 | terrapen | jsolares, i just wonder about the consistency |
| 20:55.32 | Katty | i've talked to one guy who sets it up and says that my hardware tends to like debian the best |
| 20:55.36 | bjohnson | followed by slackware and redhat types |
| 20:55.39 | denon | terrapen: the api enforces consistency |
| 20:55.43 | Inv_arp | Katty: debian 350mhz 128meg ram actually |
| 20:55.46 | bjohnson | Katty: he's an idot |
| 20:55.49 | bjohnson | Katty: he's an idiot |
| 20:55.50 | km- | I've run debian for many years now |
| 20:55.52 | jsolares | not only that, but you have someone in charge of the cvs |
| 20:55.54 | denon | terrapen: a strong set of standards actually keeps the core much more stable .. |
| 20:55.54 | Katty | bjohnson: actually, he's not *smile* |
| 20:56.08 | Katty | bjohnson: especially considering asterisk is just a side business for him (= |
| 20:56.09 | jsolares | Katty: actually if he's suggesting that, he is |
| 20:56.11 | CMike | hi all |
| 20:56.19 | Katty | k'then |
| 20:56.19 | bjohnson | Katty: you're hardware won't know or care what distro your running |
| 20:56.23 | km- | bjohnson: her hardware may play better with debian's stock kernel, is what he means. |
| 20:56.35 | Katty | bjohnson: what about the actual digium cards though? |
| 20:56.35 | denon | right |
| 20:56.40 | km- | bjohnson: don't be so quick to call people an idiot, it only shows an inferiority complex! |
| 20:56.53 | Inv_arp | heh |
| 20:56.57 | denon | Katty: Debian is perfect, dont give it another thought. |
| 20:57.04 | Katty | denon: ok :) |
| 20:57.08 | km- | I've seen many systems behave differently based on what stock kernel you're running |
| 20:57.14 | km- | redhat's kernel is no where close to vanilla |
| 20:57.16 | jsolares | terrapen: unless you're the mantainer of the asterisk code as a whole, you dont need to worry on the consistency ;) |
| 20:57.27 | jsolares | ubuntu!! |
| 20:57.33 | km- | haha |
| 20:57.35 | km- | GENTOO! |
| 20:57.37 | Katty | denon: I think i'm right to the point of downloading source (possibly headers) so I can get the drivers for the cards |
| 20:57.43 | km- | where's that gentoo hater's website |
| 20:57.48 | Katty | denon: it's actually a miracle i got debian installed in the first place, heh. |
| 20:57.50 | km- | theres some funny joke pics about gentoo users on it |
| 20:58.00 | jsolares | Katty: very recent hardware? |
| 20:58.00 | Sedorox | hmmm |
| 20:58.01 | tzanger | km-: what, funroll-loops.org? |
| 20:58.04 | Sedorox | gentoo!!! |
| 20:58.07 | Katty | jsolares: pardon? :) |
| 20:58.17 | km- | ahhh yeah ;) |
| 20:58.17 | Katty | jsolares: I don't know much, so you'll have to be a bit...hrmm...patient :) |
| 20:58.18 | jsolares | Katty: do you have very recent hardware on your pc? |
| 20:58.31 | jsolares | like an intel 915 board |
| 20:58.37 | Katty | jsolares: moment... |
| 20:58.39 | Juggie | grr... i cant find anything on cisco and other languages for the interface other then english |
| 20:58.40 | km- | funroll-loops.org is hillarious |
| 20:58.51 | terrapen | jsolares: i guess its just a matter of where the line is drawn between "core" stuff and user-community implemented stuff |
| 20:59.06 | km- | tzanger: hey, I didnt see if you shot me a message earlier -- do you have any idea why I'm not getting callerid from the t1? |
| 20:59.19 | km- | tzanger: do I have to set a specific option like callerid=yes in the zapata.conf, even on a T1? |
| 20:59.47 | jsolares | terrapen: not really, look at the linux kernel, core = bare minum, modules = rest; so asterisk could very well have the bare minimum to work called core, and the rest of the stuff called modules, but still mantained by the same people |
| 20:59.55 | tzanger | km-: zapata.conf should have callerid=asrecieved on the telco side |
| 21:00.04 | km- | asreceived |
| 21:00.05 | jsolares | it would make it easier for people outside of the regular coders make modules |
| 21:00.06 | km- | ok |
| 21:00.12 | km- | tzanger: not callerid=yes? |
| 21:00.21 | tzanger | no not yes |
| 21:00.24 | tzanger | callerid=asreceived |
| 21:00.35 | km- | gotcha |
| 21:00.39 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 21:00.44 | km- | for my own personal thirst of knowledge, what's the difference? |
| 21:00.48 | tzanger | and callerid="Blah COmpany <(123) 456-7890>" for the pbx side |
| 21:01.05 | Katty | way for me to miss-place my bundle of Stuff |
| 21:01.07 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmp2.dialup.mindspring.com) |
| 21:01.25 | terrapen | jsolares: gotchya |
| 21:01.53 | Katty | the hardware is fairly recent...let me dig through the database |
| 21:01.56 | buddah | good lord, t.38 fax bounty |
| 21:01.58 | jsolares | Katty: well, if it's very recent hardware you would need fairly recent builds of linux. i had several cd's of linux from debian stable to redhat 9, and none of them booted on my asterisk box |
| 21:02.00 | buddah | like $11,250 |
| 21:02.01 | terrapen | jsolares: and these modules could be implemented via APIs in any of the popular languages? |
| 21:02.11 | Beirdo | buddah: starting to get closer |
| 21:02.15 | km- | might I just make an observation |
| 21:02.15 | jsolares | that's how i see it terrapen |
| 21:02.17 | tzanger | km-: it's the difference between callerid=kittycats and callerid=puppydogs |
| 21:02.21 | tzanger | it's a totally different thing |
| 21:02.23 | tzanger | asreceived != yes |
| 21:02.30 | jsolares | lol |
| 21:02.33 | km- | whoever designed the command "restart when convenient" should get a medal |
| 21:02.47 | km- | it makes my life really easy |
| 21:02.48 | km- | hehe |
| 21:02.55 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 21:02.59 | tessier_ | Hello all! |
| 21:03.02 | Katty | jsolares: someone helped me update my kernel |
| 21:03.14 | Katty | jsolares: of course the box is turned off and sitting on a rack upstairs :) |
| 21:03.14 | tessier_ | Anyone know what the avt means in the dtmfmode setting of a cisco 7960? |
| 21:03.16 | km- | tzanger: what about usecallerid=yes |
| 21:03.20 | km- | tzanger: should I set that too? |
| 21:03.25 | jsolares | Katty: ah well then stick to that, unless you want to have headaches reinstallling another distro |
| 21:03.41 | tessier_ | For some reason the phones here can't talk to autoattendants because their dtmf is not recognized. I think they are sending something inband but mangled |
| 21:03.51 | tzanger | km-: not sure about tha tone |
| 21:03.56 | km- | eh, I'll set it |
| 21:04.01 | km- | if its a problem I'll just take it away |
| 21:04.15 | Inv_arp | woah from 6.6kbytes to 33.6 kybtes when by brother logs in to conf on my other box |
| 21:05.14 | buddah | do pap-2na's support t.38 fax? |
| 21:05.27 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 21:05.29 | Juggie | tessier, not sure but it works with default settings for me, inband & avt |
| 21:05.31 | km- | some of the users were reporting that they werte talking on the phone and the phone mysteriously disconnected on them, but, nobody can say for sure if it was the remote side hanging up or not |
| 21:05.38 | Katty | jsolares: k'then (= |
| 21:05.43 | Katty | I imagine I'll need lots of help. |
| 21:05.48 | Katty | Which is another thing that concerns me. |
| 21:05.51 | km- | users = suck. |
| 21:05.52 | Katty | I'm barely using mandrake and kde. |
| 21:06.03 | Katty | Mostly I'm just a windows tech |
| 21:06.19 | km- | Katty: you dont necessarily have to be a linux wiz to learn asterisk -- it's more like a seperate beast that just uses linux as it's home |
| 21:06.22 | Katty | I have a feeling getting a linux box with asterisk on it is going to be slightly more than I can handle in the first place |
| 21:06.27 | Katty | km-: ok :> |
| 21:06.34 | Katty | does it run inside a gui? |
| 21:06.40 | jsolares | nope |
| 21:06.42 | Katty | k |
| 21:06.49 | Katty | i was talking to another person.. |
| 21:06.50 | km- | asterisk is a console application |
| 21:06.53 | Katty | we got asterisk on a wee little cd |
| 21:07.08 | Katty | but she said that it was terribly old and i should just use debian's built in hrmm...apt-get? to get asterisk |
| 21:07.16 | JerJer | oh god |
| 21:07.17 | JerJer | evil |
| 21:07.23 | km- | apt-get is the debian package management system |
| 21:07.26 | *** join/#asterisk Moc__ (~mochouina@64.235.210.66) |
| 21:07.27 | km- | however |
| 21:07.40 | Katty | that was back when i was using the Current Stable Version, and we switch the source list to the newer one |
| 21:07.54 | Katty | Sarge to woody, i think it was |
| 21:08.05 | Katty | is apt-get not a good idea? |
| 21:08.07 | km- | woody to sarge |
| 21:08.13 | km- | well, apt-get is good for regular software |
| 21:08.21 | Katty | not asterisk, then, i take it |
| 21:08.26 | km- | but asterisk is constantly evolving, faster than packages can be made |
| 21:08.43 | km- | as soon as someone makes an asterisk package, theres already a new asterisk feature that's worth including |
| 21:09.18 | Katty | so.. |
| 21:09.30 | Katty | do you update it all the time and pray it doesn't break? |
| 21:09.37 | tessier_ | Weird that dtmf works with the local voicemail system but not the autoattendant on another companies phone system |
| 21:09.38 | denon | now you're gettin it |
| 21:09.38 | *** join/#asterisk xeet2 (~xeet2@es.jsci.net) |
| 21:09.39 | Katty | i obviously have no idea how it works :) |
| 21:09.46 | km- | well, that depends if you're using it as a hobbyist or a business |
| 21:09.53 | Katty | well... |
| 21:09.53 | Inv_arp | Katty: nah i just update if i need a new feature.. or if there is a major prob |
| 21:09.55 | denon | Katty: its best to find a revision that does everything you need it to do .. test it well .. then leave it alone |
| 21:09.58 | km- | I'm not going to be upgrading every day from CVS because I'm running a business here |
| 21:10.08 | Katty | personally, this is just a hobby thing for me |
| 21:10.15 | Katty | our company is using some third party stuff |
| 21:10.27 | Katty | and in the meantime, i'm supposed to figure out how to get the company onto our own asterisk box |
| 21:10.33 | Katty | there's no rush on it |
| 21:10.42 | km- | the only problem with 'restart when convenient' is that people are on the friggin phone all the time |
| 21:10.45 | Katty | i'd like to learn how to do it right, obviously |
| 21:10.47 | tessier_ | hmm...when I call my cell the dtmf sounds like two short little pulses |
| 21:10.57 | Grooby | is there such thing as "do it right?" |
| 21:10.59 | Grooby | hehehehe |
| 21:11.05 | Katty | Grooby: gosh i hope so :> |
| 21:11.16 | Grooby | i started doing * as a hobby too |
| 21:11.17 | km- | there are ways to make Asterisk installs look like they're done right |
| 21:11.25 | Grooby | i musta reinstall 10000 times |
| 21:11.25 | Grooby | lol |
| 21:11.27 | xeet2 | tessier: what are you calling from? and what are you calling via? pots? sip provider? |
| 21:11.27 | km- | but that takes a good bullshit artist |
| 21:11.33 | Grooby | and I am still a noob |
| 21:11.52 | Katty | hmm, so i guess this will get messy |
| 21:11.57 | Katty | as in some things will work while others won't |
| 21:11.57 | xeet2 | km-: its just like the networking world =P |
| 21:11.58 | Grooby | yup |
| 21:11.58 | km- | oh yeah |
| 21:12.05 | Katty | dumb question |
| 21:12.05 | km- | if you're doing this as a hobby |
| 21:12.12 | Katty | does the sound card have to work in the machine? |
| 21:12.13 | km- | you *WILL* find things that are broken |
| 21:12.14 | km- | its a given |
| 21:12.24 | km- | katty: you don't need a sound card for asterisk |
| 21:12.27 | *** join/#asterisk Mike_TK (~Mike@213.180.245.62) |
| 21:12.29 | Katty | km-: k'then |
| 21:12.30 | km- | its cool for intercom |
| 21:12.34 | km- | but you dont need it |
| 21:12.40 | Katty | there's a soundcard in the box, but debian didn't pick it up |
| 21:12.40 | tessier_ | xeet2: Calling from my 7960 through an asterisk box out a PRI to my cell phone |
| 21:12.52 | Katty | i might be able to figure out how to make debian see it, but i'm not sure...that's a bit beyond me right now |
| 21:12.55 | tessier_ | xeet2: Phone is currently set to its default of "avt" |
| 21:13.01 | km- | katty: baby steps. |
| 21:13.04 | tessier_ | xeet2: and I have rfc2833 in the sip.conf for this phone |
| 21:13.10 | km- | katty: linux is a lot to handle in 24 hours :) |
| 21:13.27 | tessier_ | I saw a book called "Linux in 24 hours" so it must be possible. |
| 21:13.29 | tessier_ | ;) |
| 21:13.32 | Grooby | hahahahaha |
| 21:13.35 | km- | hahahaha |
| 21:13.54 | km- | after 9 years I'm still on hour 23! |
| 21:14.01 | tessier_ | I wish I had that book 12 years ago when I started playing with Linux. Took me a good couple of years to really learn it. |
| 21:14.22 | tessier_ | Actually, not 12 years ago. |
| 21:14.27 | km- | I bought "Linux Unleashed" in '95 |
| 21:14.32 | tessier_ | I started playing with it in 93....wait, that is 12 years. Wow. |
| 21:14.35 | Katty | km-: it sure is |
| 21:14.40 | Katty | km-: i'm taking it slow :) |
| 21:14.42 | Beirdo | denon: you guys should call it obelisk |
| 21:14.48 | km- | HAHAH |
| 21:14.50 | tessier_ | Linux is no longer an "upstart" OS, that's for sure. |
| 21:14.55 | tessier_ | It's older than NT/2000 etc |
| 21:14.55 | Katty | 'upstart'? |
| 21:14.57 | xeet2 | tessier: ok, well I'd say its doing what it supposed to... Since your cell provider will actually receive the dtmf tones out of band, and your cell doesn't really know what to do with them, you don't hear it... correct me if I'm wrong, but out of band dtmf works across isdn right? |
| 21:15.16 | km- | Beirdo: are you calling us monkeys? :) |
| 21:15.22 | Beirdo | no |
| 21:15.24 | Katty | is asterisk as difficult as installing debian? :< |
| 21:15.24 | km- | oh wait, that's monolith |
| 21:15.30 | km- | katty: worse |
| 21:15.35 | Katty | eep! |
| 21:15.38 | Sedorox | lol |
| 21:15.40 | km- | katty: the hurdles are only in the beginning though |
| 21:15.41 | Katty | time to hide under the bed |
| 21:15.41 | tessier_ | xeet2: Right....however I just realized I left out a part: The asterisk is talking to a cisco 5300 via SIP. |
| 21:15.42 | Sedorox | Katty: once you get it installed... |
| 21:15.44 | nestAr | asterisk is easy to install |
| 21:15.45 | Beirdo | have you ever read the asterix comics? |
| 21:15.45 | Sedorox | and start working with it |
| 21:15.48 | Sedorox | it isn't bas |
| 21:15.49 | Sedorox | bad* |
| 21:15.50 | ScarletCrusader | YES!!!!!!!!! |
| 21:15.57 | xeet2 | I find * on gentoo the best, but then I'm biased towards gentoo anyway |
| 21:15.58 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 21:15.59 | Sedorox | I started with it 3 weeks ago and the dialplan looked greek |
| 21:16.03 | Sedorox | I now know it pretty well |
| 21:16.14 | km- | xeet2: hahahaha, I was just reading funroll-loops.org! |
| 21:16.25 | Katty | Sedorox: i can only hope i'll pick it up as quickly :) |
| 21:16.25 | Sedorox | that actual install isn't bad |
| 21:16.30 | Sedorox | its the configuration |
| 21:16.35 | km- | oh yeah, I've got all sorts of cool extension snippets |
| 21:16.39 | Sedorox | voip-info.org.. or whatever it is.. |
| 21:16.41 | xeet2 | tessier: and the 5300 goes to the pri? what is the dtmfmode set to for the 5300 peer? |
| 21:16.43 | Sedorox | GREAT site |
| 21:16.45 | Katty | i iamgine i'll be tugging on a few sleeves :) |
| 21:16.47 | tessier_ | And dtmf works fine on our local system....so I bet the cisco is fudging things up somewhere. |
| 21:16.54 | Katty | s/iamgine/imagine |
| 21:16.56 | Sedorox | hehe |
| 21:17.02 | Sedorox | this channel is also very helpful |
| 21:17.08 | ScarletCrusader | is there a command to show currently defined extension or one that have been register with asterisk? |
| 21:17.08 | jsolares | echooooo |
| 21:17.09 | tessier_ | xeet2: Yes, 5300 has the PRI plugged into it. Not sure what dtmfmode is on the 5300. Gonna check that out now with my cisco guy. Thanks! |
| 21:17.09 | km- | katty: there are always people around here who are willing to help |
| 21:17.10 | Katty | irc is always helpful |
| 21:17.12 | Sedorox | that is.. if you try all you can and are really suck :-p |
| 21:17.20 | xeet2 | tessier: look at what it is on * |
| 21:17.27 | Katty | Sedorox: i'm always stuck. hehe |
| 21:17.39 | km- | hehe |
| 21:17.42 | Sedorox | hehe |
| 21:17.46 | km- | I come here and whine about my T1 |
| 21:17.48 | Sedorox | just read through the docs |
| 21:17.52 | Sedorox | when you start working with it |
| 21:17.52 | Katty | strangely enough i was introduced to linux because of my irc addiction |
| 21:17.53 | jsolares | mhnoyes: its open you dimwits :P j/k |
| 21:17.55 | Katty | i wanted irc, all the time |
| 21:17.57 | Sedorox | do a few examples.. comes pretty easy |
| 21:17.58 | xeet2 | tessier: it might actually be working just fine, you're not really supposed to hear dtmf tones on your cell if your provider received them out of band. have you tried calling into another pbx across the pstn? |
| 21:17.59 | jsolares | -mhno |
| 21:17.59 | Sedorox | lol |
| 21:18.04 | Sedorox | I was like that |
| 21:18.07 | Katty | and so now i screen irssi |
| 21:18.07 | Sedorox | my middle school days |
| 21:18.11 | Sedorox | IRC, sleep, school... |
| 21:18.13 | Sedorox | repeat |
| 21:18.14 | Katty | Sedorox: mmhmm |
| 21:18.22 | Katty | Sedorox: now it's just irc, sleep, work, repeat |
| 21:18.23 | xeet2 | you forgot sex |
| 21:18.30 | xeet2 | wait, no |
| 21:18.33 | Sedorox | xeet2: I haven't had sex yet.. thanks :-p |
| 21:18.34 | Katty | sex? what's that? |
| 21:18.40 | Sedorox | lol |
| 21:18.41 | xeet2 | forget that, thats so pre-linux |
| 21:18.47 | Sedorox | lol |
| 21:18.55 | Katty | it /so/ is |
| 21:19.26 | Katty | well, i suppose i'll go drag the box off the rack |
| 21:19.31 | Sedorox | I have a g/f right now.. so :-p |
| 21:19.34 | heison | ~seen JerJer |
| 21:19.48 | jbot | jerjer is currently on #asterisk. Has said a total of 19 messages. Is idling for 12m 31s |
| 21:19.49 | Sedorox | hehhe |
| 21:19.49 | Grooby | vulture, what happen when you load speex? |
| 21:19.49 | Katty | Sedorox: you lucky thing you! |
| 21:19.49 | |Vulture| | Grooby: Ill pastebin it |
| 21:19.49 | Sedorox | :-p |
| 21:19.59 | terrapen | vulture, you just need speex installed, then rebuild *, then allow=speex |
| 21:20.01 | terrapen | that shouuld be it |
| 21:20.08 | |Vulture| | terracon: ahh oky |
| 21:20.10 | Sedorox | with what? lol |
| 21:20.11 | |Vulture| | thats what it is then |
| 21:20.11 | Grooby | yup |
| 21:20.17 | Grooby | well..i also had to do ldconfig |
| 21:20.22 | |Vulture| | can I get speex off of * ftp? |
| 21:20.25 | Grooby | yup |
| 21:20.29 | Grooby | speex.org |
| 21:20.29 | *** join/#asterisk ropeguru_work (~ropeguru@141.152.37.26) |
| 21:20.37 | Essobi | Anyone using anything to manage meetmes from a web interface? |
| 21:20.38 | Katty | i think i'll just leave this up for a bit |
| 21:20.38 | Grooby | i just d/l the source code |
| 21:20.46 | km- | the wife is telling me to get home |
| 21:20.48 | km- | hehe |
| 21:20.48 | Sedorox | kk |
| 21:20.53 | Sedorox | lol km- |
| 21:20.56 | km- | snowstorm here in philly |
| 21:20.59 | Grooby | terrapen, how were your speex quality? |
| 21:21.04 | terrapen | well..... |
| 21:21.06 | Sedorox | it isn't that bad, it is down there? |
| 21:21.09 | |Vulture| | installing stable |
| 21:21.10 | terrapen | i'm using it currently with NuFone |
| 21:21.10 | Sedorox | I'm from that area... |
| 21:21.11 | km- | not really a snowstorm, more like, a snow shower that everyone is like "OMG OMG 8"!" |
| 21:21.12 | Grooby | i can't make anything out what I said |
| 21:21.18 | terrapen | and it seems to sound a lot nicer than gsm |
| 21:21.20 | Sedorox | I know.. |
| 21:21.21 | km- | 8 inches is not a storm |
| 21:21.22 | Grooby | nufone..it's IAX client? |
| 21:21.24 | terrapen | so i guess i'm pretty happy |
| 21:21.28 | |Vulture| | no |
| 21:21.28 | terrapen | i still get some jitter |
| 21:21.30 | Sedorox | I heard they cancealed school for the possiblility of snow |
| 21:21.30 | km- | 14", that's a storm |
| 21:21.31 | terrapen | but that could be my upstream |
| 21:21.36 | Sedorox | we're only suppose to get 4" |
| 21:21.37 | |Vulture| | nufone is a VoIP provider |
| 21:21.37 | terrapen | i'd say speex works nicely |
| 21:21.38 | km- | yeah, that's what they did around here |
| 21:21.40 | Grooby | ah ok |
| 21:21.43 | km- | sedorox: whereabouts are you? |
| 21:21.45 | terrapen | of course, there is no phone support for it, AFAIK |
| 21:21.47 | km- | ahh yeah |
| 21:21.52 | km- | I hear cape may is going to get hammered |
| 21:21.53 | Grooby | i tried it last night and i get horrible quality using xlite |
| 21:21.56 | km- | like 10" or something |
| 21:22.00 | terrapen | except for Firefly softphone |
| 21:22.06 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 21:22.14 | terrapen | grooby, is this on your local LAN? |
| 21:22.16 | Sedorox | hehe |
| 21:22.21 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
| 21:22.22 | km- | heh, ok, time to find my way home |
| 21:22.24 | terrapen | or to a remote voip provider? |
| 21:22.26 | km- | talk to you later guys! |
| 21:22.27 | Grooby | yup |
| 21:22.32 | Grooby | local lan |
| 21:22.36 | Grooby | call 1 exten to another |
| 21:22.37 | terrapen | that's really odd |
| 21:22.42 | terrapen | it works *great* for me in that aspect |
| 21:22.49 | terrapen | what are you running? |
| 21:22.57 | terrapen | and are you doing speex<-->speex? |
| 21:23.06 | terrapen | or speex<----> * <----> gsm |
| 21:23.08 | Katty | ...or not |
| 21:23.10 | shido6 | Grooby u have a switch or a hub on that LAN of yours? |
| 21:23.11 | Beirdo | man, that echo is annoying ;) |
| 21:23.22 | Katty | i'll just stay and pester the geeks instead. |
| 21:23.30 | xeet2 | grooby: are any of your clients via wireless? |
| 21:23.31 | Grooby | switch |
| 21:23.34 | terrapen | someone on that dev conf call needs to get a nicer microphone |
| 21:23.43 | [Outcast] | exit |
| 21:23.46 | terrapen | or push the boom mic away from his mouth :) |
| 21:23.46 | Katty | here's another question |
| 21:23.47 | [Outcast] | oops |
| 21:23.51 | [Outcast] | wrong screen |
| 21:23.55 | Katty | let's pretend i've got broadband at home |
| 21:24.07 | Katty | and i somehow, by way of a miracle, get this asterisk box going |
| 21:24.07 | *** join/#asterisk cads (~gary@gary.istop.com) |
| 21:24.09 | xeet2 | *pretends* |
| 21:24.10 | Essobi | WEEE PORN |
| 21:24.11 | shido6 | Grooby full duplex sound card? noise cancelling mic? |
| 21:24.15 | Katty | and decide to set myself one up at home |
| 21:24.23 | terrapen | you shouldn't have problems with Speex on a LAN |
| 21:24.26 | terrapen | you have something else wrong |
| 21:24.33 | Katty | can i just dump the phone system? |
| 21:24.35 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 21:24.40 | Katty | use uhh, softphone of some sort |
| 21:24.40 | *** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net) |
| 21:24.46 | Grooby | i think it's full duplex |
| 21:24.52 | harryvv | Anyone seen a case where the volume from the mouthpiece of a analog phone is very quiet on the spa1k? |
| 21:24.53 | Grooby | it works fine w/ every other codec except speex |
| 21:25.01 | xeet2 | grooby: try some pings, file transfers between the same two machines/devices, see what happens |
| 21:25.13 | terrapen | grooby, which ver of speex are you running? |
| 21:25.22 | Grooby | whatever .04 |
| 21:25.24 | Grooby | not the beta |
| 21:25.25 | xeet2 | katty: what do you mean can you just dump the phone system? which phone system, your old pbx? |
| 21:25.26 | terrapen | are you using some old version or some CVS version or some version that came with your OS? |
| 21:25.27 | Grooby | but the latest stable |
| 21:25.33 | terrapen | you built the latest stable, yourself? |
| 21:25.35 | Katty | xeet2: yup :) |
| 21:25.36 | Grooby | yeah |
| 21:25.40 | *** join/#asterisk Ad-Hoc (~ad-hoc@62.1.246.83) |
| 21:25.40 | terrapen | that's really odd |
| 21:25.40 | Katty | xeet2: as in southwestern bell |
| 21:25.52 | Katty | can i use my regular broadband line? |
| 21:25.53 | terrapen | speex-1.1.6 |
| 21:25.56 | terrapen | that's what i'm running |
| 21:25.58 | terrapen | and it kicks ass |
| 21:26.00 | Grooby | let me try that |
| 21:26.07 | Katty | or do i need another dedicated broadband line specifically for the asterisk box |
| 21:26.09 | terrapen | remember to rebuild asterisk too |
| 21:26.15 | xeet2 | katty: well thats entirely up to you, but * is quite capable of doing what most other pbx's do as well or better |
| 21:26.17 | Grooby | ok |
| 21:26.19 | Grooby | brb |
| 21:26.36 | harryvv | terrapen, I have not used speeks but is it proving to be 100% reliable? |
| 21:26.49 | xeet2 | katty: try it and see =) every broadband connection is different |
| 21:26.50 | Katty | xeet2: i guess what i'm asking is if i can setup asterisk on a linux box, and use that linux box for both my phone system and regular computer |
| 21:26.53 | |Vulture| | I am building with "Stable" right now |
| 21:26.58 | xeet2 | oh |
| 21:27.14 | xeet2 | not recommended |
| 21:27.25 | |Vulture| | xeet2: was that to me? |
| 21:27.25 | Katty | 2 different boxes on the same broadband line? |
| 21:27.32 | Katty | xeet2: would that work maybe? |
| 21:27.47 | Grooby | got 1 dumb question |
| 21:27.48 | xeet2 | katty: yes, that would, just get yourself a decent broadband router, ie linksys |
| 21:27.56 | Katty | xeet2: oh ah :> |
| 21:27.56 | Grooby | can I "make install" while asterisk is running? |
| 21:27.57 | |Vulture| | wrt54g |
| 21:28.03 | Grooby | or should I shut it down |
| 21:28.07 | |Vulture| | Grooby: yes |
| 21:28.14 | Grooby | ok |
| 21:28.26 | Grooby | that's what i've been doing..but thought maybe i can "cheat" |
| 21:28.29 | Grooby | :) |
| 21:28.31 | Katty | what should my next baby step be? |
| 21:28.34 | Katty | I've got debian loaded |
| 21:28.39 | |Vulture| | you don't need to shut it down |
| 21:28.41 | Katty | and i've updated the kernel and all the packages |
| 21:28.50 | Grooby | w00t! go Katty! |
| 21:28.50 | Katty | though i might want to update it again, if i can remember how...hmm |
| 21:28.59 | Grooby | apt-get update |
| 21:29.04 | Grooby | apt-get dist-upgrade? |
| 21:29.05 | Katty | that's not kernel though |
| 21:29.11 | Katty | dist-upgrade sounds framilier |
| 21:29.15 | Sedorox | lol |
| 21:29.19 | Essobi | Why is there no decent web interfaces for MeetMe? |
| 21:29.23 | Essobi | GRR. |
| 21:29.24 | Katty | i haven't looked for about a month ;) |
| 21:29.24 | Sedorox | screw debian.. get slackware or gentoo... :-p |
| 21:29.28 | tuxinator_linux | Morning Sedorox |
| 21:29.33 | xeet2 | essobi: write one? |
| 21:29.34 | Katty | Sedorox: pffft, i'm just a little geek :) |
| 21:29.35 | Essobi | Gentoo is for ricers. |
| 21:29.35 | |Vulture| | omg my dev box sucks hardcore |
| 21:29.35 | Sedorox | Morning |
| 21:29.37 | Sedorox | lol |
| 21:29.39 | Essobi | xeet2 I will. |
| 21:29.42 | xeet2 | =) |
| 21:29.43 | Sedorox | lol |
| 21:29.45 | Sedorox | I run Gentoo |
| 21:29.46 | |Vulture| | takes forever to build * |
| 21:29.46 | Sedorox | SysInfo | System: Linux 2.6.10-cko3 | CPU: Dual Intel(R) Pentium(R) 4 CPU 3.20GHz 3200.627 MHz | Mem: 193/1031Mb (19%) | Diskspace: 92Gb Free: 54Gb | Bogomips Per CPU: 6324.22 | Screen Res: 1680x1050 | Procs: 111 | Uptime: 3:40 | Connection Device: Realtek Semiconductor Co., Ltd. RTL-8169 Gigabit Ethernet (rev 16). In: 5.64Mb Out: 0.93Mb |
| 21:29.51 | Sedorox | eh.. doesn't show... allwell |
| 21:30.03 | Essobi | how often do you re-build? |
| 21:30.08 | Essobi | :) |
| 21:30.16 | xeet2 | how can anyone not like gentoo |
| 21:30.30 | xeet2 | <awaits a storm of text |
| 21:30.32 | Katty | we don't know how to use it!!! |
| 21:30.44 | Sedorox | lol |
| 21:30.47 | Katty | heh, 2 months ago i didn't know how to use an iso ;) |
| 21:30.50 | Essobi | I know perfectly well how to use it. It was modeled after a nicer OS. FreeBSD. |
| 21:30.52 | Grooby | bjohnson, that's not true |
| 21:30.54 | Grooby | DOS rules! |
| 21:30.58 | Sedorox | Essobi: you mean update? about once a week.. or twice.. I try... |
| 21:31.00 | Essobi | DrDOS FOO |
| 21:31.03 | Katty | someone hand to hold my hand and say, k, angela, download this...and use nero to do this |
| 21:31.16 | Sedorox | lol |
| 21:31.18 | Katty | and then i was all :>>>>> and OOoooo, iso :>> |
| 21:31.22 | mikegrb | lolz |
| 21:31.26 | Essobi | Sedorox Oh you binary fetch or rebuild your updates? |
| 21:31.26 | bjohnson | Grooby: google "Every OS Sucks" by Three Dead Trolls in a Baggie .. it's a funny mp3 |
| 21:31.27 | tessier_ | xeet2: The cisco has dtmf-relay rtp-nte enabled which according to the docs is rfc2833 |
| 21:31.35 | tessier_ | bjohnson: Yes, that is funny |
| 21:31.35 | Sedorox | emerge sync |
| 21:31.37 | Sedorox | emerge -uD world |
| 21:31.38 | Sedorox | :-p |
| 21:31.41 | mikegrb | Sedorox: you sure do laugh out loud a bunch |
| 21:31.44 | tessier_ | bjohnson: Although I wish they had mentioned that Linux sucks less than most. :) |
| 21:31.45 | xeet2 | essobi: ok, I can agree with that. I was more pointing towards in the linux group |
| 21:32.03 | Sedorox | mmm yup |
| 21:32.03 | Katty | so, right. |
| 21:32.10 | Katty | should i try to update debian again? |
| 21:32.11 | marc_c | how come few use the varion cards? |
| 21:32.11 | |Vulture| | failed to load codec_speex.so |
| 21:32.16 | xeet2 | tessier: ok, what does * say for the cisco 5300 peer? |
| 21:32.18 | marc_c | it's cheaper. |
| 21:32.20 | |Vulture| | gunna try unstable version |
| 21:32.24 | Grooby | trying to find it |
| 21:32.25 | Grooby | lol |
| 21:32.25 | xeet2 | if its not matched it can be confusing |
| 21:32.25 | Sedorox | Katty: I'm assuming female? (just courious.. don't have to answer) |
| 21:32.35 | Katty | Sedorox: quite. www.brick.net/~izaah (+ |
| 21:32.36 | Katty | (= |
| 21:32.44 | tessier_ | xeet2: dtmfmode=rfc2833 |
| 21:32.44 | Grooby | vulture, which distro u using? |
| 21:32.45 | Sedorox | ah.. hot |
| 21:32.47 | Essobi | tessier You got a problem with a 5300? I just fixed up my 5400s in *. |
| 21:32.49 | harryvv | bj, seen a case where your ata OR analog phone mouthpiece was very very quiet to the calling party? |
| 21:32.51 | Sedorox | chick thats into Linux AND asterisk :-p |
| 21:32.52 | bjohnson | Grooby: dcc? |
| 21:32.59 | bjohnson | harryvv: yes |
| 21:33.04 | Grooby | sure |
| 21:33.06 | Grooby | send it this way |
| 21:33.06 | Katty | Sedorox: pffft, i'm just curious :) |
| 21:33.09 | bjohnson | harryvv: I've been playing with gain |
| 21:33.16 | tessier_ | Essobi: Having a dtmf problem somewhere. The beeps come across as two really short blips |
| 21:33.17 | Sedorox | hehe |
| 21:33.18 | Grooby | is he awake? |
| 21:33.19 | Katty | Sedorox: i'll get ticked off if you treat me like a steak :P |
| 21:33.20 | Grooby | hehehe |
| 21:33.22 | Sedorox | I have asterisk here in my dorm |
| 21:33.23 | Katty | Sedorox: so don't even start :P |
| 21:33.27 | |Vulture| | Grooby: FC3, 1.0.5, 1.0.4 speex |
| 21:33.29 | harryvv | bj in the ata or * |
| 21:33.33 | Sedorox | eh? |
| 21:33.33 | Grooby | vulture, ldconfig |
| 21:33.33 | Sedorox | mmm ok |
| 21:33.39 | Grooby | i had that problem |
| 21:33.45 | Katty | Sedorox: no drool, kthx |
| 21:33.47 | Sedorox | like I said.. I was just courious.. I wouldn't treate you any difference |
| 21:33.49 | Sedorox | no... |
| 21:33.50 | Katty | s/drool/drooling |
| 21:33.52 | Sedorox | like I said.. I have a g/f |
| 21:33.58 | Katty | excellent |
| 21:34.02 | Sedorox | and in my eyes.. no one is hotter then her |
| 21:34.02 | Sedorox | hehe |
| 21:34.03 | |Vulture| | Grooby: it returned nothing |
| 21:34.07 | Grooby | that's fine |
| 21:34.07 | Sedorox | (sorry if it offends you) |
| 21:34.09 | Grooby | now try to run it |
| 21:34.10 | xeet2 | tessier: hmmm, ok, try no'ing out that config line, I think thats for cas signalling |
| 21:34.13 | Katty | not at all (= |
| 21:34.15 | Sedorox | :-p |
| 21:34.20 | Essobi | Tessier what codec? |
| 21:34.25 | Katty | but i am in the market for a girlfriend |
| 21:34.26 | Sedorox | I just like to see women working with this kinda stuff |
| 21:34.27 | Grooby | i was having the same problem here on CentOS |
| 21:34.29 | xeet2 | tessier: on the 5300 that is |
| 21:34.29 | Essobi | SIP? Reinvites? |
| 21:34.33 | Sedorox | hehe |
| 21:34.37 | chaoscon | Sedorox: mine wants to learn it ;) |
| 21:34.41 | Sedorox | hehe |
| 21:34.41 | Sedorox | nice |
| 21:34.44 | tessier_ | xeet2: I just called from my cell phone to my cisco 7960 and I don't get beeps on the 7960 when I press buttons on my cell phone but I do get the short beeps on my cell when I push buttons on the 7960. |
| 21:34.54 | Katty | Sedorox: are you normally in here? |
| 21:35.02 | Sedorox | yea.. when I'm on irc I'm here... |
| 21:35.03 | Grooby | basically the /usr/local/xxxx/ (forgot what it was) wasn't in so path |
| 21:35.07 | Beirdo | Katty: many of us like having geek chicks around, it keeps us more honest :) |
| 21:35.09 | Katty | i must find a Main Contac to pester the hell out of |
| 21:35.12 | tessier_ | Essobi: All ulaw here. No reinvites. |
| 21:35.20 | xeet2 | tessier: yeah that sounds like the 5300 is reproducting dtmf tones on the pri, while transmitting them out of band at the same time |
| 21:35.22 | Essobi | HRM. |
| 21:35.29 | Sedorox | honest.. and sane... and well.. mature |
| 21:35.29 | Sedorox | lol |
| 21:35.33 | Sedorox | Katty: thats fine |
| 21:35.34 | |Vulture| | Grooby: http://pastebin.ca/6376 |
| 21:35.36 | Essobi | got the inline turned on? |
| 21:35.37 | Sedorox | I like to help where Ican... so... |
| 21:35.54 | xeet2 | tessier: you don't need to reproduce dtmf on a pri, that command is usually used on cas circuits or analog |
| 21:35.55 | Grooby | vulture, that was the error i was getting |
| 21:36.01 | Sedorox | I know how it is to start and be clueless on stuff.. and have someone to help you along.. so I like to do the same to others |
| 21:36.04 | Beirdo | Sedorox: yeah, that too |
| 21:36.06 | Essobi | tessi Got inline DTMF turned on? |
| 21:36.09 | *** join/#asterisk cogi (~root@titanic.pjwstk.edu.pl) |
| 21:36.12 | Katty | :> |
| 21:36.15 | xeet2 | essobi: tessier is doing out of band dtmf |
| 21:36.22 | tessier_ | Essobi: Should be all rfc2833 |
| 21:36.29 | Essobi | Do inband if you're running ULAW. |
| 21:36.29 | Beirdo | knowing that women are around will make us act more like gentlemen :) |
| 21:36.33 | tessier_ | We use a lot of g729 also so I want everything rfc2833 |
| 21:36.34 | Katty | should i apt-get asterisk or not? |
| 21:36.38 | Grooby | i asked on the usergroup |
| 21:36.40 | Sedorox | hehe |
| 21:36.41 | Grooby | let me dig up that email on the fix |
| 21:36.46 | tessier_ | The phones here in house are ulaw but most of our phones are g729 |
| 21:36.47 | |Vulture| | Grooby: thanx |
| 21:36.53 | roamer323 | geek chick - haha , thought it was geekette :-D |
| 21:36.55 | xeet2 | essobi: why? out of band dtmf is always better if you can do it |
| 21:36.55 | Sedorox | dunno.. I don't use debian.. I know people that do... check to see what version it is.. if its 1.0.5.. yes |
| 21:36.56 | Katty | use the one provided on the cd that's horribly old? |
| 21:37.04 | Essobi | xeet2: that's the problem. He can't. |
| 21:37.05 | Essobi | :) |
| 21:37.07 | xeet2 | only reproduce tones closest to the destination |
| 21:37.12 | xeet2 | well, we're working on that part =) |
| 21:37.15 | Sedorox | if it isn't 1.0.5.. I would say go download it |
| 21:37.15 | Essobi | I think debians only up to 1.0.3 |
| 21:37.16 | *** join/#asterisk cogi (~cogi@titanic.pjwstk.edu.pl) |
| 21:37.21 | tessier_ | Essobi: I should be able to though. It's gotta be a confused config somewhere. |
| 21:37.21 | Katty | roamer323: you need a special drive for a geekette :P |
| 21:37.25 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 21:37.36 | xeet2 | tessier: try removing that config line on the 5300 |
| 21:37.37 | Essobi | sounds like it |
| 21:37.43 | Sedorox | lol |
| 21:37.43 | Essobi | what's your dial peer look like? |
| 21:37.45 | cogi | hi |
| 21:37.55 | Grooby | vulture, got my message? |
| 21:37.57 | xeet2 | its making the 5300 reproduce dtmf tones, and send out of band dtmf on the pri |
| 21:37.58 | Katty | Sedorox: will you need to know what my kernel version is to tell me which one to download? |
| 21:38.08 | tessier_ | xeet2: Ah...I'll try that. |
| 21:38.11 | Katty | Sedorox: is it based on i386 or 64, etc |
| 21:38.16 | tessier_ | xeet2: I am surprised it would let us do both at once. |
| 21:38.26 | Sedorox | no.. there is one download on asterisk's site |
| 21:38.28 | cogi | does someone have new asterisk-oh323 package? www.inaccessnetworks.com is down... and i've just get access to h332 gateway and would like to test it |
| 21:38.30 | xeet2 | it technically doesn't, its kind of broken |
| 21:38.34 | Katty | Sedorox: k'then |
| 21:38.35 | Sedorox | unless you want the addons or sounds.. then its 2 or three :-p |
| 21:38.38 | xeet2 | cisco? broken? who knew!?! |
| 21:38.39 | Essobi | xeet2 thats cisco for you |
| 21:38.45 | Sedorox | but yea.. its just a tarball (.tar.gz) |
| 21:38.48 | tessier_ | I gotta run for a bit, I'll get our cisco guy to make that change when I get back and let you know how it goes. Thanks for the tips! |
| 21:38.49 | Sedorox | one-size-fits-all |
| 21:38.53 | Sedorox | kinda... |
| 21:38.55 | Katty | one-size-fits-4 |
| 21:38.59 | Essobi | but I suppose it could be feasible to have a need for it. |
| 21:39.01 | Sedorox | lol |
| 21:39.02 | tuxinator_linux | Katty is cute geeket |
| 21:39.03 | xeet2 | tessier: sure np let us know |
| 21:39.04 | Katty | from the women's department at walmart |
| 21:39.15 | Katty | tuxinator_linux: so they claim...i'm still not convinced :P |
| 21:39.29 | Essobi | tuxinator_linux Maha.. I converted my woman. She won't let me work on her PC anymore. she does it herself now. |
| 21:39.44 | tuxinator_linux | My wife much prefers linux |
| 21:39.46 | Beirdo | Essobi: left too much pr0n behind? |
| 21:39.50 | Essobi | lol |
| 21:39.53 | ariel_ | argh I hate voip faxing.... |
| 21:39.55 | Katty | i think anyone who switches from windows to linux will be converted |
| 21:39.55 | tuxinator_linux | but she's not to geeky yet |
| 21:40.09 | Katty | i'd switch for screen alone :> |
| 21:40.16 | Katty | erm |
| 21:40.17 | Beirdo | not having to reboot daily.. Mmmmm. |
| 21:40.19 | Essobi | She called me a work one day.. asking me where the lapping compound and the wet/dry sandpaper was |
| 21:40.20 | znoG | its easier to NOT have a wife; no problem with which OS each one prefers. |
| 21:40.22 | Katty | ScarletCrusader: oops (= |
| 21:40.25 | [Outcast] | :q |
| 21:40.33 | Sedorox | Katty: I do second tuxinator_linux... dont' let anyone tell you that your not... |
| 21:40.42 | znoG | it's best to have a number of girlfriends, they have no right to say which OS they prefer. |
| 21:40.47 | tuxinator_linux | i don't really care for the goth look |
| 21:41.10 | xeet2 | lol |
| 21:41.10 | Beirdo | znoG: bah |
| 21:41.10 | tuxinator_linux | but the other pics are cute |
| 21:41.10 | Katty | tuxinator_linux: then don't look ;) |
| 21:41.10 | Sedorox | lol.... I love the goth.. but |
| 21:41.10 | Katty | tuxinator_linux: i'm more psuedogoth/indie now |
| 21:41.11 | Sedorox | ANYWAY.... |
| 21:41.26 | Inv_arp | whos is the main person talking in dev? |
| 21:41.33 | tuxinator_linux | I like the cute Mandy Moore type (like my wife) |
| 21:41.47 | Sedorox | tuxinator_linux: lol |
| 21:42.18 | BrianR___ | inbound DTMF from my fxo -> cisco 7940 is muted. I thought it was normal. |
| 21:42.18 | Katty | i have a thing for geek females too |
| 21:42.18 | tuxinator_linux | I don't know why she choose me ;-) |
| 21:42.18 | Katty | especially the ones who have as much hair as i do |
| 21:42.22 | Grooby | hmm |
| 21:42.23 | BrianR___ | Since the tones on the fxo are converted to out of band... The out of band signals may be getting received at the cisco, but it's under no obligation to play them. |
| 21:42.27 | Grooby | my spa2k went crazy |
| 21:42.30 | Sedorox | I probably have more hair then you... but eh |
| 21:42.35 | cogi | so no one can help me with the asterisk-oh323 package ? |
| 21:42.35 | Sedorox | I havemore hair them my g/f |
| 21:42.35 | xeet2 | brianr: fxo > zaptel card > * > SIP > 7940? |
| 21:42.37 | Katty | Sedorox: oooooh, how much? :> |
| 21:43.05 | Sedorox | when my back is straight.. it goes to the bottom of my shoulder blades... |
| 21:43.05 | Katty | Sedorox: if you stand up, where does it fall to? |
| 21:43.08 | BrianR___ | xeet2: well.. fxs > zaptel fxo card > asterisk > SIP > 7940 |
| 21:43.13 | Katty | you don't have a lot of hair |
| 21:43.19 | Sedorox | was longer.. then my mom 'trimed it' |
| 21:43.21 | Katty | mine goes well below the hips :) |
| 21:43.24 | Sedorox | for a guy it is :-p |
| 21:43.26 | Sedorox | nice |
| 21:43.26 | Beirdo | I used to have hair half way down my back |
| 21:43.28 | Katty | it sure is |
| 21:43.30 | Katty | and hair is dreamy |
| 21:43.35 | Sedorox | lol |
| 21:43.37 | xeet2 | brianr: yeah thats normal, there's no need for the 7940 to play dmtf tones... Are you having a problem with * recognizing dtmf? |
| 21:43.38 | Grooby | terrapin, 1.1.6 works a hell lot better |
| 21:43.43 | Beirdo | now it's about 3mm long |
| 21:43.45 | xeet2 | er, dtmf |
| 21:43.52 | Katty | i never can find males with long hair out here |
| 21:43.59 | BrianR___ | xeet2: Nah. Someone else was mentioning it as if it's broken. I don't think it necessarily a problem. |
| 21:44.03 | Sedorox | what state you in anyway? |
| 21:44.05 | Beirdo | it was all falling out anyways. |
| 21:44.07 | Katty | missouri |
| 21:44.13 | cads | Does anyone have any experience with TDM04B's? |
| 21:44.13 | Katty | right smack in the middle of the bible belt |
| 21:44.15 | Sedorox | ahh ok |
| 21:44.17 | Sedorox | lol |
| 21:44.20 | BrianR___ | it may be a problem if certain ATA's don't play the tones. BUt for a desk set, I'd just assume not have the noise in my ear. |
| 21:44.23 | Katty | which eliminates nearly all chances of finding a bi female too |
| 21:44.27 | roamer323 | Grooby - spa2k spasms? |
| 21:44.27 | Katty | i need to move |
| 21:44.32 | xeet2 | brianr: its by design, * see's the dtmf tone and takes it out of the rtp stream |
| 21:44.34 | Sedorox | I was gonna say.. if you wee up here.. I might know someone for ya :-p |
| 21:44.35 | Beirdo | Katty: at least one that will admit it |
| 21:44.38 | Sedorox | were* |
| 21:44.46 | tuxinator_linux | Katty: Whats wrong with us? |
| 21:44.47 | Grooby | yeah...had to restart that booger |
| 21:44.48 | Grooby | hehehe |
| 21:44.53 | Katty | Beirdo: ya, there are a couple on okcupid (= |
| 21:44.55 | Katty | tuxinator_linux: hmm? |
| 21:44.57 | Grooby | but i am impressed by speex at this point |
| 21:45.10 | Grooby | just dunno how well it will work when I am away on client site |
| 21:45.12 | xeet2 | brianr: for a device like that, you would want to set the dtmfmode to inband, which would force it to be in the audio |
| 21:45.16 | Katty | tuxinator_linux: wrong with who? |
| 21:45.31 | Beirdo | men are pigs. :) |
| 21:45.35 | BrianR___ | xeet2: Only if the ATA itself doesn't do the DTMF. |
| 21:45.36 | tuxinator_linux | Katty: guys |
| 21:45.42 | Katty | tuxinator_linux: not a single thing (= |
| 21:45.46 | roamer323 | Hey Kat - what are you looking to hookup with your * ? |
| 21:45.49 | Katty | or maybe everything ;) |
| 21:45.53 | xeet2 | brianr: right. |
| 21:45.55 | BrianR___ | xeet2: Since inbound DTMF is going to get mangled if the codec is not ulaw/alaw |
| 21:46.00 | Sedorox | depends on the guy... |
| 21:46.02 | Katty | roamer323: moment, i'll go grab one of the phones |
| 21:46.23 | tzanger | anyone with a name like katty you know you'll get tha tkind of answer |
| 21:46.32 | xeet2 | brianr: yes, I was thinking of a situation where you would have one of those ata devices connected to an older pbx |
| 21:46.32 | BrianR___ | got my asterisk to talk to fwd today.. nifty. |
| 21:46.41 | *** join/#asterisk Blackvel (~blackvel@dsl-082-082-059-189.arcor-ip.net) |
| 21:46.45 | znoG | Katty: come to argentina - it's practically impossible finding a male with shorter hair than any woman |
| 21:47.03 | mishehu | boom |
| 21:47.16 | cads | Does anyone have any experience with TDM04B's? |
| 21:47.23 | znoG | not cause they like it, i think it's just people's tight financial position that they can't afford a haircut :) |
| 21:47.26 | BrianR___ | xeet2: Yes.. Applications like legacy voicemail, old style answering machines, those little IVR boxes that monitor the temperature at a vacation home, etc. |
| 21:47.27 | xeet2 | cads: whats the question/problem? |
| 21:47.29 | Katty | Soundpoint IP 500 |
| 21:47.39 | Beirdo | heya mishehu |
| 21:47.41 | Katty | there is another phone...uhh...somewhere |
| 21:47.52 | mishehu | cads: why not just *ask* your question? |
| 21:47.53 | Beirdo | someone was mocking your domainname earlier |
| 21:47.54 | xeet2 | brianr: icky, I hate migration problems |
| 21:47.54 | roamer323 | Kat: cool hardware |
| 21:47.56 | Sedorox | lol |
| 21:48.04 | Sedorox | I wish I had IP phones laying around like that |
| 21:48.10 | BrianR___ | xeet2: Heh. I'm doing an integration with a norstar MICS... |
| 21:48.17 | mishehu | Beirdo: ah, who? |
| 21:48.18 | cads | We have been trying to get a TDM04B to behave reliably...tried a total of 4 cards with the same result. |
| 21:48.18 | BrianR___ | xeet2: I have everything but caller id working using only analog lines |
| 21:48.22 | xeet2 | well its not like they're that much more expensive than a regular analog phone anymore |
| 21:48.23 | Katty | snomphone 190 is another one |
| 21:48.23 | Beirdo | geek chick with geek toys. nice :) |
| 21:48.32 | *** join/#asterisk Twister (~jase@216.30.232.106) |
| 21:48.34 | Beirdo | mishehu: I don't remember :) |
| 21:48.37 | Katty | there's a couple digium cards in the box |
| 21:48.39 | roamer323 | Kat appears to be a well2do geekette irc quotient ...+20 |
| 21:48.43 | Katty | but, again, it's sitting on a rack |
| 21:48.46 | *** join/#asterisk Leland (~leland@ws2.discpro.org) |
| 21:48.49 | xeet2 | brianr: any echo issues? |
| 21:48.55 | Leland | evening all |
| 21:48.59 | Katty | and i'm sure not getting the box off the rack without a little help |
| 21:49.01 | mishehu | Beirdo: you must tell me so that I may force them to view other domains with "goat" in them |
| 21:49.01 | Sedorox | 0_o |
| 21:49.02 | xeet2 | cads: define "reliably" whats going on? |
| 21:49.06 | Beirdo | heh |
| 21:49.07 | Sedorox | Katty: send some this way :-p |
| 21:49.12 | Katty | Sedorox: :<< |
| 21:49.13 | Beirdo | holy frigging echo |
| 21:49.15 | Katty | Sedorox: mine! |
| 21:49.17 | mishehu | cads: what is the issue? |
| 21:49.19 | Sedorox | :( |
| 21:49.21 | cads | They work for about a week, and then they become unstable... |
| 21:49.28 | Katty | Sedorox: you can come /here/ and get it yourself :P |
| 21:49.34 | Katty | Sedorox: and setup asterisk for me while you're at it :> |
| 21:49.38 | mishehu | cads: you mean as in "stop answering" ? |
| 21:49.38 | Leland | does anyone know of any "wallboard" applications for monitoring asterisk queues via the manager interface? |
| 21:49.39 | cads | We hear static on the lines and then the lines refuse to pick up. |
| 21:49.39 | roamer323 | Kat... you're amongst starving * developers with no hardware :-( |
| 21:49.40 | Sedorox | lol |
| 21:49.45 | BrianR___ | xeet2: None. |
| 21:49.46 | Sedorox | yea.. really |
| 21:49.48 | Katty | aww :< |
| 21:49.48 | xeet2 | cads: mmm, and when you reboot it all is back to normal? |
| 21:49.52 | BrianR___ | Going to get a PRI soon to get the caller id working, but.. |
| 21:49.57 | Katty | luckily my company paid for it |
| 21:50.05 | Katty | i'm so lucky to be at this company in the first place |
| 21:50.07 | mishehu | cads: known issue, though mine tend to work for 2-6 weeks before that happens. (minus the static bit) |
| 21:50.15 | cads | Sometimes all it takes is an unload/reload of the modules. |
| 21:50.19 | Katty | i don't deserve the position as network admin, heh, i'm barely a tech |
| 21:50.25 | cads | Sometimes a full power-cycle. |
| 21:50.37 | Katty | oh |
| 21:50.39 | xeet2 | cads: have the same problem, I usually just reboot any box with those cards about every night or every other night |
| 21:50.44 | Katty | isn't there a molex power connect on the cards? |
| 21:50.44 | Sedorox | wow... |
| 21:50.50 | Sedorox | some of them yes |
| 21:50.52 | Katty | the..hrmm..i should go dig up the paperwork on that card |
| 21:50.57 | tzanger | Katty: on the TDM400Ps yes |
| 21:51.00 | tzanger | and if they're not like 1st-gen |
| 21:51.07 | mishehu | cads: what happens if you stop asterisk, modprobe -r wctdm, wait a sec, modprobe wctdm, ztcfg, and relaunch asterisk? |
| 21:51.27 | xeet2 | mishehu: does digium know whats causing this? |
| 21:51.42 | cads | The module reloading works, but then the same issues appear, usually within the hour. |
| 21:52.01 | mishehu | xeet2: from what I know, yes. I have a replacement card from them with beta modules on it, haven't had a chance to down the server and swap it out though |
| 21:52.06 | xeet2 | cads: can you cron a nightly reboot until the bug is resolved? |
| 21:52.07 | *** join/#asterisk viLeR (~miv@aurora.telesat.com.co) |
| 21:52.17 | *** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com) |
| 21:52.19 | mishehu | cads: that souds like a different issue. what type of hardware are you using this on? |
| 21:52.35 | mishehu | xeet2: that's digusting. |
| 21:52.37 | mishehu | nightly reboot. |
| 21:52.48 | mishehu | I only down servers when absolutely necessary. |
| 21:52.50 | cads | I have replicated the results on three differnt machines. |
| 21:52.55 | Blackvel | who has bristuff 0.2.0-RC7f (asteirsk 1.0.5) running and has problem like me that DISA doesn't take anymore extension numbers without waiting for dail tone? |
| 21:52.57 | xeet2 | the pbx's reboot with all the windows boxes! |
| 21:53.01 | mishehu | rarely when I'm not on location. |
| 21:53.17 | cads | Could it be a line issue? |
| 21:53.42 | xeet2 | mishehu: I only have two of these, and haven't had time to mess with it... nightly reboot made an ok way around it |
| 21:53.43 | mishehu | cads: yes, it could. normal behavior is that the cards should resume normal operation after kernel mod reloading |
| 21:53.49 | |Vulture| | windows + pbx == evil! |
| 21:53.57 | Katty | well, after yanking the case off, the pretty puppy says Tiger 320 |
| 21:54.07 | Katty | and digium on the back |
| 21:54.08 | Grooby | vulture, got it to work? |
| 21:54.18 | |Vulture| | asterisk is rebuilding |
| 21:54.23 | Grooby | hehehe ok |
| 21:54.25 | mishehu | |Vulture|: windows + pbx == new way to unwittingly get infected and spam your friends |
| 21:54.25 | |Vulture| | its the dev machine so it takes awhile |
| 21:54.26 | Katty | there are 4 red ...card things |
| 21:54.28 | Grooby | what kinda machine is it running on? |
| 21:54.29 | xeet2 | mishehu: my issue is the same as cads, I have to reboot the entire box to bring it back up |
| 21:54.30 | Katty | are those digital? |
| 21:54.35 | Sedorox | no |
| 21:54.39 | Sedorox | their either fxo or fxs |
| 21:54.43 | Grooby | P 75mhz? |
| 21:54.43 | |Vulture| | mishehu: that would be a sweet ass virus |
| 21:54.44 | Grooby | :P |
| 21:54.44 | Sedorox | I forget what the red are |
| 21:54.44 | cads | Is rebooting nightly a common solution? |
| 21:54.52 | Katty | hmmmmm. |
| 21:54.56 | |Vulture| | Grooby: please! its a P400 |
| 21:54.56 | mishehu | |Vulture|: don't give anybody any bright ideas |
| 21:55.01 | Grooby | hehehehe |
| 21:55.03 | Katty | i'll go ask on another server |
| 21:55.05 | Katty | later |
| 21:55.06 | mishehu | xeet2: what revision? |
| 21:55.11 | Sedorox | lol |
| 21:55.13 | Sedorox | ok... |
| 21:55.14 | xeet2 | cads: not a good one, but if you're ok with it it would get around the issue for now |
| 21:55.15 | RGi_- | dows fax and asterisk with Cisco ata 182 adapters work ? good/bad ? |
| 21:55.15 | JerJer | cads: windows? sure |
| 21:55.19 | |Vulture| | Grooby: but my Dual Xeon 2.8 shipped out of Dell today :) |
| 21:55.23 | Katty | Sedorox: i talk on slashnet.org too :) |
| 21:55.27 | Grooby | :P |
| 21:55.28 | xeet2 | mishehu: e, with 4 fxs mods |
| 21:55.29 | Sedorox | ahhh ok |
| 21:55.39 | Sedorox | can't do alt+3 for them huh? |
| 21:55.41 | mishehu | RGi_-: fax and anything needs g711, and extremely low latency |
| 21:55.42 | Sedorox | and alt+4 for us? |
| 21:55.49 | Katty | you mean in irssi? |
| 21:55.53 | Sedorox | si |
| 21:55.55 | BrianR___ | xeet2: And I'm using the really cheap X100P cards from eBay. $10/ea. |
| 21:55.55 | RGi_- | mishehu :hmmf.. how low ? |
| 21:56.02 | Katty | Sedorox: umm, i don't know how :< |
| 21:56.02 | mishehu | xeet2: hmm, we have 1 e/f revision, and one h revision. both operate identically. |
| 21:56.08 | |Vulture| | CLONES! |
| 21:56.11 | Katty | Sedorox: i know how to connect to multiple servers in mirc |
| 21:56.20 | Katty | Sedorox: but not irssi yet |
| 21:56.22 | mishehu | RGi_-: recommended to be on a LAN. I've never tested over a wan |
| 21:56.29 | roamer323 | BrianR___ is there any call progress problem with those X100P clones? |
| 21:56.30 | JerJer | BrianR___: then don't bitch when you have problems |
| 21:56.32 | chaoscon | Katty: /connect <server> |
| 21:56.40 | Sedorox | hehe |
| 21:56.43 | cads | I'm not sure of the card revision, but we just received the replacements 4 weeks ago. |
| 21:56.43 | JerJer | when not if |
| 21:56.46 | Sedorox | <PROTECTED> |
| 21:56.47 | Sedorox | :-p |
| 21:56.49 | Katty | chaoscon: surely not for multiple servers |
| 21:56.57 | Katty | isn't /server -n or something |
| 21:56.58 | chaoscon | Katty: yes for multiple servers |
| 21:57.00 | chaoscon | no |
| 21:57.03 | mishehu | cads: it should say when you modprobe |
| 21:57.04 | mishehu | in dmesg |
| 21:57.06 | RGi_- | mishehu : hmmf.. well.. I have a phone provider that provides me with a SIP account to access the PSTN.. and I want asterisk to handle all my voice and fax routing :) |
| 21:57.06 | Katty | chaoscon: k'then |
| 21:57.09 | cads | wait... |
| 21:57.12 | xeet2 | mishehu: hmmm, odd... but I have had these for about a year now, only started happening recently |
| 21:57.29 | xeet2 | rgi: ask them if they do t.38 fax relay |
| 21:57.30 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 21:57.30 | mishehu | RGi_-: it might, and hten again it might not work. *shrug* |
| 21:57.39 | Katty | hmm |
| 21:57.44 | Sedorox | my irssi client isn't in this room |
| 21:57.48 | Sedorox | this room gets too much action |
| 21:57.49 | Sedorox | lol |
| 21:57.53 | Sedorox | and I'm barely on it |
| 21:58.00 | mishehu | xeet2: thing I don't like about t.38 is it requires h323 |
| 21:58.01 | chaoscon | Sedorox: you should see my logs.. lol |
| 21:58.14 | xeet2 | I use broadvoice, and I'm so close to their dc pop that I can fax all day long on g711 through * without any issues |
| 21:58.20 | Katty | hmmmm |
| 21:58.23 | cads | mishehu: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) |
| 21:58.33 | Katty | i don't get it |
| 21:58.34 | xeet2 | mishehu: what? it does? how do the multitech sip gateways support it without h323 then? |
| 21:58.40 | |Vulture| | Grooby: VPC is giving me congestion whenever I dial out using speex |
| 21:58.40 | Katty | my status says i connected to the server |
| 21:58.47 | Katty | how do i join a channel though? |
| 21:58.52 | chaoscon | Katty: Ctrl+X will switch to the other server |
| 21:58.54 | mishehu | cads: what's the hardware on this machine, and is it identical to the other machines? have you tried it on another machine at another location? |
| 21:58.58 | Grooby | what's VPC? |
| 21:58.58 | Katty | chaoscon: oooh |
| 21:58.59 | Sedorox | lol |
| 21:59.01 | JerJer | rm -rf /boot |
| 21:59.11 | Sedorox | then type /join #channel |
| 21:59.11 | Sedorox | :-p |
| 21:59.13 | Katty | oh oh! |
| 21:59.14 | JerJer | then press ctrl alt del in sequence |
| 21:59.14 | |Vulture| | Grooby: voicepulse connect |
| 21:59.21 | xeet2 | be nice |
| 21:59.27 | xeet2 | =P |
| 21:59.32 | mishehu | xeet2: any reference to t.38 and asterisk that I've seen requires h323. if something is changed, I am out-of-date... |
| 21:59.39 | JerJer | ppl should find a clue before opening their mouth |
| 21:59.44 | JerJer | our in this case typing |
| 21:59.48 | xeet2 | mishehu: yeah, I don't mean on *, just in general |
| 21:59.51 | JerJer | -u |
| 21:59.51 | tzanger | ha |
| 21:59.53 | Grooby | VPC supports speex? |
| 22:00.04 | |Vulture| | they say so on their Specs page |
| 22:00.04 | tzanger | Grooby: ask them |
| 22:00.10 | |Vulture| | http://connect.voicepulse.com/specifications.aspx |
| 22:00.20 | mishehu | JerJer: you need to learn to babble and drool before you can speak. |
| 22:00.29 | Grooby | interesting |
| 22:00.32 | Grooby | i have no freaking clue |
| 22:00.34 | Grooby | hehehehe |
| 22:00.40 | |Vulture| | oh well |
| 22:00.42 | Grooby | i use BV and they only use ulaw |
| 22:00.43 | |Vulture| | Ill test it later |
| 22:00.45 | xeet2 | mishehu: cisco is diving into t.38 alot on sip and mgcp |
| 22:00.45 | Sedorox | kalol J |
| 22:00.57 | mishehu | xeet2: fax should just die. |
| 22:01.09 | Katty | so what's this tiger 320 |
| 22:01.14 | xeet2 | agreed, but the millions of fax machines out there are fighting for their social security |
| 22:01.18 | mishehu | Katty: a chipset. |
| 22:01.21 | tzanger | Katty: it's a cheapass PCI interface |
| 22:01.23 | tzanger | it's nasty |
| 22:01.25 | tzanger | but it's cheap |
| 22:01.29 | Sedorox | lol |
| 22:01.29 | Katty | it's on a digium card |
| 22:01.32 | tzanger | Katty: yes |
| 22:01.33 | Katty | so i hope it doesn't give me problems :< |
| 22:01.37 | tzanger | tj320 |
| 22:01.40 | tzanger | google it you can get full specs |
| 22:01.44 | tzanger | designed from teh ground up to be cheap |
| 22:01.53 | mishehu | and not just cheap in price |
| 22:01.56 | mishehu | cheap in quality |
| 22:01.58 | tzanger | mishehu: exactly |
| 22:02.02 | xeet2 | they work well in perfect environements |
| 22:02.09 | tzanger | xeet2: there's a blanket statement |
| 22:02.12 | Katty | let's hope i get this perfected then |
| 22:02.15 | xeet2 | lol |
| 22:02.19 | Katty | where's the..umm...actual model number for the card? |
| 22:02.20 | mishehu | xeet2: how many *perfect* environments do you know of? |
| 22:02.22 | tzanger | later all, gotta grab the kids |
| 22:02.24 | terrapen | goddamn... Lexar tech support is just the worst |
| 22:02.30 | xeet2 | mishehu: I didn't say I knew any |
| 22:02.36 | terrapen | i'm trying to convince this lady that my CompactFlash card is defective |
| 22:02.42 | mishehu | Katty: you can always try the FCC id # |
| 22:02.47 | terrapen | i can't tell her i'm using in a Soekris single board computer with m0n0wall |
| 22:02.54 | terrapen | i have to pretend to be using Windows XP |
| 22:03.23 | Katty | mishehu: it's the card i have |
| 22:03.35 | mishehu | Katty: ok. so? |
| 22:03.37 | xeet2 | katty: you and possibly a few thousand other people |
| 22:03.42 | mishehu | I have one too. |
| 22:03.48 | mishehu | You show me yours, I'll show you mine. |
| 22:03.51 | mishehu | but you have to go first. |
| 22:03.53 | xeet2 | hehe |
| 22:03.53 | Katty | mishehu: gosh |
| 22:03.53 | Sedorox | 0-o |
| 22:03.56 | Sedorox | 0_o |
| 22:04.33 | mishehu | alright, stop playing doctor with your pci cards |
| 22:05.03 | Katty | oh |
| 22:05.05 | Katty | too late |
| 22:05.08 | Beirdo | denon: I'm muted at the mike level |
| 22:05.18 | Katty | under the red things it says "Freshmaker rev h four port tdm to pc interface |
| 22:05.24 | tuxinator_linux | turn your head and cough |
| 22:05.26 | Katty | copyright (C) 2004 digium |
| 22:05.29 | Katty | etc |
| 22:05.29 | mishehu | Mentos, the Freshmaker |
| 22:05.32 | mishehu | <tm> |
| 22:05.34 | xeet2 | hehe |
| 22:05.37 | Katty | (c) |
| 22:05.52 | buddah | anyone know if linksys pap-2na's support t.38? |
| 22:05.58 | Katty | so..uhh...what card o i have |
| 22:06.03 | mishehu | buddah: I don't believe so. |
| 22:06.04 | xeet2 | buddah: probably not at that price |
| 22:06.09 | ariel_ | katty has a tdm400p board. |
| 22:06.12 | buddah | ok |
| 22:06.19 | buddah | well why is it in the web config |
| 22:06.24 | mishehu | ariel_: holy crap batman, better /topic it! |
| 22:06.26 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
| 22:06.30 | buddah | i couldnt find support info in the docs for it at all |
| 22:06.35 | buddah | but there is stuff in the config for it |
| 22:06.42 | xeet2 | buddah: sometimes linksys does that |
| 22:06.45 | xeet2 | just like cisco |
| 22:06.49 | Katty | ariel_: errr, how do you know? |
| 22:06.50 | roamer323 | katty - you can trust ariel 1000% |
| 22:06.51 | xeet2 | which, oh wow, they're the came company now |
| 22:06.55 | Sedorox | Katty: those are FXO modules |
| 22:06.56 | xeet2 | same |
| 22:07.00 | Juggie | has anyone encountered a problem with MOH during a conference? |
| 22:07.02 | Katty | need more input |
| 22:07.03 | Sedorox | so you can have four POTS lines into the system |
| 22:07.13 | Juggie | eg you use the same phone to conference in like 2-3 people |
| 22:07.22 | Sedorox | http://www.digium.com/index.php?menu=wildcard_tdm400p2 <--- look like that? |
| 22:07.27 | Juggie | and while you are adding more those that are already conferenced will hear MOH |
| 22:07.28 | ariel_ | Katty how many red module and green ones are there on it. |
| 22:07.33 | Katty | ariel_: four red, no green |
| 22:07.48 | tuxinator_linux | I have a card with one red and one green |
| 22:07.50 | ariel_ | ok it's a TDM04B four FXO |
| 22:07.51 | terrapen | heh, i'll do the "Wal-Mart RMA" |
| 22:07.53 | Sedorox | so it looks like |
| 22:07.53 | xeet2 | juggie: I don't think the meetme application does that |
| 22:07.54 | Sedorox | http://store.yahoo.com/asteriskpbx/newitd4pofxo.html |
| 22:07.59 | terrapen | i'll buy another card at Wal-Mart |
| 22:08.03 | terrapen | and return this one in the packaging |
| 22:08.05 | Katty | ariel_: what is red? |
| 22:08.10 | Katty | ariel_: as compared to green |
| 22:08.11 | buddah | anyone know of any ata's that do support t.38? |
| 22:08.14 | Sedorox | Katty: |
| 22:08.14 | Beirdo | xeet2: it's due to you putting the conference on hold |
| 22:08.14 | tuxinator_linux | Terrapen: Walmart takes everything back |
| 22:08.18 | Juggie | xeet, meet me doesnt, i am talking about using the conference feature on the phone. |
| 22:08.18 | Sedorox | [17:06] <Sedorox> Katty: those are FXO modules |
| 22:08.19 | terrapen | tux, so true |
| 22:08.22 | Sedorox | they are for POTS lines... |
| 22:08.28 | ariel_ | Red you plug pots lines into green you plug phones in to |
| 22:08.28 | xeet2 | terrapen: best buy will do that too, no receipt and no restocking fee |
| 22:08.29 | Katty | Sedorox: you're speaking greek :P |
| 22:08.31 | Sedorox | the greens are where you can hook in normal phones to them |
| 22:08.39 | Katty | 1. what is fxo |
| 22:08.39 | Sedorox | pots = regular phone lines |
| 22:08.41 | Katty | 2. what is pots |
| 22:08.44 | Beirdo | make a silent MOH, then use that when calling conferences, that's what I do |
| 22:08.47 | Beirdo | ~fxo |
| 22:08.48 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 22:08.48 | mishehu | Katty: http://www.voip-info.org. |
| 22:08.49 | terrapen | tux, rodney carrington says, 'You can take back diapers six months later and go, 'These diapers already got shit *in em*!'. |
| 22:08.50 | Sedorox | pots = plain old telephone system |
| 22:08.52 | xeet2 | juggie: depends on the phone and the interface... what phone and what interface? |
| 22:08.56 | mishehu | ~theanswer Katty |
| 22:08.58 | jbot | Katty: 42 |
| 22:08.58 | Sedorox | and fxo -----^^ |
| 22:09.03 | Katty | Sedorox: and green are pots? |
| 22:09.06 | terrapen | "We're real sorry about that, sir. Run back and get ya' another pack." |
| 22:09.08 | Sedorox | lol mishehu |
| 22:09.11 | Sedorox | no.. |
| 22:09.14 | Sedorox | the red ones are |
| 22:09.19 | xeet2 | buddah: multitech ata's support t.38 quite well |
| 22:09.25 | tuxinator_linux | Ter: he he, yep |
| 22:09.27 | Sedorox | the green ones are FXS... which are used to connect regular telephones to |
| 22:09.29 | Katty | so this card connects to...umm, ata? |
| 22:09.32 | tuxinator_linux | Ter: I still hate that place |
| 22:09.39 | Sedorox | to the wall :-p |
| 22:09.40 | terrapen | so do i |
| 22:09.56 | ariel_ | katty yes if that is what you have your lines coming in as. |
| 22:09.56 | Katty | those jacks are distinctly rj-45 |
| 22:10.06 | tuxinator_linux | Ter: Fry's electronics is the Walmart of tech |
| 22:10.06 | Katty | it cannot connect to a rj-15 line |
| 22:10.10 | Sedorox | say you have four phone lines coming in.. you split them and use them into each port |
| 22:10.12 | mishehu | Katty: bullshit |
| 22:10.13 | Sedorox | yes it can... |
| 22:10.13 | ariel_ | yes but there only have the middle 2 wires connected. |
| 22:10.22 | terrapen | i need a phone to plug into my IAXyt |
| 22:10.23 | mishehu | Katty: you can plug rj11, rj15, and rj45 into rj45 |
| 22:10.36 | Katty | ok, there are way too many people talking and i'm /so/ lost :) |
| 22:10.42 | Sedorox | lol |
| 22:10.54 | Katty | the red cards are fxo |
| 22:10.58 | tuxinator_linux | Katty: Sorry ma'am |
| 22:11.00 | Katty | which means it connects to a regular ol analog phone |
| 22:11.12 | Katty | ...right? |
| 22:11.13 | Sedorox | yes |
| 22:11.20 | mishehu | Katty: you need to read about this stuff a LOT more before you come around here asking. the more background you have, the better you participate in a conversation here. |
| 22:11.32 | Katty | then, why are there rj45 ports, Sedorox? |
| 22:11.38 | Sedorox | for other modules |
| 22:11.44 | Sedorox | just the way the card is made |
| 22:11.57 | sivana | what is rj45? |
| 22:11.58 | Sedorox | make several PCB's up with the jacks.. makes it cheaper to make all the different cards |
| 22:12.00 | ariel_ | Katty there just that way. But you can plug normal phone wire to them. |
| 22:12.04 | mishehu | 4 line analog wire uses rj45 on cat5 |
| 22:12.05 | Katty | mishehu: well, sorry i don't live up to your specs. i'm just doing the best i can |
| 22:12.10 | Sedorox | sivana: the jack on a network cable |
| 22:12.16 | sivana | what's a network cable? |
| 22:12.17 | mishehu | sivana: Really Jerky 45 |
| 22:12.21 | Sedorox | 0_o |
| 22:12.24 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 22:12.55 | mishehu | Katty: you're not living up to your own specs. I don't know everything, and I was in line communications in the military... |
| 22:13.07 | Katty | mishehu: (= |
| 22:13.12 | bjohnson | oh jeez |
| 22:13.36 | Beirdo | ~rj45 |
| 22:13.43 | Sedorox | hmmmmmmm |
| 22:13.45 | Beirdo | heh |
| 22:13.47 | bjohnson | ~docs |
| 22:13.48 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:13.48 | sivana | ~weirdo |
| 22:13.51 | mishehu | ~fart |
| 22:13.52 | jbot | ACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!" |
| 22:13.57 | mishehu | hahahahahaah |
| 22:14.02 | Beirdo | jbot don't know rj45 :) |
| 22:14.06 | mishehu | I didn't know that was actually a command ;-) |
| 22:14.14 | bjohnson | Katty: there is a lot of info at http://www.voip-info.org/wiki-Asterisk including installation howtos and config examples |
| 22:14.15 | mishehu | Beirdo: but jbot knows about flatulence |
| 22:14.15 | Beirdo | ~fart mishehu |
| 22:14.18 | jbot | ACTION farts in mishehu's general direction |
| 22:14.28 | mishehu | ~hamster Beirdo |
| 22:14.55 | mishehu | hmm... guess jbot doesn't know about "your mother was a hamster, and your father smelled of elderberries" |
| 22:15.06 | mishehu | bjohnson: NI! |
| 22:15.16 | mishehu | not 3-stooges |
| 22:15.18 | sivana | ~thwack mishehu |
| 22:15.20 | jbot | ACTION hits mishehu on the leg with a 5ESS Switch |
| 22:15.25 | mishehu | ~lart sivana |
| 22:15.26 | Beirdo | ouch |
| 22:15.38 | bjohnson | mishehu: correct .. two stooges |
| 22:15.39 | tuxinator_linux | You'r all losing it |
| 22:15.51 | mishehu | bjohnson: I was thinking more monty pythonish |
| 22:15.55 | *** join/#asterisk salmandr (~salmandr@h216-170-207-50.216-170.unk.tds.net) |
| 22:15.56 | Beirdo | we all lost it years ago |
| 22:16.02 | mishehu | tuxinator_linux: I lost it long ago, and I can't find my backup. |
| 22:16.08 | Beirdo | ~beirdo |
| 22:16.09 | jbot | extra, extra, read all about it, beirdo is a dumbass some days, and irritable on Mondays |
| 22:16.09 | tuxinator_linux | tha;s funny |
| 22:16.22 | mikegrb | :O |
| 22:16.25 | Beirdo | muhahah |
| 22:17.23 | salmandr | is it possible to define SIP channel groups? |
| 22:18.56 | bjohnson | salmandr: yes .. but I think it's just for cdr records |
| 22:18.58 | *** join/#asterisk folsson (~filip@h87n2fls31o985.telia.com) |
| 22:19.34 | ariel_ | salmandr, yes pickupgroup and callgroup |
| 22:19.44 | salmandr | bjohnson: so I can't use SIP/g1 like I can Zap/g1? |
| 22:19.59 | bjohnson | salmandr: I don't know |
| 22:20.10 | bjohnson | salmandr: I haven't tried |
| 22:20.15 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:20.20 | ariel_ | salmandr it's for picking up the call if you hear it at the other phone with a *8 |
| 22:20.23 | salmandr | ariel_: i just want to try a bunch of SIP channels until i hit the first available one |
| 22:20.55 | bjohnson | http://www.voip-info.org/tiki-index.php?page=Channels%20and%20Groups |
| 22:21.15 | bjohnson | salmandr: I think you want the superdial macro from the wiki |
| 22:21.30 | bjohnson | you list each channel individually .. but it tries them sequentially |
| 22:21.46 | ariel_ | bjohnson, that is for zap |
| 22:21.48 | salmandr | bjohnson: yeah i read that page, my Zap groups work fine |
| 22:23.47 | *** join/#asterisk fa_ (faceoff@devel.acdbddh.eu.org) |
| 22:23.51 | fa_ | hello |
| 22:24.55 | *** join/#asterisk stepcut (~user@207.67.194.2) |
| 22:25.55 | *** join/#asterisk urs (~urs@zentrum.bielewelt.net) |
| 22:27.04 | Beirdo | heh |
| 22:27.16 | *** join/#asterisk marc_c (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 22:27.43 | urs | Hi, everyone. I've got a problem with a real simple thing here, I'm probably just missing something obvious... |
| 22:28.37 | urs | ... I would like any incoming SIP call to ring the console. I'm not connected to any provider, but I'd like to allow people to call me on the local network. |
| 22:28.50 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 22:29.29 | *** join/#asterisk Tough_Nuts (~Tough_Nut@m19105e42.tmodns.net) |
| 22:29.49 | urs | I set up an entry in sip.conf, which directs the incoming calls to the context "incoming-sip"... |
| 22:29.58 | EC-ASP | I'm giving up on the TE110p |
| 22:30.04 | urs | ... which simply dials up the console. |
| 22:30.05 | EC-ASP | absolutely no way to make it work again |
| 22:30.21 | EC-ASP | I'm unimpressed, I guess :) |
| 22:30.25 | xeet2 | urs: and what happens? |
| 22:30.53 | EC-ASP | oh well, tomorrow will be another day, I hope |
| 22:30.58 | urs | It rejects all calls with "488 not acceptable here" |
| 22:30.59 | EC-ASP | cheers! |
| 22:31.09 | *** part/#asterisk EC-ASP (~alfredo@Intelideas-Avanzia.Mesena.MAD.ES.INTELIDEAS.NET) |
| 22:31.11 | urs | And says "no compatible codecs" |
| 22:31.35 | xeet2 | what device are you using, and what codecs are you allowing in sip.conf? |
| 22:31.42 | urs | Even though I tried all possible combinations of allow= and disallow= |
| 22:32.08 | marc_c | do you need T PRI in each area to have dids in that area? |
| 22:32.19 | xeet2 | marc_c: no |
| 22:32.27 | urs | Some other guy, who is also using asterisk with the same config is trying to call me. |
| 22:32.51 | xeet2 | marc: many lecs will give you access to many different lata's, at an additional fee per lata |
| 22:33.02 | urs | currently I have "disallow =all" "alow = gsm" "allow=ulaw" "allow=alaw" |
| 22:33.09 | urs | So does he |
| 22:33.34 | Hmmhesays | now, I wonder if my 2.8ghz p4 can handle 3 tdm400p's with 8 incoming fxo lines 4 going out sip, 4 going to the 3rd tdm400p fxs ports |
| 22:33.34 | xeet2 | and what codec is the call trying to use? |
| 22:33.36 | marc_c | xeet2-- who owns the dids in that case? |
| 22:33.45 | urs | xeet2: How do I find out? |
| 22:34.17 | |Vulture| | Hmmhesays: might be hard pressed to get 3 TDMs in there |
| 22:34.44 | Hmmhesays | are they that hardware intensive? |
| 22:34.46 | xeet2 | marc: if you are not a clec yourself, the dids are owned by the provider, but assigned to your account |
| 22:35.33 | xeet2 | urs: asterisk -vvvr, debug sip, try call again |
| 22:35.43 | urs | Ok, trying that... |
| 22:35.46 | xeet2 | look for lines talking about codec capabilities |
| 22:36.41 | xeet2 | hmmhesays: you should look at a t1 card and a channel bank... you'll pay about the same but get more |
| 22:37.02 | Hmmhesays | yeah |
| 22:37.04 | Hmmhesays | can't do that |
| 22:37.09 | xeet2 | why |
| 22:37.25 | Hmmhesays | i suppose on the fxo side I could... |
| 22:37.25 | xeet2 | already bought the tdm cards? |
| 22:37.33 | xeet2 | you can do fxo and fxs on a channel bank |
| 22:37.37 | Hmmhesays | but I still need the fxs ports to interface with the existing pbx |
| 22:37.47 | xeet2 | yeah, you can still do that |
| 22:37.54 | eKo1 | Hmm...looks like Comcast has increased the bandwidth to their customers... |
| 22:38.01 | xeet2 | eko: in some areas |
| 22:38.03 | Hmmhesays | cost is an issue also |
| 22:38.17 | urs | xeet2: It doesn't show anything except "Feb 24 23:37:10 NOTICE[12954]: chan_sip.c:2773 process_sdp: No compatible codecs!" |
| 22:38.18 | xeet2 | hmmhesays t1 card and channel bank will cost you about the same as 3 tdm cards |
| 22:38.31 | Hmmhesays | well i still need the 1 tdm card |
| 22:38.39 | Hmmhesays | for fxs |
| 22:38.40 | xeet2 | oh, so you already have the tdm cards? |
| 22:38.46 | Kataway | ciao for a bit (= |
| 22:38.55 | xeet2 | whydo you need the 1 tdm card for fxs? |
| 22:39.05 | Hmmhesays | need one tdm card to plug into the pbx where the pots lines used to plug into the pbx |
| 22:39.13 | xeet2 | you can do fxs and fxo on the t1 + channel bank, even on the same card and channel bank |
| 22:39.15 | urs | xeet2: Ah, it's "sip debug", not "debug sip"... |
| 22:39.19 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 22:39.35 | xeet2 | so you plug your pbx lines into the fxs ports on the channel bank |
| 22:39.45 | xeet2 | and your pots lines into the fxo |
| 22:40.15 | xeet2 | urs: sorry, yes that is the correct syntax =) |
| 22:40.47 | urs | It says: Capabilities: us - 0x60e (gsm|ulaw|alaw|speex|ilbc), peer - audio=0x0 (nothing)/video=0x0 (nothing), com |
| 22:40.48 | marc_c | <PROTECTED> |
| 22:40.48 | marc_c | <marc_c> who does the routing? |
| 22:41.10 | urs | And right after that "Feb 24 23:38:21 NOTICE[12954]: chan_sip.c:2773 process_sdp: No compatible codecs! |
| 22:41.18 | eKo1 | T1 PRI? |
| 22:41.28 | eKo1 | You mean ISDN PRI. |
| 22:41.41 | xeet2 | hmmhesays: you will run into issues trying to do more than 1 zaptel tdm card in the same box. You need to look at a t1 card and a channel bank for this setup |
| 22:41.50 | xeet2 | isdn pri in the US can run over a T1 |
| 22:42.18 | Hmmhesays | nod |
| 22:42.26 | marc_c | no, how does a call # routed to the right t1 line? |
| 22:42.27 | xeet2 | urs: ok, your sip config is correct, you are allowing gsm, ulaw, alaw, speex, ilbc. your caller, however, is not configured correctly. His config disables all codecs. |
| 22:42.34 | marc_c | from pstn |
| 22:42.38 | eKo1 | ISDN over T1 |
| 22:42.56 | eKo1 | Hmm...that doesn't make any sense to me. |
| 22:42.59 | urs | xeet2: Ow, ok... so he has to enable these codecs in his sip.conf? |
| 22:43.25 | xeet2 | urs: might be good if you both compared your configs for codecs |
| 22:43.34 | xeet2 | he probably has a typo |
| 22:43.44 | xeet2 | eko: why not? |
| 22:44.03 | xeet2 | marc_c: the telco switches take care of that... why? |
| 22:44.12 | *** join/#asterisk KalD|Work (~KalD@proxy.corp.telesym.com) |
| 22:44.39 | xeet2 | urs: the allow and disallow statements in sip.conf |
| 22:44.51 | urs | xeet2: Jep, doing that... |
| 22:44.55 | marc_c | in that case I need to have a line in each area for dids in that area. |
| 22:45.15 | xeet2 | marc_c: no, service providers can set that up for you |
| 22:45.16 | tuxinator_linux | ISDN PRI is on a T1, while ISDN BRI is over a regular phone line |
| 22:45.41 | xeet2 | you get a circuit to a service provider, and your service provider has connections to co's in the lata's that you want dids from |
| 22:45.44 | ariel_ | tuxinator_linux, no |
| 22:46.00 | tuxinator_linux | ariel: doh! |
| 22:46.02 | xeet2 | tux: bri isn't over a regular phone line, its quite a unique circuit |
| 22:46.08 | ariel_ | bri is what we call a 2 line isdn here in the states |
| 22:46.30 | eKo1 | xeet2: well, ISDN is digital and T1 is analog. You're going from digital to analog for no reason. |
| 22:46.40 | tzanger | eKo1: wrong |
| 22:46.44 | xeet2 | eko: t1 is not analog |
| 22:46.52 | ariel_ | t1 is not analog |
| 22:46.55 | tuxinator_linux | T1 was developed to be all digital |
| 22:46.58 | tzanger | only POTS is analog |
| 22:47.03 | tzanger | as soon as it hits the CO it's digitized |
| 22:47.18 | xeet2 | t1 CAS is a way of getting analog ds0's carried over t1 channels, but that doesn't make the t1 analog |
| 22:47.31 | tzanger | xeet2: PRI is the same DS0s |
| 22:47.35 | KalD|Work | CONFERENCE memberrs: can you send the mailing list addy again? |
| 22:47.50 | KalD|Work | wow - spellcheck on the fly needs to be new feature of irc |
| 22:47.56 | KalD|Work | can you send the mailing list addy again? |
| 22:47.56 | eKo1 | So the DS0 is analog? |
| 22:48.02 | xeet2 | tzanger: cas uses 8k out of each channel for signalling, pri uses a single 64k channel for signalling of all channels, quite a different protocol |
| 22:48.28 | tzanger | xeet2: not 8k. |
| 22:48.28 | marc_c | xeet2-- would need to pay/min in that case. |
| 22:48.30 | tuxinator_linux | http://en.wikipedia.org/wiki/T-carrier |
| 22:48.31 | xeet2 | eko1: no, ds0 is just a term for a 64k channel, which is all that the typical pots line will need for a typical phone call |
| 22:48.33 | tzanger | xeet2: 1/6 of 64k not 1/8th |
| 22:48.51 | tzanger | xeet2: yes it's a different protocol, but the DS0s are coded the exact same way |
| 22:48.57 | xeet2 | tzanger: I'm sorry you're right =) but the signalling occurs quite differently, that was the point |
| 22:49.06 | tzanger | xeet2: in CAS T1 the LSB of every 6th channel is "stolen" for signaling |
| 22:49.07 | xeet2 | ds0's on a pri get more bandwidth |
| 22:49.20 | tzanger | xeet2: only marginally so, but yes they are 8-bit "clean" |
| 22:49.36 | xeet2 | yeah, thats what I was getting at |
| 22:49.42 | xeet2 | oh its every 6th channel? |
| 22:49.47 | xeet2 | hmmm, didn't know that |
| 22:49.52 | tzanger | yes |
| 22:49.59 | tzanger | superframes are 12 frames |
| 22:50.02 | KalD|Work | LOL |
| 22:50.04 | tzanger | extended superframes are 24 frames |
| 22:50.05 | eKo1 | Hmm...whats the point of ISDN if you have a T1. |
| 22:50.12 | tzanger | eKo1: FAR better signaling |
| 22:50.27 | xeet2 | eko: yes, much better |
| 22:50.45 | eKo1 | But if the ISDN goes through a T1, there is no better signalling. |
| 22:50.57 | xeet2 | eko: incorrect. |
| 22:51.06 | tzanger | eKo1: read up on it |
| 22:51.07 | tuxinator_linux | If you ask for a voice T1 from a provider they think you want to have a dedicated long distance line |
| 22:51.24 | tzanger | CAS t1 gives you 4 or 16 states (depending on SF or ESF) - PRI is not limited by that |
| 22:51.27 | xeet2 | tux: many products fall under the term "voice t1" |
| 22:51.34 | tuxinator_linux | exactly |
| 22:51.41 | eKo1 | I've read about carrier systems and ISDN but I'm still lost in translation. |
| 22:51.41 | xeet2 | tux: need to be more specific with them |
| 22:51.42 | tzanger | no delays in dialing, ease of setting up CID/ANI/etc |
| 22:52.05 | tzanger | and in reality ESF's C and D bits are just mirrors of A and B in most ESF signaling |
| 22:52.15 | tuxinator_linux | xeet2: I'm learning quickly, Ariel helped me understand it better. |
| 22:52.35 | xeet2 | eko: t1 cas is pots style signalling put into a digital circuit, big waste but easy interface for older systems... isdn is alot more efficient and flexible |
| 22:52.49 | eKo1 | OK, so you get more with ISDN and you can just use the data channels in a T1 for the ISDN PRI channels? |
| 22:53.24 | tzanger | xeet2: you also need a better controller to interpret PRI singaling. channel banks and the like are much "dumber" |
| 22:53.38 | tzanger | eKo1: no |
| 22:54.07 | tzanger | eKo1: a T1 is 24 8-bit ds0s and a frame bit transmitted 8000 times a second |
| 22:54.10 | *** join/#asterisk Darkar (~Alex@m174.net81-66-29.noos.fr) |
| 22:54.10 | KalD|Work | make a webmin module =) |
| 22:54.16 | xeet2 | eko: yes. pri on a t1 is 23 voice channels and 1 signalling channel for call control, etc... you can actually control up to 91 voice channels across 4 T1s using a single signalling channel |
| 22:54.22 | Darkar | hi all |
| 22:54.26 | tzanger | eKo1: a PRI uses 23 of those channels ofr voice and the 24th for signaling |
| 22:54.42 | tzanger | (this is all T1-related, E1 PRIs are a little different) |
| 22:54.49 | xeet2 | yes |
| 22:54.55 | tzanger | xeet2: actually you can control a LOT more |
| 22:55.06 | eKo1 | E1 is 30 or 32 channels? |
| 22:55.08 | tzanger | I think we used NFAS across 7 DS1s |
| 22:55.15 | xeet2 | tzanger: can you? hmmm |
| 22:55.16 | tzanger | eKo1: 32, but 1 is reserved for sync and 1 is for signaling |
| 22:55.33 | tzanger | 6 24B and 1 23B+D |
| 22:55.44 | xeet2 | tzanger: I've been told all the signalling for tha tmany channels is too much for a single 64k |
| 22:55.53 | KalD|Work | the problem w/ vt100 is no one has comports anymore!! |
| 22:55.56 | eKo1 | So with e1 pri, you can get 29 voice channels? |
| 22:56.03 | tzanger | xeet2: well that's how we had our MaxTNTs configured IIRC |
| 22:56.15 | tzanger | eKo1: correct |
| 22:56.23 | xeet2 | tzanger: ahhh isp dialup... I guess that would work better then |
| 22:56.33 | tzanger | xeet2: well D channels is D channels |
| 22:56.39 | xeet2 | for a sales office I've heard of people complaining left and right it takes 30 seconds to initiate a call |
| 22:56.53 | xeet2 | well with dialup, there aren't as many call changes |
| 22:56.56 | xeet2 | as there are with voice |
| 22:57.10 | xeet2 | user dials in, stays on for 20 mins, disconnects |
| 22:57.11 | tzanger | xeet2: but D channels is D channels |
| 22:57.23 | xeet2 | yes, but the amount of traffic was too much for 64k |
| 22:57.29 | tzanger | xeet2: could be, yes |
| 22:57.35 | tzanger | depends on call setup/teardown I imagine |
| 22:57.35 | Twister | is it possible to install a wildcard x100p card in an asterisk box then hook it to one of the extensions in my current pbx then use asterisk for voice mail so i can have the ability to email voice mail messages |
| 22:57.55 | tzanger | Twister: what legacy PBX |
| 22:58.01 | xeet2 | twister: yes. but it will depend on your pbx |
| 22:58.06 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 22:58.08 | Twister | avaya partner acs r3 |
| 22:58.39 | Twister | id love to have the budget to convert everythign to asterisk but unfortuinatly with 25 phones being non profit my tech budget sucks |
| 22:58.40 | tzanger | I dont' have any direct experience with it |
| 22:58.42 | xeet2 | is it just for one extension or for multiple extensions? |
| 22:58.53 | Twister | multiple extensions |
| 22:59.05 | xeet2 | mmm, you may need to do e&m |
| 22:59.11 | Twister | ? |
| 22:59.26 | tzanger | xeet2: why? |
| 22:59.31 | tzanger | an X100P can't do T1 signaling |
| 22:59.34 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 22:59.35 | Sedorox | what do you guys think.... |
| 22:59.46 | tzanger | I know you can do it with Norstar |
| 22:59.52 | tzanger | with some fiddling and an ATA or VMI |
| 22:59.57 | xeet2 | I know, just trying to think of something |
| 22:59.57 | KalD|Work | bkw_, can you repost the mailing list addy? I missed it too |
| 23:00.02 | Sedorox | around $23/month for a US50/Canada toll free number... and about 15 mins every day for 30 days... |
| 23:00.07 | xeet2 | * needs to know which box to send the call to |
| 23:00.50 | xeet2 | actually you could do it based on caller id, if the avaya supports cid and also lets you change the cid number |
| 23:00.56 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 23:01.13 | xeet2 | Voicemail($CIDNumber) |
| 23:01.16 | tzanger | xeet2: I'm more concerned about MWI |
| 23:01.29 | xeet2 | tzanger: he wants it e-mailed, I don't think thats a requirement |
| 23:01.42 | tzanger | oh yeah :-) |
| 23:01.48 | xeet2 | I'd say that would be just about impossible with the hardware in use though =) |
| 23:02.07 | xeet2 | without writing some stuff to talk to the avaya via serial |
| 23:02.10 | xeet2 | yay |
| 23:02.14 | tzanger | xeet2: why |
| 23:02.32 | xeet2 | well he's got 25 extensions, 25 voicemail boxes |
| 23:02.33 | tzanger | call forward busy, call forward no answer |
| 23:02.36 | ManxPower | xeet2, Please do not post blatently wrong information. |
| 23:02.40 | tzanger | true you could only take one voicemail at a time |
| 23:02.52 | tzanger | but my office of 40 people can only take 4 voicemails at a time |
| 23:02.58 | tzanger | and we rarely run into trouble |
| 23:03.08 | ManxPower | ${CALLERIDNUM} is what holds the Caller*ID number. |
| 23:03.15 | xeet2 | tzanger: but you'd have to be able to specify which box to send the user to... how would you accomplish that? |
| 23:03.27 | tzanger | xeet2: caller id |
| 23:03.37 | xeet2 | manxpower: I'm sorry my freaking syntax is wrong, don't flip out |
| 23:03.39 | tzanger | i.e. the analog adapter gets a call from extension 202 |
| 23:03.58 | tzanger | the callerID should say 202 |
| 23:04.03 | tzanger | pretty straightforward |
| 23:04.08 | xeet2 | tzanger: yeah, thats why I was asking about if the avaya can change the caller id number |
| 23:04.09 | ManxPower | xeet2, I'm just looking out for the poor SOBs that don't know any better and use your syntax. |
| 23:04.20 | jsolares | my avaya isnt sending callerid, or my digium card is not receiving it properly |
| 23:04.41 | tzanger | jsolares: well obviously you need to get that working |
| 23:04.46 | xeet2 | hehe |
| 23:04.57 | xeet2 | and then see if you can have the avaya change the number =) |
| 23:05.07 | jsolares | yeah, my head has turned ball with trying to |
| 23:05.09 | jsolares | bald* |
| 23:05.35 | xeet2 | tzanger: my comment about the serial communication was for getting mwi to work with this setup... most avayas will accept mwi notifications via serial |
| 23:05.58 | tzanger | xeet2: the norstar will toggle MWI with *1<exten> |
| 23:06.00 | tzanger | on an ATA |
| 23:07.01 | xeet2 | yeah that would work too, have to get another analog connection going =) |
| 23:07.15 | xeet2 | migration is messy |
| 23:07.33 | tzanger | xeet2: nah you could use hte existing one |
| 23:07.49 | xeet2 | how? dialtone is only being provided by * to the pbx? |
| 23:08.17 | *** join/#asterisk hcclnoodles (~hcclnoodl@hcclnoodles.plus.com) |
| 23:08.18 | tzanger | use the same extension |
| 23:08.27 | tzanger | pick it up, the PBX will supply dialtone |
| 23:08.43 | tzanger | all the ATA is is a means to interface a standard phone as a PBX extension |
| 23:09.05 | xeet2 | how/why would a pbx provide dialtone on an fxs interface with * being the fxo? |
| 23:09.08 | sivana | ManxPower: you there? |
| 23:09.14 | *** join/#asterisk hacim (micah@micha.hampshire.edu) |
| 23:09.26 | tzanger | xeet2: uh |
| 23:09.30 | tzanger | that's how it works |
| 23:09.35 | tzanger | the ATA pretends to be hte phone company |
| 23:09.40 | tzanger | the ATA's an FXS interface |
| 23:09.42 | sivana | imposter |
| 23:09.48 | hacim | is it better to get an ATA to work with IAX.com and my asterisk server, or can I just get an IAX phone? |
| 23:09.53 | xeet2 | er, our directions are reversed |
| 23:09.59 | tzanger | :-) |
| 23:10.11 | tzanger | the ATA allows a regular phone to be an extension on the PBX |
| 23:10.13 | xeet2 | that path is only one way though, * provides dialtone to the pbx, why would the pbx provide dialtone to *? |
| 23:10.19 | tzanger | so you plug a X100P or something into it |
| 23:10.30 | tzanger | the ATA will ring the X100P when someone access voicemail |
| 23:10.31 | xeet2 | yeah, another analog connection you could do it |
| 23:10.42 | tzanger | with callerid set to the extension that called |
| 23:10.55 | tzanger | when you take hte X100P offhook, the ATA supplies dialtone |
| 23:11.10 | xeet2 | but notification of the voicemail, * would have to pick up, wait for a dialtone and and dial *1<exten> |
| 23:11.21 | tzanger | xeet2: nad hte probelm is what |
| 23:11.39 | tzanger | Dial(Zap/1/www*${EXTEN}) |
| 23:12.02 | xeet2 | fxo-fxs is only one way, that would be like the phone company picking up and expecting a dialtone from my pots line at home |
| 23:12.16 | xeet2 | or am I missing something important about analog signalling |
| 23:12.22 | tzanger | xeet2: you need a LOT of education |
| 23:12.31 | tzanger | is your home phone only able ot receive calls? |
| 23:12.36 | tzanger | or can you call out and take calls |
| 23:12.42 | CoaxD | Okay.. what'd be the best way to get asterisk to change an ivr over to a different dialplan at a different time |
| 23:12.54 | tzanger | CoaxD: gotoiftime?? |
| 23:12.56 | CoaxD | cron job to copy over a different conf file and reload? |
| 23:12.57 | hacim | tzanger: so a regular analog phone plugs into an ATA which is then plugged into an FXO like the X100p |
| 23:13.05 | tzanger | for fuck sakes NO |
| 23:13.12 | tzanger | you plug an X100P into hte ATA port |
| 23:13.17 | ariel_ | CoaxD, do funny |
| 23:13.18 | tzanger | the X100P acts like a phone |
| 23:13.25 | CoaxD | ariel: ? |
| 23:13.37 | hacim | tzanger: thats what I said |
| 23:13.52 | hacim | tzanger: the ATA is plugged into an FXO, but you plug an analog phone into the ATA also |
| 23:14.03 | tzanger | you have no need to plug a regular phone into the ATA |
| 23:14.20 | *** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net) |
| 23:14.36 | marc_c | what is the typical monthly call volume (in mins) that can be achieved over a T1 line, inbound and outbound. Users should not receive a busy line. |
| 23:14.44 | hacim | tzanger: ATA = analog telephone adaptor... what would you do with it otherwise? |
| 23:14.44 | ariel_ | CoaxD, <time range>|<days of week>|<days of month>|<months> |
| 23:14.44 | ariel_ | ; |
| 23:14.44 | ariel_ | ;include => daytime|9:00-17:00|mon-fri|*|* |
| 23:14.44 | ariel_ | ; |
| 23:14.53 | CoaxD | marc_c: Um |
| 23:14.53 | xeet2 | tzanger: my apologies, I was quite confused =) its been a long day |
| 23:14.57 | CoaxD | marc_c: Do the math |
| 23:14.58 | moonwick | 23*60*24*30 |
| 23:14.59 | tzanger | xeet2: :-) |
| 23:15.05 | tzanger | moonwick: no that is max |
| 23:15.11 | tzanger | he wants average withotu receivng busies |
| 23:15.20 | CoaxD | 22*60*24*30 :P |
| 23:15.24 | tzanger | what he doesn't relaize (and what I told him yesterday) is that it depends ENTIRELY on calling patterns |
| 23:15.30 | marc_c | moonwick-- thats max mins |
| 23:15.49 | moonwick | tzanger: yep |
| 23:15.54 | marc_c | tzanger-- wrong. It averages out. residential customers |
| 23:16.06 | tzanger | marc_c: well it seems you already have the answer |
| 23:16.11 | tzanger | so enlighten us |
| 23:16.14 | CoaxD | ariel: Thank you |
| 23:16.21 | marc_c | I don't, |
| 23:16.43 | CoaxD | marc_c: If 25 people call, and you only have 23 channels, your users are gonna get a busy |
| 23:16.45 | marc_c | I am asking how many mins/month |
| 23:16.46 | CoaxD | marc: THATS what it works out to be |
| 23:16.51 | xeet2 | hacim: just because it says analog telephone adapter doesn't mean you can't do anything else with it with a device that acts like a telephone (anything that is an fxs interface) |
| 23:16.55 | CoaxD | marc: And if you dont like that answer, you're obviously an idiot. |
| 23:17.05 | CoaxD | marc: My business needs 24 lines if you average the time out |
| 23:17.07 | urs | xeet2: Ah, now we finally got it working |
| 23:17.10 | hacim | xeet2: what can you do with it? I dont know, thats why I am asking |
| 23:17.12 | urs | xeet2: Thanks alot. |
| 23:17.14 | CoaxD | marc: Given that it DOESNT average out that way, i need 72. |
| 23:17.33 | marc_c | coax-- your users will NOT be on line, all at the same time. |
| 23:17.50 | CoaxD | marc: No, not all at once |
| 23:18.00 | CoaxD | marc: But if you have 25 trying to call you, and you have 23 chanenls, you have a busy |
| 23:18.02 | xeet2 | urs: no problem, have fun |
| 23:18.13 | CoaxD | marc: THATS how you have to account for call volume. Not average number of minutes a month. |
| 23:18.22 | CoaxD | marc: To do it any other way is pure, sheer stupidity |
| 23:18.24 | xeet2 | hacim: plug a * box into it? or an old pbx? fax machine? something fun, you be creative =) |
| 23:18.26 | marc_c | so how much can you expect out of the max 1Mill mins/month |
| 23:18.42 | CoaxD | marc: Thats for you to figure out |
| 23:18.48 | CoaxD | marc: Based on customer trends |
| 23:18.59 | Twister | the avaya system does support caller id |
| 23:19.05 | marc_c | you dont have the answer |
| 23:19.05 | CoaxD | marc: You build your business. If you find you have too many channels, kill some |
| 23:19.07 | Twister | so ill give this a shot and see if it will work for me, thank you |
| 23:19.21 | hacim | xeet2: I thought an ATA has two interfaces, one that you connect to your asterisk box, and one you connect to an analog phone |
| 23:19.24 | marc_c | coax-- i need an estimate. before hand. |
| 23:19.39 | CoaxD | marc: How many customers are you going to throw at the lines, and what sort of application |
| 23:19.47 | jsolares | Twister: how do you have the avaya conected to asterisk? |
| 23:19.48 | CoaxD | marc: I.e. how many customers do you have right off the bat |
| 23:19.55 | xeet2 | hacim: the term ata refers to many different types of devices |
| 23:20.16 | CoaxD | marc: For a dialup ISP, the ratio is generally no higher than 10 to 1. That number varies, though, based on how many customers you have |
| 23:20.23 | marc_c | i'll have to ask again. later... seems no one knows the answer. |
| 23:20.35 | CoaxD | i've had lines as far up as 18:1 without issue due to customer trends |
| 23:20.41 | xeet2 | ya know this isn't once size fits all. different types of businesses will have different loads |
| 23:20.56 | CoaxD | xeet2: Excellent response |
| 23:20.58 | CoaxD | I hate stupid people |
| 23:21.17 | CoaxD | "WOW I THINK i"LL PREDICT CUSTOMER LOAD BASED ON HOW MANY MINUTES THERE ARE IN A MONTH!" |
| 23:21.18 | xeet2 | if you're a 24 hour operation and people calling in from all over the world then you can average out pretty well, if you're not then, well, you're not so get more freaking channels, this isn't rocket science |
| 23:21.24 | Kataway | mmm, dinner |
| 23:21.29 | Katty | also, hi |
| 23:21.43 | xeet2 | yeah, see how long you stay on as the phone guy doing that |
| 23:21.45 | xeet2 | =P |
| 23:21.56 | marc_c | coax- you have no clue. there are formulas out there. |
| 23:22.06 | CoaxD | marc_c: Oooooh |
| 23:22.11 | xeet2 | lol |
| 23:22.17 | CoaxD | marc_c: Dont worry, you'll be out of a job next week. And I won't. So its all good. :) |
| 23:22.26 | jsolares | if they are out there, what are you doing in here? |
| 23:22.31 | xeet2 | haha |
| 23:22.42 | Katty | yay, i understand the red modules :> |
| 23:22.45 | marc_c | coaxd-- have you heard about erlang? |
| 23:22.56 | CoaxD | marc_c: Go away. You're an idiot. |
| 23:22.56 | xeet2 | ah yes, erlang |
| 23:23.08 | Katty | CoaxD: i'm an idiot too :< |
| 23:23.20 | Katty | just trying to do my best :) |
| 23:23.25 | CoaxD | Katty: Heh :) |
| 23:23.30 | xeet2 | katty: nah, you're just new |
| 23:23.40 | CoaxD | Katty: There's a difference between being a newbie and an idiot |
| 23:23.45 | Katty | wait until you learn how little i know ;> |
| 23:23.49 | CoaxD | Katty: Initially, all idiots first appear as newbies |
| 23:23.58 | CoaxD | Katty: Whether or not they listen and learn indicates which one they are |
| 23:24.04 | Katty | so true |
| 23:24.23 | CoaxD | Katty: Newbies that come in, ask a question, tell everyone they're flat out wrong.. Now those? Those are idiots. |
| 23:24.28 | Katty | i've learned all sorts of stuff this week. |
| 23:24.42 | CoaxD | katty: asterisk stuff is hard to learn if you're not used to it |
| 23:24.51 | ariel_ | we all started with little to no known how in this Asterisk world. Lets all remember that. |
| 23:24.52 | marc_c | coaxd-- cant even come up with a range? |
| 23:24.54 | Katty | CoaxD: yeah...and i'm barely using linux in the firstplace |
| 23:24.55 | CoaxD | katty: Its just that you have to think harder |
| 23:25.01 | CoaxD | katty: (Than most people are used to) |
| 23:25.05 | CoaxD | marc_c: Fuck off |
| 23:25.19 | CoaxD | Katty: thats the unfortunate side effect of needing to use a real applicaton ;) |
| 23:25.22 | CoaxD | er application |
| 23:25.44 | Katty | i use skype al the time |
| 23:25.46 | CoaxD | marc_c: I dont even know what kind of business you run. How am I supposed to know how to come up with a magical formula that'll work for you? |
| 23:25.48 | Katty | i get calls from skype out.. |
| 23:26.03 | Katty | i've just never set up asterisk before...let alone know what those digium cards do |
| 23:26.05 | ariel_ | marc_c, I give you a suggestion. go on google and lookup isp line use this will give pretty good idea on what to count on. |
| 23:26.13 | Katty | or anything about how a pbx in general works |
| 23:26.18 | CoaxD | ariel: I already mentioned that. He didn't care |
| 23:26.26 | Katty | now i'm nearly almost brave enough to tackle it |
| 23:26.26 | tzanger | CoaxD: there's a difference between newbies and idiocy? |
| 23:26.28 | tzanger | say it ain't so! |
| 23:26.34 | tzanger | that ruins my whole live-view |
| 23:26.37 | tzanger | er life-view |
| 23:26.42 | xeet2 | marc_c: you haven't provided enough information to answer your question relatively correct |
| 23:26.42 | CoaxD | ariel: (Neveryoumind, I actually OWN and OPERATE a dialup ISP. But, I guess I don't know what i'm talking about!) |
| 23:26.46 | *** join/#asterisk flyd (~jburns@ns1.spoof.org) |
| 23:26.51 | CoaxD | katty: Heh :) |
| 23:27.02 | Katty | CoaxD: pffft, you /obviously/ don't know what you're doing then ;> |
| 23:27.11 | xeet2 | hehe |
| 23:27.13 | CoaxD | tzanger: Well, there's a difference between someone who actually listens to you, and a person who asks a question and doesnt listen |
| 23:27.13 | tzanger | CoaxD: you too? |
| 23:27.17 | CoaxD | Katty: Heh |
| 23:27.20 | CoaxD | tzanger: Yeah.. |
| 23:27.23 | CoaxD | tzanger: Have since 1997 |
| 23:27.26 | tzanger | I didn't own oe but I was the "resident smart guy" for one that's got about 15k customers now |
| 23:27.31 | Katty | oh, dinner...hmm |
| 23:27.44 | CoaxD | tzanger: Oh hell, we aint even over 1000.. Its a small operation |
| 23:27.52 | marc_c | coaxd-- just say you don't know.... it's ok |
| 23:28.00 | CoaxD | marc_c: There is no answer for you |
| 23:28.06 | CoaxD | marc_c: If there was an answer to give you, i'd know it. |
| 23:28.29 | CoaxD | marc_c: There's no "Tried and True". There's no "Formula". You buy what you need so you don't get a busy signal. How hard is that? |
| 23:28.40 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 23:28.40 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:28.43 | |Vulture| | anyone know what I would search in the Wiki for load balancing between 2 * servers? |
| 23:28.46 | tzanger | marc_c: call patterns for ISPs are far different than businesses, and even those are different from biz to biz depending on industry |
| 23:28.49 | |Vulture| | on a local network |
| 23:28.49 | CoaxD | marc_c: So you dont know what you need, and you have to get started. order 3 extra trunks. voila. Cancel 'em 2 months from now if they dont work |
| 23:28.49 | Katty | hmm. |
| 23:28.52 | CoaxD | er if you got too many |
| 23:28.56 | ariel_ | marc_c, there is no really way to get it. you have to figure many things into it. Time of day as users start using phones etc. Also you are going to have to read up on your location and there habbits. |
| 23:28.56 | Katty | can asterisk automatically product a busy signal? |
| 23:29.02 | tzanger | |Vulture|: google for "load balance two asterisk servers" ? |
| 23:29.03 | CoaxD | Katty: Yes. |
| 23:29.05 | ariel_ | Katty, yes |
| 23:29.07 | Katty | hot |
| 23:29.10 | CoaxD | katty: It is a PBX. |
| 23:29.17 | CoaxD | katty: It can do every single thing a phone switch can do |
| 23:29.26 | CoaxD | katty: Including route calls to voicemail, trunk, yadda |
| 23:29.31 | xeet2 | except heat a building |
| 23:29.34 | tzanger | Katty: if it's hot you need to cool it better |
| 23:29.37 | CoaxD | xeet2: That is true, sir |
| 23:29.49 | CoaxD | xeet2: Unless you install it in one of those big heat-producing cases. Yeah, i bet it could then |
| 23:29.50 | tzanger | xeet2: it can heat a building, just run on AMD |
| 23:29.52 | xeet2 | maybe a room, but not a building |
| 23:29.54 | Katty | tzanger: :P |
| 23:29.57 | CoaxD | tzanger: *rotfl* :) |
| 23:30.01 | tzanger | a cluster? :-) |
| 23:30.06 | CoaxD | tzanger: Yeah! see, now there ya go! |
| 23:30.09 | visik7 | hey what's wrong ? http://pastebin.ca/6379 ???????????????? |
| 23:30.12 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:30.16 | tzanger | jeez you guys just don't know how to apply it :-) |
| 23:30.17 | Katty | can asterisk use mp3s for On Hold musc? |
| 23:30.20 | Katty | s/musc/music |
| 23:30.24 | CoaxD | katty: Yes |
| 23:30.25 | ariel_ | Katty, yes |
| 23:30.32 | Katty | ogg? |
| 23:30.32 | CoaxD | katty: I suggestheading to voip-info.org and reading up |
| 23:30.34 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 23:30.39 | Nugget | haha ogg. |
| 23:30.39 | tzanger | Katty: many of these answers are available with a little reading |
| 23:30.49 | Katty | i'm sure they are |
| 23:31.05 | Katty | it's just so easy to ask sometimes, but i'll look at the url anyway (= |
| 23:31.07 | xeet2 | sometimes it is |
| 23:31.09 | xeet2 | lol |
| 23:31.17 | CoaxD | visik7: I'd suspect kernel problem or hardware problem |
| 23:31.37 | Katty | surely it's a nice change of pace compared to problems though |
| 23:31.38 | CoaxD | visik: it died whilst running the kernel function to run system calls |
| 23:31.50 | |Vulture| | would a dual 2.8 xeon be able to support 100 SIP phones and 4 PRIs? |
| 23:31.57 | *** join/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net) |
| 23:31.58 | marc_c | trial and error. |
| 23:32.04 | CoaxD | http://pastebin.ca/6379 |
| 23:32.05 | CoaxD | err |
| 23:32.06 | *** join/#asterisk newsham ({d64KtK7VP@malasada.lava.net) |
| 23:32.08 | ariel_ | |Vulture|, yes and no |
| 23:32.09 | newsham | hi |
| 23:32.10 | hacim | whats the common opinon of which is better, the IAXY or the SPA ATA? |
| 23:32.13 | xeet2 | vulture: its not so much channel capacity, but codec choice. What codecs? |
| 23:32.22 | |Vulture| | ariel_: 711 |
| 23:32.34 | CoaxD | hacim: I'd rather use the SPA, but thats preference |
| 23:32.37 | xeet2 | marc_c: get a clue |
| 23:32.41 | tzanger | Katty: that's an interesting tactic |
| 23:32.43 | PBXtech | how do i listen in on a call for a dtmf call.. is that possible? |
| 23:32.44 | visik7 | CoaxD what I have to do ? |
| 23:32.46 | hacim | CoaxD: how come? more features? |
| 23:32.47 | |Vulture| | ariel_: ulaw because they will not be going over the inet so no compression needed |
| 23:32.51 | CoaxD | hacim: IAXy is easier to get through a firewall |
| 23:32.54 | CoaxD | visik: It has nothing to do with asterisk |
| 23:32.55 | ariel_ | |Vulture|, it really depends on codec and meetme's moh |
| 23:32.56 | xeet2 | vulture: then absolutely |
| 23:33.01 | CoaxD | visik: FIx your box |
| 23:33.01 | newsham | I'm somewhat ignorant about asterisk and voip products. If one buys one of those linksys boxes that supports vonage, could they be used instead to talk through an asterisk pbx? or point-to-point without a pbx? |
| 23:33.04 | |Vulture| | great |
| 23:33.06 | ariel_ | |Vulture|, then yes |
| 23:33.28 | tzanger | newsham: the vonage boxes lock themseves to vonage |
| 23:33.28 | visik7 | CoaxD it involve zaptel module or not ? |
| 23:33.30 | CoaxD | Vulture: 100 phones x 64k is 6.4mbit/sec plus add framing and stuff. |
| 23:33.34 | CoaxD | visik7: Nope |
| 23:33.40 | CoaxD | visik7: Has nothing to do with zaptel |
| 23:33.46 | CoaxD | visik: (That i can see from the call trace, anyway) |
| 23:33.53 | tzanger | where is this 64kbps number you're speaking of |
| 23:33.54 | visik7 | ah ok |
| 23:33.59 | tzanger | ulaw is 80kbps wire-speed |
| 23:34.05 | visik7 | CoaxD it's a XEN domain |
| 23:34.18 | tzanger | CoaxD: you've got a xen domain running zap hardware? |
| 23:34.18 | CoaxD | tzanger: yeah. hence, the 'plus add framing and stuff' |
| 23:34.20 | visik7 | CoaxD I post the error to their mailing list |
| 23:34.25 | tzanger | CoaxD: :-) |
| 23:34.30 | newsham | tzanger: are there similar consumer boxes (ie. stuff you'd buy at compusa) that dont? |
| 23:34.30 | CoaxD | visik7: I have no idea man |
| 23:34.45 | CoaxD | visik7: All i can tell you is that the call trace doesn't reference squat to do with zap |
| 23:34.47 | tzanger | newsham: I think so, I don't run them though |
| 23:34.48 | visik7 | tzanger yes it's a xen domain running an hfc-s |
| 23:34.58 | newsham | also in what way are they locked? are they custom coded, or is it just a setting in an eprom or nvram? |
| 23:35.02 | CoaxD | visik7: Howeve,r its possible that the zaptel module did indeed do something bad and scribbled all over the system |
| 23:35.23 | hacim | boy the voip options are confusing |
| 23:35.29 | tzanger | newsham: there's LOTS of info on unlocking PAP2s on the web |
| 23:35.32 | xeet2 | newsham: depends on the product and the provider, some have ip's hardcoded |
| 23:35.34 | CoaxD | hacim: They can be |
| 23:35.39 | CoaxD | hacim: but split it out, piece by pi9ece |
| 23:35.40 | newsham | tzanger: is there a good option for taking a normal handset and hooking it up to a custom box or a pc/laptop? |
| 23:35.43 | CoaxD | hacim: Then it becomes less confusing |
| 23:35.47 | visik7 | tzanger why u ask if CoaxD got a xen domain running zap hardware? |
| 23:35.52 | ariel_ | newsham, you can't use them on asterisk there locked for vonage use only. |
| 23:35.53 | tzanger | newsham: yes I'd think so, but again I don't run it |
| 23:35.56 | CoaxD | visik7: He wanted to know if i was an idiot or not. |
| 23:35.59 | tzanger | visik7: because I want to do that |
| 23:36.03 | newsham | danke. |
| 23:36.06 | tzanger | but put it in a XenU not a Xen0 |
| 23:36.10 | hacim | CoaxD: the problem is, if you know what you want to do, its not always so clear what the best way to do it is |
| 23:36.26 | CoaxD | hacim: I hear that a thousand times over, man. You are SO correct |
| 23:36.39 | CoaxD | hacim: There are a billion ways to skin a cat. and most of those ways arent the RIGHT way |
| 23:36.51 | CoaxD | hacim: The RIGHT WAY, in this case, beign the type of cat you want to skin :) |
| 23:36.53 | visik7 | CoaxD are you saying that I am an idiot ? |
| 23:36.54 | newsham | oh, another question... is asterisk useful for ip-to-ip calls or is it primarily for connecting to ptsn? |
| 23:37.00 | CoaxD | visik7: Do you speak english? |
| 23:37.04 | xeet2 | and some people will tell you to count the minutes in a month to determine the best way to skin said cat |
| 23:37.10 | visik7 | CoaxD more or less |
| 23:37.11 | ariel_ | voip options are getting out of hands. You should see all the vendors offereing voip service at the show today.. |
| 23:37.13 | CoaxD | visik7: If you don't, i'll understand. No, I did not call you an idiot |
| 23:37.30 | CoaxD | visik7: I'm insinuating that tzanger was asking me due to the fact that I gave you a definitive answer, and he wasn't confident in it |
| 23:37.32 | tzanger | ariel_: how is it |
| 23:37.51 | CoaxD | visik7: He wanted to know if I was truely giving you the right answer |
| 23:37.57 | ariel_ | newsham, asterisk is a pbx/voip server not a softphone and yes it can connect ip to ip. |
| 23:37.57 | visik7 | ah |
| 23:38.22 | hacim | CoaxD: lets say you wanted to do cheap calls, do most people go the spa-1000 and voicepulse route? |
| 23:38.25 | xeet2 | newsham: asterisk can do anything, thats why its called * |
| 23:38.41 | tzanger | xeet2: haha |
| 23:38.45 | ariel_ | tzanger, it was nice to see so many new vendors and lots of new toys. The new BT phone looks nice. But there still not shipping yet. |
| 23:38.48 | CoaxD | hacim: I would do asterisk, spa-1000, and voicepulse connect |
| 23:38.57 | Katty | xeet2: can * make hot cocoa? |
| 23:38.57 | CoaxD | hacim: Only, i'd always buy an spa-2000 |
| 23:39.05 | tzanger | ariel_: yeah I'd like a nice BT phone |
| 23:39.05 | CoaxD | hacim: SPA-2000's are $60 on ebay + $10 shipping right now |
| 23:39.06 | xeet2 | katty: if you run it on an amd box |
| 23:39.09 | hacim | there should be like a common scenario breakdown somewhere |
| 23:39.12 | Katty | :>>> |
| 23:39.14 | hacim | CoaxD: thats a pretty good price |
| 23:39.16 | ariel_ | I like the flip screen |
| 23:39.19 | CoaxD | hacim: Indeed ti is |
| 23:39.19 | tzanger | I have a motorola BT headset but all the softphones suck |
| 23:39.25 | CoaxD | hacim: I'm thinking of buying one, myself |
| 23:39.26 | tzanger | Katty: yes it can |
| 23:39.32 | tzanger | see my comment about running it on AMD earlier |
| 23:39.33 | CoaxD | hacim: I have a Sipura 2000 and it works WONDERFULLY |
| 23:39.35 | newsham | ok, lets say I want to set up a system where 5 people in disparate locations have a handset<->computer interface, and they should all be able to dial and connect to one another and have options like vmb. would that be a single centralized asterisk? |
| 23:39.39 | Katty | k'then |
| 23:39.42 | visik7 | tzanger anyway I put the hfc into a domU and run * on it |
| 23:39.43 | CoaxD | hacim: I have not had even one ounce of trouble with it |
| 23:39.46 | newsham | would they each need an asterisk or just a soft phone? |
| 23:39.55 | tzanger | visik7: how are you accessing a PCI device directly in a XenU? |
| 23:39.55 | xeet2 | katty: fries too, just have to get a powerful enough case fan |
| 23:40.03 | ariel_ | Ok folks see you later I have to make dinner and feed my baby. See you later. |
| 23:40.04 | CoaxD | hacim: Then again, I've heard great reviews of several ATAs |
| 23:40.08 | tzanger | later air |
| 23:40.09 | tzanger | er ariel_ |
| 23:40.13 | CoaxD | hacim: The IAXy is a great little box, albeit very overpriced |
| 23:40.14 | visik7 | tzanger xen is able to do that |
| 23:40.21 | hacim | CoaxD: it seems like the SPA-2000 is the reference that all compare against |
| 23:40.21 | newsham | xen is nifty. |
| 23:40.23 | CoaxD | hacim: (I haven't tried it, but i've read great reviews on it) |
| 23:40.24 | Katty | buh bye, ariel_ (= |
| 23:40.27 | Katty | thanks for info earlier! |
| 23:40.30 | tzanger | visik7: I must have an early version of Xen :-) |
| 23:40.30 | CoaxD | hacim: Yeah, becuase most people get SPA-2000 |
| 23:40.30 | visik7 | tzanger hide the device to dom0 and set the domU to get it |
| 23:40.37 | CoaxD | hacim: Its the most commonly available product |
| 23:40.43 | tzanger | visik7: what about CPU time? |
| 23:40.47 | hacim | CoaxD: the only reason why I consider the IAXy is because it cuts out the SIP, which seems unnecessary if I am going to use voicepulse connect |
| 23:40.49 | CoaxD | hacim: The thing is, you can have 2 fxs ports plugged into 2 separate phones, connected to 2 different sip clients |
| 23:40.54 | tzanger | visik7: can you say that the 'asterisk' domain gets CPU whenever it needs it? |
| 23:41.01 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 23:41.05 | tzanger | hacim: iaxy has its own set of problems |
| 23:41.09 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 23:41.09 | CoaxD | hacim: Oh. Yeah, so you wouldnt need to use asterisk at all |
| 23:41.14 | CoaxD | hacim: You could just hook up your VPC |
| 23:41.15 | tzanger | test it thoroughly before sending it across the sea :-) |
| 23:41.16 | newsham | tzanger: there are several schedulers that you can assign |
| 23:41.17 | visik7 | tzanger u can limit the cpu of the domU running * |
| 23:41.25 | hacim | CoaxD: VPC? |
| 23:41.27 | tzanger | visik7: I don't want to limit it :-) |
| 23:41.32 | CoaxD | hacim: actually, i do think you need asterisk to use IAXy |
| 23:41.35 | visik7 | ok so don't limit it |
| 23:41.37 | CoaxD | hacim: Please, though, don't quote me on that |
| 23:41.52 | CoaxD | hacim: VPC == Voice Pulse COnnect |
| 23:42.01 | CoaxD | hacim: (http://connect.voicepulse.com) |
| 23:42.02 | hacim | oh right, yeah I think you'd need asterisk |
| 23:42.11 | *** join/#asterisk therouterboy (~icechat5@pcp0011553856pcs.anapol01.md.comcast.net) |
| 23:42.11 | visik7 | tzanger anyway I use Xen 2.0.4 |
| 23:42.14 | CoaxD | hacim: yeah because of the iaxy provisioning stuff |
| 23:42.16 | tzanger | I think I'm on 2.0.0 |
| 23:42.25 | visik7 | olso 2.0.0 can do that |
| 23:42.27 | tzanger | actually 2.0.0 has a problem with reducing memory footprint |
| 23:42.28 | hacim | CoaxD: however, isn't there a way to cut out the ATA altogether and just get an IAX phone? |
| 23:42.33 | *** part/#asterisk newsham ({d64KtK7VP@malasada.lava.net) |
| 23:42.44 | CoaxD | hacim: There are some in the works, but nothing concrete yet |
| 23:42.54 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 23:42.54 | CoaxD | hacim: For instance, http://www.farfon.com |
| 23:43.00 | tzanger | CoaxD: when they ship |
| 23:43.03 | CoaxD | hacim; They're gonna manufacture on a small scale |
| 23:43.11 | tzanger | CoaxD: I'll have to try it later |
| 23:43.12 | CoaxD | hacim: but they aren't shipping production units yet |
| 23:43.13 | hacim | CoaxD: I guess you might loose some configurability that way? |
| 23:43.15 | tzanger | er not CoaxD, visik7 |
| 23:43.24 | tzanger | xen rox muh sox |
| 23:43.26 | Goshen | Asterisk Management Portal - AMP doesn't have its own channel does it? |
| 23:43.27 | visik7 | tzanger but I got that oops 2 hours ago and I'm not sure if Xen is ready for a production system |
| 23:43.38 | tzanger | visik7: well 2.0.0's running production |
| 23:43.44 | xeet2 | bbl |
| 23:43.45 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 1PM CST MARCH 3rd -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
| 23:43.45 | CoaxD | hacim: The thing is, i'd MUCH rather have an ATA |
| 23:43.50 | CoaxD | hacim: I do not want a phone I dont like |
| 23:43.59 | eipi | anyone is workign with odbc? |
| 23:44.01 | CoaxD | hacim: With an ATA, i get to choose whichever phone i want. (Which might well be a CORDLESS phone) |
| 23:44.08 | visik7 | tzanger dunno, I'm waiting for a reply by the xen team |
| 23:44.24 | tzanger | :-) |
| 23:44.25 | tzanger | what |
| 23:44.26 | tzanger | about |
| 23:44.30 | hacim | CoaxD: very true |
| 23:44.32 | CoaxD | hacim: problem is, with a sip PHONE, you might get some buttons that have bells and whistles that you cant really duplicate on a regular phone |
| 23:44.41 | CoaxD | hacim: i.e. sip/iax/whatever |
| 23:45.02 | hacim | CoaxD: i just wish I had a really small phone (like cell phone size) that I could take with me, with my ATA |
| 23:45.10 | visik7 | what about what ? |
| 23:45.17 | CoaxD | hacim: Hmm. they do make phones like that |
| 23:45.22 | CoaxD | hacim: Cordless, of course |
| 23:45.30 | hacim | CoaxD: really? hmm I haven't found one yet |
| 23:45.35 | CoaxD | hacim: I saw a few at walmart |
| 23:45.39 | *** part/#asterisk urs (~urs@zentrum.bielewelt.net) |
| 23:45.40 | CoaxD | hacim: That was a long while ago, tho |
| 23:45.45 | hacim | CoaxD: its kinda lame to carry around an ATA and a giant pushbutton phone |
| 23:45.51 | CoaxD | hacim: Yes indeed |
| 23:46.13 | CoaxD | hacim; There's nothing out there that'll do EXACTLY what you want, but i'm sure yoiu can find a small-footprint cordless phone |
| 23:46.18 | CoaxD | hacim: Trouble is, the bases arent that small yet |
| 23:46.29 | tzanger | CoaxD: just need to look around |
| 23:46.52 | hacim | CoaxD: yeah, thats the thing, you get a cordless and you are then carrying around an ATA + cordless base + cordless phone |
| 23:47.13 | CoaxD | hacim: The thing is, man, an ATA isnt really meant for carrying around from location to location |
| 23:47.15 | hacim | CoaxD: when I go somewhere, I want to travel with my voip action, but I dont want to unload a suitcase of equipment |
| 23:47.17 | visik7 | it's late I have to get up in 5 hours |
| 23:47.19 | CoaxD | hacim: I mean it *CAN* be used for that |
| 23:47.24 | visik7 | bye |
| 23:47.32 | CoaxD | hacim: Not that you're whacky for wanting to use it for that or anything |
| 23:47.40 | tzanger | I want to get headset bluetooth support for linux working |
| 23:47.40 | hacim | CoaxD: what would you do if you wanted to do that? |
| 23:47.44 | CoaxD | hacim: Just need small footprint phone gear to make it practical, |
| 23:47.44 | CoaxD | etc |
| 23:47.48 | tzanger | then I could use iaxcomm with linux and the bt101 |
| 23:47.53 | CoaxD | hacim: I'd bitch about the same damn things you are ;) |
| 23:48.01 | CoaxD | hacim: You know, you might want to check out the wireless phone by pulver |
| 23:48.05 | CoaxD | hacim: Its 802.11b |
| 23:48.06 | tzanger | NO YOU DON"T |
| 23:48.08 | tzanger | it sucks ass |
| 23:48.11 | hacim | haha |
| 23:48.26 | hacim | tzanger: what would you recommend for travelling voip? |
| 23:48.29 | tzanger | gets hot, poor battery life, shitty display |
| 23:48.33 | hacim | ATA + phone? |
| 23:48.33 | CoaxD | tzanger: Lame |
| 23:48.36 | CoaxD | tzanger: :( |
| 23:48.42 | tzanger | CoaxD: yeah, I was really disappointed |
| 23:48.46 | tzanger | hacim: I don't know |
| 23:48.48 | hacim | thats good to know |
| 23:48.52 | CoaxD | hacim: a better product needs to be made for roaming voip, for sure |
| 23:48.55 | CoaxD | cuz one doesnt really exist |
| 23:48.58 | tzanger | I'd probably use my bt headset if I could get it working well in linux |
| 23:49.09 | tzanger | firefly and windows is alright but the windows bt stack seems more unstable than linux's |
| 23:49.14 | CoaxD | hacim: Perhaps a softphone, given that you almost always will need a computer too? |
| 23:49.18 | hacim | CoaxD: I dont really need wireless voip, I just need to head to san francisco and get to my friend's place, and plug into his network and have my phone |
| 23:49.29 | CoaxD | hacim: I hear that |
| 23:49.41 | CoaxD | hacim: It is also possibel to forward calls to your cel with a voicepulse account |
| 23:49.46 | CoaxD | er possible |
| 23:49.49 | hacim | tzanger: thats my problem, there is no good iax client in linux |
| 23:49.54 | Katty | 'voicepulse'? |
| 23:49.54 | CoaxD | hacim: You pay double, but its worth it sometimes |
| 23:50.03 | tzanger | hacim: iaxcomm is alright |
| 23:50.06 | CoaxD | katty: Voicepulse is a voice over IP to regular telephone network gateway |
| 23:50.12 | tzanger | and it is written by the guy who put the new jitter buffer in * |
| 23:50.29 | jsolares | what is the difference between loop start, ground start, and kewl start |
| 23:50.37 | CoaxD | katty: it allows you to make phone calls over the pstn (the regular telephone network) |
| 23:50.40 | Katty | CoaxD: err, would that be the red modules? |
| 23:50.46 | CoaxD | katty: haha. no. |
| 23:50.49 | Katty | k'then |
| 23:50.53 | Katty | don't try explaining it to me |
| 23:50.53 | hacim | its the blue pill |
| 23:50.53 | CoaxD | katty: HOw old are you, if you don't mind me asking? |
| 23:51.00 | Katty | i've had all the input i can handle for one day :) |
| 23:51.04 | JerJer | i'm 12 |
| 23:51.04 | Katty | i'm twenty |
| 23:51.13 | tzanger | jerjer don't like |
| 23:51.15 | CoaxD | Katty: Cool. :) Are you a windows geek, or? |
| 23:51.19 | tzanger | you're 8 and that's being generous |
| 23:51.29 | CoaxD | JerJer: Shush yer damn mouth. she's a 20 year old chick. If you shut your mouth enough, you might actually get to see one naked someday. |
| 23:51.35 | JerJer | oh yeah that's right 8 year old body 12 year old winky |
| 23:51.37 | Katty | CoaxD: that's what i learned on, yes. slowly been converting to linux for the sake of screening irssi |
| 23:51.39 | CoaxD | jerjer: (i.e. instead of scaring them away.) |
| 23:51.41 | tzanger | CoaxD: ... wtf |
| 23:51.41 | hacim | hrm, well I think it might be voicepulse+ata2000+home asterisk box for me |
| 23:51.51 | tzanger | irssi works wonders |
| 23:51.53 | JerJer | CoaxD: I have a live human girlfriend |
| 23:51.59 | Katty | CoaxD: and discovering some wonderful things on the way :} |
| 23:52.00 | hacim | mmm, irssi |
| 23:52.01 | CoaxD | Jerjer: hehe cool :) |
| 23:52.08 | CoaxD | Katty: Awesome! |
| 23:52.11 | JerJer | to which kram facilitated meeting |
| 23:52.16 | tzanger | irssi was the reason for the rise of the roman empire |
| 23:52.22 | hacim | CoaxD: does voicepulse end up being cheaper than just doing somethng like broadvocie? |
| 23:52.38 | CoaxD | hacim: I'd rather use NuFone, if it were me |
| 23:52.40 | hacim | tzanger: irssi+bitlbee |
| 23:52.42 | CoaxD | hacim: cheaper rates |
| 23:52.45 | tzanger | hacim: haven't tried bitlbee yet |
| 23:52.48 | tzanger | I use Psi for my IM |
| 23:52.49 | CoaxD | hacim: http://www.nufone.net |
| 23:52.57 | CoaxD | JerJer, i'll expect my check next week |
| 23:53.07 | JerJer | yeah its already in the mail |
| 23:53.10 | CoaxD | JerJer: Sweet |
| 23:53.24 | hacim | CoaxD: hmm, a minute ago you were suggesting voicepulse, now nufone? |
| 23:53.29 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 23:53.30 | CoaxD | hacim: No, i didnt suggest voicepulse |
| 23:53.37 | CoaxD | hacim: You brought it up, and so i assumed that it was what you were using |
| 23:53.38 | tzanger | I use nufone almost exclusively |
| 23:53.41 | CoaxD | hacim: I would never suggest voicepulse |
| 23:53.46 | hacim | CoaxD: ah! ok :) |
| 23:53.51 | CoaxD | hacim: Too many problems |
| 23:53.57 | CoaxD | hacim: They might've fixed some of those, but.. |
| 23:54.01 | hacim | CoaxD: I had no idea, I just assumed people used voicepulse |
| 23:54.09 | CoaxD | hacim: I have. They're not a bad telco at all |
| 23:54.11 | hacim | tzanger: nufone pretty decent? |
| 23:54.14 | CoaxD | hacim: I just prefer nufone. Better service. |
| 23:54.21 | CoaxD | hacim: I actually get to talk to the owner once in a while |
| 23:54.27 | CoaxD | hacim: Although he's a total jackass at times |
| 23:54.32 | tzanger | hacim: its *excellent*... it si *NOT* for newbies though |
| 23:54.32 | CoaxD | JerJer: :) |
| 23:54.42 | JerJer | just don't expect NuFone to call you back or answer email |
| 23:54.45 | tzanger | I am the unofficial (and unpaid) nufone frontline support :-) |
| 23:54.53 | hacim | hahah |
| 23:54.57 | *** part/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 23:54.58 | CoaxD | hacim: (Just FYI, JerJer owns NuFone :) |
| 23:55.03 | jsolares | tzanger, go make them gimme my DID |
| 23:55.03 | hacim | ok, I'm revising... spa-2000+nufone |
| 23:55.06 | jsolares | :) |
| 23:55.11 | tzanger | jsolares: sorry I don't do incoming |
| 23:55.14 | tzanger | only termination help :-) |
| 23:55.23 | jsolares | how convenient :p |
| 23:55.23 | tzanger | jsolares: got a ticket #? |
| 23:55.23 | hacim | jerjer the 12 yer old star warez character owns nufone? |
| 23:55.33 | jsolares | not yet |
| 23:55.37 | tzanger | hacim: no that's jarjar |
| 23:55.44 | tzanger | jsolares: if you haven't got a ticket# yo uhaven't got a gripe |
| 23:55.47 | hacim | oh, jarjar and juarez |
| 23:55.55 | jsolares | i'm too lazy, and it's only an added bonus |
| 23:56.03 | jsolares | when i do need it, i'll get the ticket |
| 23:56.10 | tzanger | support@nufone.net |
| 23:56.23 | hacim | where does nufone have dids? |
| 23:56.27 | jsolares | michigan |
| 23:56.31 | hacim | only? |
| 23:56.33 | tzanger | email them and say "jerjer's a dirty whore, he took my money and won't give me my did" |
| 23:56.44 | jsolares | hacim: as far as i know, yes |
| 23:56.51 | denon | hey, doan' be mockin jerjer |
| 23:56.54 | hacim | hrm, thats not very convenient |
| 23:56.54 | jsolares | lol tzanger |
| 23:57.03 | jsolares | get an 1800 did |
| 23:57.14 | tzanger | jsolares: you'll get a ticket# and if you don't get a live human response in 24h after getting the ticket, come in here and Jerjer will personally smack his support team |
| 23:57.24 | tzanger | I'm still waiting for the webcam so I can see that in realtime |
| 23:57.24 | jsolares | ohhh neat |
| 23:57.25 | hacim | jsolares: whats incoming cost on a 800 DID? |
| 23:57.32 | denon | no charge, just minutes |
| 23:57.34 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 23:57.37 | jsolares | hacim: i think 0.02$ per minute |
| 23:57.52 | hacim | nufone planning on getting other DIDs? |
| 23:58.20 | eipi | anyone is working with odbc? |
| 23:58.57 | hacim | jsolares: yeah but having an 800 DID means you get charged incoming minutes, which you dont if you have a local DID, right? |
| 23:59.05 | jsolares | right |
| 23:59.18 | hacim | which means, its more expensive unless you are in michigan |
| 23:59.20 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
| 23:59.24 | jsolares | but it's an 1800 man |
| 23:59.24 | tzanger | bbl, bathing th ekids |
| 23:59.40 | tzanger | hacim: depends |
| 23:59.42 | jsolares | i'm -> ' ' <- that close to getting an 1800 DID |
| 23:59.46 | tzanger | the DIDs I lease are per-minute |
| 23:59.53 | tzanger | take as many concurrent calls as you want |
