| 00:00.55 | cloud9 | Shaun2222 : yea.. I followed a step by step guide that for the most part seems to be working.. just some glitches.. thanks anyway. I'll get it. |
| 00:01.15 | linuxtuxi1 | syzygyBSD: just tested it out to be 100% sure...and I can confirm wget www.google.com gives me back the expected result |
| 00:01.23 | mercestes | cloud9: In res_mysql.conf you should have localhost and a point to your sock. It's complaining that the name "mysql" can't be found in ODBC so remove the odbc crap and just use the direct APi's in Res_mysql.conf |
| 00:01.43 | cloud9 | mercestes : even if my database isn't on the localhost? |
| 00:02.01 | mercestes | cloud9: Umm....no..then you point it at your database host..lol |
| 00:02.09 | cloud9 | cool, lol |
| 00:02.55 | cloud9 | [general] |
| 00:02.55 | cloud9 | dbhost = 192.168.1.201 |
| 00:02.56 | cloud9 | dbname = asterisk |
| 00:02.56 | cloud9 | dbuser = root |
| 00:02.56 | cloud9 | dbpass = billy123 |
| 00:02.56 | cloud9 | dbport = 3306 |
| 00:03.04 | mercestes | Nice. |
| 00:03.08 | cloud9 | soo |
| 00:03.16 | mercestes | Should I go into why you shouldn't have done that?? |
| 00:03.21 | cloud9 | sorry |
| 00:03.33 | mercestes | don't say sorry to me...lol..change your pass. |
| 00:03.39 | mercestes | and dont' use root, make another user. |
| 00:03.57 | Shaun2222 | can you do dial(${args}) and have ARGS set to "SIP/111|30|t" anybody know if that will work? |
| 00:03.58 | CunningPike | NormanASD: Have you set up hints? |
| 00:04.00 | cloud9 | you won't find my asterisk or db ip. it's not anywhere near here. |
| 00:04.13 | cloud9 | but anyway |
| 00:06.13 | NormanASD | no, I thought hints were only for presence, which is not what I want. |
| 00:06.25 | brian | hey, just wondering if there is anyway at all to listen for DTMF on a meetme conference using fastagi? |
| 00:07.02 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 00:07.02 | *** join/#asterisk Vegar (i=vegar@unaffiliated/vegar) [NETSPLIT VICTIM] |
| 00:07.02 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 00:07.02 | *** join/#asterisk Rhizome (n=Rhizome@host-81-191-147-145.bluecom.no) [NETSPLIT VICTIM] |
| 00:07.02 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 00:07.15 | stephane | re |
| 00:08.16 | CunningPike | NormanASD: OK - I thought you were asking about SUBSCRIBE/NOTIFY |
| 00:08.33 | *** join/#asterisk oomph-work2 (n=jimmy@65.216.185.17) |
| 00:09.39 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 00:10.26 | clyrrad | Does anyone know of a canadian voip provider than can register and port area code 905 DID's? |
| 00:11.30 | NormanASD | CunningPike: I was, but SUBSCRIBE/NOTIFY is much more general than for presence. It can, according to the RFC, be used for any arbitrary events. |
| 00:12.08 | CunningPike | NormanASD: True - I assumed you were using it for presence. Most questioners here are - what are you using it for> |
| 00:12.08 | CunningPike | ? |
| 00:12.37 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
| 00:13.06 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 00:15.32 | *** part/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net) |
| 00:16.45 | NormanASD | CunningPike: I want to pass some state from one peer to another, in this case application state. |
| 00:17.18 | NormanASD | While the RFC will allow one to do this, Asterisk in the middle, won't. I could do this if I disabled the noreinvite, I suspect, since the two peers would talk direct, but I don't want to do that. |
| 00:18.05 | NormanASD | I'm wondering if the SIP INFO command will work instead. |
| 00:18.16 | file | Asterisk isn't a SIP proxy... it won't deliver messages directly between devices |
| 00:18.33 | lullabud | i have a question concerning data routing between SIP phones connected to asterisk servers. |
| 00:19.42 | lullabud | if i have two or more phones in a private network, would it be advantageous to have an asterisk server in there as opposed to having the phones talk across the internet to an asterisk server somewhere else? |
| 00:20.08 | lullabud | that is, does the data flow directly from one phone to another, or does it go through the asterisk server? |
| 00:22.52 | NormanASD | file: thanks, that's what I was starting to suspect. I guess I'll have to either go SIP peer to peer bypassing asterisk for this type of call, or use another protocol behind the scenes. |
| 00:27.00 | NormanASD | Alas, I guess I need to add a header to the INVITE so the two peers will know who they are talking to, and then modify the dialplan to read the variable and pass it to the other peer. |
| 00:29.08 | *** join/#asterisk yardB (n=yardie@c-68-44-44-42.hsd1.nj.comcast.net) |
| 00:29.13 | *** join/#asterisk killercoder (n=killerco@CPE000c41aac393-CM001225009430.cpe.net.cable.rogers.com) |
| 00:29.54 | yardB | awfully quiet here! |
| 00:30.07 | Un1x | hey |
| 00:32.13 | *** join/#asterisk yassine (n=yassine@xdsl-87-78-32-31.netcologne.de) |
| 00:32.32 | Un1x | i just noticed |
| 00:32.35 | Un1x | g729 module |
| 00:32.41 | Un1x | isn't available for athalon in 64 bit |
| 00:32.50 | Un1x | btw is 1.4 stable yet |
| 00:32.56 | yassine | anyone of you guys from germany using an "AVM b1 Aktive ISDN Karte" ? |
| 00:32.56 | Un1x | or at least reliable not to crash |
| 00:36.19 | *** join/#asterisk syberdave (i=asdf@syberdave.net) |
| 00:37.12 | tsurk0 | hello everybody |
| 00:37.12 | syberdave | is there a way to set the caller ID number going to a SIP provider to be routed to PSTN? |
| 00:37.40 | yardB | i want to test a SIP phone for incoming call ..can i get a voluteer to call? |
| 00:37.44 | tsurk0 | i'm using IAX to connect to an asterisk over internet - could you tell me how is the password transmited? |
| 00:37.49 | tsurk0 | is it encrypted? |
| 00:39.03 | yardB | yes? |
| 00:40.14 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 00:42.59 | *** join/#asterisk apardo (n=apardo@87.217.145.139) |
| 00:47.26 | *** join/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
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| 00:54.09 | *** join/#asterisk glasco36 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 00:54.22 | yardB | useless |
| 00:54.39 | *** join/#asterisk blitzrage (n=blitzrag@CPE0016b614c984-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 00:57.22 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:57.24 | Un1x | hello |
| 00:57.28 | Un1x | is anyone around |
| 00:57.29 | Un1x | ? |
| 00:57.33 | Un1x | i get this after installing zaptel |
| 00:57.34 | Un1x | http://pastebin.ca/261984 |
| 00:57.37 | Un1x | and doing |
| 00:57.40 | Un1x | modprob wctdm |
| 00:57.43 | Un1x | thats the error i get |
| 00:58.40 | *** join/#asterisk gr1ncheux_ (n=devine@AStDenis-105-1-43-115.w80-8.abo.wanadoo.fr) |
| 01:03.10 | yardB | Anyone with a sip PHONE ? |
| 01:03.19 | Un1x | hmm whoever, msged me i closed the window accidenly, anywya... as you see above i said zaptel ^^^ wich means i have a Analouge phone not a SIP |
| 01:05.02 | yardB | <PROTECTED> |
| 01:05.09 | Strom_C | yardB: I have several |
| 01:05.31 | *** join/#asterisk Agrizzi (n=Brandon@69-165-243-236.atlsfl.adelphia.net) |
| 01:05.35 | yardB | cool Strom_C |
| 01:07.32 | yardB | Strom_C i meant test |
| 01:08.29 | *** part/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 01:10.47 | *** part/#asterisk glasco36 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 01:14.36 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 01:14.55 | *** join/#asterisk MrTelephone (n=DeaLER25@204.244.209.131) |
| 01:15.10 | MrTelephone | So is anyone here working on any channel protocols? |
| 01:19.47 | rob0 | Un1x: the error said to "see dmesg". Did you? It wasn't in your paste. |
| 01:21.28 | Un1x | http://pastebin.ca/261997 |
| 01:21.32 | Un1x | the paste was wrong |
| 01:21.49 | MrTelephone | why is here no NCS -> MGCP proxy servers anywhere |
| 01:21.50 | MrTelephone | damn |
| 01:22.15 | Un1x | http://pastebin.ca/262000 |
| 01:22.18 | Un1x | here is the dmesg |
| 01:22.57 | MrTelephone | Un1x what kind of hardphones are you using? |
| 01:24.10 | rob0 | Looks like for some reason crc_ccitt didn't load. |
| 01:25.35 | rob0 | wctdm needs zaptel needs crc_ccitt , so they should load in reverse order (crc_ccitt, zaptel, wctdm). |
| 01:25.36 | Un1x | a regular phone but i'm trying to compile zaptel |
| 01:25.42 | Un1x | i havbent got to asterisk or libpri yet |
| 01:25.56 | rob0 | do you need libpri? |
| 01:25.57 | Un1x | rob0 how do i get crc_ccitt to load |
| 01:26.05 | Un1x | Yes, im using the TDM400P |
| 01:26.09 | *** part/#asterisk syberdave (i=asdf@syberdave.net) |
| 01:26.12 | *** join/#asterisk keith80403 (n=keith804@24-56-189-80.co.warpdriveonline.com) |
| 01:26.39 | rob0 | Un1x: it should Just Work ... it always did for me. Maybe you didn't enable crc_ccitt in your kernel? "modprobe -v crc_ccitt"?? |
| 01:27.30 | Un1x | root@Canucks2:~/zaptel-1.2.11# modprobe -v crc_ccitt |
| 01:27.30 | Un1x | FATAL: Module crc_ccitt not found. |
| 01:27.30 | Un1x | root@Canucks2:~/zaptel-1.2.11# |
| 01:27.31 | *** join/#asterisk dean_ (n=chatzill@115-127.187-72.tampabay.res.rr.com) |
| 01:27.38 | Un1x | yea the module isn't found so is that the problem? |
| 01:28.09 | rob0 | indeed |
| 01:28.24 | Un1x | thanks rob0 |
| 01:29.17 | *** part/#asterisk dean_ (n=chatzill@115-127.187-72.tampabay.res.rr.com) |
| 01:30.35 | *** join/#asterisk hyphen (n=hyphen@c-71-224-213-97.hsd1.pa.comcast.net) |
| 01:31.53 | sbingner | how would I do this: $[${DB(REMOTEUSERS/${EXTEN}/timeout)} ? ${DB(REMOTEUSERS/${EXTEN}/timeout)} :: 30] |
| 01:32.10 | sbingner | it's giving an error if it doesn't exist |
| 01:37.52 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:38.33 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:42.37 | sbingner | $[$[${DB(REMOTEUSERS/${EXTEN}/timeout)} != ""] ? $[${DB(REMOTEUSERS/${EXTEN}/timeout)}] :: 30] <-- works, but gives 2 ast_yy errors |
| 01:43.04 | bkw_ | sbingner, |
| 01:43.13 | bkw_ | wrap it in () |
| 01:43.17 | bkw_ | it will stop bitching |
| 01:43.29 | bkw_ | (${DB(REMOTEUSERS/${EXTEN}/timeout)}) |
| 01:44.22 | sbingner | cool, thanks |
| 01:44.24 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 01:45.21 | bkw_ | also doesn't $DB_RESULT exist? |
| 01:45.25 | bkw_ | you can use that for the second leg of that |
| 01:45.30 | bkw_ | so you don't have to do the lookup twice |
| 01:45.39 | SplasPood | hrm you can do that? 3 levels of db? |
| 01:45.53 | bkw_ | yes |
| 01:45.56 | SplasPood | ie, family/key/otherkey ? |
| 01:45.58 | bkw_ | its key val |
| 01:46.06 | bkw_ | SplasPood, its all just one long ass key |
| 01:46.16 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.159.Dial1.SanJose1.Level3.net) |
| 01:46.17 | litage | what are the most probable causes of audio cutting in and out and stuttering for outgoing sip calls and incoming zap calls? |
| 01:46.17 | bkw_ | test/12342345/2342/32345/23463/46346/56/457 |
| 01:46.21 | bkw_ | would be perfectly valid |
| 01:46.24 | SplasPood | hrm, ok gotcha |
| 01:46.27 | SplasPood | i use - for that |
| 01:46.30 | SplasPood | <PROTECTED> |
| 01:46.35 | bkw_ | hehe yah |
| 01:46.36 | SplasPood | I will use / from now on |
| 01:46.44 | bkw_ | litage, are you doing any reloads or high verbose? |
| 01:47.13 | SplasPood | btw, while people are alive/responding... anyone know the difference between the debug = in asterisk.conf and verbose = ? |
| 01:47.29 | bkw_ | SplasPood, last I seen debug wasn't used much if any |
| 01:47.33 | bkw_ | verbose was where it was at |
| 01:47.38 | sbingner | lol it still bitches |
| 01:47.44 | bkw_ | sbingner, then they broke it :P |
| 01:48.02 | bkw_ | wrap it in a ${ISNULL()} |
| 01:48.03 | bkw_ | maybe? |
| 01:48.08 | sbingner | oo :) |
| 01:48.21 | litage | bkw_: nope |
| 01:48.22 | bkw_ | I know the () works in my version |
| 01:48.27 | SplasPood | bkw_: yea the new doc/asterisk-conf or whatever pointed me to this debug... first I'd heard of it.. other than the debug entry in logger.conf |
| 01:48.40 | bkw_ | SplasPood, don't think much uses it yet |
| 01:48.50 | bkw_ | debug was an after thought |
| 01:49.03 | litage | bkw_: also have very low call volume...1 call every 1-10 minutes |
| 01:49.15 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 01:49.30 | SplasPood | bkw_: ooo.. #exec, thats new, no? |
| 01:49.53 | bkw_ | na been around for a few |
| 01:49.56 | bkw_ | anthm coded that btw |
| 01:50.12 | bkw_ | litage, interrupt issues |
| 01:50.17 | MrTelephone | litage, did u check ur irqs |
| 01:50.37 | bkw_ | cat /proc/interrupts |
| 01:50.38 | MrTelephone | some guy said to turn off hyperthreading if your running ht enabled processor |
| 01:50.47 | bkw_ | MrTelephone, that doesn't matter really |
| 01:50.53 | MrTelephone | ok |
| 01:51.04 | MrTelephone | why do atas and handsets sound better than sip hardphones? |
| 01:51.10 | MrTelephone | when using tdm cards |
| 01:51.16 | bkw_ | MrTelephone, because you have a sucky hard phone? |
| 01:51.21 | bkw_ | My 7970 does great |
| 01:51.28 | bkw_ | along with my grandstream BT101 |
| 01:51.39 | bkw_ | I do g722 with my grandstream.. sounds awesome |
| 01:51.43 | SplasPood | bkw_: ahh never knew about it before... did an svn update the other day and found the new doc for asterisk.conf |
| 01:51.44 | litage | bkw_: yeah, /proc/interrupts looks fine: http://rafb.net/paste/results/QMhz5h53.html |
| 01:52.10 | MrTelephone | I have polycom 501s and cisco 7920's |
| 01:52.35 | litage | MrTelephone: the server's got a Celeron CPU, so it doesn't have hyperthreading |
| 01:52.38 | bkw_ | MrTelephone, my goodness those are good phones |
| 01:52.39 | MrTelephone | and the tdm circuits sound crappy. now I know i have tdm issues because the phone lines suck. but on the other hand the atas+handsets sound better as if the handsets filter out crap |
| 01:52.49 | bkw_ | MrTelephone, what card you running? |
| 01:53.27 | MrTelephone | im running 1 digium 4 port card (wctdm) and 2 tdm2400 cards w/o echo cancel (tdm24xxp) |
| 01:53.32 | MrTelephone | all in different machines |
| 01:54.03 | sbingner | bkw_: lol, that works for the logical part, but then it still whines about the assignment value it's not using (1 ? 30 :: ) |
| 01:54.22 | MrTelephone | It's such a wierd problem I don't know if I can even find out where to start looking to fix it |
| 01:54.38 | sbingner | bkw_: long time no see btw heh |
| 01:54.50 | litage | bkw_: any suggestions as to how i can debug audio cutting in and out and stuttering for outgoing sip calls and incoming zap calls? |
| 01:54.56 | MrTelephone | and some days there are more echo than other days... |
| 01:55.13 | Supaplex | monkies have invaded our phone system! |
| 01:55.18 | Supaplex | ;) |
| 01:55.25 | MrTelephone | I hear sangoma cards have automatic impedence adjusting? |
| 01:55.46 | MrTelephone | I feel like I wasted 3 grand going with digium if thats the case.. |
| 01:55.56 | MrTelephone | not everyone lives 2 blocks away from the CO |
| 01:56.07 | sbingner | I may just have to use Set |
| 01:56.24 | sbingner | and gotoif heh |
| 01:57.50 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 01:57.59 | MrTelephone | litage what is your hit count with zttest tool? |
| 01:58.06 | MrTelephone | 99.8% +? |
| 01:58.38 | MrTelephone | also make sure your computer is grounded out properly?? not sure how to make sure of that |
| 01:58.52 | litage | MrTelephone: 99.987793% 100.000000% 100.000000% 100.000000% 100.000000% 100.000000% |
| 01:58.53 | *** join/#asterisk Agrizzi (n=Brandon@69-165-243-236.atlsfl.adelphia.net) |
| 01:59.00 | Agrizzi | hello everyone |
| 01:59.11 | bkw_ | MrTelephone, what cards do you have? |
| 01:59.13 | Agrizzi | dose anyone know of a web sip phone |
| 01:59.14 | MrTelephone | and its cutting out on your? what kind of phone? did you try different voice codecs? |
| 01:59.35 | MrTelephone | bkw_: tdm2400 and tpe400 or something |
| 01:59.44 | MrTelephone | whatever the 4 port fxo card is |
| 01:59.56 | bkw_ | MrTelephone, yah give up now. those suck |
| 02:00.05 | MrTelephone | ligta, is your cpu usage going crazy? |
| 02:00.11 | litage | MrTelephone: grounding would occur via the powersupply |
| 02:00.15 | MrTelephone | bkw_: what do u suggest? |
| 02:00.27 | bkw_ | MrTelephone, well what symtoms do you see? |
| 02:00.30 | bkw_ | or hear I should say |
| 02:00.52 | MrTelephone | I have poor outgoing volume, poor incoming volume at default settings |
| 02:01.08 | *** join/#asterisk blitzrage (n=blitzrag@CPE0016b614c984-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 02:01.17 | MrTelephone | slight echo after running fxotune -s -b 1 -e 4 (example) |
| 02:01.22 | _DAW | Hey everyone. I all of a sudden have started receiving warnings from asterisk saying "Ring/Off-hook in strange state 6 on channel 5". Nothing has changed. The channels are T1 kewlstarts through a channelbank (FXO) to the pstn. Any suggestions? |
| 02:01.30 | MrTelephone | echo comes back when line impedence changes due to weather conditions |
| 02:01.34 | *** join/#asterisk kph100 (n=kph100@206-248-156-211.dsl.teksavvy.com) |
| 02:01.36 | kph100 | hello, |
| 02:01.38 | [TK]D-Fender | MrTelephone : A200d FTW |
| 02:01.45 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 02:01.47 | MrTelephone | thanks TK |
| 02:01.55 | kph100 | I need help with asterisk queues. What does the 'announce ' field do? |
| 02:02.19 | [TK]D-Fender | MrTelephone : I just did a replacement for a high-profile client and the difference as night & day. |
| 02:02.29 | MrTelephone | I don't want to purchase differnet cards yet because I want to make sip trunks between the offices when I get a centralized digital phone trunk (PRI) |
| 02:02.52 | Agrizzi | dose anyone know of a web sip phone |
| 02:03.10 | MrTelephone | TKDFender, do digium work in good conditions though? how come noone mentions that they are no good in rural areas? |
| 02:03.12 | kph100 | anyone knows how to use 'announce' in queues.conf? |
| 02:03.49 | MrTelephone | rural meaning impedence changes and high dB loss |
| 02:05.19 | MrTelephone | TK, what were the problems before you switched the cards? |
| 02:06.46 | *** join/#asterisk hyphen (n=hyphen@c-71-224-213-97.hsd1.pa.comcast.net) |
| 02:08.02 | *** part/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 02:09.16 | [TK]D-Fender | MrTelephone : Crackly lines, nasty gain manipulation for clarity (which didn't help echo), a bit of echo though not that bad. |
| 02:09.26 | [TK]D-Fender | MrTelephone : Primarily audio quality. |
| 02:09.49 | MrTelephone | it was bad eh.. I mean I ironed out most of the echo but they still notice it and complain about it |
| 02:10.16 | MrTelephone | and "hard to hear" but when you increase the phone volume you increase the background noise volume |
| 02:10.37 | MrTelephone | if your on the phone the echo may be ok but when you tap a keyboard you can hear the reflection |
| 02:11.14 | MrTelephone | so sometimes there is echo, when there isn't you still get echo from distance noises that occur around you |
| 02:11.25 | MrTelephone | very poor hardware, I do not reccommend it |
| 02:11.58 | MrTelephone | maybe for city installations where the lines are good |
| 02:13.38 | MrTelephone | It's just that my repuation is getting battered because I implemented a cost effective sollution for 3 areas and all of them dislike the echo |
| 02:13.44 | MrTelephone | customers are not very forgiving |
| 02:14.08 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:14.21 | MrTelephone | Now I'm afraid to purchase a pri card. Should a person go with echo cancel or not? People say there is no echo on PRI? |
| 02:14.45 | MrTelephone | Thanks TK for the insight |
| 02:14.47 | [TK]D-Fender | MrTelephone : Hell yes you want EC on-board.... |
| 02:15.01 | MrTelephone | but the difference in cost is quadrouble |
| 02:15.07 | sbingner | this was the only thing I could get to work properly without errors, since it's a number it was good: $[${ISNULL(${DB(REMOTEUSERS/${EXTEN}/timeout)})} ? 30 :: $[0 + 0${DB(REMOTEUSERS/${EXTEN}/timeout)}]] |
| 02:15.12 | MrTelephone | 2000 thousand for EC on PRI? |
| 02:15.32 | MrTelephone | TK I can't respond to you |
| 02:15.42 | [TK]D-Fender | MrTelephone : And far from quadruple. |
| 02:16.03 | MrTelephone | software should be able to handle 1 pri? |
| 02:16.16 | [TK]D-Fender | MrTelephone : Gotcha (on the PM part). You must not be registered to freenode properly. |
| 02:16.19 | MrTelephone | I purchased ec on the tdm2400 digium and it had no effect |
| 02:16.25 | MrTelephone | k |
| 02:16.29 | [TK]D-Fender | MrTelephone : How many PRI's? |
| 02:16.43 | MrTelephone | I want to have the abillity to expand to 90 homes |
| 02:16.49 | MrTelephone | 24 lines should do it |
| 02:16.57 | MrTelephone | right now I'm signing for 15 channels and 30 DID's |
| 02:17.06 | [TK]D-Fender | MrTelephone : A single PRI then.... |
| 02:17.16 | MrTelephone | EC on a single PRI? |
| 02:17.44 | *** join/#asterisk jeebusmobile (n=jeebusmo@cpe-75-80-231-237.dc.res.rr.com) |
| 02:18.31 | MrTelephone | You think I will get a good buck for the digium cards on ebay? |
| 02:19.56 | [TK]D-Fender | MrTelephone : Who's to say.... |
| 02:20.34 | *** join/#asterisk blitzrage (n=blitzrag@CPE0016b614c984-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 02:20.48 | MrTelephone | TDM2400 isn't even on a thick enough board to support its own weight |
| 02:21.03 | MrTelephone | when you look at it, its about to break in half from its own weight |
| 02:21.05 | MrTelephone | oh well |
| 02:21.12 | MrTelephone | I'll wait for the pri |
| 02:22.04 | MrTelephone | I do have a concern with asterisk sip trunking... if you have a link between 2 boxes and you have a call limit of 5. is that 5 x 260 bye packets or do they send bigger packets |
| 02:22.19 | MrTelephone | 1300 I guess it will be. |
| 02:24.11 | [TK]D-Fender | Packet size is constant. |
| 02:24.30 | [TK]D-Fender | MrTelephone : And I'm not sure what you're getting at |
| 02:27.13 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 02:32.51 | JT | MrTelephone: fibreglass won't just snap like that |
| 02:35.17 | MrTelephone | means that if you have 5 phone calls you have 5x260 packets sent every 20ms? |
| 02:35.25 | MrTelephone | so your packet per second count is really high |
| 02:35.52 | MrTelephone | if the packets were multiplexed then it won't put much stress on the routers and etc |
| 02:36.16 | bkw_ | where are you getting those numbers? |
| 02:36.28 | bkw_ | and wtf are you talking about multiplexed? |
| 02:36.34 | bkw_ | the packet is still a small packet |
| 02:36.48 | bkw_ | its not like you're going to jam multiple ones into the packet |
| 02:37.00 | *** join/#asterisk mmurdock_Laptop (n=compaq@adsl-072-156-061-242.sip.bct.bellsouth.net) |
| 02:37.59 | MrTelephone | you may wonder what I am talking about, but the IAX protocol does this. |
| 02:38.06 | MrTelephone | if you have a timing device |
| 02:38.51 | MrTelephone | maybe its no significance |
| 02:39.16 | JT | you said SIP before |
| 02:39.20 | JT | make up your mind |
| 02:39.20 | MrTelephone | but when you run a cable isp a lot of small traffic can bog a network |
| 02:39.44 | bkw_ | rtp can be trunked |
| 02:40.02 | MrTelephone | in asterisk? |
| 02:40.12 | bkw_ | nope |
| 02:40.15 | bkw_ | but it can |
| 02:40.22 | bkw_ | it will be able to be done in freeswitch when we get time |
| 02:40.34 | MrTelephone | your a programmer for freeswitch? |
| 02:41.47 | *** join/#asterisk haidozo (n=mark@m208-127.dsl.rawbw.com) |
| 02:42.00 | bkw_ | MrTelephone, yes |
| 02:42.48 | MrTelephone | I'm developing NCS 1.0 for chan_mgcp.c in asterisk.. more like manipulating someone elses patch |
| 02:42.56 | bkw_ | NCS? |
| 02:43.20 | bkw_ | Network-Based Call Signaling Protocol Specification |
| 02:43.27 | MrTelephone | PacketCable variant of MGCP.. very close to MGCP except the header has NCS 1.0 in it and the Line packages are a little different |
| 02:43.59 | MrTelephone | NCS uses one line package L/signal |
| 02:44.38 | MrTelephone | I have the cable modem mtas working with the patch at asterisk.urtho.net BUT there is a couple HOOKSTATE issues |
| 02:44.59 | bkw_ | the MGCP in asterisk isn't complete |
| 02:45.11 | bkw_ | its only one side of the equation |
| 02:45.19 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 02:45.27 | MrTelephone | you mean it won't talk to other MGCP gateways? |
| 02:45.52 | bkw_ | right |
| 02:45.55 | MrTelephone | the protocol is very easy to follow too.. but I'm an amateur at C and it will take me a while to make adjustments |
| 02:46.34 | bkw_ | "I no longer invest my time in developing solutions for Asterisk platform. |
| 02:46.34 | bkw_ | My new company can help You scale Your Asterisk setup but I strongly believe that Asterisk |
| 02:46.34 | bkw_ | is the most expensive platform for large rollouts in terms of TCO (and not initial investments)." |
| 02:46.39 | MrTelephone | It's a lot easier than SIP.. |
| 02:46.42 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 02:46.44 | bkw_ | thats so freakin true |
| 02:46.48 | rpm | what could be the cause of app_voicemail.c infrequently dropping off the end of voicemail messages? i am using IAX trunking to my ITSP, recording messages in gsm and wav49 |
| 02:46.53 | bkw_ | http://asterisk.urtho.net/tiki-index.php |
| 02:46.55 | MrTelephone | Why is that true? |
| 02:47.07 | bkw_ | it just is dear |
| 02:47.13 | MrTelephone | you can hav eyour network guy do all the phone stuff |
| 02:47.29 | MrTelephone | maybe it is... |
| 02:47.54 | MrTelephone | a company outside of town here spent 2 million on their phone system |
| 02:48.04 | MrTelephone | for a pbx with extensions |
| 02:48.18 | MrTelephone | I couldn't beleive it |
| 02:48.19 | bkw_ | and i'll bet they'll end up loosing more in sales and man time trying to make asterisk work in a large scale env. |
| 02:48.29 | bkw_ | you do NOT bet the farm on asterisk |
| 02:48.37 | bkw_ | you DO NOT run critical business thru it |
| 02:48.40 | bkw_ | NEVER depend on it. |
| 02:48.42 | bkw_ | thats it. |
| 02:48.51 | bkw_ | bottom line is 99% of the time it works greaet |
| 02:48.53 | bkw_ | but that 1% |
| 02:48.53 | MrTelephone | how about freeswitch? |
| 02:48.58 | bkw_ | will get you killed |
| 02:49.02 | [TK]D-Fender | :D |
| 02:49.07 | MrTelephone | freeswitch is 100%? |
| 02:49.17 | bkw_ | MrTelephone, its getting there rapidly |
| 02:49.21 | MrTelephone | your biased as your developing a competitive piece of software |
| 02:49.26 | MrTelephone | haha |
| 02:49.29 | bkw_ | MrTelephone, na |
| 02:49.38 | bkw_ | You'll see in time. |
| 02:49.55 | bkw_ | I don't expect anyone to trust what I say.. but in time most people that have to dive in feel the same way. |
| 02:50.15 | bkw_ | Asterisk is a Journey |
| 02:50.20 | MrTelephone | if you have a solid working version of asterisk that works for a year without problems how can it fail? |
| 02:50.25 | [TK]D-Fender | bkw_ : Is it anywhere near ready for public consumption at this point? |
| 02:50.38 | bkw_ | [TK]D-Fender, we do have people using it in production |
| 02:50.44 | bkw_ | MrTelephone, how many calls? |
| 02:50.59 | bkw_ | how many calls per second? |
| 02:51.00 | MrTelephone | very small applications |
| 02:51.19 | MrTelephone | 20 calls per day |
| 02:51.20 | bkw_ | see it fits well in those niche tasks... let me fire my torture test at it and it will FALL over. |
| 02:51.29 | MrTelephone | hahaha |
| 02:51.33 | bkw_ | I can bet money on it |
| 02:51.48 | [TK]D-Fender | bkw_ : And some people use * in production :) |
| 02:51.57 | MrTelephone | it must be able to handle calls |
| 02:52.03 | MrTelephone | people use it for call centers and etc |
| 02:52.22 | bkw_ | [TK]D-Fender, yes but can you shoot 1600 calls @ 800 calls persecond at Asterisk? |
| 02:52.53 | MrTelephone | iptel.org sip proxy server |
| 02:53.01 | bkw_ | yes but that doesn't handle the media |
| 02:53.05 | bkw_ | we do |
| 02:53.24 | MrTelephone | bkw, can you incorporate NCS 1.0? |
| 02:53.34 | [TK]D-Fender | bkw_ : Probably, but the exit wound might be nasty ;) |
| 02:54.02 | *** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 02:54.13 | *** join/#asterisk t (n=t@port-212-202-198-94.dynamic.qsc.de) |
| 02:54.48 | MrTelephone | damn I need to register so I can talk privately |
| 02:54.55 | bkw_ | haha |
| 02:55.03 | bkw_ | [TK]D-Fender, exit wond? in what respect? |
| 02:55.37 | MrTelephone | I can't beleive asterisk doesn't have full mgcp support yet |
| 02:55.38 | MrTelephone | :( |
| 02:55.50 | [TK]D-Fender | MrTelephone : Go write it and contribute! |
| 02:56.51 | MrTelephone | I'm working on it |
| 02:56.58 | bkw_ | [TK]D-Fender, its not easy |
| 02:56.59 | [TK]D-Fender | bkw_ : With respect to your fully-automatic call-cannon :D |
| 02:57.28 | bkw_ | haha true |
| 02:58.00 | MrTelephone | like how do you jump into development when your project is that big |
| 02:58.02 | MrTelephone | jeez |
| 02:58.18 | MrTelephone | im looking all over for the asterisk static vars |
| 02:58.26 | MrTelephone | and pointers |
| 03:05.56 | MrTelephone | freeswitch is going to handle 1000 calls per second? |
| 03:08.50 | kph100 | does 'announce' field in queues.conf refer to a sound file? |
| 03:10.08 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-29-253.houston.res.rr.com) |
| 03:15.53 | *** join/#asterisk yassine_ (n=yassine@xdsl-87-78-20-194.netcologne.de) |
| 03:17.36 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 03:29.34 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 03:32.37 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 03:35.34 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 03:37.54 | *** join/#asterisk brookshire (i=mbrooks@65.172.243.127) |
| 03:38.29 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 03:41.35 | icyfire0573 | Does anyone know how to make the "Linksys SPA941" hangup after the other party hangs up ? |
| 03:45.52 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 03:50.19 | icyfire0573 | sbingner is being flippant :-); My grandstream and polycom phones gracefully hangup when the call is terminated. However the SPA just starts making that bad dialtone sound and its really annoying because I have to go over to thephone and hang it up. |
| 03:53.19 | *** join/#asterisk itguru (n=guru@host86-147-3-247.range86-147.btcentralplus.com) |
| 03:54.42 | itguru | Hi guys - i'm an asterisk virgin - 2 things - i got 2 hours to lose it, and 3 simple tasks - can you help me? |
| 03:55.06 | [TK]D-Fender | icyfire0573 : Kinda stupid that way. They still treat their "hardphones" like ATA w/ analog phones attached.... |
| 03:55.26 | [TK]D-Fender | itguru : Ask a specific question and you might get a specific answer |
| 03:55.54 | icyfire0573 | Yea, the only thing that I REALLY like about the linksys is the solid buttons. |
| 03:56.23 | icyfire0573 | It dosen't even have a backlight and it costs more than any of the grandstreams. |
| 03:56.32 | icyfire0573 | How are the Polycoms with backlights? |
| 03:57.08 | itguru | [TK]D-Fender-> Tru, very true, okay, I got a fresh box, with debian installed, i just installed asterisk on it. I have three SIP accounts, which normally hook up to software in XP in order to recieve calls. I want those calls to be routed to my asterisk box, and have my software phone connect to my box to recieve calls |
| 03:57.10 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 03:57.26 | sbingner | isn't the bad dialtone sound a feature? |
| 03:57.43 | *** part/#asterisk hyphen (n=hyphen@c-71-224-213-97.hsd1.pa.comcast.net) |
| 03:57.46 | sbingner | lets you know the phone isn't on the hook but it's not making a call? :b |
| 03:58.11 | [TK]D-Fender | icyfire0573 : Only the IP 650 will have one so far and very overpriced. The SPA-942 has a backlight (nice I hear), but the SPA lose on functionality big time. If you seriously need a backlight look for an Aastra 480i |
| 03:58.15 | sbingner | or does this happen when it's in speakerphone mode or something |
| 03:58.38 | [TK]D-Fender | itguru : Ok, you have our complete permission to do so :) |
| 03:58.49 | icyfire0573 | speakerphone mode. |
| 03:59.03 | [TK]D-Fender | sbingner : No, its nags you like a joe-blow phone at the telco. |
| 03:59.14 | icyfire0573 | yea, exactly [TK]D-Fender |
| 03:59.30 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 03:59.38 | itguru | [TK]D-Fender-> Okay, I see you have a sense of humour!! Okay, okay, I've play fair - and get my hands dirty myself! |
| 03:59.50 | itguru | I've = I'll |
| 03:59.55 | icyfire0573 | I don't need a backlight enough to spend 250$ on it. Its just something I would really like. But I'm poor and its only for my house so I can deal. |
| 04:00.04 | itguru | And when I have a specific question, I'll jump in here! |
| 04:00.04 | itguru | lol |
| 04:00.52 | icyfire0573 | The AASTRA phones look a lot like the Altigen phones we have at work. |
| 04:01.35 | itguru | [TK]D-Fender-> Okay, the question I should ask first - how do I configure * to connect to my SIP providers? |
| 04:01.52 | itguru | [TK]D-Fender-> Or should I say, tell me where to look, so I can figure it out myself :) |
| 04:02.14 | rob0 | sip.conf pages at the wiki |
| 04:02.30 | rob0 | (and sip.conf itself) |
| 04:04.32 | itguru | rob0-> That was to me... right? |
| 04:05.48 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 04:06.36 | rob0 | "vi /etc/asterisk/sip.conf" and soon enough it should be apparent. :) s/vi/$SOME_NICER_EDITOR/ |
| 04:07.06 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 04:07.16 | rob0 | Be sure to clean up the keyboard later. |
| 04:08.55 | itguru | This is gonna hurt .. I think i need more lube! |
| 04:13.39 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:21.22 | [TK]D-Fender | itguru : ... |
| 04:21.24 | [TK]D-Fender | ~book |
| 04:21.36 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:22.02 | [TK]D-Fender | icyfire0573 : You cen get the 480i for about $175 US |
| 04:22.32 | [TK]D-Fender | icyfire0573 : The Polycom IP 501 is a very nice phone and also goes for about $170. |
| 04:22.44 | *** join/#asterisk mrc3_ (n=mrc3@189.157.116.253) |
| 04:24.24 | icyfire0573 | Alright, I only did a quick google to find that first price. |
| 04:24.46 | *** join/#asterisk Meins (n=Meins@M797P013.adsl.highway.telekom.at) |
| 04:25.04 | Meins | Hello! |
| 04:27.12 | mrc3_ | hi all! asterisk keeps my x100p off-hook after using the zap channel for anything. where should i be looking for errors? |
| 04:32.01 | mrc3_ | i would say it is not hanging up correctly, because i see "Hungup 'Zap/1-1'" on the console, but line stays off-hook |
| 04:33.38 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 04:42.24 | itguru | Does DeStar have a defualt username and password? |
| 04:43.39 | [TK]D-Fender | itguru : you may want to try their home page or related groups. This isnt a place for * GUI support. |
| 04:44.02 | itguru | [TK]D-Fender-> thier pages give no info on that matter |
| 04:47.53 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 04:47.54 | DavoFrom818 | hi |
| 04:48.20 | DavoFrom818 | can someone help me, my on hold music sound is so low how can i increase the sound volume on the on hold? |
| 04:49.03 | icyfire0573 | where is the music on hold coming from? |
| 04:49.10 | DavoFrom818 | from the pbx |
| 04:49.28 | icyfire0573 | Thats great. Line In? MP3 directory? |
| 04:49.31 | DavoFrom818 | yes |
| 04:49.34 | DavoFrom818 | MP3 |
| 04:49.48 | icyfire0573 | how do the mp3s sound at normal volume when you play them to yourself? |
| 04:49.55 | DavoFrom818 | good |
| 04:49.59 | DavoFrom818 | nice and loud |
| 04:50.08 | icyfire0573 | are you sure thats not your volume settings? |
| 04:50.33 | icyfire0573 | what are you using for you config? |
| 04:50.34 | DavoFrom818 | voldume settings in config? |
| 04:50.44 | icyfire0573 | quitemp3, mp3 mp3nb quietmp3nb ? |
| 04:51.17 | DavoFrom818 | where is that supposed to be? |
| 04:51.29 | icyfire0573 | mode= |
| 04:51.29 | DavoFrom818 | all i have is mode and directory |
| 04:51.45 | DavoFrom818 | mode=files |
| 04:51.50 | icyfire0573 | alright, thats fine |
| 04:51.56 | DavoFrom818 | ok |
| 04:52.45 | icyfire0573 | Thats all I have here. (except for random=yes ) |
| 04:53.00 | DavoFrom818 | no random is not on |
| 04:53.18 | brookshire | try changing mode=files to mode=loud in musiconhold.conf |
| 04:53.18 | icyfire0573 | its not necessary, thats just what I happen to have. |
| 04:53.24 | DavoFrom818 | so what do i do |
| 04:54.33 | icyfire0573 | try brookshire's advice and use the mode=mp3 to see if it will play back louder |
| 04:54.55 | icyfire0573 | other options are to use SOX to convert to GSM or some other format and amplify the sound during the conversion |
| 04:55.04 | *** join/#asterisk NormanASD_ (n=norman@206.135.58.98) |
| 04:55.17 | icyfire0573 | I'm off to bed now. Good Luck. |
| 04:57.43 | *** part/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 04:59.44 | *** join/#asterisk DrAk0SX (n=luisjose@190.38.151.37) |
| 04:59.51 | sloth | is ${CALLERID(num)} available from within a macro? |
| 04:59.57 | itguru | okay, i think I'm getting someone where. If I have an external SIP account, I can get asterisk to connect to it by adding "register => username:password@sipprovideripaddress " to sip.conf - correct? |
| 05:02.08 | brookshire | sloth: i believe it should be.. if not.. you can pass it |
| 05:02.23 | sloth | thnx, ill try it now. |
| 05:02.54 | brookshire | itguru: connect in what way? |
| 05:03.20 | brookshire | there are there different ways to setup sip |
| 05:03.44 | brookshire | incoming, outgoing, and both |
| 05:04.00 | itguru | brookshire-> well, basically, I'm an Aserisk virgin - so, I have to learn this REALLY quick! |
| 05:04.11 | itguru | brookshire-> It's an incoming phone number |
| 05:04.14 | brookshire | is this for a sip phone? |
| 05:04.51 | [TK]D-Fender | itguru : If you're in that big a hurry, hire a consultant to do it for you. otherwise start with THE BOOK, and then move on to the WIKI |
| 05:04.53 | [TK]D-Fender | ~book |
| 05:04.59 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:04.59 | [TK]D-Fender | ~wikis |
| 05:05.01 | jbot | from memory, wikis is http://www.voip-info.org |
| 05:06.10 | itguru | [TK]D-Fender-> I have 24 hours to install a working asterisk box - I found out @ 2.30 AM GMT, as in london time, I'm in london, and I don't have the time to hire consultants, besides, I love a challenge |
| 05:06.52 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 05:07.40 | [TK]D-Fender | itguru : Why such a tight deadline, and why is it you think you can learn faster yourself then letting someone who knows what they're doing do it from start to finish? |
| 05:08.18 | brookshire | itguru: i never use register => for sip |
| 05:08.35 | brookshire | basically.. you need to create a context named with your username |
| 05:08.41 | brookshire | [username] |
| 05:09.03 | [TK]D-Fender | brookshire : Guess you only deal with nice ITSP's that allow fixed IP's :) |
| 05:09.37 | [TK]D-Fender | brookshire : Only ran into one like that myself. Freaked me out at first. |
| 05:09.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 05:09.57 | brookshire | nah.. you just set it up as friend.. and it just works :) |
| 05:10.11 | brookshire | usually |
| 05:10.23 | itguru | [TK]D-Fender-> i don't decide my deadlines, I get the phone call, I get to work - the info I have is sketchy @ the moment, I do know this is pro bono |
| 05:10.42 | itguru | and that it's important, and as a favour |
| 05:10.54 | itguru | but trust me, I am open to all recommendadtions |
| 05:11.12 | [TK]D-Fender | itguru : If you can't treat it as such and take the proper time then something very wrong with this plan. |
| 05:11.13 | brookshire | itguru: so is for a phone or is this from an itsp? |
| 05:11.45 | [TK]D-Fender | brookshire : He's talking about setting up * as a PBX between a few ITSP's and his soft-phone |
| 05:12.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:13.40 | itguru | [TK]D-Fender-> I could approach this like a conventional contract, but I don't have the time right about now. I need something that can be demonstrated at about 11am, which is about 6 hours away, and then after that, I can sit down and implement it properly |
| 05:14.36 | [TK]D-Fender | itguru : This whole setup will take a lot more than you have time for. Either hire someone to do it for you or pick a dumbed-down GUI'd up "ready-to-eat-then-rejugitate-faster-than-nicole-ritchie" solution like Trixbox. then after you're done with that and have satisfied people and time to actually LEARN something come back and be ready to learn from scratch. |
| 05:14.40 | itguru | but for now - I can understand that my approach is wrong, but not everyone has the luxuary of time on thier hands.... right now, I'm in one of those *SHIT happens* scenarios! :) |
| 05:15.02 | *** join/#asterisk sphilp (n=sphilp@c-71-205-146-117.hsd1.mi.comcast.net) |
| 05:15.17 | [TK]D-Fender | itguru : Sorry, but learning to do it from scratch will take a LOT longer than that. Too many config files to consider. |
| 05:15.20 | brookshire | yeah.. asterisk is not easy to learn in one day |
| 05:15.43 | brookshire | i think it took me a solid week of playing around with it.. just to get my first phone call through it |
| 05:15.49 | [TK]D-Fender | itguru : Why would you be busting your hump pro-bono on this? |
| 05:16.06 | *** join/#asterisk NormanASD (n=norman@206.135.58.98) |
| 05:16.07 | brookshire | if you want something immediately working.. try something like asteriskNow or trixbox |
| 05:16.08 | itguru | [TK]D-Fender-> I would love to do that, and as a matter of fact, I have those ISO images, right here, the box I'm working on, isn't right next to me, it's 50 miles away, on site, with a basic debian install on it, and I'm connected to it via SSH - otherwise, I would have gone with that soultion |
| 05:16.55 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 05:16.56 | [TK]D-Fender | itguru : EVEN BETTER! Yay for you! Here's a paddle, cause you've got a LOT of creek to backtrace on now... |
| 05:17.09 | itguru | [TK]D-Fender-> I have no idea! lol ! I guess it's because I've been telling myself for months to learn how to use this thing, and now that I've been thrown in the deep end, I'm trying to swim |
| 05:18.18 | itguru | But trust me, I'll get it working, if only to prove my boss wrong! and besides, my GF was already super pissed of with me, so bedtime wasn't really that much fun |
| 05:18.54 | itguru | dammit!! just had a brain wave |
| 05:18.55 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 05:19.10 | itguru | Thanks [TK]D-Fender you gave me an idea |
| 05:19.30 | [TK]D-Fender | itguru : Try crack. That outta straighten that little "blip" back to a nice comfortable flat-line ;) |
| 05:19.43 | itguru | lol |
| 05:20.51 | puzzled | morning |
| 05:21.05 | blitzrage | evening |
| 05:21.54 | [TK]D-Fender | blitzrage : I don't want to meet your mom! |
| 05:22.00 | [TK]D-Fender | *darn* |
| 05:22.11 | [TK]D-Fender | another good punch-line GONE! |
| 05:23.10 | DrAk0SX | Nov 30 01:17:14 WARNING[22997]: chan_sip.c:9845 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 05:23.19 | DrAk0SX | this error means? |
| 05:23.39 | file | [TK]D-Fender: I just want |
| 05:23.48 | puzzled | DrAk0SX: upgrade :) |
| 05:24.07 | [TK]D-Fender | file : ! ! ! |
| 05:24.14 | [TK]D-Fender | file : Safe at first! |
| 05:24.34 | DrAk0SX | puzzled, im on 1.2.13 |
| 05:24.40 | file | [TK]D-Fender: :D |
| 05:25.10 | itguru | [TK]D-Fender-> Okay, this is my new plan, inspired by you - okay, I'm firing up a virtual machine right now, and I'm going to do a trixbox install, and get it working all nice - THEN, burn the sucker to DVD, take my DVD, lappy, and various live CD's I have on site - ghost the VM over to the real machine, boot it up, check it works, which it should, and sigh a huge fat breath of relief, and laugh at my boss |
| 05:25.50 | [TK]D-Fender | ~glwaot |
| 05:25.53 | DrAk0SX | I don't know if what im trying to do is correct, I have a SIP account on that server and I want to register my account to my asterisk server so I can managed the incomming and outgoing calls from my asterisk. |
| 05:26.18 | puzzled | DrAk0SX: have you searched google for the error? |
| 05:26.40 | DrAk0SX | puzzled, but is possible what i want to do? |
| 05:27.01 | puzzled | DrAk0SX: yes |
| 05:27.08 | [TK]D-Fender | itguru : Ok well good luc, there will still be a learning curnce to learn its way of work, plus the actualy manipulation time, but its more realistic than learning from scratch, though slower that hiring a consultant. |
| 05:27.31 | puzzled | installing Trixbox is one thing. Configuring it is a whole different story |
| 05:27.40 | [TK]D-Fender | itguru : "Good Luck With All Of That". You've got an interesting challenge, you may need it. |
| 05:28.01 | itguru | puzzled-> thanks for the support :) |
| 05:28.20 | file | [TK]D-Fender: sleep? good idea |
| 05:28.20 | puzzled | itguru: just telling you how I experienced that beast |
| 05:28.47 | puzzled | file: I thought you abolished sleep?! |
| 05:29.08 | itguru | puzzled-> I know, i'm sorry, i'm really jacked up on coffee, and my GF is pissed of with me, and I have about 6 hours to pull a miracle out my ass |
| 05:30.02 | puzzled | itguru: good luck :) |
| 05:30.17 | [TK]D-Fender | file : +/- 30 mins |
| 05:30.24 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 05:30.28 | BSDTech | eveing |
| 05:30.47 | BSDTech | where can I svn all the current sounds |
| 05:30.48 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 05:30.54 | BSDTech | I cant find a svn link |
| 05:31.05 | itguru | And then, once miracle is done, need to buy girlfriend a shiny thing, with price tag of a small second hand family car, continue to laugh at boss because of the miracle, and also make sure miracle doesn't fall apart on first day :) |
| 05:31.14 | *** join/#asterisk Meins_ (n=Meins@M782P023.adsl.highway.telekom.at) |
| 05:31.19 | puzzled | BSDTech: i think the developer section on asterisk.org tells you where to find them |
| 05:32.39 | BSDTech | nope not seeing it |
| 05:32.56 | CunningPike | ~wglwat |
| 05:32.58 | jbot | i guess wglwat is well, good luck with all that |
| 05:32.58 | *** part/#asterisk sphilp (n=sphilp@c-71-205-146-117.hsd1.mi.comcast.net) |
| 05:33.02 | CunningPike | ;) |
| 05:33.11 | DrAk0SX | Failed to authenticate on INVITE to '"Luis Jose Da Silva" |
| 05:33.13 | DrAk0SX | hmm |
| 05:33.43 | itguru | Okay, guys, i'm building my plan of attack - first round of configuration will be me configuring two softphones, on two computers, to call each other, via asterisk - that should tell me everything is working, right? |
| 05:33.43 | [TK]D-Fender | CunningPike : Thanks for the reminder ;) |
| 05:33.48 | CunningPike | :D |
| 05:34.27 | itguru | then, I plan on configuring an external SIP account to connect to one of the softphones, to make sure, outside calls can make it into my box |
| 05:34.34 | [TK]D-Fender | itguru : at least that part. You'll have ALL sorts of little things to figure out hard & fast. |
| 05:34.47 | BSDTech | I need all the current sounds |
| 05:35.10 | BSDTech | 1.2.1 tar is old I know more sounds have been added |
| 05:35.23 | BSDTech | but not finding the svn link |
| 05:35.24 | puzzled | then just get trunk |
| 05:35.39 | BSDTech | there is no trunk for sounds |
| 05:35.43 | itguru | i have my energy drinks, and my iPod - I'M READY!!!! |
| 05:35.54 | BSDTech | there is no sounds src dir on svn |
| 05:36.04 | puzzled | ah ok |
| 05:36.07 | BSDTech | that I can find |
| 05:36.42 | BSDTech | thats why I am asking |
| 05:37.02 | Un1x | w0ah i didn't even know Blueray Burners are already for sale |
| 05:37.02 | Un1x | lol |
| 05:37.04 | Un1x | im buying one :D |
| 05:37.58 | file | http://ftp.digium.com/pub/telephony/sounds/ |
| 05:38.32 | [TK]D-Fender | Un1x : AND ADDORABLE PRICED TOO! (ymmv) |
| 05:39.37 | BSDTech | ok thanks |
| 05:40.12 | BSDTech | I have everythign but freepbx upand running |
| 05:40.21 | BSDTech | and tomarrow I work on it |
| 05:40.31 | BSDTech | then I have a rocking system |
| 05:40.40 | itguru | blueray burner?! |
| 05:42.27 | [TK]D-Fender | itguru : Back to work slacker! |
| 05:43.30 | BSDTech | sorry correction I also have to patch asterisk-addons-1.4 for bsd |
| 05:43.37 | BSDTech | then I will have a rocking system |
| 05:43.49 | Un1x | frepbx sucks |
| 05:44.01 | BSDTech | 2.2 rocks |
| 05:44.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:44.07 | BSDTech | its much better |
| 05:44.09 | Un1x | 2.2? |
| 05:44.18 | Un1x | err oh freepbx |
| 05:44.22 | BSDTech | yes they are now on 2.2 rc1 |
| 05:44.31 | Un1x | heh i hatre it man if you want a GUI soooo bad why not go with Ast-GUI |
| 05:44.32 | BSDTech | yes I am porting it |
| 05:44.55 | Un1x | maybe i should startup a bounty for the BSDdev team |
| 05:44.58 | BSDTech | astgui is 2 years behind freepbx and lacks sql |
| 05:45.11 | Un1x | first one to port a stable version of zaptel for freebsd gets certtain ammount of $$ |
| 05:45.22 | Un1x | maybe that owuld incourage some devs to do it so i can get myass off slackware :P |
| 05:45.32 | Un1x | heh |
| 05:45.36 | BSDTech | its already in the works and there is a current zaptel on bsd its in the ports |
| 05:45.50 | BSDTech | they are working on 1.4 zaptel now |
| 05:45.55 | BSDTech | to late |
| 05:46.05 | BSDTech | there is a asterisk-bsd dev team |
| 05:46.11 | Un1x | panasonic makes them for about 700$ itguru |
| 05:46.12 | BSDTech | and a mailing list |
| 05:46.24 | [TK]D-Fender | ok, bedtime for Bonzo here... |
| 05:46.27 | Un1x | bsdtech asterisk already runs on BSD perfectly.. |
| 05:46.29 | Un1x | im talking zaptel |
| 05:46.52 | BSDTech | zaptel is in the ports |
| 05:47.04 | BSDTech | and they are working to port the 1.4 zaptel now |
| 05:47.21 | BSDTech | /usr/ports/misc/zaptel |
| 05:47.29 | BSDTech | been aroiund awhile |
| 05:47.36 | BSDTech | we are upto 1.0.1 |
| 05:47.38 | Un1x | Yes, but its not stable man learn to read |
| 05:47.39 | BSDTech | on it |
| 05:47.40 | Un1x | its still beta |
| 05:47.51 | BSDTech | no its outa beta |
| 05:48.03 | Un1x | since when because i just looked at pages few days ago |
| 05:48.06 | Un1x | and it said still beta |
| 05:48.10 | BSDTech | I have 4 rhino cards on bsd and the drivers work fine |
| 05:48.14 | Un1x | and everyone responded its not stable,,, and it crashes |
| 05:48.21 | Un1x | not for the TDM400P |
| 05:48.27 | BSDTech | the page has not been updated in so long |
| 05:48.50 | BSDTech | get the svn |
| 05:49.01 | Un1x | I see. |
| 05:49.12 | Un1x | oh well i already installed asterisk + zaptel on slackware now... |
| 05:49.18 | Un1x | on the 26 kernel |
| 05:49.27 | Un1x | no point removing it after all that work last night just to switch to bsd |
| 05:49.28 | BSDTech | I will get access this next week and update the page |
| 05:49.46 | Un1x | :O |
| 05:50.03 | BSDTech | I am doing a slack box with the bsd layout |
| 05:50.09 | Un1x | oh well still ive already done kernel on this one not going to waste another few hours switch to bsd make custom kernel include dual core support etc |
| 05:50.16 | Un1x | when its already been done :P |
| 05:50.16 | BSDTech | but my plan is to doop trixbox on bsd |
| 05:50.23 | Un1x | screw trixbox :P |
| 05:50.38 | BSDTech | no there are some new tools that are great |
| 05:50.43 | Un1x | wow BD burners are slow |
| 05:50.47 | Un1x | burning speed = 2x |
| 05:50.49 | BSDTech | beta2 of 2.0 changes things |
| 05:50.55 | Un1x | # 2X BD-R, BD-RE, BD-R DL, BD-RE DL |
| 05:50.55 | Un1x | # 8X DVD+R |
| 05:50.55 | Un1x | # 8X DVD±RW |
| 05:50.55 | Un1x | # 6X DVD-RW |
| 05:50.55 | Un1x | # 5X DVD-RAM |
| 05:51.27 | BSDTech | well back to patching and porting |
| 05:51.45 | BSDTech | thams for pointing to the sounds I can make a sounds port now also |
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| 05:55.32 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 06:00.05 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 06:00.20 | niZon | anyone setup the contact list on a polycom IP301? |
| 06:00.33 | *** part/#asterisk Meins_ (n=Meins@M782P023.adsl.highway.telekom.at) |
| 06:02.41 | CunningPike | niZon: What's your question? |
| 06:04.24 | niZon | just wondering how to get the directory stuff to actually save when entering it on the phone |
| 06:04.34 | niZon | I can go through and create a contact, but it won't save |
| 06:09.46 | *** join/#asterisk stephane (n=stephane@nostromo.cabale.net) |
| 06:10.47 | stephane | jour |
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| 06:14.05 | brookshire | nizon: see provisioning :) |
| 06:14.15 | niZon | hmm |
| 06:14.18 | brookshire | directory.xml |
| 06:14.26 | brookshire | enable updates |
| 06:14.36 | brookshire | the polycom will save on reboot |
| 06:14.58 | brookshire | enable updates on the server.. i mean |
| 06:15.07 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 06:15.08 | wasim | yusuf is 6 runs short of viv richards all time most runs in a calendar year record :) |
| 06:15.10 | brookshire | so.. like.. setup a public read/writeable ftp |
| 06:15.18 | brookshire | with your configs |
| 06:15.38 | kph100 | each time a new file is added to musiconhold directory, asterisk needs to be reloaded.... is there a way around this? |
| 06:16.04 | niZon | yeah i have the ftp (well, tftp, i'm lazy) |
| 06:16.05 | kph100 | if a wav file is removed from the dir, then musiconhold app fails. |
| 06:16.09 | niZon | and it gets the configs |
| 06:16.13 | shellshark | kph100: add a cron script to reload asterisk once an hour? |
| 06:16.13 | niZon | just trying to find that setting |
| 06:16.26 | brookshire | tftp is not the same as ftp but it should work the same |
| 06:17.01 | kph100 | shellshark-- any other ways? |
| 06:17.19 | *** join/#asterisk hfb (n=hfb@adsl-75-4-22-136.dsl.irvnca.sbcglobal.net) |
| 06:17.22 | brookshire | kph: stream your music on hold? |
| 06:17.54 | shellshark | kph100: you could always write a shell script to do adding/removing of MOH files... it would copy the file to the correct directory, as well as reload res_musiconhold.so |
| 06:18.12 | shellshark | that would probably be a bit more effective |
| 06:18.20 | brookshire | shellshark: or write a script that monitors for changes |
| 06:18.40 | shellshark | brookshire: but that's using excess CPU |
| 06:18.52 | kph100 | a delete of one musiconhold file needs a asterisk reload? |
| 06:19.14 | shellshark | brookshire: as you'd need to write a daemon that periodically probed for added or deleted files, then take action |
| 06:19.31 | shellshark | brookshire: if you wrote a script to add / delete files for you, there would be a lot less overhead |
| 06:19.48 | kph100 | thats really not good. |
| 06:20.16 | shellshark | ah yes, streaming is also a very effective solution |
| 06:20.30 | kph100 | streaming from where? |
| 06:21.38 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:21.48 | shellshark | local machine |
| 06:21.54 | shellshark | setup icecast or something |
| 06:22.03 | brookshire | kph100: have you tried native music on hold? |
| 06:22.15 | kph100 | native/ no, |
| 06:22.27 | brookshire | it's another music on hold engine |
| 06:22.54 | kph100 | needs reload ? |
| 06:22.59 | brookshire | i have no idea |
| 06:23.02 | brookshire | worth a try? |
| 06:23.10 | shellshark | yeah it does need a reload |
| 06:23.15 | shellshark | i use native here |
| 06:23.22 | clyrrad | if you change moh engine you definaly need to reload |
| 06:23.28 | shellshark | native and files are the same thing |
| 06:23.43 | *** join/#asterisk apardo (n=apardo@87.217.145.139) |
| 06:23.46 | clyrrad | native works well for me |
| 06:23.48 | shellshark | clyrrad: your statement is not relative to our conversation ;) |
| 06:24.18 | brookshire | you could always submit a patch :) |
| 06:24.46 | kph100 | musiconhold.conf should be made realtime. |
| 06:24.57 | brookshire | why? |
| 06:25.01 | brookshire | what purpose? |
| 06:25.45 | shellshark | realtime musiconhold.conf would not solve the problem |
| 06:26.09 | *** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
| 06:26.16 | kph100 | instead of defining classes. actual path to audio file would be stored in db table. |
| 06:27.38 | wasim | congrats yusuf! |
| 06:28.12 | wasim | 1712 runs at an average higher than bradmans! |
| 06:28.21 | Qwell | kph100: We're working on something that would make that possible |
| 06:28.39 | Qwell | oh, wait, nm |
| 06:28.46 | Qwell | just the path to the file? |
| 06:29.17 | Ambrose | Can anyone recommend a good external FXO ? |
| 06:29.19 | brookshire | qwell: work on stuff! |
| 06:29.30 | kph100 | the playback cmd uses filename as parameter. |
| 06:29.45 | Qwell | brookshire: tired |
| 06:29.52 | brookshire | lame! |
| 06:29.56 | Qwell | quite |
| 06:30.15 | brookshire | qwell: find me a fedora core whatever box to play on :) |
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| 06:32.06 | Shaun2222 | i there anyway at all to tell how a dial() was awnswered.. |
| 06:32.32 | Shaun2222 | for example i have a macro that i use but i want the macro to stop if it detects that it's in voicemail and wasnt answered by a person |
| 06:33.17 | wasim | yeah, you want NVFaxDetect or something of that sort |
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| 07:06.06 | Zefk | Hi, I'm running asterisk 1.2.13 on FC5 with B410P board. I use only port 1 of the board. Inbound and outbound calls via BRI port is working. I have the following message when asterisk starts: mISDN dss1 fromup without proc pr=10180 dinfo(0). Could be anything wrong? Thx. |
| 07:06.06 | jeff1 | test |
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| 07:12.52 | j0 | is there any program for viewing sip traces? even something with syntax highlighting would be nice |
| 07:16.09 | jeff1 | try wireshark at http://www.wireshark.org |
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| 07:26.29 | j0 | jeff1: is there a way to do remote captures of a linux box with it? i know its possible with windows |
| 07:26.42 | j0 | i've been coyping tcpdump logs over and then viewing in wireshark, but it's time consuming |
| 07:27.00 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:27.34 | j0 | arg.. here we go again.. asterisk just killed my box on a reload |
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| 07:35.32 | parag_ast | Hi, If i want that a asterisk trunk should keep registering after every 1 min to the remote asterisk server then what do i need to set in my iax.conf ?? |
| 07:35.47 | parag_ast | is there any registry time out field is there |
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| 08:04.53 | DavoFrom818 | hi |
| 08:05.06 | DavoFrom818 | anyone here use sunrocket? |
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| 08:46.42 | mesfet | Hi! A question regarding SIP protocol: can asterisk send to the SIP phone a line of information to be displayed? I mean information displayed during dialplan parsing. |
| 08:47.33 | mesfet | Information like "Transfer call to blablabla" or the name of the called number, .... |
| 08:47.50 | dlynes_laptop | mesfet: yes, but it depends on whether your phone supports it or not, as to whether anythign will happen with it |
| 08:48.03 | dlynes_laptop | mesfet: you would use the sip header functions |
| 08:48.14 | mesfet | dlynes_laptop: Very good! |
| 08:48.28 | shellshark | dlynes_laptop: would a polycom phone support such things? |
| 08:48.36 | dlynes_laptop | mesfet: Aastra phones support it; I think Polycom and Cisco also support it |
| 08:48.55 | dlynes_laptop | mesfet: but don't quote me on the polycom and cisco...it's just an educated guess |
| 08:49.02 | shellshark | interesting |
| 08:49.25 | shellshark | could you make the phone display something without having to call the phone or have the phone be on an active call? |
| 08:49.47 | shellshark | send an INVITE to it, for example, with just an arbitrary line of text? |
| 08:50.05 | tm | Hiho! |
| 08:50.10 | tm | Spricht hier wer deutsch? :-) |
| 08:50.44 | mesfet | dlynes_laptop: please could you tell me which is the SIP header to be set to display something? |
| 08:50.48 | shellshark | tm: mich deutsch ist schisse ;) |
| 08:50.54 | mesfet | SIPAddHeader(what????) |
| 08:51.04 | shellshark | tm: spreche englisch? |
| 08:51.22 | dlynes_laptop | mesfet: it's going to be dependent on your particular phone |
| 08:51.29 | tm | shellshark: :P |
| 08:51.30 | dlynes_laptop | mesfet: sip_header is very phone-specific |
| 08:52.36 | brian | hey, just wondering if there is anyway at all to listen for DTMF on a meetme conference using fastagi? |
| 08:52.36 | dlynes_laptop | mesfet: or sipaddheader, that is |
| 08:52.54 | mesfet | Ok. |
| 08:53.10 | mesfet | I'll try to googling some information, and let you know. |
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| 08:53.19 | mesfet | dlynes_laptop: many thanks for instance. |
| 08:53.50 | dlynes_laptop | mesfet: you can try checking voip-info.org |
| 08:54.02 | dlynes_laptop | mesfet: there's one or two references on there about how to use it for certain phones |
| 08:54.48 | itguru | is there anyway to tell if my truck is connecting to my external SIP provider so that incoming calls will work? |
| 08:54.54 | itguru | *trunk |
| 08:56.10 | *** part/#asterisk Shi_Chan (i=da@61.68.31.193) |
| 08:56.12 | mesfet | itguru: sip show peers ? |
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| 08:59.18 | brian | Why doesn't meetme work with DTMF |
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| 09:01.49 | itguru | mesfet-> I don't have that command |
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| 09:09.34 | E-bola | Hey gguys quick question |
| 09:09.34 | E-bola | my sip->pstn provider no wants me to stop registering with their server |
| 09:09.34 | E-bola | and just accept incoming calls |
| 09:09.42 | E-bola | i can see they send me an invite to a certain contact like randomaccountid@my_ip |
| 09:09.43 | E-bola | how do i get asterisk to work with this kind of setup? |
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| 09:23.17 | dlynes_laptop | E-bola: to just accept connections without a registration? |
| 09:23.40 | E-bola | dlynes_laptop: yes |
| 09:23.48 | E-bola | they simply send me invite's |
| 09:23.53 | dlynes_laptop | Just set it up normally |
| 09:24.03 | E-bola | whats normal? :) |
| 09:32.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 09:32.14 | brian | Why doesn't meetme work with DTMF |
| 09:40.30 | *** join/#asterisk xnon (n=xnon@200.8.87.11) |
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| 09:56.31 | E-bola | dlynes_laptop: ? |
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| 09:56.44 | dlynes_laptop | E-bola: sorry...been kinda busy |
| 09:56.50 | dlynes_laptop | E-bola: getting a build to work on solaris |
| 09:57.01 | E-bola | no worries |
| 09:57.36 | E-bola | this may be a a rather dumb question but.... i just wanted asterisk to accept invites of a form dfgsdfhakjsdfg43545@1.2.3.4 |
| 09:57.48 | E-bola | do i need to specify it somewhere in sip.conf? |
| 09:57.59 | E-bola | cuz its apparently not enough to just have the number matched in extension.conf |
| 09:58.29 | dlynes_laptop | dfgsdfhakjsdfg43545 if I remember correctly is the username for the sip connection |
| 09:58.43 | dlynes_laptop | i.e. the value in the [...] in the sip.conf file |
| 10:00.41 | Chris-NB | hi |
| 10:01.09 | *** join/#asterisk Asttt (n=asterisk@193.110.8.51) |
| 10:02.04 | Chris-NB | my asterisk is connected via E1 to an alcatel 4400. if I try to make a call from a sipphone via alcatel out to the pstn, the alcatel does not rescieve that the sipphone when the sipphone hangs the call up |
| 10:02.37 | Chris-NB | the call is still beeing established or continuing |
| 10:02.43 | Chris-NB | anybody discovered this? |
| 10:02.48 | dlynes_laptop | E-bola: sip.conf is for incoming calls, not extension.conf |
| 10:03.33 | dlynes_laptop | Chris-NB: that's probably a setting in your zaptel.conf file, or your alcatel |
| 10:04.32 | E-bola | dlynes_laptop: well u control incoming calls with extension.conf |
| 10:04.41 | dlynes_laptop | E-bola: no you donm't |
| 10:04.54 | dlynes_laptop | E-bola: You just decide where they go with extensions.conf |
| 10:04.55 | E-bola | the dialplan? |
| 10:05.01 | dlynes_laptop | E-bola: they're not validated in extensions.conf |
| 10:05.03 | E-bola | yes thats controling what happens with them |
| 10:05.06 | E-bola | nm |
| 10:05.20 | dlynes_laptop | E-bola: they're validated in the channel driver config file |
| 10:05.29 | E-bola | just a matter of lingo |
| 10:05.39 | E-bola | but ok so i need to make entries for each contact my isp sends me? |
| 10:05.58 | dlynes_laptop | E-bola: you generally need to define one sip entry for the ISP |
| 10:06.07 | dlynes_laptop | E-bola: they'll pass you a username and password, usually |
| 10:06.19 | E-bola | i have 2 entries for each of my isp's |
| 10:06.24 | E-bola | since they loadbalance that way |
| 10:06.27 | dlynes_laptop | E-bola: ok |
| 10:06.29 | E-bola | but thats only for outgoing calls |
| 10:06.42 | dlynes_laptop | E-bola: it'll probably suffice for incoming calls, also |
| 10:06.54 | dlynes_laptop | E-bola: define the isp as a 'friend', instead of as a 'peer' |
| 10:07.00 | E-bola | --- (23 headers 13 lines) --- |
| 10:07.00 | E-bola | Ignoring this INVITE request |
| 10:07.06 | E-bola | atm its ignoring the invites form the isp... |
| 10:07.19 | dlynes_laptop | E-bola: type=friend |
| 10:07.48 | E-bola | the text in the 2 [ and ] doesnt matter? |
| 10:08.14 | dlynes_laptop | E-bola: shouldn't, no...it should be the same as whatever it currently is (assuming it's currently being used |
| 10:08.23 | E-bola | ok testing... |
| 10:08.55 | E-bola | nope no go |
| 10:09.27 | E-bola | Found no matching peer or user for '212.98.67.12:5060 |
| 10:09.46 | dlynes_laptop | E-bola: paste your sip.conf file, and scrub your passwords out of it |
| 10:09.46 | E-bola | hmm ok |
| 10:09.54 | E-bola | they dont call me from the same ip as iu call through |
| 10:10.14 | dlynes_laptop | E-bola: set up some other sip entries then |
| 10:10.30 | E-bola | in the progress of doing just that :) |
| 10:10.34 | E-bola | i guess ile skip secret |
| 10:10.38 | E-bola | and put insecure very |
| 10:11.16 | zapp-branigan | hi, when i compile the asterisk in fedora 6 give a error linux/compiler.h not found because the fedora not use now the glibc-kernheaders and use the kernel-headers, how can compile this ? |
| 10:11.55 | zapp-branigan | i how is editing the /lib/modules/`uname -r`/build/include/linux |
| 10:12.07 | zapp-branigan | autoconf.h |
| 10:12.22 | zapp-branigan | but what line i must to comment? |
| 10:14.03 | zapp-branigan | :( |
| 10:14.55 | E-bola | Found no matching peer or user for '212.98.65.12:5060' |
| 10:15.15 | E-bola | even if i have an entry in sip.conf |
| 10:15.17 | dlynes_laptop | E-bola: pastebin your sip.conf and your log file |
| 10:15.22 | E-bola | dlynes: ok |
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| 10:20.36 | brian | Why doesn't meetme work with DTMF |
| 10:21.03 | *** join/#asterisk CleanerX (n=nix@p54A397D9.dip0.t-ipconnect.de) |
| 10:21.07 | brian | i want to make my meetme conference say the call count when the user presses 8 (or whatever key) how can I accomplish this? |
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| 10:27.10 | dlynes_laptop | brian: customize the app_meetme.c code? |
| 10:29.03 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 10:29.22 | brian | exten => _X,1,AGI(agi://localhost:4575) <-- shouldn't this respond to any extension pressed in? |
| 10:29.29 | brian | (1 digit long) |
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| 10:35.33 | xnon | anybody here speak spanish!?? |
| 10:35.48 | xnon | i need any help with a problem friends |
| 10:35.59 | xnon | my english is not so good |
| 10:36.32 | xnon | i have a big problem with de hangup in my PSTN line connected to a TDM12B |
| 10:36.36 | xnon | Digium |
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| 10:40.53 | dlynes_laptop | wazza problem with the tdm12b, xnon? |
| 10:41.04 | xnon | yes friend! :( |
| 10:41.35 | xnon | with the polarity call |
| 10:41.35 | xnon | i think so |
| 10:41.49 | xnon | i was use the patch for this |
| 10:42.01 | dlynes_laptop | shouldn't need a patch |
| 10:42.02 | xnon | but my asterisk aplication is broken now |
| 10:42.09 | dlynes_laptop | xnon: which country are you in? |
| 10:42.17 | xnon | Venezuela :P |
| 10:42.28 | xnon | Venezuela Latinamerica |
| 10:42.42 | dlynes_laptop | Ok, so the phone lines there...are the analog lines the same as north america? |
| 10:42.51 | *** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com) |
| 10:42.58 | dlynes_laptop | btw |
| 10:43.07 | DrAk0SX | its a spain line |
| 10:43.07 | dlynes_laptop | venezuela's south america, not latin america :) |
| 10:43.21 | dlynes_laptop | DrAk0SX: ? |
| 10:43.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:43.54 | DrAk0SX | dlynes_laptop, the pstn line he's trying to use is a spain's line, in spain. |
| 10:44.04 | dlynes_laptop | DrAk0SX: oh |
| 10:44.25 | xnon | dlynes_laptop, DrAk0SX is a good friend! |
| 10:44.30 | dlynes_laptop | DrAk0SX: so iow, it might not have disconnect supervision then, eh? |
| 10:44.33 | DrAk0SX | dlynes_laptop, and the problem is when someone call and hangup it doesnt register the hangup even and the call stay open for minutes |
| 10:45.01 | xnon | DrAk0SX, explicale al pana que eres un buen amigo mio que trabajamos juntos pero que tu tienes la lengua inglesa y yo la japonesa con mayonesa jejeeje |
| 10:45.04 | dlynes_laptop | DrAk0SX: I think i recall that some countries in the EU don't have disconnect supervision avaialble on their lines |
| 10:45.12 | xnon | el es un pana de Canada |
| 10:45.23 | xnon | es full pana siempre me da una mano cuando puede |
| 10:45.36 | dlynes_laptop | xnon: what about Canada? |
| 10:45.53 | DrAk0SX | dlynes_laptop, he says you are from Canada |
| 10:45.57 | dlynes_laptop | ah |
| 10:46.04 | xnon | ;) |
| 10:46.06 | dlynes_laptop | i thought that's what that was, but wasn't sure |
| 10:46.19 | dlynes_laptop | He is a person from Canada (direct translation), right? |
| 10:46.20 | DrAk0SX | dlynes_laptop, he is kinda trying to introduce us hehe |
| 10:46.37 | DrAk0SX | dlynes_laptop, something like that , yes |
| 10:46.43 | dlynes_laptop | not sure what pana is, but it's a guess |
| 10:47.03 | DrAk0SX | dlynes_laptop, pana is la friend, pal. |
| 10:47.07 | dlynes_laptop | ah |
| 10:47.14 | dlynes_laptop | I thought friend was amigo? |
| 10:47.23 | DrAk0SX | dlynes_laptop, so, how we can solve that problem? |
| 10:47.34 | xnon | jejejejeeeje PANA = GOOD FRIEND |
| 10:47.41 | dlynes_laptop | He needs to find out if disconnect supervision is on the line or not |
| 10:47.50 | dlynes_laptop | And if it isn't, try to get it put on there |
| 10:47.59 | dlynes_laptop | Then make sure he's using kewlstart signalling |
| 10:48.10 | dlynes_laptop | 'ks' in asterisk-speak |
| 10:48.42 | DrAk0SX | dlynes_laptop, thats an external device? |
| 10:49.03 | dlynes_laptop | DrAk0SX: disconnect supervision? |
| 10:49.06 | DrAk0SX | yes |
| 10:49.54 | dlynes_laptop | No, it's a feature your telco can put on the line |
| 10:50.00 | Aurs | does that have anything to do with the BAD! BAD! BAD! errormsg? |
| 10:50.01 | dlynes_laptop | It's standard on north american lines |
| 10:50.12 | dlynes_laptop | but it's not standard on a lot of european lines |
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| 10:52.26 | *** join/#asterisk syzygyBSD__ (n=chatzill@poplar.matraex.com) |
| 10:52.27 | DrAk0SX | dlynes_laptop, what if they dont? |
| 10:52.39 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 10:52.43 | dlynes_laptop | DrAk0SX: ask them to put it on |
| 10:53.06 | dlynes_laptop | DrAk0SX: if they can't put it on, you're not going to be able to detect a hangup |
| 10:53.23 | dlynes_laptop | DrAk0SX: so then, you'll need to figure out some kind of kludge to handle it |
| 10:53.51 | DrAk0SX | I see |
| 10:54.24 | *** join/#asterisk KSKS (n=KLBKvTur@85.101.29.119) |
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| 10:56.40 | dlynes_laptop | DrAk0SX: do you understand everything, then? |
| 10:58.16 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 10:58.27 | DrAk0SX | I guess so |
| 10:59.09 | dlynes_laptop | You don't sound too confident :) |
| 11:00.07 | DrAk0SX | Just I don't think telefonica is going to put anything on that line :/ |
| 11:00.24 | dlynes_laptop | why not? |
| 11:00.29 | dlynes_laptop | they suck really really bad? |
| 11:01.08 | DrAk0SX | for what i've heard , yes , but... who knows xnon is going to call in few min. |
| 11:01.27 | dlynes_laptop | DrAk0SX: tell him to scream at them in Italian |
| 11:02.30 | brian | can you use something like _X in a context name? |
| 11:02.57 | dlynes_laptop | can't recall if it can begin with '_', or not, but it can certainly have '_''s in it |
| 11:03.10 | brian | I mean like...a wildcard number |
| 11:03.16 | dlynes_laptop | huh? |
| 11:03.26 | brian | I don't want to create 20 contexts that all have the same stuff in them. |
| 11:03.38 | dlynes_laptop | brian: so use contextual includes then |
| 11:03.49 | brian | contextual includes? |
| 11:04.03 | dlynes_laptop | include => thisdialplancontext |
| 11:04.10 | brian | But I still have to use 20 different sections |
| 11:04.17 | dlynes_laptop | so? |
| 11:04.22 | brian | I don't like it :P |
| 11:04.28 | dlynes_laptop | deal with it |
| 11:04.33 | dlynes_laptop | or don't create so many sections |
| 11:04.52 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-115.user.ono.com) |
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| 11:08.08 | DrAk0SX | hshs |
| 11:08.14 | DrAk0SX | will do |
| 11:08.49 | dlynes_laptop | hshs == hehe? |
| 11:09.28 | dlynes_laptop | xnon: como esta? |
| 11:10.21 | dlynes_laptop | DrAk0SX: so what country are you from? |
| 11:10.50 | xnon | yo muy bien y tu como estas? |
| 11:10.54 | Aurs | dlynes_laptop: hshs = haha, of course |
| 11:11.00 | xnon | jejejeej |
| 11:11.12 | Aurs | and jeje = hehe |
| 11:11.17 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 11:11.21 | xnon | hehehehehe |
| 11:11.23 | dlynes_laptop | muy bueno |
| 11:11.38 | dlynes_laptop | gracias senor |
| 11:11.57 | xnon | en un momento llamo a Telefonica para pedir el soporte que nos comentas |
| 11:11.59 | dlynes_laptop | Aurs: well, I knew jeje was the same as hehe |
| 11:12.01 | Aurs | que? |
| 11:12.04 | DrAk0SX | dlynes_laptop, Venezuela |
| 11:12.14 | Aurs | (quote fawlty towers) |
| 11:12.16 | dlynes_laptop | DrAk0SX: ah...you and xnon work together? |
| 11:12.27 | dlynes_laptop | Aurs: bbc sitcom |
| 11:12.30 | DrAk0SX | yup |
| 11:12.37 | xnon | yes |
| 11:12.51 | Aurs | dlynes_laptop: yep. and there is a funny spanish guy in that series |
| 11:12.53 | Aurs | hehe |
| 11:12.58 | Aurs | "que?" |
| 11:13.10 | xnon | Aurs where u from? |
| 11:13.16 | dlynes_laptop | que is french, not spanish |
| 11:13.34 | wasim | que cera cera is french? |
| 11:13.45 | dlynes_laptop | you mean que sera sera? |
| 11:13.54 | dlynes_laptop | yes |
| 11:14.03 | wasim | ah |
| 11:14.11 | dlynes_laptop | very famous song |
| 11:14.14 | Aurs | xnon: norway |
| 11:14.21 | wasim | i always thought it was spanish for some reason |
| 11:14.23 | RoyK | wasim prater bare tull likevel :) |
| 11:14.24 | dlynes_laptop | Holly Cole Trio did a great remake of it |
| 11:14.32 | xnon | ok |
| 11:14.40 | wasim | never saw the movie either |
| 11:14.41 | Aurs | so how do you type "que?" (what?) in spanish, then |
| 11:15.01 | DrAk0SX | Aurs, "qué" |
| 11:15.02 | RoyK | que |
| 11:15.04 | wasim | 'tis ok, i'm fasting today |
| 11:15.08 | RoyK | oh |
| 11:15.21 | RoyK | wasim: i thought the ramadan was over..... |
| 11:15.31 | wasim | RoyK: yeh, voluntary |
| 11:15.42 | Aurs | life in the fasting lane |
| 11:15.43 | Aurs | :P |
| 11:15.47 | RoyK | weight loss? :) |
| 11:15.49 | *** join/#asterisk leoncamel (n=leoncame@219.238.107.107) |
| 11:15.53 | wasim | RoyK: :) bingo! |
| 11:16.13 | wasim | RoyK: and also make up for 15 years of missed fasts |
| 11:16.16 | RoyK | Let Them Eat Cake |
| 11:16.34 | RoyK | wasim: by fasting a year or so? :) |
| 11:16.36 | DrAk0SX | I'm starving.... |
| 11:16.46 | E-bola | when my ISP send me an invite |
| 11:16.49 | wasim | RoyK: 3 times a week, 3 to 4 years |
| 11:16.51 | E-bola | i match it by the ip its comming from |
| 11:16.53 | xnon | DrAk0SX, crm@902 esta vacio |
| 11:16.55 | dlynes_laptop | I thought ramadan was on for another week or two yet? |
| 11:16.58 | E-bola | but my asterisk server sends back a 404 not found |
| 11:16.59 | xnon | DrAk0SX, solo le cambiare el pass |
| 11:17.03 | E-bola | what is it that isnt found? |
| 11:17.12 | wasim | dlynes_laptop: nope, ended about 4 weeks ago |
| 11:17.19 | dlynes_laptop | ah |
| 11:17.28 | dlynes_laptop | but muslims don't drink beer |
| 11:17.34 | wasim | rignes, we do! |
| 11:17.35 | dlynes_laptop | or at least they're not supposed to :) |
| 11:17.46 | RoyK | wasim: the breakfast? the time you break the fast? ringnes, though |
| 11:18.06 | Aurs | ringnes? |
| 11:18.14 | wasim | RoyK: sunrise to sunset |
| 11:18.18 | RoyK | hehe |
| 11:18.22 | RoyK | breakfast :) |
| 11:18.42 | Aurs | wasim: .no? |
| 11:18.57 | wasim | Aurs: non |
| 11:19.00 | E-bola | http://paste.uni.cc/11766 |
| 11:19.04 | E-bola | can soebody please take a look? |
| 11:19.06 | Aurs | but? ringnes? |
| 11:19.12 | E-bola | im trying to understand why its doing a 404 |
| 11:19.12 | *** join/#asterisk bluemono (n=matthewo@host-212-158-219-181.bulldogdsl.com) |
| 11:19.14 | wasim | Aurs: can't help it, the viking gave a gift |
| 11:19.35 | Aurs | RoyK, du må jo sende dahls! |
| 11:19.46 | bluemono | hello |
| 11:20.23 | bluemono | just a friendly hi, I'm just entering the world of asterisk :) |
| 11:20.44 | Aurs | hello bluemono |
| 11:20.46 | wasim | bluemono: bonjour |
| 11:21.34 | dlynes_laptop | yo quero taco bell! |
| 11:22.48 | RoyK | Aurs: Aass, kanskje. Det er visst populært på amerikanske homsebarer :) |
| 11:22.58 | dlynes_laptop | E-bola: because it's trying to go to an extension that doens't exist in your context 'Incoming' |
| 11:23.05 | RoyK | wasim == paki |
| 11:23.18 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 11:23.24 | dlynes_laptop | E-bola: Looking for K0000150333369975 in Incoming (domain 85.81.181.54) |
| 11:23.25 | dlynes_laptop | # |
| 11:23.29 | shellshark | wasallam |
| 11:23.49 | dlynes_laptop | E-bola: you don't have an extension of 'K0000150333369975' defined in your 'Incoming' dialplan context |
| 11:24.04 | E-bola | dlynes_laptop: just figured it out |
| 11:24.25 | E-bola | As a general question to everyone: Isnt it a bit insecure to let any ip send u invite's? |
| 11:26.13 | *** join/#asterisk viperdude (n=viperdud@84-45-129-190.no-dns-yet.enta.net) |
| 11:27.28 | dlynes_laptop | E-bola: depends...is your outgoing context and your incoming context the same context? |
| 11:28.12 | Aurs | RoyK: lol |
| 11:29.10 | oej | RoyK: Du bör nog lägga in patchen jag just committade till 1.2 |
| 11:29.14 | dlynes_laptop | wtf? |
| 11:29.17 | dlynes_laptop | mischan |
| 11:29.30 | oej | Talking in a secret language |
| 11:29.52 | shellshark | looks like dutch to me |
| 11:29.56 | shellshark | or norsk ;) |
| 11:30.15 | oej | Actually Swedish - to a norwegian. |
| 11:31.42 | shellshark | swedish, dutch, norsk.... they all look the same to me ;) |
| 11:32.33 | jm|work | all greek? |
| 11:32.49 | E-bola | dlynes_laptop: no they are different |
| 11:33.15 | RoyK | oej: kan jeg få? er det om det kræsjen? |
| 11:33.21 | dlynes_laptop | E-bola: shouldn't be an issue, then |
| 11:33.48 | E-bola | dlynes_laptop: alright |
| 11:33.52 | oej | RoyK: Det var något jag hittade på ett system med 1.0 idag, men också kan drabba 1.2 |
| 11:33.56 | oej | enkel patch |
| 11:33.58 | dlynes_laptop | E-bola: there's probably still security implications, but whatever security implications there might be, there's also issues with people not being able to send you calls if you tighten it up too much, too |
| 11:34.01 | oej | Ett tacken i en rad |
| 11:34.53 | E-bola | dlynes_laptop: i guess. It just seemed more secure to use register lines |
| 11:35.56 | Aurs | oej: bra at bugfixer er forbeholdt de som kan lese svensk ;) |
| 11:36.08 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 11:36.19 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 11:36.26 | oej | Aurs: gotta keep some thing secret :-) |
| 11:37.29 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 11:38.09 | Hello2007 | when i delete my voice mail using a phone, are they also physically deleted from the * server? |
| 11:39.28 | shellshark | yes |
| 11:39.30 | oej | yes, if you use the voicemailmain menus, Hello2007 |
| 11:39.46 | Hello2007 | ok, thanks |
| 11:39.49 | E-bola | dlynes_laptop: my isp just told me they will be sending me calls from a random range of ip addresses |
| 11:39.57 | E-bola | how do i make a contact to accept anything? |
| 11:40.06 | E-bola | host => dynamic? |
| 11:40.17 | dlynes_laptop | correct |
| 11:40.27 | E-bola | mmm and that doesnt depend on register lines? |
| 11:40.28 | Hello2007 | but you choose to undelete the deleted mail? |
| 11:40.28 | *** join/#asterisk lat1234 (n=lat@61.9.4.58) |
| 11:40.34 | lat1234 | hello |
| 11:40.34 | E-bola | host = dynamic|hostname|IPAddr : How to find the client - IP # or host name. If you want the phone to register itself, use the keyword dynamic instead of Host IP. |
| 11:40.36 | dlynes_laptop | but if it's going to be dynamic |
| 11:40.41 | dlynes_laptop | they have to register |
| 11:40.43 | Hello2007 | but you can to undelete the deleted mail? |
| 11:40.45 | E-bola | then it wont work |
| 11:40.48 | dlynes_laptop | I think |
| 11:40.55 | lat1234 | anyone who knows how to run asterisk as a service/background in linux? |
| 11:40.59 | lat1234 | anyone who knows how to run asterisk as a service/background in linux? |
| 11:41.03 | dlynes_laptop | someone else might be able to give you a more certain answer, though |
| 11:41.11 | dlynes_laptop | lat1234: safe_asterisk |
| 11:41.17 | Hello2007 | but how can you choose to undelete a delete mail? |
| 11:41.35 | *** join/#asterisk glasco36 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 11:41.58 | dlynes_laptop | Hello2007: you don't |
| 11:42.05 | dlynes_laptop | Hello2007: it's not windows |
| 11:42.07 | Aurs | dlynes_laptop: yes, you do |
| 11:42.14 | dlynes_laptop | Aurs: eh? |
| 11:42.22 | Aurs | "press 7 to delete... " *7* "press 7 to undelete" |
| 11:42.22 | dlynes_laptop | Aurs: after you've permanently deleted it? |
| 11:42.29 | Hello2007 | in the voicemail menu ,it is possible to choose: undelete a deleted mail |
| 11:42.50 | Aurs | but I think that option is gone the next time you dial in |
| 11:42.55 | Aurs | isn't it? |
| 11:43.02 | dlynes_laptop | Hello2007: i guess you're talking about undeleting before you hang up |
| 11:43.02 | Hello2007 | so if they physically removed , this option should not exist,no??? |
| 11:43.10 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:43.10 | dlynes_laptop | Hello2007: thought you were talking about if you dialed back in again |
| 11:43.16 | Hello2007 | ah , ok only before i hang up |
| 11:43.28 | Hello2007 | ok, thanks |
| 11:43.35 | lat1234 | hello |
| 11:43.39 | lat1234 | anyone who knows how to run asterisk as a service/background in linux? |
| 11:43.46 | brian | is there anyway to dynamically create a context? |
| 11:43.48 | dlynes_laptop | Hello2007: it gets put into your tmp directory or something when you delete it |
| 11:43.52 | Aurs | lat1234: [12:40] <dlynes_laptop> lat1234: safe_asterisk |
| 11:43.57 | dlynes_laptop | Hello2007: but when you hang up, it gets deleted from there |
| 11:44.07 | Hello2007 | ok,i get it |
| 11:44.13 | brian | I mean like...a context in a database... |
| 11:44.25 | brian | Like SQLite |
| 11:44.52 | dlynes_laptop | brian: well, you could do it using realtime, I guess, but you'd need to write your own application module or something to achieve it, unless someone else has already written it |
| 11:45.20 | brian | There is no dynamic context module available? |
| 11:45.46 | dlynes_laptop | brian: google voip-info.org and find out |
| 11:46.21 | Aurs | your existing context should be so dynamic itself, that you don't need to add contexts ;) |
| 11:46.29 | brian | Aurs: what do you mean? |
| 11:46.44 | brian | Aurs: Well, I have context that are static room_1 - room_19 |
| 11:47.01 | dlynes_laptop | brian: it's called a joke :) |
| 11:47.07 | brian | But I also want to add the ability for non-public rooms. |
| 11:47.33 | brian | And I assumed the only way for me to retain the room number when exiting and re-entering AGI is to put it in the context... |
| 11:47.43 | brian | That is a correct assumption? |
| 11:48.27 | brian | FastAGI handles a lot of the functions, but it just can't handle everything |
| 11:48.37 | brian | Because it's limited |
| 11:48.58 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:49.17 | brian | The way I did it is pretty neat I think. |
| 11:50.24 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 11:50.42 | sergee | Any GrandStream GXV3000 users here? |
| 11:50.49 | oej | brian: You can create contexts from ami and cli I believe |
| 11:50.51 | oej | Hmmm |
| 11:51.21 | *** join/#asterisk Agrizzi (n=Brandon@69-165-243-236.atlsfl.adelphia.net) |
| 11:51.39 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:52.09 | RoyK | oej: did you find out anything more about that crash? |
| 11:52.57 | oej | RoyK: Not yet |
| 11:53.06 | oej | brian: You can add extensions but not contexts from the cli |
| 11:56.30 | *** join/#asterisk DerPraktikant (n=Tgu@pD95DEF39.dip.t-dialin.net) |
| 11:57.29 | DerPraktikant | hi! i got an problem with compiling bristuff with my asterisk |
| 11:58.17 | stephane | re |
| 11:58.24 | DerPraktikant | wb |
| 11:58.44 | DerPraktikant | i use suse linux and got my asterisk at work for local voip calls |
| 11:59.12 | DerPraktikant | now i wanted to connect the asterisk to an normal pbx by isdn |
| 11:59.19 | dlynes_laptop | bonjour, stephane |
| 11:59.22 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 12:00.01 | DerPraktikant | after the installation of bristuff-0.2.0 asterisk doesnt start anymore |
| 12:00.12 | DerPraktikant | can anybody help me? |
| 12:00.59 | florz | DerPraktikant: You possibly should actually describe your problem and not just mention that you do have a problem. |
| 12:01.08 | DerPraktikant | asterisk gives this erros: [ Booting............Nov 30 11:37:50 WARNING[15707]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/cdr_custom.so: undefined symbol: ast_register_file_version |
| 12:01.08 | DerPraktikant | Nov 30 11:37:50 WARNING[15707]: loader.c:440 load_modules: Loading module cdr_custom.so failed! |
| 12:02.19 | DerPraktikant | the cdr_custom needs the resource :asterisk:/usr/lib/asterisk/modules # ldd cdr_custom.so |
| 12:02.19 | DerPraktikant | <PROTECTED> |
| 12:02.19 | DerPraktikant | <PROTECTED> |
| 12:02.19 | DerPraktikant | <PROTECTED> |
| 12:02.33 | dlynes_laptop | DerPraktikant: did you forget to have 'autoload=yes' in your modules.conf file? |
| 12:03.23 | DerPraktikant | autoload= yes |
| 12:04.08 | DerPraktikant | like u can see in the text above , he needs the linux-gate.so.1 |
| 12:04.18 | DerPraktikant | but that file doenst axist on my pc |
| 12:04.22 | DerPraktikant | *exist |
| 12:04.26 | Aurs | brian: you could always have a web gui that stores contexts in a db table, and export that to file, then do extensions reload |
| 12:04.32 | dlynes_laptop | DerPraktikant: are you using a precompiled asterisk? |
| 12:04.51 | DerPraktikant | no |
| 12:05.02 | DerPraktikant | i compiled it myself |
| 12:05.17 | Aurs | "enter name of new context" - click - done. |
| 12:05.28 | florz | DerPraktikant: Do you need cdr_custom? |
| 12:05.55 | Aurs | if "room_1" and "room_20" are "equal", that is a easy way to do it |
| 12:06.03 | DerPraktikant | do u know for what cdr_custom is for? |
| 12:06.25 | dlynes_laptop | DerPraktikant: if you actually compiled it yourself, it shouldn't be trying to link to a non-existent elf library |
| 12:06.52 | florz | DerPraktikant: Well, then you probably don't need it, so try not loading it. |
| 12:06.54 | DerPraktikant | hm |
| 12:07.06 | bluemono | <-- newbie help needed |
| 12:07.08 | dlynes_laptop | noload => cdr_custom.so |
| 12:07.17 | DerPraktikant | i try it mom |
| 12:07.23 | DerPraktikant | thx ftw |
| 12:07.23 | bluemono | how do i login? |
| 12:07.29 | dlynes_laptop | bluemono: to what? |
| 12:07.33 | florz | I guess, it could be left over from the previous asterisk installation, not actually from the most recent compile ... |
| 12:07.42 | bluemono | i'm new to asterisk....linux too |
| 12:08.11 | bluemono | i've been tasked to learn linux,asterisk and trixbox lol |
| 12:08.38 | florz | DerPraktikant: Maybe the timestamp of the file gives you a hint as to whether this is the case? |
| 12:08.43 | wasim | bluemono: first go learn linux, then learn asterisk |
| 12:08.55 | bluemono | I've just switched the server on and it's asking for asterisk login |
| 12:09.05 | florz | .o( And then forget about trixbox? =:-) |
| 12:09.27 | Zefk | Hi, I'm running asterisk 1.2.13 on FC5 with B410P board. I use only port 1 of the board. Inbound and outbound calls via BRI port is working. I have the following message when asterisk starts: "mISDN dss1 fromup without proc pr=10180 dinfo(0)". Could be anything wrong? |
| 12:09.38 | bluemono | do i just insert the password here or do i need to tell it what account to login under ie. admin? |
| 12:09.53 | DerPraktikant | [ Booting..................Nov 30 13:09:09 WARNING[15889]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_chanspy.so: undefined symbol: ast_config_AST_MONITOR_DIR |
| 12:09.53 | DerPraktikant | Nov 30 13:09:09 WARNING[15889]: loader.c:440 load_modules: Loading module app_chanspy.so failed! |
| 12:10.45 | Aurs | bluemono: sounds like you are in a shell login prompt |
| 12:10.52 | *** join/#asterisk AK (n=ak@28.228.210.62.te-dns.org) |
| 12:11.33 | Aurs | and that doesn't really have anything to do with asterisk. you really should focus on learning some linux first |
| 12:11.56 | Aurs | it will be like learning to swim before learning to float.. or something |
| 12:12.25 | bluemono | ah ok thanks for the advice Aurs |
| 12:12.44 | Aurs | and what linux system do you have installed, bluemono+ |
| 12:12.47 | bluemono | i'm guessing the nect few months are going to be full of headaches |
| 12:13.10 | bluemono | CentOS release 4.4 final? |
| 12:13.14 | Aurs | ok |
| 12:13.42 | Aurs | you can login with username root, and the root password you entered during the installation |
| 12:13.53 | *** join/#asterisk santibiotico (n=santi@37.Red-83-36-42.dynamicIP.rima-tde.net) |
| 12:13.54 | santibiotico | hi |
| 12:15.14 | santibiotico | i'm witing an ivr menu for ISDN incoming calls...and the problem i'm having is the following: if i call through a cell phone or an analog phone to the ISDN number i can move through the menus without problem by pressing '1' or '2' or whatever |
| 12:15.18 | bluemono | lol the root password my boss gave is incorrect...oh dear not a good start |
| 12:15.44 | RoyK | bluemono, Aurs: http://karlsbakk.net/fun/dirty-advice.txt |
| 12:15.47 | Aurs | bluemono: did you not install linux? |
| 12:15.50 | DerPraktikant | ok i tryed it , to unload the problem has no sense because he opens everytime a new that cant be loaded |
| 12:15.55 | Aurs | oioioi |
| 12:15.57 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 12:15.57 | santibiotico | but if i call through a VoIP phone of my network connected to the same asterisk box, the asterisk does not recognise the keys dialed in the ivr menu |
| 12:16.08 | Aurs | bluemono: DO NOT listen to advice from RoyK! hehehe |
| 12:16.15 | RoyK | :) |
| 12:16.26 | santibiotico | does anybody know what could be happening? |
| 12:16.27 | AK | changing root psw isn't a probleme on linux or windows |
| 12:16.29 | viperdude | santibiotico: check DTMFmode on the asterisk /SIp phones |
| 12:16.34 | bluemono | no I've just been given the task to configure it after the person who installed it has left the company |
| 12:16.46 | Aurs | that is pure evil. hehe |
| 12:16.59 | dlynes_laptop | Aurs: that's why the other guy left |
| 12:17.00 | Aurs | bluemono: ok |
| 12:17.08 | santibiotico | viperdude: it's set to rfc2833 |
| 12:17.37 | bluemono | @AK do i not need the old password to login first bvefore i change it? |
| 12:17.41 | viperdude | santibiotico: try auto |
| 12:17.44 | Aurs | bluemono: if you want to learn linux, you really should install it yourself :) |
| 12:18.12 | DerPraktikant | florz are u still there? |
| 12:18.15 | AK | bluemono : what distrib? |
| 12:18.39 | Aurs | AK: he mentioned centos 4.4 |
| 12:18.48 | heh_v_water | bluemono, you can start linux in single-user mode and change root password probably |
| 12:19.13 | bluemono | ah that would help |
| 12:19.21 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 12:19.30 | shellshark | unless your boot loader is password-protected, then you'll need a livecd :) |
| 12:19.33 | AK | Aurs, sry i'm not reading everything, I don't know about centos but you can try to boot on a live cd |
| 12:19.38 | AK | mount the partition |
| 12:19.43 | shellshark | AK: yep |
| 12:19.44 | AK | and do a passwd |
| 12:19.47 | Aurs | probably password... yes.. what shellshark said.. hehe |
| 12:19.48 | shellshark | well |
| 12:19.51 | shellshark | chroot first |
| 12:19.54 | shellshark | then passwd |
| 12:20.06 | Aurs | this guy have never touched a linux system guys |
| 12:20.07 | AK | haha, i'm not reading |
| 12:20.18 | shellshark | otherwise it will change the password on the working environment (livecd) |
| 12:20.18 | AK | g2g |
| 12:20.28 | bluemono | *blushes* |
| 12:20.32 | Aurs | bluemono: the easiest way (and you also learn from it) would be to reinstall ;) |
| 12:20.50 | shellshark | ugh |
| 12:20.54 | DerPraktikant | do anybody know what this could mean: undefined symbol: ast_register_file_version |
| 12:20.55 | shellshark | that's not the easiest way |
| 12:21.18 | Aurs | of course it is. for a guy that don't know linux |
| 12:21.18 | shellshark | that's the cheater way to do things |
| 12:21.20 | dlynes_laptop | DerPraktikant: it means it doesn't have the shared object loaded into memory that contains that symbol |
| 12:21.23 | Aurs | my mom can install linux |
| 12:21.35 | dlynes_laptop | Aurs: your mom is bill gates |
| 12:21.41 | Aurs | (or probably not) |
| 12:21.43 | jmesquita | Anyone have a problem with bug #0007765???? |
| 12:21.44 | bluemono | it's a bit maddening that i have a server built with an os and asterisk on it but now i've to wipe it clean and re-install |
| 12:21.55 | brian | And I assumed the only way for me to retain the room number when exiting and re-entering AGI is to put it in the context... |
| 12:21.56 | Aurs | dlynes_laptop: oh, we're doing "yo mama" jokes, are we? :P |
| 12:21.58 | bluemono | just cus i don't have the root password |
| 12:22.00 | brian | That is a correct assumption? |
| 12:22.11 | dlynes_laptop | bluemono: Just do a chroot like shellshark suggested |
| 12:22.12 | DerPraktikant | how can i change this error? |
| 12:22.12 | shellshark | brian: no, you can use variables |
| 12:22.14 | Aurs | bluemono: ok, then do not reinstall |
| 12:22.19 | *** join/#asterisk kuto (n=kuto@125.60.241.24) |
| 12:22.20 | brian | shellshark: How would I use variables? |
| 12:22.35 | shellshark | brian: read the docs for set |
| 12:22.37 | brian | shellshark: Are the variables per-session? |
| 12:22.39 | shellshark | on voip-info.org |
| 12:22.39 | Aurs | brian: room number can be saved in ie astdb |
| 12:22.43 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 12:22.45 | shellshark | brian: sure |
| 12:22.50 | Aurs | so you can use DB to get room number for a given ext |
| 12:22.50 | brian | shellshark: The reason I didn't use variables is because i thought they were global |
| 12:23.03 | bluemono | Thanks guys/girls i'll spend the rest of the day getting my head around linux i guess |
| 12:23.14 | shellshark | brian: you can make them global, or set them per-session also |
| 12:23.14 | brian | Aurs: I'm using FastAGI |
| 12:23.26 | shellshark | bluemono: any questions come back and ask :) |
| 12:23.30 | Aurs | brian: ok, and that does not read variables? |
| 12:23.31 | brian | Aurs: And what is astdb anyways |
| 12:23.36 | brian | Aurs: Yes, it does |
| 12:23.37 | DerPraktikant | dlynes_laptop: who can i change this? |
| 12:23.56 | Aurs | brian: the DB function. saves to a bercley database.. /var/something/asterisk/something/astdb |
| 12:24.00 | dlynes_laptop | bluemono: Just boot up with a linux bootable cd, mount your root partition from your hard drive, do a chroot do that mounted directory, type 'passwd', and then type in the new root password |
| 12:24.02 | brian | Aurs: But for some reason...like say I do MeetMeCount|var, when the conference is empty it sets the variable to 1 |
| 12:24.10 | dlynes_laptop | bluemono: then reboot, remove the cd from the drive, and you'll be good to go |
| 12:24.17 | dlynes_laptop | bluemono: no need to reformat and reinstall |
| 12:24.20 | shellshark | Aurs: berkely* |
| 12:24.31 | Aurs | ok, berkely |
| 12:24.36 | brian | I don't want to use Berkeley, I want to use SQLite |
| 12:24.55 | viperdude | dlynes_laptop: the user bluemono has never used linux before... might be a bit too much for him |
| 12:24.56 | Aurs | ok, then use SQLite |
| 12:25.07 | shellshark | brian: just use session variables then, pick them up in your AGI, then use whatever DB backend from the AGI you want to |
| 12:25.11 | bluemono | dlynes_laptop: your a star :) |
| 12:25.17 | brian | I don't think variables will work because I have to disconnect and reconnect the FastAGI |
| 12:25.21 | shellshark | bluemono: and no credit to me, eh? :) |
| 12:25.28 | shellshark | brian: so? |
| 12:25.41 | shellshark | brian: as long as you don't jump context they'll still be set |
| 12:25.41 | brian | There are session variables in asterisk? |
| 12:25.43 | dlynes_laptop | bluemono: like i said...shellshark already told you...I just gave you more details |
| 12:25.50 | bluemono | sure shellshark |
| 12:25.51 | RoyK | jmesquita: i'd try to downgrade to 1.2.8, 1.2.9, 1.2.9.1 and so on to locate it. perhaps try to upgrade to 1.2.13 |
| 12:25.52 | shellshark | brian: didn't I just say that? :) |
| 12:25.54 | Aurs | brian: channel vars are alive as long as the channel is (i think?) |
| 12:25.57 | brian | I didn't know all that. |
| 12:26.05 | dlynes_laptop | viperdude: It's a lot less for him than trying to figure out how to install linux and set it up |
| 12:26.09 | brian | But it's still complicating... |
| 12:26.11 | DerPraktikant | dlynes_laptop: who can i change this? who can i load that shared object? |
| 12:26.13 | shellshark | Aurs: as long as you don't jump context |
| 12:26.19 | brian | Well...I guess not |
| 12:26.22 | shellshark | brian: it's very simple, read the docs |
| 12:26.22 | brian | I guess that will work |
| 12:26.23 | brian | Let's see |
| 12:26.27 | viperdude | dlynes_laptop: depends I thought trixbox was boot from CD and go |
| 12:26.35 | brian | so I use the AGI command SET VARIABLE right? |
| 12:26.35 | shellshark | viperdude: it is |
| 12:26.41 | jmesquita | RoyK: I have it on 1.2.13 now |
| 12:26.43 | dlynes_laptop | viperdude: he's using trixbox? |
| 12:26.45 | shellshark | brian: set it from your dialplan |
| 12:26.48 | bluemono | i have the trixbox cd here |
| 12:26.48 | jmesquita | RoyK: And I still have this annoying bug |
| 12:26.48 | viperdude | thats what he said |
| 12:26.54 | brian | shellshark: My dialplan? |
| 12:26.57 | Aurs | brian: you can set it before AGI runs |
| 12:26.57 | jmesquita | RoyK: The problem is that this is a hard problem to track |
| 12:27.06 | shellshark | brian: that's what i said ;) |
| 12:27.08 | brian | Aurs: Err, I have to set it before AGI runs? |
| 12:27.09 | dlynes_laptop | viperdude: but even still, he'd still have to figure out where the config files are, and back them up and that kinda thing |
| 12:27.13 | jmesquita | RoyK: I haven't been able to really identify how and why these missing events occur |
| 12:27.14 | Aurs | brian: extensions.conf |
| 12:27.17 | bluemono | trixbox v1.2.3 |
| 12:27.22 | shellshark | brian: you don't HAVE to, but that's common practice |
| 12:27.25 | brian | Mother #$!% |
| 12:27.27 | dlynes_laptop | viperdude: it's more work to reconfigure trixbox than doing a simple chroot |
| 12:27.43 | brian | shellshark: Well, in my case... |
| 12:27.44 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 12:27.48 | Aurs | hehe, I won't force you to do it brian. just a possible solution |
| 12:27.51 | viperdude | dlynes_laptop: i have never used it so I bow to your knowledge :-) |
| 12:27.52 | bluemono | chroot is like change directory? |
| 12:27.53 | brian | shellshark: They might not be in a room at all. |
| 12:27.57 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:28.11 | shellshark | brian: I know nothing of your setup, so saying "room" means nothing to me ;) |
| 12:28.18 | brian | shellshark: meetme conference |
| 12:28.18 | Aurs | brian: then the variable would be empty. and your AGI should be able to handle that |
| 12:28.18 | shellshark | brian: as does "they" |
| 12:28.33 | shellshark | Aurs: yep |
| 12:28.49 | brian | Aurs: So set an example variable? |
| 12:29.02 | Aurs | with SQLite? never used that one |
| 12:29.24 | brian | I don't think I need a SQL database for this part. |
| 12:29.41 | DerPraktikant | does anybody know who i can an isdn-card with an cologne-chip? |
| 12:29.59 | Aurs | no, and DB (astdb/berkley) is not sql |
| 12:30.01 | brian | I know that |
| 12:30.09 | dlynes_laptop | viperdude: neither have I |
| 12:30.13 | brian | I'm talking about SQL within my FastAGI application :P |
| 12:30.15 | Aurs | you can easily save and read vars with DB function |
| 12:30.22 | dlynes_laptop | viperdude: but even if it's just simple asterisk, and not trixbox |
| 12:30.32 | Aurs | exten => _X.,10,DBget(mb1=${ext}/room) |
| 12:30.33 | brian | I'm not going to use DB functions within Asterisk, i'm going to do that within FastAGI |
| 12:30.33 | heh_v_water | bluemono, since your new to linux if you don't know what a command does man it meaning.. man chroot |
| 12:30.33 | dlynes_laptop | viperdude: He's still gotta figure out how to back up his config files from that box |
| 12:30.41 | DerPraktikant | the device is zaphfc but the version of zaptel which comes with the asterisk 1.2 dont support it |
| 12:30.43 | Aurs | it's not DBget anymore, but similar syntax |
| 12:30.56 | brian | Aurs: It doesn't have to be persistent. |
| 12:31.06 | kuto | hi all, has anyone have a solution with this bug? http://bugs.digium.com/view.php?id=6691 <= i tried following what is stated but i got no luck. |
| 12:31.07 | viperdude | dlynes_laptop: i guess...sounded to me like it was a fresh install with no config made |
| 12:31.16 | brian | Aurs: If the FastAGI (or asterisk) goes down it just goes back to having 0 people in it. |
| 12:31.43 | brian | Aurs: So saving it in a database isn't neccessary |
| 12:31.50 | Chris-NB | anybody knows what that err means? |
| 12:31.56 | Chris-NB | zt_pri_error: 1 updating callstate, ourcallstate 1 to 6 |
| 12:31.58 | brian | Aurs: What would need to be saved in a database is like a voice mail PIN |
| 12:32.01 | Aurs | well you have to save it somewhere if you want to check for it |
| 12:32.04 | dlynes_laptop | viperdude: oh |
| 12:32.16 | brian | Aurs: Yeah, into the session variable. |
| 12:32.23 | dlynes_laptop | viperdude: i thought it was a live system |
| 12:32.32 | brian | SET roomnumber 1 or whatever |
| 12:32.38 | dlynes_laptop | bluemono: is this trixbox install a live system, or is it a fresh system that's never been used? |
| 12:32.41 | brian | then reference it like ${roomnumber} right? |
| 12:33.17 | viperdude | dlynes_laptop: dunno i could be wrong.. |
| 12:33.58 | shellshark | anyone here a mason? :) |
| 12:34.00 | brian | What is NULL in Asterisk? |
| 12:34.03 | bluemono | dlynes_laptop I'm guessing it hasn't been used yet |
| 12:34.06 | *** part/#asterisk glasco36 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 12:34.35 | bluemono | since we have another system running Voip-live to run the phones at the mo |
| 12:34.37 | dlynes_laptop | viperdude: bluemono no I've just been given the task to configure it after the person who installed it has left the company |
| 12:34.44 | dlynes_laptop | viperdude: that's why I thought he'd used it :) |
| 12:34.51 | dlynes_laptop | bluemono: i c |
| 12:35.00 | dlynes_laptop | bluemono: yeah...the better advice was to reinstall then |
| 12:35.01 | viperdude | ok |
| 12:35.09 | dlynes_laptop | bluemono: forget the chroot |
| 12:35.15 | bluemono | ok |
| 12:35.21 | dlynes_laptop | bluemono: then you learn what's included with linux, and learn the basics first |
| 12:35.24 | *** join/#asterisk ftexcom (n=info@14.Red-80-26-4.staticIP.rima-tde.net) |
| 12:35.25 | ftexcom | whoa |
| 12:35.32 | ftexcom | My zap dosn't detect pickup on zap |
| 12:35.45 | bluemono | steep learning curve then eh guys |
| 12:35.49 | bluemono | :) |
| 12:35.53 | shellshark | kinda :) |
| 12:36.02 | viperdude | bluemono: all good things are |
| 12:36.55 | Aurs | gotta go. good luck brian ;) |
| 12:37.03 | bluemono | viperdude: true |
| 12:37.08 | heh_v_water | bluemono, it's worth the effort |
| 12:37.16 | dlynes_laptop | bluemono: asterisk is a like a nice lady |
| 12:37.31 | heh_v_water | linux has many interesting and amazing things to offer |
| 12:37.31 | dlynes_laptop | bluemono: you need to wine and dine her before you take her up to your place for a romp in the sack |
| 12:37.44 | *** join/#asterisk beyond (n=beyond@c9346fb2.virtua.com.br) |
| 12:37.47 | bluemono | lol |
| 12:38.26 | viperdude | dlynes_laptop: you obviously aint met some of the women i know ;-) |
| 12:38.59 | dlynes_laptop | Ok...let me rephrase |
| 12:39.16 | dlynes_laptop | A nice lady that you would take home to introduce to your mother |
| 12:39.23 | viperdude | lol |
| 12:39.38 | dlynes_laptop | not a nice lady that you won't call back the next day :) |
| 12:39.49 | dlynes_laptop | or that you have to pay $150 for the night :) |
| 12:39.53 | bluemono | <--loads the cd and holds his breath...wish me luck guys |
| 12:40.09 | viperdude | yeah but asterisk can go zombie on ya |
| 12:40.51 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 12:40.59 | viperdude | anyone here going to Von in San Jose? |
| 12:44.12 | bluemono | is it right that the server doesn't need to be really powerfull? |
| 12:44.38 | bluemono | it's on a dual P4 2.6ghz with 512mb Ram lol |
| 12:45.14 | viperdude | bluemono: depends on a number of factors... |
| 12:45.17 | SheriF_SpacE | any problmes with SMP machines ? xeon or anything and asterisk ? |
| 12:45.29 | viperdude | number concurrent calls, transcoding, etc etc |
| 12:46.07 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:46.50 | bluemono | i see, so the larger the number of employee's the "meatier" the box needs to be |
| 12:47.18 | viperdude | generally speaking yes... |
| 12:47.23 | viperdude | how many you got? |
| 12:47.44 | E-bola | if its pure sip |
| 12:47.50 | bluemono | 50 here and about another 15-20 at a remote site |
| 12:47.51 | E-bola | the server more or less cant be too small |
| 12:48.07 | viperdude | you should be fine with that number |
| 12:48.25 | viperdude | maybe boost ram to 1gig |
| 12:48.55 | viperdude | i run 220 extensions on a similar setup |
| 12:49.06 | bluemono | so accentuate the Ram rather than the processor power |
| 12:49.20 | viperdude | depends if you are transcoding |
| 12:50.29 | viperdude | bluemono: check http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 12:50.44 | bluemono | ok thanks |
| 12:51.21 | bluemono | do you also have a url for list of linux commands and basic meanings plz? |
| 12:52.05 | heh_v_water | bluemono, thats a hooooge list |
| 12:52.06 | viperdude | bluemono: google is your friend |
| 12:52.15 | dlynes_laptop | SheriF_SpacE: nope |
| 12:52.23 | dlynes_laptop | SheriF_SpacE: lots of peeps running asterisk on xeons |
| 12:52.25 | SheriF_SpacE | sweeeeeeeeeeeeet |
| 12:52.30 | dlynes_laptop | SheriF_SpacE: I'm running it on a dual piii |
| 12:52.31 | SheriF_SpacE | peeps ? |
| 12:52.37 | wasim | bluemono: there is just 1 command, ok, 2 that you need to know ... (rm and dd) |
| 12:52.38 | bluemono | yeah i guess i would just google it. |
| 12:52.54 | dlynes_laptop | SheriF_SpacE: I've seen a number of people in this channel having problems running asterisk with hyperthreading enabled, though |
| 12:52.58 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 12:53.22 | wasim | bluemono: whenever you are on irc and someone like coppice tells you to rm or dd something, be very wary |
| 12:54.17 | bluemono | lol yeah, i was guessing he was doing the "make fun of the newbie" trick there |
| 12:55.11 | heh_v_water | bluemono, the hting to remember is if somebody says typ ethis command.. before you type the command type man *command* first |
| 12:55.25 | heh_v_water | that will show you what it does |
| 12:55.27 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 12:55.34 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:56.29 | bluemono | cool tip heh_v_water cheers |
| 12:59.58 | *** join/#asterisk delphus (n=delphus@mail.delphus.org) |
| 13:00.01 | *** join/#asterisk k84 (n=pyro@86.84-48-44.nextgentel.com) |
| 13:00.02 | bluemono | well the server is rebuilt now but i've learnt nothing about linux/asterisk/trixbox |
| 13:00.23 | bluemono | cd just formatted and installed it all for me |
| 13:00.24 | tcseke | Hello, |
| 13:00.25 | tcseke | I've a problem about monitoring with app_meetme |
| 13:00.27 | tcseke | I've a local and a zap channel in a meetme conference, and i'd like to monitor the zap channel, i tried monitor and mixmonitor. |
| 13:00.28 | tcseke | with both of them the result the local channel is not recorded, |
| 13:00.31 | delphus | question: does anyone here have sip lines with inphonex and can receive calls ? |
| 13:01.06 | *** part/#asterisk k31th (n=keith@cartman.nzsolutions.net) |
| 13:01.18 | heh_v_water | bluemono, but you now have root password |
| 13:01.28 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 13:02.50 | tzafrir | bluemono, to learn about linux and asterisk, don't use trixbox |
| 13:03.17 | ftexcom | anyone can give me some advice? |
| 13:03.21 | bluemono | ok so uninstall trixbox? |
| 13:03.39 | heh_v_water | ftexcom, yes don't step on the soft brown rocks |
| 13:03.50 | ftexcom | very ironic |
| 13:03.53 | bluemono | never eat yellow snow? :O |
| 13:04.22 | ftexcom | My zap dosn't detect pickup when calling to a mobile phone |
| 13:04.30 | ftexcom | regional settings are correct |
| 13:04.55 | *** join/#asterisk Tili (n=tili@202.133.67.127) |
| 13:05.45 | heh_v_water | bluemono, what you must understand is trixbox is built for asterisk, is pretty new and small distribution.. I couldnt tell you how stable it is or how well they keep up on security |
| 13:06.12 | heh_v_water | bluemono, you can do everything trixbox does with a more mainstream linux distribution |
| 13:06.26 | ftexcom | the curious thing |
| 13:06.32 | ftexcom | is that I'm able to talk |
| 13:07.09 | *** join/#asterisk daniloegea (n=daniloeg@200.146.119.3.static.gvt.net.br) |
| 13:07.22 | *** part/#asterisk daniloegea (n=daniloeg@200.146.119.3.static.gvt.net.br) |
| 13:07.23 | bluemono | heh_v_water: any recommendations on linux version? |
| 13:07.31 | dlynes_laptop | bluemono: slackware |
| 13:07.48 | heh_v_water | geezus |
| 13:07.52 | bluemono | sounds v.dodgy |
| 13:08.00 | k84 | gentoo |
| 13:08.08 | dlynes_laptop | sourcemage |
| 13:08.14 | k84 | lfs |
| 13:08.17 | HarryR | go with the biggies, RHEL/CentOS, SLES, Gentoo :) |
| 13:08.29 | dlynes_laptop | gentoo's a biggy? |
| 13:08.32 | k84 | Yes |
| 13:08.33 | HarryR | yeah |
| 13:08.33 | heh_v_water | bluemono, stick to mainstream ones that are easier to work on.. fedora, debian, gentoo |
| 13:08.58 | k84 | I can only say one thing, stick with gentoo! :-) listen to heh_v_water & HarryR They are smart ppl :-) |
| 13:08.59 | dlynes_laptop | I thoguht all you guys were insisting on him learning linux? |
| 13:09.06 | dlynes_laptop | slackware is the best distro for that |
| 13:09.14 | dlynes_laptop | it forces you to learn how to do everything manually |
| 13:09.16 | dlynes_laptop | no gui tools :) |
| 13:09.22 | k84 | dlynes_laptop, you dont learn linux if you dont have to do anything, no pain no gain! |
| 13:09.30 | k84 | dlynes_laptop, You have to learn from a to z :) |
| 13:09.34 | HarryR | but has really shoddy package management, a bad upgrade path and a small pool of maintainers/supporters |
| 13:09.35 | heh_v_water | dlynes_laptop, you dont give a first time pilot the space shuttle man.. get a grip |
| 13:09.37 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 13:09.40 | dlynes_laptop | k84: exactly...so use slackware :) |
| 13:10.15 | HarryR | dlynes_laptop, why Slackware over Gentoo? |
| 13:10.25 | k84 | dlynes_laptop, no :-) Hmm if you look at it like that i'd say slackware has much more "ncurses" /gui's then let's say gentoo |
| 13:10.25 | dlynes_laptop | no special reason |
| 13:10.30 | dlynes_laptop | I just like slackware :) |
| 13:10.32 | k84 | let me guess |
| 13:10.39 | HarryR | ok so you'r heavily bias |
| 13:10.52 | dlynes_laptop | k84: you were thinking bitchx? |
| 13:11.04 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 13:11.05 | k84 | dlynes_laptop, i was thinking slackware or not :-) |
| 13:11.12 | dlynes_laptop | k84: slackware 10.2 |
| 13:11.12 | heh_v_water | here's the hting.. you have to go out and try some.. it's like trying on a pair of shoes.. the one you like and makes sense to you.. stick to it |
| 13:11.45 | k84 | dlynes_laptop, so how did that slack dude come along, can remember he had some health issues years ago? |
| 13:11.45 | coppice | wasim: any what exactly is wrong with rm -r / that a reinstall won't cure? |
| 13:11.45 | dlynes_laptop | k84: although, i've got solaris 9 running on the machine under this one |
| 13:11.45 | tzanger | morning coppice |
| 13:11.48 | coppice | hi |
| 13:11.49 | heh_v_water | I started out on redhat and mandrake and suse.. debian is the best I've used.. maybe it will be for you too, maybe it wont |
| 13:11.52 | dlynes_laptop | k84: Patrick's fine now |
| 13:12.07 | k84 | k84, Great :-) |
| 13:12.07 | dlynes_laptop | k84: a slackware user pointed him to a good doctor that was able to get him fixed up |
| 13:12.23 | k84 | k84, Think i was on slack myself at that time, so i read his letter :D |
| 13:12.32 | k84 | k84, didnt know any good doctor though :-) |
| 13:12.36 | dlynes_laptop | k84: enjoy talking to yourself? :) |
| 13:12.50 | k84 | dlynes_laptop, enjoy discussing distro with you :-) |
| 13:13.02 | dlynes_laptop | but you keep prefacing every reply with your name :) |
| 13:13.02 | k84 | dlynes_laptop, Damn autocomplete :-) |
| 13:13.19 | k84 | str_replace? :) |
| 13:13.28 | dlynes_laptop | s/k84/dlynes_laptop/ |
| 13:13.29 | dlynes_laptop | ? |
| 13:13.35 | k84 | regex is fine |
| 13:13.43 | k84 | yes |
| 13:14.23 | HarryR | http://www.zend.com/de/phpide |
| 13:14.28 | HarryR | uh, wrong chan |
| 13:15.32 | *** join/#asterisk A_Thief (n=sam@mbl-99-58-31.dsl.net.pk) |
| 13:17.23 | A_Thief | i'm calling from one asterisk server to another asterisk server. both have the same range of extensions defined. i get a "failed to authenticate" error msg from the receiving asterisk server. |
| 13:17.31 | dlynes_laptop | thief! |
| 13:17.52 | A_Thief | has anyone faced something like this before? the problem is asterisk is only looking at the username and not at the domain |
| 13:17.59 | bluemono | certainly given me a lot of food for thought there guys |
| 13:18.06 | dlynes_laptop | A_Thief: you either needed to register, or you passed an incorrect username or password |
| 13:19.13 | bluemono | i think i need a seperate box to figure which one is best for me, but i'll stick with this setup as i don't really want to challenge the boss right now |
| 13:19.29 | *** join/#asterisk alerios (n=alerios@69.79.145.100) |
| 13:19.35 | *** join/#asterisk nix (i=nix@spirit.infernix.net) |
| 13:21.33 | A_Thief | dlynes_laptop: actually, an asterisk to asterisk call works if i call 201@AsterA from 401@AsterB (since user 401 does not exist on AsterA), but if i call 201@AsterA from 201@AsterB, it says failed to authenticate |
| 13:22.28 | dlynes_laptop | A_Thief: need to make sure their passwords match, then |
| 13:23.34 | *** join/#asterisk |oranjia| (n=kvirc@dsl-243-129-164.telkomadsl.co.za) |
| 13:23.42 | |oranjia| | hello peeps :) |
| 13:23.49 | dasenjo | Hi! someone know a way of integrate a harphone with a softphone in the same extension? |
| 13:24.06 | A_Thief | dlynes_laptop: if the passwords were incorrect, then the call to 201 from 401 would not go thru. |
| 13:24.37 | dlynes_laptop | A_Thief: password for 201@AsterA must match password for 201@AsterB |
| 13:24.38 | |oranjia| | I haveing trouble getting asterisk to see a #... so when something like this is in the dialplan : exten=>*33*23# etc the hash doesn't get picked up |
| 13:24.43 | |oranjia| | I am using Realtime |
| 13:26.00 | A_Thief | dlynes_laptop: that should not be the case if some other server is calling with the same username (but different domain), but either way, the password DO match in my case |
| 13:26.22 | dlynes_laptop | A_Thief: another server wouldn't be passing a username and password |
| 13:26.45 | dlynes_laptop | A_Thief: but your astera and asterb are authenticating to each other |
| 13:27.47 | A_Thief | dlynes_laptop: both asterisks are connect thru openser. both asterisks r registered to openser |
| 13:28.12 | dlynes_laptop | A_Thief: no idea then...i know nothing about openser or ser |
| 13:28.35 | A_Thief | dlynes_laptop: well, the problem is that asterisk needs to match the domain part of the URI too |
| 13:28.46 | dlynes_laptop | again, no idea |
| 13:28.52 | A_Thief | dlynes_laptop: thanks :) |
| 13:29.03 | dlynes_laptop | but tehn again, i'm concentrating on other things atm |
| 13:31.15 | dlynes_laptop | DerPraktikant: btw...forgot to mention |
| 13:31.24 | dlynes_laptop | DerPraktikant: that shared object you couldn't find |
| 13:31.36 | dlynes_laptop | DerPraktikant: it's a virtual shared object that's in your kernel-space |
| 13:31.50 | dlynes_laptop | DerPraktikant: perhaps you haven't included that option in your kernel |
| 13:34.04 | *** join/#asterisk Robyn (n=Ebola@host86-136-86-71.range86-136.btcentralplus.com) |
| 13:36.47 | *** join/#asterisk _ebola (n=freenode@veryniceshop.com) |
| 13:37.24 | Chris-NB | can someone plz look at this http://phpfi.com/180078 and probably can tell me whats the problem there? |
| 13:38.15 | bluemono | <--struck gold with the Asterisk 376 page Book \o/\o/ |
| 13:39.42 | |oranjia| | for some reason my sip client is sending my asterisk box %23 instead of a hash |
| 13:39.57 | |oranjia| | So i am guess that asterisk is not to blame |
| 13:40.18 | |oranjia| | hash=pound |
| 13:40.21 | |oranjia| | =# |
| 13:41.49 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 13:42.41 | *** join/#asterisk SomethingISODD (n=Somethin@209.226.89.101) |
| 13:43.01 | SomethingISODD | hello all question does anyone know of any programs that will work as a server and handle iptv with channel guides and such |
| 13:45.09 | jmesquita | SomethingISODD: Wow, that would be interesting :D |
| 13:45.39 | SomethingISODD | jmesquita: well it would be if i could find something that actually worked lol |
| 13:45.53 | jmesquita | SomethingISODD: Have you tried anything yet? |
| 13:46.05 | brian | asterisk programming is fun |
| 13:46.06 | brian | :D |
| 13:46.38 | DerPraktikant | dlynes_laptop , thanks but i got my own way now , deinstalled and used another version, it work , thanks 4tw |
| 13:46.44 | jmesquita | SomethingISODD: All I cound found at first search are clients |
| 13:47.09 | SomethingISODD | jmesquita: ya a few vlc mythtv a few others but none handle it to the scale i need |
| 13:47.16 | SomethingISODD | myth is just a client |
| 13:47.20 | SomethingISODD | vlc is a server |
| 13:47.27 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 13:48.47 | DerPraktikant | the version of bristuff must be equal to the preinstalled asterisk |
| 13:48.58 | jmesquita | SomethingISODD: How big do you need? |
| 13:49.15 | brian | mine is 1 foot long |
| 13:49.23 | DerPraktikant | sometimes i wish the developers where skilled in writing understandable manuals |
| 13:49.24 | brian | my cat5e cable that is |
| 13:50.12 | SomethingISODD | well i live out of town, on wireless internet connection i want to rebroadcast my local stations to me home |
| 13:51.56 | jmesquita | SomethingISODD: Ever heard of icecast? |
| 13:52.17 | brian | anyone here hear of talkee |
| 13:52.21 | SomethingISODD | no is it any good? |
| 13:52.40 | brian | SomethingISODD: yes! |
| 13:52.57 | santibiotico | can anyone help me with IVR ?? |
| 13:52.59 | brian | SomethingISODD: just take out that pesky ogg vorbis and replace it with mpeg4 or something |
| 13:53.05 | SomethingISODD | ok i will take a look thanks. |
| 13:53.08 | nicox | what do you need? |
| 13:53.34 | jmesquita | SomethingISODD: Hummm, I guess its only audio |
| 13:53.34 | santibiotico | i want sip users to be able to interact with an IVR menu defined in the same asterisk the users are logged in |
| 13:53.55 | santibiotico | however, if i associate sip users to a context (ie: local-users) |
| 13:54.14 | santibiotico | then, when i call the extension the ivr is located in |
| 13:54.30 | santibiotico | and when i try to dial one ivr menu option (ie: '1') |
| 13:55.02 | santibiotico | the systems, obviously, tries to find a dialplan for extension 1 in the context "local-users" |
| 13:55.24 | santibiotico | however, in that context there is no dialplan for extension '1', as it is an IVR menu option |
| 13:56.17 | santibiotico | apart from changing the context, (i don't want to..) is there any way to solve this problem? |
| 13:57.41 | santibiotico | :?? |
| 13:58.06 | nicox | did you hear the ivr menu already? |
| 13:59.42 | santibiotico | yes |
| 14:00.19 | *** join/#asterisk Asttt (n=asterisk@193.110.8.51) |
| 14:00.22 | nicox | do you try another client |
| 14:00.26 | santibiotico | i hear the message i've recorded |
| 14:00.28 | nicox | sorry, |
| 14:00.32 | nicox | try another client |
| 14:00.38 | santibiotico | let me explain you... |
| 14:00.52 | santibiotico | if i call from a cell to an isdn where i place the ivr...it all goes ok |
| 14:01.44 | santibiotico | if i call from an external sip phone, it also goes ok |
| 14:01.59 | santibiotico | the problem is with sip phones in my system |
| 14:02.10 | santibiotico | because when i dial a ivr option, such as '1' or '2' |
| 14:02.24 | santibiotico | the system treats it as if i was dialing an extension |
| 14:02.28 | nicox | i think thats a problem of dtmf signalling |
| 14:02.40 | santibiotico | i've tried all methods of dtmf |
| 14:02.43 | santibiotico | rtf2833 |
| 14:02.45 | santibiotico | auto |
| 14:02.47 | santibiotico | inband |
| 14:02.48 | santibiotico | etc... |
| 14:02.51 | nicox | yes |
| 14:02.56 | nicox | try inband |
| 14:02.56 | santibiotico | with no success at all |
| 14:03.04 | santibiotico | i've already done it |
| 14:03.43 | *** join/#asterisk dr0ne (n=fn@CABLE-72-53-45-212.cia.com) |
| 14:03.50 | nicox | hm... only in asterisk, or did you changed it also on the phone? |
| 14:04.11 | santibiotico | only in * |
| 14:04.26 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 14:04.55 | nicox | try to change it on the phone |
| 14:05.01 | nicox | thats important |
| 14:05.07 | santibiotico | i've just done it |
| 14:05.14 | santibiotico | and i get the same |
| 14:05.28 | santibiotico | asterisk tries to reach extension '1' when i dial '1' as an IVR option |
| 14:06.18 | nicox | try to option INFO on the phone |
| 14:07.21 | santibiotico | i've just tried |
| 14:07.25 | santibiotico | and the same |
| 14:07.29 | santibiotico | :( |
| 14:08.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:09.53 | santibiotico | any other idea? |
| 14:10.09 | santibiotico | is there any way to tell asterisk not to treat the dialed code as an extension¿ |
| 14:11.10 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.52 | *** join/#asterisk andresmujica (n=andresmu@201.244.196.229) |
| 14:14.25 | *** join/#asterisk gdh_ (i=foobar@bum.net) |
| 14:14.35 | *** join/#asterisk banik (n=Main@203.208.196.140) |
| 14:14.58 | banik | Hello everybody |
| 14:15.11 | *** join/#asterisk AK (n=ak@28.228.210.62.te-dns.org) |
| 14:15.56 | nicox | do you tried relaxdtmf? |
| 14:16.03 | banik | I'm getting this error "UnsupportedMedia Type" |
| 14:16.16 | banik | what does it mean? |
| 14:16.33 | banik | codec mismatch? |
| 14:17.08 | banik | when ever i call to my quintum my asterisk box reports this error "UnsupportedMedia Type" |
| 14:17.24 | banik | any body know why does it occure? |
| 14:17.34 | gdh_ | Anyone who can help with this the prob here: http://pastebin.ca/262406 ? |
| 14:17.38 | ManxPower | banik: that is usually a codec issue |
| 14:17.40 | *** join/#asterisk seele_ (n=seele@208.35.117.246) |
| 14:17.47 | banik | Thanks |
| 14:17.59 | banik | Thanks ManxPower |
| 14:18.33 | seele_ | please help .... I have this warning http://pastebin.ca/262405 in my /asterisk/full log .... how can I solve this? |
| 14:18.59 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 14:19.49 | banik | How many calls can handle an asterisk box simutaneously? |
| 14:19.52 | pjz | my asterisk server got rebooted last night and now every vmail box that had a custom greeting no longer has sone |
| 14:20.01 | pjz | banik: depends on the size of the box |
| 14:20.05 | banik | my box is p4 2.4 GH, 2GB RAM |
| 14:20.23 | pjz | banik: several dozen at minimum, I think |
| 14:20.43 | banik | several means? |
| 14:20.45 | banik | 3/4? |
| 14:20.52 | pjz | banik: at least |
| 14:21.02 | gdh_ | banik: http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 14:21.08 | pjz | banik: there's some info on voip-info.org about sizing your system |
| 14:21.13 | banik | pjz: great |
| 14:22.04 | pjz | my asterisk server got rebooted last night and now every vmail box that had a custom greeting no longer has one... the files are there in /var/spool/asterisk/voicemail/default/, but they're not being used for some reason ; any ideas how to turn them back on if the file exisgts? |
| 14:22.25 | *** join/#asterisk sloth_ (n=josh@pool-162-84-157-242.ny5030.east.verizon.net) |
| 14:23.15 | banik | anybody have digium E1 interface experience? |
| 14:23.36 | viperdude | banik: I have a digium card connect to a E1 |
| 14:23.36 | banik | if i campare with quintum E1 and digium E1 which one will be better |
| 14:24.05 | banik | viperdude: in terms of performance |
| 14:24.51 | viperdude | dont know i use digium |
| 14:24.57 | pjz | banik: I've got a digium TE110P, though I'll admit it's plugged into a T1 instead of an E1 |
| 14:25.32 | banik | pjz: How is the performance? |
| 14:25.53 | pjz | banik: just fine; It's ona 3GHz P4 w/ 1GB RAM |
| 14:25.55 | banik | pjz: have you faced any problem? |
| 14:26.06 | banik | pjz: Thanks |
| 14:26.09 | pjz | banik: not with the hardware :) |
| 14:27.21 | seele_ | please help my queue freeze, after awhile lets pass calls soon to begin to pass them slowly , and my queue is full |
| 14:28.32 | banik | A box with 3GHz P4/ 2GB RAM, How many E1 it can handles? |
| 14:28.38 | *** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net) |
| 14:29.00 | pjz | banik: did you see the dimensioning page that gdh_ pointed out? |
| 14:29.11 | banik | watching |
| 14:30.54 | gdh_ | And while we're here, does http://pastebin.ca/262406 ring any bells (ahem) ? |
| 14:32.44 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:32.55 | banik | somebody worked with h323 & asterisk? |
| 14:33.30 | zoa | yes, it never worked the way it should :) |
| 14:33.43 | gdh_ | H.323 is just asking for a world of pain. |
| 14:34.20 | kristalino | does anyone here use asterisk with a wengo account ? |
| 14:34.34 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:34.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:34.55 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 14:35.11 | pjz | my asterisk server got rebooted last night and now every vmail box that had a custom greeting no longer has one... the files are there in /var/spool/asterisk/voicemail/default/, but they're not being used for some reason ; any ideas how to turn them back on if the file exists? |
| 14:35.37 | gdh_ | pjz: permissions? the files are readable by root only, but asterisk is now running as non-root? |
| 14:35.52 | Corydon76-home | pjz: check your formats line in voicemail.conf. Does it specify the format that exists? |
| 14:35.58 | pjz | gdh_: nope, everything's still owned by asterisk (which the server is running as) |
| 14:36.01 | *** join/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 14:36.10 | gdh_ | pjz: shame :/ just a wild guess.. |
| 14:36.22 | Corydon76-home | pjz: check the permissions on parent directories |
| 14:36.23 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:36.40 | WGFreewill | is dundi stable? for a dual pbx install with phones failing over to the second pbx |
| 14:36.45 | Corydon76-home | pjz: if the asterisk user can't view a parent directory, it doesn't matter what the permissions are |
| 14:38.46 | Corydon76-home | gdh_: change those commas to '|' |
| 14:39.20 | Corydon76-home | gdh_: commas in extensions.conf get translated to '|' on load. No such translation happens in the version of AEL in 1.2 |
| 14:39.36 | pjz | okay, perms are definitely fine |
| 14:39.37 | AK | WGFreewill with dns roundrobin? |
| 14:39.51 | *** join/#asterisk santiago (n=santiago@208.195.214.146) |
| 14:39.54 | WGFreewill | or device multiple server support |
| 14:40.37 | gdh_ | Corydon-w: :)) was just in the middle of trying that :D |
| 14:40.56 | gdh_ | Corydon-w: Thank you for that, though - I have this incredible ability to solve problems only after I've asked in a public forum |
| 14:41.04 | gdh_ | <slump> =) |
| 14:42.02 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 14:42.17 | pjz | format=wav49|gsm |
| 14:42.52 | AK | i've tested with a friend dnsroundrobin and dundi, and it worked, but no clue if it's stable |
| 14:42.55 | pjz | but it's not using the {busy,greet,unavail}.wav files |
| 14:43.53 | WGFreewill | There is this paper http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf |
| 14:44.08 | WGFreewill | SIP Agents Using Backup Registration Server, that is my case |
| 14:44.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:46.01 | pjz | Corydon76-home: any other ideas? |
| 14:47.30 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 14:47.42 | fourcheeze | hi, which module contains Playtones ? |
| 14:47.48 | awannabe | does anyone have some dial-plan strings that work good with the snoms? ive made my own and cant get it to work correctly! |
| 14:48.04 | fourcheeze | or is there a way to tell a channel to always play the tones rather than send the signals? |
| 14:48.16 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 14:48.24 | fourcheeze | awwanabe it depends a lot on where you are |
| 14:48.28 | *** join/#asterisk denon23532 (i=denon@synapse.subneural.net) |
| 14:48.31 | fourcheeze | awannabe: ^^ |
| 14:49.28 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 14:49.30 | awannabe | fourcheeze: yeah, i cant get them to work in conjunction, its very weird |
| 14:50.41 | mattfletcher | hello, i have a question on dialplan logic. can anyone help me? i find that if any phone is busy on a hunt group (or a dial command to multiple phones) that phone's busy message will be played, rather than trying the next phone. ideas anyone? |
| 14:50.49 | awannabe | they work by themselves, then once i start to add them together, it breaks |
| 14:50.53 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 14:54.43 | gdh_ | Corydon-w: Would this also be likely to fail under AEL in 1.2? switch(${MACRO_EXTEN:0:1}) { case.... case... etc.} |
| 14:55.06 | b11d | can anyone here recommend a good headset which is compatible with a poly 501? |
| 14:55.30 | pjz | b11d: the recruiter at our office likes her plantronics |
| 14:55.44 | awannabe | fourcheeze: you got a example i could look at and try to pick apart |
| 14:55.45 | b11d | i've heard those were good |
| 14:55.53 | b11d | do you know what model she uses? |
| 14:57.22 | heh_v_water | I havent had any luck whatsoever trying to compile iaxclient.. has anyone done it recently using the provided svn location? |
| 14:57.29 | awannabe | b11d: plantronics are great, they got about 300 differnet models though, heh |
| 14:58.16 | b11d | weak ! |
| 14:58.23 | mattfletcher | hello, i have a question on dialplan logic. can anyone help me? i find that if any phone is busy on a hunt group (or a dial command to multiple phones) that phone's busy message will be played, rather than trying the next phone. ideas anyone? |
| 14:58.30 | b11d | I mean, I'm glad they are a good brand.. i'll investigate them.. 300 models though.. dammit! |
| 14:58.43 | b11d | hunt groups.. are those still around? |
| 14:59.24 | awannabe | b11d: they got tons, wires, wireless, bluetooth adapter type, RF, all kinds |
| 14:59.25 | mattfletcher | well i mean agent queues i suppose |
| 14:59.47 | ManxPower | I use the Planronics M175 for phones with a 2.5mm jack |
| 15:00.25 | b11d | I guess I'll have to buy a few different models and let my end users make the pick. |
| 15:00.33 | *** join/#asterisk svenna__ (n=svenna@p548D23DB.dip0.t-ipconnect.de) |
| 15:00.47 | pjz | diffderent people like differnt kinds of ear hangers |
| 15:01.07 | ManxPower | The M175 supports over-the-ear and over-the-head |
| 15:01.34 | b11d | i'll have to go check that oot.. |
| 15:01.38 | ManxPower | good sound quality and has a mute button and volime congtrol |
| 15:01.56 | b11d | m175 eh.. |
| 15:02.10 | b11d | god headsets are so gay though |
| 15:02.12 | b11d | :P |
| 15:03.20 | b11d | hm.. under options for choosing a headset it says "Wireless" and "Cordless" -- wtf is the difference |
| 15:03.42 | zoa | haha |
| 15:05.02 | b11d | sigh.. volume control & mute options for poly 501's only give me 3 poor options |
| 15:05.35 | seele_ | http://pastebin.ca/262405 ? |
| 15:05.58 | b11d | codec issues |
| 15:05.59 | b11d | hah |
| 15:07.15 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 15:08.02 | seele_ | how can I solve this?? |
| 15:08.50 | bluemono | what's the correct command to edit the hosts file? |
| 15:09.03 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:09.21 | b11d | vi /etc/hosts |
| 15:09.21 | b11d | ? |
| 15:09.22 | awannabe | bluemono: its just a text file, /etc/hosts you mean? |
| 15:09.39 | bluemono | yeah i think so |
| 15:09.43 | b11d | ee = greatest editor ever, btw. |
| 15:10.29 | bluemono | i've done the netconfig and given the server a static local ip address |
| 15:10.45 | bluemono | but the server isn't on the network |
| 15:10.54 | b11d | set a default route? |
| 15:11.01 | b11d | is it plugged into the right port? |
| 15:11.02 | b11d | etc |
| 15:11.58 | bluemono | so i was thinking it's because it's not on the domain? |
| 15:12.16 | *** join/#asterisk dasuberdavid (n=edwardfa@199.227.185.35) |
| 15:12.45 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust4.leed.cable.ntl.com) |
| 15:14.00 | b11d | being part of a domain has nothing to do with basic IP connectivity |
| 15:14.11 | bluemono | ok |
| 15:14.15 | b11d | i mean, it may be possible that you're doing something unique there.. |
| 15:14.39 | b11d | where maybe domain authentication opens allows your box to traverse a firewall or something |
| 15:14.41 | b11d | but.. |
| 15:15.07 | b11d | what error do you get when you ping your gateway? |
| 15:15.29 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:15.31 | bluemono | from the asterisk box? |
| 15:15.41 | b11d | yes |
| 15:15.52 | bluemono | what i was trying to do was get the mail setup |
| 15:16.00 | b11d | ok.. take these questions elsewhere |
| 15:16.20 | bluemono | but this is an asterisk channel right? |
| 15:16.24 | b11d | this is a pbx support channel, not basic email & network connectivity support |
| 15:16.34 | bluemono | ok bud |
| 15:16.44 | b11d | but then again.. im certainly not the authority around here |
| 15:16.47 | b11d | so .. do what you will :) |
| 15:16.53 | bluemono | lol |
| 15:17.24 | bluemono | ok lets go down the pbx route |
| 15:17.34 | mercestes | bluemono: What distro are you using? |
| 15:17.44 | bluemono | CentOS |
| 15:17.48 | bluemono | 4.4 |
| 15:18.00 | mercestes | ... Ok, you know about the CentOS bug, right? |
| 15:18.08 | bluemono | eeek. |
| 15:18.11 | bluemono | no |
| 15:18.12 | b11d | a lot of people use CentOS eh.. |
| 15:18.15 | b11d | i need to look at that |
| 15:18.21 | mercestes | !centos |
| 15:18.33 | mercestes | Where is the bot? |
| 15:18.34 | b11d | ~centos |
| 15:18.42 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos |
| 15:18.43 | b11d | ? |
| 15:18.43 | b11d | hrm |
| 15:18.47 | bluemono | mercestes: what bug do you refer to |
| 15:18.48 | b11d | ah ha |
| 15:18.53 | mercestes | ~centos bug |
| 15:19.00 | Aurs | ~centosbug |
| 15:19.02 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 15:19.16 | mercestes | There. |
| 15:19.28 | b11d | damn linux |
| 15:19.37 | mercestes | bluemono: Give nullmailer a try. It's rediculously simple to setup and will work to deliver voicemail notifications |
| 15:19.54 | *** join/#asterisk vgster` (n=vgster@cpc2-ledn1-0-0-cust4.leed.cable.ntl.com) |
| 15:20.44 | bluemono | *pats jbot on the head* |
| 15:20.57 | bluemono | cheers aurs |
| 15:21.12 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:21.40 | Aurs | hello bluemono |
| 15:22.19 | bluemono | hi aurs, my heads throbbing :) i'm on information overload right now |
| 15:22.48 | b11d | thats the best.. |
| 15:22.55 | b11d | it also means its time to take a break / sleep |
| 15:23.48 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:25.29 | *** part/#asterisk banik (n=Main@203.208.196.140) |
| 15:26.15 | *** join/#asterisk doc9 (n=doc9@201.238.226.125) |
| 15:26.26 | doc9 | Hello |
| 15:27.05 | doc9 | please somebody have problem with TE110P and this message |
| 15:27.16 | doc9 | chan_zap.c:8207 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 |
| 15:27.19 | b11d | you want someone to have a problem? shame on you :) |
| 15:27.33 | b11d | No more alarm.. isnt that good? |
| 15:27.51 | doc9 | soo, exist other alarm |
| 15:27.58 | doc9 | WARNING[18268]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 15:28.27 | doc9 | i try with span=1,0,ccs,hbd3 1,1,ccs... |
| 15:28.36 | doc9 | signalling = pri_net |
| 15:28.38 | b11d | I wish I knew what to do to help you .. I dont.. |
| 15:28.38 | doc9 | pri_cpe |
| 15:28.39 | b11d | sorry |
| 15:29.00 | b11d | you got the correct encoding & framing info from the telco? |
| 15:29.10 | b11d | Cafe Del Mar is a good band.. |
| 15:29.11 | doc9 | yes is correct with telco |
| 15:29.13 | b11d | not that its related |
| 15:29.29 | b11d | hrm.. strange indeed doc9.. |
| 15:29.37 | b11d | I dont know.. I'm sorry! |
| 15:29.42 | doc9 | wait please |
| 15:29.45 | b11d | im not going anywhere |
| 15:30.10 | *** join/#asterisk DrAk0SX (n=luisjose@190.38.151.37) |
| 15:30.15 | doc9 | ok |
| 15:30.17 | doc9 | look |
| 15:30.29 | doc9 | i had a pc with other card TE110P |
| 15:30.34 | doc9 | and work ok |
| 15:30.38 | *** join/#asterisk vgster^ (n=vgster@cpc2-ledn1-0-0-cust4.leed.cable.ntl.com) |
| 15:30.49 | doc9 | and now replace this card for other new |
| 15:31.02 | mercestes | doc9: Maybe you should switch the cards back. |
| 15:31.06 | b11d | yeah |
| 15:31.10 | b11d | just to double-check |
| 15:31.17 | *** join/#asterisk _cleric_ (n=dacleric@p54822462.dip0.t-ipconnect.de) |
| 15:31.21 | doc9 | but this new card and using the same config that last card |
| 15:31.28 | doc9 | not work |
| 15:31.33 | doc9 | and only message is |
| 15:31.35 | b11d | doc.. it says this on asteriskguru |
| 15:31.35 | b11d | You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with hdparm (make your drivers slower) or disable that card, and take a new one. |
| 15:31.47 | b11d | from here: http://www.asteriskguru.com/tutorials/e1t1.html |
| 15:31.49 | doc9 | Nov 30 12:24:01 NOTICE[18268]: chan_zap.c:8207 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 |
| 15:31.52 | b11d | i dont know if you saw that or not |
| 15:32.03 | doc9 | yes |
| 15:32.20 | doc9 | i know how to config the te110p , te205p ,etc |
| 15:32.26 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:32.36 | b11d | yeah its not about how to config it.. its about troubleshooting the issues |
| 15:32.42 | doc9 | but i think that |
| 15:32.48 | doc9 | the card is bad |
| 15:32.54 | b11d | and if it we're config'd right.. then you'd not have these issues.. unless its a hardware issue, in which case, you're screwed |
| 15:32.56 | b11d | until you get a new card |
| 15:33.13 | doc9 | this I can be true |
| 15:33.26 | doc9 | b11d i have two card te110p |
| 15:33.35 | b11d | yep..; |
| 15:33.38 | doc9 | and the two show the same problem |
| 15:33.45 | b11d | oh.. i thought you said one worked |
| 15:34.28 | doc9 | yes the card work is a 3rd card |
| 15:34.32 | b11d | ohh |
| 15:34.33 | doc9 | is another |
| 15:34.52 | doc9 | i had 3 card te110p |
| 15:35.03 | doc9 | the first works ok, no problem |
| 15:35.27 | *** join/#asterisk Sedorox (n=brandon@smartserv/cna/Sedorox) |
| 15:35.34 | syzygyBSD | hmm, were there connection problems last night? |
| 15:35.35 | doc9 | i buy two more and the two no works with the same computer , the same telco & the same config zaptel/zapata |
| 15:36.06 | doc9 | i use the last zaptel and libpri |
| 15:36.23 | doc9 | make with make linux & try with make linux26 |
| 15:36.40 | doc9 | i use kernel 2.6.xx in Debian sarge |
| 15:36.57 | b11d | i would call Digium and ask them.. maybe the cards have different firmware? |
| 15:37.07 | AK | anyone has already try astmanproxy? |
| 15:37.07 | doc9 | mmm |
| 15:37.25 | doc9 | and this can be a problem? |
| 15:37.35 | b11d | potentially.. |
| 15:37.42 | b11d | i mean.. you ARE experiencing problems that dont make sense.. |
| 15:37.43 | b11d | so.. |
| 15:37.44 | b11d | maybe |
| 15:38.04 | doc9 | mmm |
| 15:38.34 | doc9 | and if it was the problem of the firmware |
| 15:38.37 | Zefk | I'm running asterisk 1.2.13 on FC5 with B410P board. I use only port 1 of the board. Inbound and outbound calls via BRI port is working. I have the following message when asterisk starts: "mISDN dss1 fromup without proc pr=10180 dinfo(0)". Could be anything wrong? |
| 15:38.51 | doc9 | then I have to change the cards |
| 15:39.08 | doc9 | b11d :) |
| 15:39.20 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 15:39.31 | b11d | hehe |
| 15:39.41 | riddlebox | in 1.4, has the gotoif changed? |
| 15:39.51 | ManxPower | riddlebox: it should not have |
| 15:39.51 | b11d | you wouldnt have to change them, maybe digium would have an executable you'd just run to flash them.. |
| 15:40.00 | b11d | but I dunno.. the firmware argument is pretty thin.. |
| 15:40.06 | ManxPower | contact dihium |
| 15:40.07 | b11d | i'd still call Digium. |
| 15:40.12 | ManxPower | they do provide support for their cards |
| 15:40.33 | b11d | yeah it might not be a free call from where doc9 is from though |
| 15:40.40 | b11d | maybe you shoudl email them :) |
| 15:40.58 | riddlebox | ManxPower, I have this line but every call comes in and just goes to 20 ,gotoif($[{CALLERID(num)} = xxxxxxxxxx]?20:10) |
| 15:41.32 | doc9 | ok thank you b11d & ManxPower |
| 15:41.50 | b11d | np |
| 15:41.53 | b11d | hope it helps |
| 15:42.06 | awannabe | anyone have any sample dial-plan regex for snoms? |
| 15:43.01 | doc9 | ok bye |
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| 15:48.33 | fourcheeze | awannabe: no, I tend to remove the dial plans from snoms and let people press the OK key |
| 15:48.51 | fourcheeze | anyone know how I get to use Playtones? |
| 15:49.07 | fourcheeze | i.e. is it with some module? |
| 15:49.13 | awannabe | yeah, these people want it, heh |
| 15:49.21 | fourcheeze | weird |
| 15:49.22 | fourcheeze | ;-) |
| 15:50.22 | fourcheeze | awannabe: does .* work? |
| 15:50.47 | awannabe | .*? in the dialplan string? |
| 15:51.30 | awannabe | see i just had the auto dial after 2 seconds on, but i cant do that, cause when you have two calls on hold and go to transfer one, it bridges the two calls, heh |
| 17:04.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:04.30 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 17:08.26 | sam33 | is there anybody that make it work (asterisk + fax)? |
| 17:08.47 | ManxPower | sam33: yes. Use a PSTN line. |
| 17:09.05 | ManxPower | For most of our Asterisk deployments (5 or 6 so far) we have the fax line outside of Asterisk |
| 17:09.14 | ManxPower | solves all these issues. |
| 17:10.42 | Strom_C | sam33: I've also had success in a pure-tdm environment bridging from a channel bank to a PRI on the same quad-span t1 card |
| 17:12.49 | ManxPower | Strom_C: do you think he is going to be an arm flapper? |
| 17:13.13 | Strom_C | ....a what? |
| 17:13.20 | Strom_C | *puzzled look* |
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| 17:14.14 | danbrwn | i am confused about how asterisk displays data on phones, could someone explain this? |
| 17:14.20 | ManxPower | An arm flapper. Someone that thinks they can fly by flapping their arms. Regardless that everyone tells them that you can't fly by flapping your arms. The arm flapper things that is he tries hard enough he can do the implossible. Fortuantly they are reasonably harmless, but they can be damn funny to watch. |
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| 17:14.54 | ManxPower | Someone that spends a week trying to get fax working over GSM would be an arm flapper |
| 17:15.20 | ManxPower | danbrwn: Asterisk does not display information on phones. Phones do that. |
| 17:15.26 | Strom_C | ManxPower: heheh |
| 17:15.36 | Qwell[] | ManxPower: It does on chan_skinny |
| 17:15.44 | Qwell[] | display information on the phones, that is |
| 17:15.50 | ManxPower | Qwell |
| 17:16.25 | ManxPower | Qwell: It very well may SEND info to the phone, but I doubt Asterisk has anything to do with actually displaying it. |
| 17:17.05 | Qwell[] | ManxPower: No, it does. There are 3 messages that can change the display directly |
| 17:17.06 | brian | :( |
| 17:17.50 | brian | netsplities |
| 17:17.50 | danbrwn | ManxPower: i called tech support at polycom and asked what gets displayed on the display and they said the menus and such were configured by asterisk, through what mechanism |
| 17:17.50 | Qwell[] | brian: That was nothing |
| 17:17.50 | Qwell[] | danbrwn: That's a lie, I'm fairly certain |
| 17:17.50 | brian | HI |
| 17:17.51 | brian | HOW ARE YOU QWELL[] |
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| 17:17.54 | ManxPower | danbrwn: they lied. |
| 17:17.54 | Qwell[] | well...not a lie |
| 17:17.55 | Qwell[] | They're just wrong |
| 17:17.55 | ManxPower | it is configured via the polycom config fileles |
| 17:17.55 | Qwell[] | The menus (directory, etc) is elsewhere |
| 17:18.13 | brian | There's a difference between a lie and bullshit. |
| 17:18.13 | ManxPower | my typing really sucks when I have a feaver |
| 17:18.14 | brian | But both smell the same. |
| 17:18.14 | Qwell[] | ManxPower: fever :p |
| 17:18.20 | brian | ManxPower: Ebony...fever? |
| 17:18.26 | ManxPower | I think that if you ask a vendor about their product and they tell you something wrong you can only call it a lie. |
| 17:18.33 | danbrwn | so, who creates the configuration |
| 17:18.41 | Qwell[] | ManxPower: whatever it is, it's wrong |
| 17:18.46 | brian | i create the configuration |
| 17:18.54 | *** join/#asterisk LordBacon (n=frb@mail1.dahnyoga.com) |
| 17:18.58 | danbrwn | brian: how |
| 17:19.00 | ManxPower | danbrwn: download the polycom admin guide. read it. Then read it again |
| 17:19.02 | brian | i'm joking |
| 17:19.03 | brian | :( |
| 17:19.16 | brian | i don't even know what configuration you're talking about |
| 17:19.26 | LordBacon | greetings, I'm a newb, what's the quickest way for me to get asterisk configured for an office environment? |
| 17:19.34 | ManxPower | The display of menus can be changed via the localazation features. |
| 17:19.43 | syzygyBSD | LordBacon: hire a consultant |
| 17:19.45 | brian | LordBacon: RUN! |
| 17:19.57 | brian | :P |
| 17:20.02 | brian | i mean |
| 17:20.02 | Qwell[] | LordBacon: If you don't mind spending some money, syzygyBSD is right. |
| 17:20.06 | brian | Not run... |
| 17:20.06 | LordBacon | can't hire a consultant, I've got a lot of linux experience, just never touched VoIP |
| 17:20.09 | ManxPower | LordBacon: Asterisk is a highly complex system. |
| 17:20.15 | Qwell[] | otherwise, there it quite a bit of documentation that you can read |
| 17:20.17 | Qwell[] | ~book |
| 17:20.18 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:20.18 | Qwell[] | ~docs |
| 17:20.22 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:20.23 | *** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 17:20.24 | ManxPower | ~mailinglist |
| 17:20.32 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:20.35 | brian | LordBacon: You can leave security holes in your PBX if you aren't careful mister |
| 17:20.50 | Qwell[] | or, and, also? |
| 17:20.54 | LordBacon | I found documentation, but there are like 42 different "guis" to configure asterisk |
| 17:20.57 | syzygyBSD | well, you asked the quickest way... Asterisk is fairly easy to setup, read the book ^^^ first link from jbot |
| 17:21.00 | Qwell[] | That's like triply redundant |
| 17:21.02 | brian | LordBacon: poke poke poke i poke you poke |
| 17:21.04 | Qwell[] | (file, 5 for 5) |
| 17:21.27 | brian | I never read the documentation, syzygyBSD |
| 17:21.30 | brian | I just figure it out myself. |
| 17:21.32 | brian | It's funner. |
| 17:21.41 | syzygyBSD | um.. never? |
| 17:21.47 | syzygyBSD | not even voip-info? |
| 17:21.48 | brian | If I don't know how to do something I bug people on IRC. |
| 17:21.53 | brian | Or I search Google. |
| 17:21.54 | brian | :D |
| 17:21.58 | Qwell[] | file: I'll be at 10 in no time |
| 17:22.07 | syzygyBSD | well, google links to "documentation" |
| 17:22.07 | LordBacon | I bug people on irc to give me the good docs, since google turns up way too many |
| 17:22.16 | brian | i mean the main documentation |
| 17:22.24 | file | Qwell[]: your use of the myriad of words in the english language is truly perplexing... |
| 17:22.30 | syzygyBSD | LordBacon: voip-info.org |
| 17:22.33 | Qwell[] | file: omg, you b0rked it |
| 17:22.38 | danbrwn | LordBacon: There seems to be a lot of canned packages just waiting to be bought, some will sell you everything you need and have gui interfaces I think. If i just wanted to set up a one time thing i think id look at that |
| 17:22.55 | brian | LordBacon: you should give me your money |
| 17:22.55 | syzygyBSD | well, I have never read the book either, but voip-info is very very useful |
| 17:22.56 | LordBacon | the issue with canned, is that I have to put it on my current gateway |
| 17:23.07 | LordBacon | brian: I have no budget for this, only time |
| 17:23.09 | brian | LordBacon: I'm from nigeria can you pay me using western union |
| 17:23.11 | brian | :x |
| 17:23.31 | syzygyBSD | lol brian |
| 17:23.43 | brian | :-D |
| 17:23.55 | LordBacon | I'm seriously tempted to just create a virtual machine and run trixbox |
| 17:24.03 | brian | trixbox? |
| 17:24.11 | brian | you're a jerk |
| 17:24.13 | brian | :( |
| 17:24.22 | danbrwn | LordBacon: you might want to read the information cited earlier, it might not be a good idea to try to shoehorn asterisk into an existing system. It seams that many think it is intensive on processing |
| 17:24.27 | brian | i'm leaving you for the FreeBSD daemon LordBacon :( |
| 17:24.33 | [TK]D-Fender | Poor little GUI chumps... don't expect much help here.... |
| 17:24.33 | LordBacon | eh? |
| 17:24.47 | brian | Use FreeBSD sucka |
| 17:25.05 | danbrwn | [TK]D-Fender: dont expect help, you got that right! |
| 17:25.06 | syzygyBSD | [TK]D-Fender: hmm, is it sad I don't remember the last time I saw a linux gui? |
| 17:25.16 | LordBacon | danbrwn: processor isn't a problem, the system I'm shoehorning just runs squid, and postfix |
| 17:25.26 | brian | syzygyBSD: hooray for BSD |
| 17:25.40 | brian | i'm running linux on this machine though |
| 17:25.45 | brian | too lazy to switch |
| 17:25.54 | brian | i installed linux when i was young and stupid! |
| 17:25.55 | syzygyBSD | lol.. well, I don't really run BSD on many production servers, just happens to be my nick from years ago |
| 17:26.12 | brian | Linux kind of sucks though. |
| 17:26.15 | syzygyBSD | back with freebsd 4.5 was bleeding edge |
| 17:26.21 | [TK]D-Fender | syzygyBSD: Dunno. Depends what you imply by "linux GUI". I never use "X" on my servers persaonlly, just SSH to bash. Except on my home PVR server, but I guess that'd kinda HAVE to be an exception :) |
| 17:26.23 | syzygyBSD | s/with/when |
| 17:26.32 | brian | synthetiq: I remember when FreeBSD 3 came out :D |
| 17:26.40 | [TK]D-Fender | syzygyBSD: By rest assured its configured by text files direct :D |
| 17:26.42 | Qwell[] | [TK]D-Fender: bah |
| 17:26.47 | Qwell[] | [TK]D-Fender: mplayer -vo fb :P |
| 17:26.51 | syzygyBSD | [TK]D-Fender: :P |
| 17:27.06 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:27.06 | brian | syzygyBSD: what do you run WINDOWS??? |
| 17:27.06 | [TK]D-Fender | Qwell[]: OH? yay! GUI free! |
| 17:27.19 | *** part/#asterisk danbrwn (n=danny@216.77.58.40) |
| 17:27.21 | [TK]D-Fender | Qwell[]: Assuming it works with my TV-out :) |
| 17:27.25 | syzygyBSD | brian: that I do on my desktop |
| 17:27.42 | Qwell[] | [TK]D-Fender: if you can see your console, it should work |
| 17:27.43 | file | [TK]D-Fender: ! |
| 17:27.45 | brian | synthetiq: :( |
| 17:27.46 | Qwell[] | I've never personally used it, but hey |
| 17:27.56 | Qwell[] | [TK]D-Fender: also, try...umm...what's the option |
| 17:28.00 | Qwell[] | -vo aa ? |
| 17:28.12 | syzygyBSD | hah, ascii art |
| 17:28.15 | Qwell[] | :D |
| 17:28.20 | syzygyBSD | at least do the color ascii art |
| 17:28.26 | file | Qwell[]: fancy word, right now! |
| 17:28.27 | Qwell[] | How do you do that? |
| 17:28.32 | Qwell[] | file: nah |
| 17:28.52 | syzygyBSD | don't remember off the top of my head, something like ac though |
| 17:29.21 | syzygyBSD | caca |
| 17:31.29 | brian | syzygyBSD: you probably run vista you cruel person |
| 17:31.40 | syzygyBSD | nope |
| 17:32.12 | brian | don't lie to me! |
| 17:32.19 | syzygyBSD | however when I have to manage clients servers I need some tools that are only on windows |
| 17:32.52 | brian | windows vista was released in india |
| 17:33.07 | brian | the indians are going to their 2 weeks courses to become experts of windows vista now |
| 17:33.16 | syzygyBSD | the only economy that could afford it |
| 17:33.55 | brian | lots of outsourcing shops will buy vista |
| 17:34.19 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 17:34.20 | *** join/#asterisk sloth (n=josh@mail.rssalesco.com) |
| 17:36.31 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-f22db975d42354b7) |
| 17:41.20 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 17:44.31 | af_ | what is xml phone book download on gxp2000? |
| 17:46.12 | *** join/#asterisk Math` (n=privmath@modemcable184.59-131-66.mc.videotron.ca) |
| 17:49.08 | *** join/#asterisk saftsack (n=saftsack@pD9E06FF8.dip.t-dialin.net) |
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| 17:52.55 | *** join/#asterisk ManxPowe1 (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 17:53.54 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 17:54.39 | *** join/#asterisk Shazaumm (i=Alexandr@201.35.254.22) |
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| 17:56.07 | [Wiebel] | hmmz |
| 17:56.23 | [Wiebel] | what can it be that If a incomming caller hangs up , and asterisk doesnt disconnect the line? |
| 17:56.29 | [Wiebel] | the other way around works fine |
| 18:00.10 | Strom_C | let me guess - you have analog trunks |
| 18:00.36 | brian | analog? |
| 18:00.37 | brian | ewwww |
| 18:01.35 | brian | if you call anywhere with a analog trunk with skype skype will drop you a dialtone |
| 18:01.37 | awannabe | mmmm dirty FXO ports |
| 18:01.38 | toggy | can anyone help me out ? im trying to get chan_capi installed |
| 18:01.49 | brian | toggy: try harder |
| 18:01.53 | toggy | hehe |
| 18:02.04 | brian | you can do it toggy! |
| 18:03.18 | [TK]D-Fender | file: ! ! ! |
| 18:03.38 | *** join/#asterisk Cybix (n=Sobaka@82-204-26-196.dsl.bbeyond.nl) |
| 18:03.38 | file | [TK]D-Fender: is it unusually warm up your way? |
| 18:03.43 | Cybix | Hello |
| 18:03.45 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 18:03.46 | [TK]D-Fender | file: 18 or so :) |
| 18:03.49 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:03.54 | *** join/#asterisk diablopico (n=diablopi@ip68-101-129-147.sd.sd.cox.net) |
| 18:03.55 | [TK]D-Fender | file: Tomorrow... -2 :( |
| 18:03.55 | teknoprep | what is a good 1-800 service? |
| 18:04.11 | *** join/#asterisk ronmac (n=rmcdanie@216.109.9.131) |
| 18:04.11 | file | [TK]D-Fender: the weather is so very very weird... haven't had any snow storms yet either! |
| 18:04.14 | teknoprep | or 1-888.. i want to call forward a 1800 or 1888 number to a number of my choice |
| 18:04.19 | toggy | well i get som f... errors on the making of the capi trunk /usr/include/asterisk dir not found |
| 18:04.21 | diablopico | hello..... can anyone help with aquireing codec g723.1 |
| 18:04.26 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 18:04.46 | ronmac | Can anyone tell me what the default username and pass is for AsteriskNow?? |
| 18:05.05 | Cybix | I have made a script (in extensions.conf) for incoming calls. The script works fine, only I'd like to print a message on the disply of the sip phone where the call is forwarded to. Is it possible, if so what command can I use for that? |
| 18:05.26 | brian | ronmac: It's boomboomnowmakeitsaywayohhh |
| 18:05.44 | [TK]D-Fender | ronmac: Please read the channel topic. |
| 18:06.05 | brian | ronmac: yeah, read the topic man!! |
| 18:06.48 | ronmac | sorry, i'll jump to the GUI site i guess... |
| 18:06.55 | CunningPike | No more caffeine for brian |
| 18:06.58 | [Wiebel] | Strom_C: no sip |
| 18:07.05 | [Wiebel] | Strom_C: and it's only with one of the 3 sip lines I have |
| 18:07.24 | Strom_C | [Wiebel]: what's on the other end of the sip trunks? |
| 18:07.33 | [Wiebel] | Strom_C: budgetphone :) |
| 18:07.39 | [Wiebel] | and voipbuster |
| 18:07.43 | Strom_C | oh christ |
| 18:07.44 | [Wiebel] | but budgetphone is the one with issues |
| 18:07.53 | [Wiebel] | that's my main nr |
| 18:07.55 | *** part/#asterisk ronmac (n=rmcdanie@216.109.9.131) |
| 18:08.03 | [Wiebel] | voipbuster is only used for outgoing , free, calls |
| 18:08.14 | Strom_C | the solution in this case is "get an ITSP that doesn't completely suck" |
| 18:08.18 | *** join/#asterisk rr-- (n=rr@cpe-66-69-217-206.austin.res.rr.com) |
| 18:08.26 | [Wiebel] | why does budget phone suck? |
| 18:08.56 | Strom_C | the law of voip: choose two of the following options: (1) cheap (2) reliable (3) high-quality |
| 18:09.13 | [Wiebel] | budgetphone isnt that cheap |
| 18:09.21 | [Wiebel] | allthough the name may suspect otherwise :P |
| 18:09.30 | [Wiebel] | it's pretty default |
| 18:09.46 | [Wiebel] | and it worked before |
| 18:09.49 | Strom_C | if they're expensive /and/ unreliable, then RUN |
| 18:10.02 | [Wiebel] | it work for allmost a year without issues |
| 18:10.09 | [Wiebel] | Now I have this , rather small, issue |
| 18:10.15 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:10.24 | [Wiebel] | And i'm not sure the issues is a budgetphone issue |
| 18:10.27 | [Wiebel] | could be local |
| 18:10.34 | Strom_C | did you change anything at all locally? |
| 18:10.50 | [Wiebel] | dtmf setting in sip.conf |
| 18:10.52 | [Wiebel] | that's it |
| 18:11.04 | [Wiebel] | but I reinstalled my linksys with openwrt |
| 18:11.15 | [Wiebel] | so that's to much of a coinsidence |
| 18:11.25 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 18:11.29 | Strom_C | what were you running it on before? |
| 18:11.33 | [Wiebel] | dd-wrt :) |
| 18:11.52 | hmmhesays | asterisk dp needs an if then type statement |
| 18:11.54 | Strom_C | oy. |
| 18:12.55 | FuriousGeorge | hey all |
| 18:12.55 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:13.11 | Dr-Linux | is Asterisk2billing program is opensource? |
| 18:13.18 | *** join/#asterisk |Vulture| (n=Vulture@101.222.121.70.cfl.res.rr.com) |
| 18:13.26 | rr-- | is it worth the extra $300 to get hardware echo cancellation on a sangamo A2xxxx card ... can't asterisk or whatever sip server do the echo cancellation in software? |
| 18:14.38 | [Wiebel] | hmmm |
| 18:14.46 | [Wiebel] | debugging doesnt show anything when the remote side hangs up the phone |
| 18:14.50 | [Wiebel] | that's rather odd |
| 18:14.59 | brian | rr--: obviously if it's in the hardware that means less work for the software |
| 18:15.22 | Cybix | I bought some phonenumers from budgetphone, I use it for inbound calls, works perfect for me and it costs EUR 10,- per year/per number. Doesn't sound too expensive to me |
| 18:15.34 | brian | budgetphone? |
| 18:15.40 | brian | how many inbound channels cy3o3 |
| 18:15.43 | brian | Cybix i mean |
| 18:16.14 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 18:17.12 | [TK]D-Fender | rr--: Yes, its entirely worth it. Try without and see how well Zaptel fares and oy may wish you just bought it right from the start. |
| 18:19.09 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 18:19.17 | *** join/#asterisk mroberto (n=root@S0106004063d8e527.ed.shawcable.net) |
| 18:20.15 | mroberto | I am looking for a fxo card analog simple and works good ?? Cheap too what do you guys recommend? |
| 18:21.08 | [TK]D-Fender | mroberto: Analog. Good. Cheap. Pick TWO. |
| 18:21.23 | *** part/#asterisk LordBacon (n=frb@unaffiliated/frb) |
| 18:21.33 | [TK]D-Fender | mroberto: If you want something decent its going to cost a bit. |
| 18:21.36 | mroberto | TWO? |
| 18:22.44 | mroberto | Well lets say I am not running asterisk for a phone system other than call users couple days before the event they register and play a msg |
| 18:22.50 | mroberto | So I am like using 10% of asterisk power |
| 18:23.13 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:25.26 | *** join/#asterisk KeRneL (n=Zengin@88.241.108.232) |
| 18:26.41 | Strom_C | here's a nub question - when setting up a queue with rrmemory, is there an option which will eliminate the delay between one member timing out and the next member being called? |
| 18:26.49 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 18:27.02 | CunningPike | mroberto: You're using 100% of your FXO port's power - that's why you need to pay for a good one |
| 18:28.31 | mroberto | So what would be a good FXO card? |
| 18:28.49 | mroberto | Also anybody interested in doing custom work for asterisk that I need done. |
| 18:29.13 | shellshark | mroberto: what kind of custom work? |
| 18:29.43 | Aurs | Strom_C: perhaps retry=0? |
| 18:30.20 | Strom_C | no, i've already got that one set |
| 18:30.31 | CunningPike | mroberto: I'm inclined to use a good external gateway (SPA-3000, for example) over a card |
| 18:31.29 | mroberto | Shellshark: Well I not sure how to do what I need to do but it's sorta like a reminder system for people |
| 18:31.43 | mroberto | CunningPike : Why would you recoomend something external |
| 18:31.59 | CunningPike | mroberto: Easier to setup/replace etc |
| 18:32.06 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-183.mtco.com) |
| 18:32.08 | *** join/#asterisk alexis101 (n=as@ip216-239-93-50.vif.net) |
| 18:32.13 | shellshark | mroberto: check messages please |
| 18:32.17 | shellshark | zmef420: heya |
| 18:32.23 | alexis101 | hi , is there anyone that can help me with a little call transfer problem ? |
| 18:33.00 | mroberto | ok thank you |
| 18:35.16 | alexis101 | i can't make a supervised call transfer this is what i got in my asterisk console when i try: http://pastebin.ca/262594 |
| 18:37.47 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.165) |
| 18:38.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:45.12 | *** part/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 18:47.32 | heh_v_water | just installed trixbox to see what it does.. interesting little setup.. seems somewhat insecure though |
| 18:47.35 | *** join/#asterisk AuPix (n=root@adsl-04-85.abel.net.uk) |
| 18:52.14 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 18:55.22 | *** join/#asterisk ucfMethod (n=ucfmetho@c-69-143-112-70.hsd1.va.comcast.net) |
| 18:55.24 | ucfMethod | hey |
| 18:56.34 | *** join/#asterisk savoy (n=chatzill@tomcat.celinaisd.com) |
| 18:57.49 | *** join/#asterisk alerios (n=alerios@69.79.145.100) |
| 19:01.37 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 19:02.14 | b11d | boring |
| 19:02.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-37-212.red.bezeqint.net) |
| 19:02.53 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 19:05.22 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 19:07.42 | syzygyBSD | hmm, so I have 1 credit to have allison record something that has to be used today, any ideas? |
| 19:08.21 | *** join/#asterisk tim0123 (n=cash247@24-182-105-104.dhcp.ftwo.tx.charter.com) |
| 19:08.21 | ucfMethod | whats the website you used to contact her.. |
| 19:08.37 | syzygyBSD | thevoice.digium.com |
| 19:08.38 | tim0123 | Whats up everybody |
| 19:08.39 | rob0 | thevoice ? |
| 19:08.42 | syzygyBSD | ya |
| 19:08.44 | ucfMethod | nice... thanks |
| 19:08.55 | syzygyBSD | last day to use credits on that site... |
| 19:09.21 | ucfMethod | is she closing shop? |
| 19:09.21 | tim0123 | Anybody ever use hudlite |
| 19:09.21 | syzygyBSD | you can't buy any more off there, but you can get them from the digium store |
| 19:09.23 | rob0 | Weasels have eaten her voice. |
| 19:09.48 | syzygyBSD | already got one about Weasles and a phone system |
| 19:09.54 | ucfMethod | "this is not the website you were looking for" |
| 19:09.55 | syzygyBSD | and one about a mad donkey |
| 19:10.10 | syzygyBSD | ohh.... could be a good one... |
| 19:10.32 | syzygyBSD | nothing to see hear, keep moving |
| 19:13.00 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 19:13.10 | syzygyBSD | "you have been charged for a collect call from the county jail. to get charged twice press 1, to connect press 2" |
| 19:13.44 | ucfMethod | I have been converting old movie clips to gsm ... funny ones from Alien, Terminator, HAL 9000 etc.... |
| 19:13.50 | ucfMethod | syzygyBSD: haha |
| 19:14.05 | rob0 | http://www.theivrvoice.com/ her own site |
| 19:14.22 | syzygyBSD | ya, but i haven't found out how to get IVR recordings off there |
| 19:15.54 | Strom_C | ucfMethod: gsm? yech! |
| 19:15.57 | Strom_C | :) |
| 19:16.31 | *** part/#asterisk tim0123 (n=cash247@24-182-105-104.dhcp.ftwo.tx.charter.com) |
| 19:16.33 | syzygyBSD | ya, 8000 mono wav is so much better... |
| 19:17.08 | *** join/#asterisk blepsoaf (n=newbie@454a200a.cst.lightpath.net) |
| 19:17.17 | ucfMethod | i just figured since all the rest were gsm, i would keep it the same. |
| 19:17.53 | syzygyBSD | I keep mine in wav so I know which are mine and which are default |
| 19:18.02 | blepsoaf | hello all, does anyone use polycom telephones? If so, when using a 4 digit extension in the sip.conf, when making an outbound call how do you fill in the rest of the number IE xxx-xxx-extension. |
| 19:18.04 | Strom_C | i keep mine in a separate directory so I know which are mine :) |
| 19:18.09 | *** join/#asterisk TonyM_ (n=TonyM@softins.claranet.co.uk) |
| 19:18.26 | Strom_C | blepsoaf: that question doesn't make sense. |
| 19:18.31 | ucfMethod | blepsoaf: say again... i have 35+ Polycom 501's |
| 19:18.40 | syzygyBSD | blepsoaf: exten => _XXXX,1,Dial(zap/1/345${EXTEN}) |
| 19:18.45 | ucfMethod | blepsoaf: but i dont understand your question |
| 19:20.00 | syzygyBSD | wait.. you mean for caller id? |
| 19:20.18 | blepsoaf | well I guess in order to dial extension to extension you have to have the caller ID as "John Doe" <1234> - but when making an outbound call to the PSTN it will show 1234 |
| 19:20.33 | blepsoaf | yes sorry, I shold have put caller ID :P |
| 19:21.08 | blepsoaf | making the redial's and call lists easier to use |
| 19:21.31 | *** join/#asterisk anthonyc (i=anthony@fl-69-68-136-133.sta.embarqhsd.net) |
| 19:21.38 | anthonyc | Hi Qwell :) |
| 19:21.40 | anthonyc | Heh |
| 19:21.45 | blepsoaf | should I just re-write the caller ID in an outbound dial macro? |
| 19:21.47 | syzygyBSD | exten => _NXXXXXXXXX,1,Set(Callerid(number)=801333${CALLERID(NUMBER)}) |
| 19:21.51 | syzygyBSD | or something... |
| 19:21.55 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-28-34.telkomadsl.co.za) |
| 19:22.25 | blepsoaf | but what happens when you have XXX-956 && XXX-434 exchanges |
| 19:22.30 | syzygyBSD | or just set the callerid correctly in sip.conf so internal calls have the whole number |
| 19:22.46 | X-Gen | hey freaks |
| 19:22.58 | syzygyBSD | hey Gen-X |
| 19:23.01 | *** join/#asterisk frenzy (n=frenzy@196.46.104.215) |
| 19:23.12 | frenzy | where do I change the sender email for voicemail? |
| 19:23.34 | syzygyBSD | voicemail.conf the general section |
| 19:23.40 | frenzy | currently sends as asterisk@HOSTNAME which is boucning as that email doesnt exist... |
| 19:23.44 | blepsoaf | syzygyBSD: BUT, when redialing - if the whole number is there it will fail because of not dialing 9 & would also use an unnecessary zap channel |
| 19:23.44 | syzygyBSD | serveremail = admin@mybox.com |
| 19:23.59 | frenzy | hmmm wired.. I have that configured but doenst use that |
| 19:24.06 | syzygyBSD | blepsoaf: so fix your extensions.conf so that isn't a problem |
| 19:24.20 | frenzy | wierd** |
| 19:24.51 | syzygyBSD | is it in the general section? |
| 19:25.00 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 19:25.16 | syzygyBSD | frenzy: are you using sendmail? |
| 19:25.20 | *** join/#asterisk andresmujica (n=andresmu@201.244.196.229) |
| 19:25.32 | frenzy | nop |
| 19:25.43 | syzygyBSD | then what? |
| 19:25.57 | frenzy | am piping from sendmail to my mail server application |
| 19:25.59 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:26.35 | frenzy | btw i have freepbx installed... |
| 19:26.46 | blepsoaf | hmm I guess I'm not sure how I would do that then, IE how to make asterisk understand the call is originating from SIP and to dial direct vs dialing outbound to PSTN |
| 19:26.48 | frenzy | anychance it could be overidding settings? |
| 19:26.55 | syzygyBSD | oh.. then I have no clue, I just know how asterisk works |
| 19:27.29 | syzygyBSD | blepsoaf: any chance you can pastebin your extensions.conf? |
| 19:27.43 | blepsoaf | of course, thanks for taking the time to help |
| 19:28.04 | rob0 | frenzy: just a WAG here, but did you restart after the change to voicemail.conf ? |
| 19:28.22 | anthonyc | anyone here have time to work on a VOIP box? |
| 19:28.24 | ucfMethod | frenzy: a reload from asterisk console would work too |
| 19:28.28 | anthonyc | from scratch, need a simple incoming and outgoing # |
| 19:28.33 | anthonyc | and follow-me |
| 19:28.34 | anthonyc | for a company |
| 19:28.42 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 19:28.52 | ucfMethod | anthonyc: contract work? |
| 19:29.28 | anthonyc | not high budget |
| 19:29.28 | anthonyc | :/ |
| 19:29.35 | anthonyc | need a dedicated server? heh |
| 19:29.48 | syzygyBSD | what do you mean follow-me? |
| 19:30.27 | anthonyc | For billing press 1, for sales press 2 |
| 19:30.33 | anthonyc | Need a good professional voice etc |
| 19:30.33 | ucfMethod | anthonyc: linear dialing? |
| 19:30.38 | anthonyc | Im not sure what it is |
| 19:30.39 | Strom_C | that's IVR, not follow-me |
| 19:30.41 | anthonyc | Its setup and working |
| 19:30.45 | syzygyBSD | theivrvoice.com |
| 19:30.50 | anthonyc | but the got that was working on it |
| 19:30.53 | anthonyc | got fired. |
| 19:30.55 | ucfMethod | anthonyc: call desk first, then call cell, then call home ????? |
| 19:30.58 | anthonyc | and I dont know where to start |
| 19:31.12 | anthonyc | ucfMethod that could be an option |
| 19:31.17 | anthonyc | but Im talking about ivr i guess |
| 19:31.22 | anthonyc | may I PM you? |
| 19:31.25 | syzygyBSD | ~book |
| 19:31.33 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:31.33 | syzygyBSD | ~docs |
| 19:31.35 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 19:31.53 | frenzy | restarted asterik... (Y) |
| 19:31.54 | syzygyBSD | anthonyc: those will get you started, unless you are looking for a contractor |
| 19:32.55 | *** join/#asterisk AdmoIRC (n=Miranda@CPE-65-27-25-141.kc.res.rr.com) |
| 19:33.51 | blepsoaf | http://pastebin.ca/262648 |
| 19:34.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:35.12 | hmmhesays | i wish if statements could take regular expressions |
| 19:35.51 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:37.18 | syzygyBSD | blepsoaf: add "include => incoming_context" to your us-outbound-dialing context |
| 19:38.44 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-b07756a9f91e90ee) |
| 19:39.38 | bradoaks | syzygyBSD: that last link for astmasters, looks like spam. |
| 19:39.45 | bradoaks | from jbot. |
| 19:40.00 | syzygyBSD | well I don't have control over jbot... |
| 19:40.18 | bradoaks | does any *one*, or do we all? |
| 19:40.36 | syzygyBSD | I think just the ops |
| 19:40.40 | syzygyBSD | ~help |
| 19:43.22 | bradoaks | okay, i was able to teach it to not return that spam link. |
| 19:43.36 | [TK]D-Fender | jbot : no, documentation is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:43.37 | jbot | I think you lost me on that one, [TK]D-Fender |
| 19:43.49 | [TK]D-Fender | jbot : documentation is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:43.51 | jbot | [TK]D-Fender: I think you lost me on that one |
| 19:43.59 | [TK]D-Fender | forgot the syntax |
| 19:44.10 | bradoaks | ~docs |
| 19:44.11 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:44.26 | [TK]D-Fender | ~docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:44.29 | jbot | i already had it that way, [TK]D-Fender |
| 19:44.39 | syzygyBSD | lol |
| 19:44.45 | bradoaks | [TK]D-Fender: i first had to have it forget docs then tell it the corrected info. |
| 19:45.01 | bradoaks | but i did it in a /msg |
| 19:45.02 | [TK]D-Fender | ~docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:45.03 | jbot | okay, [TK]D-Fender |
| 19:45.16 | [TK]D-Fender | bradoaks: I did that as you were typing it :) |
| 19:45.18 | [TK]D-Fender | ~docs |
| 19:45.19 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:47.32 | blepsoaf | anyone know how to set a default customer ringer on a polycom phone, that way everyone is forced to have the same ring tone? |
| 19:47.49 | blepsoaf | default custom ringer I meant |
| 19:48.13 | blepsoaf | i know how to add a new ring tone to the phone, just dont know how to make that default |
| 19:48.52 | BSDTech | ok I need input pls |
| 19:49.04 | BSDTech | anyone here use app_confrence vs meet me |
| 19:49.15 | BSDTech | what are the diffs |
| 19:49.16 | [TK]D-Fender | blepsoaf: I always set it on the phone after boot. Every attempt I did in provisioning failed. And the phone can ALWAYS override your provisioning. |
| 19:49.25 | BSDTech | I cant find any docs that compaire |
| 19:49.37 | hmmhesays | well that was a serious pain in the ass |
| 19:50.19 | blepsoaf | hmm, ok TK - I heard you have the CTU ringer, could you send me that wav file? |
| 19:52.20 | [TK]D-Fender | blepsoaf: Later, sure. |
| 19:52.57 | [TK]D-Fender | BSDTech: app_conference doesn't require a Zaptel Timer. Ther are a few other differences but I don't recall the details. |
| 19:53.24 | [TK]D-Fender | BSDTech: app_conference doesn't require a Zaptel timing source. There are few other differences, but I don't recall offhand |
| 19:53.27 | BSDTech | ok |
| 19:53.32 | BSDTech | so is it better |
| 19:53.54 | BSDTech | I now it comes in vicidial |
| 19:53.59 | BSDTech | thats why I ask |
| 19:54.13 | BSDTech | call them |
| 19:54.23 | BSDTech | and ask the boss |
| 19:54.32 | BSDTech | he is a nice guy and will get things worked out |
| 19:54.49 | BSDTech | I think they are having warehouse issues and he wants to know of any issues |
| 19:55.06 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 19:55.12 | BSDTech | I had ordered 45 phones and I ordered nextday shipping and it took a week to get them |
| 19:55.19 | BSDTech | he refunded the shipping |
| 19:55.33 | *** join/#asterisk SplasPood (n=jwb@64.90.191.180.nyinternet.net) |
| 19:55.51 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 19:55.57 | wunderkin | which boss? ive left a vm for cory a few days ago with no reply, and i called garrett and it just went to voicemail... i tried calling and asking for a support person that can deal directly with polycom, and the guy i spoke with didnt really seem to know and was going to have engineering call me but ive been waiting a month for a call from them already |
| 19:56.11 | wunderkin | i had 2 unanswered firmware requests, plus other things |
| 19:56.35 | BSDTech | Garret I think his name was |
| 19:56.49 | BSDTech | what firm ware for the polycom ? |
| 19:57.05 | BSDTech | you looking for 2.0.1 |
| 19:57.12 | [TK]D-Fender | 2.0.3 is out.... |
| 19:57.20 | BSDTech | ahh ok |
| 19:57.21 | wunderkin | i didnt leave a voicemail for him, i guess i can try... i have 2.0.2 now.. i had to get it elsewhere...... blah... ive had a problem with 1.6.7, 2.0.1, and 2.0.2.. im thinking maybe it is a config problem but not sure why |
| 19:57.23 | wunderkin | really?? ugggh |
| 19:57.36 | BSDTech | I have 2.0.1 |
| 19:57.41 | BSDTech | but would like 2.0.3 |
| 19:57.46 | BSDTech | let me see |
| 19:58.59 | [TK]D-Fender | wunderkin: What kind of problem? |
| 19:59.53 | *** part/#asterisk mroberto (n=root@S0106004063d8e527.ed.shawcable.net) |
| 20:00.12 | wunderkin | [TK]D-Fender, i asked you a little about it a day or so ago... intermittant reboot.. i get one way audio (they can hear me, but i cant hear them - on an outgoing call).. then it just reboots.... it is happening on almost every call on my phone now, in the office, there were at least 2 or 3 phones that had the problem, at least intermittantly.... incoming or outgoing calls... even with only 1 call on the line |
| 20:00.51 | Strom_C | wunderkin: just for shits and giggles, do you have canreinvite=no in the phone's sip.conf entry? |
| 20:01.02 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:01.28 | [TK]D-Fender | wunderkin: Oh yeah. Mondo FUBAR'd. RMA that poor phone |
| 20:01.31 | wunderkin | yup.. i do... |
| 20:01.56 | wunderkin | they were all ordered around the same time, and all from voipsupply so they were probably in the same batch... |
| 20:02.27 | [TK]D-Fender | wunderkin: I've had a few phones go flakey on me. happens. |
| 20:02.40 | anthonyK | allison is nice |
| 20:02.42 | anthonyK | heh |
| 20:02.43 | wunderkin | yeah... that sucks... ive just been trying hard to confirm that my config files and everything is good |
| 20:03.00 | syzygyBSD | hmm, still haven't come up with something for her to record |
| 20:03.24 | BSDTech | if you get it let me know |
| 20:03.27 | anthonyK | she sounds very nice |
| 20:03.38 | syzygyBSD | if I had more credits I would have her record part of the Dr Online song |
| 20:03.56 | wunderkin | im sure i wont be able to get 2.0.3 from them, since i requested 2.0.1 and 2.0.2 from them and never got it |
| 20:04.16 | syzygyBSD | how can i get the firmware for my polycom 501's? |
| 20:04.45 | wunderkin | i figured it was the phone or a config file, since it happens on 1.6.7, 2.0.1, and 2.0.2 firmware... i think ive only heard the problem on 2 or 3 phones.. out of 25..? |
| 20:06.47 | wunderkin | syzygyBSD, well you are supposted to get it from your reseller but it is on www.freedomfiles.net/polycom or something like that... |
| 20:07.28 | syzygyBSD | well, I dont' have a reseller |
| 20:07.45 | wunderkin | ... how did you get the phone then? fingers or ebay? |
| 20:07.53 | syzygyBSD | my boss... |
| 20:08.06 | wunderkin | i see.. 5 finger discount |
| 20:08.25 | syzygyBSD | pretty much, I don't really need 2 phones on my desk |
| 20:08.44 | syzygyBSD | but they are connected to 4 servers |
| 20:09.11 | [TK]D-Fender | syzygyBSD: Just get 1 IP 601. That will cover them all. |
| 20:09.51 | ucfMethod | syzygyBSD: which version do u need? |
| 20:10.28 | syzygyBSD | well, i don't really need any, it works fine with what I have now, but the newest version wouldn't hurt |
| 20:11.23 | wunderkin | i would like to find 2.0.3 since im already using 2.0... although i was told not to use 2.0 yet... but oh well, i was trying to fix this bug.. |
| 20:11.27 | *** join/#asterisk hypnox (n=dan@lleuad.ocq.omnicea.net) |
| 20:11.44 | ucfMethod | i think im still using 1.4.1 |
| 20:11.48 | ucfMethod | jeez, im behind |
| 20:11.56 | hypnox | can AGI stream file be made to play a file from a location other than the default sounds directory? |
| 20:13.15 | blepsoaf | hypnox: yes |
| 20:13.22 | blepsoaf | use languages |
| 20:14.12 | *** join/#asterisk LGND_A-Tuin (n=A-Tuin@212.41.185.81) |
| 20:14.13 | hypnox | thx |
| 20:14.35 | syzygyBSD | languages? i thought you could just do an absolute path? |
| 20:14.44 | hypnox | i thought you could, but its not working for me |
| 20:14.48 | syzygyBSD | been a while since I wrote that program though |
| 20:15.07 | blepsoaf | well I use languages so I dont have to do the full path |
| 20:15.27 | stephane | soirt |
| 20:16.25 | syzygyBSD | Remember, the file extension must not be included in the filename |
| 20:18.13 | hypnox | ah yeah, thats better |
| 20:18.54 | zapp-branigan | hi, when i compile the asterisk in fedora 6 give a error linux/compiler.h not found because the fedora not use now the glibc-kernheaders and |
| 20:18.54 | zapp-branigan | use the kernel-headers, how can compile this ? |
| 20:19.28 | zapp-branigan | in the i how is editing the /lib/modules/`uname -r`/build/include/linux/autoconf.h |
| 20:19.48 | zapp-branigan | is to comment some linine or something else ? |
| 20:20.33 | zapp-branigan | :( |
| 20:25.48 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 20:26.33 | *** join/#asterisk linuxtuxi1 (n=kku@212.117-242-81.adsl-dyn.isp.belgacom.be) |
| 20:36.13 | *** join/#asterisk santiago (n=santiago@208.195.214.146) |
| 20:41.47 | RoyK | http://www.hugi.is/hahradi/bigboxes.php?box_id=51208&f_id=1648 |
| 20:42.18 | *** join/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 20:42.30 | *** join/#asterisk sloth_ (n=josh@mail.fex.org) |
| 20:43.38 | syzygyBSD | "Never talk to a woman about the half a pie she just ate" |
| 20:44.46 | [TK]D-Fender | syzygyBSD: Yeah, wait till she starts setting eyes on YOUR half ;) |
| 20:45.23 | syzygyBSD | lol, trying to find something allison will record without too much protest, or will she record anything? |
| 20:46.30 | linuxtuxi1 | Hi syzygyBSD, I'm ready for some firewall action :) |
| 20:47.04 | syzygyBSD | That means no spittin', swearin', fartin', or picking your ass |
| 20:47.13 | syzygyBSD | firewall! |
| 20:47.41 | RoyK | syzygyBSD: http://karlsbakk.net/fun/asterisk-installation.wav |
| 20:48.25 | syzygyBSD | mmmhmmm |
| 20:48.44 | Nivex | *laugh* |
| 20:51.03 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 20:52.08 | *** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net) |
| 20:52.53 | *** join/#asterisk Skarmeth (n=Skarmeth@201009008073.user.veloxzone.com.br) |
| 20:52.59 | _DAW | When is the temporary greeting used in * voicemail? |
| 20:53.32 | _DAW | It is playing the temporary message when I use u(mailbox)@(context) |
| 20:53.45 | syzygyBSD | anyone at digium know if the files will still be hosted at thevoice.digium.com after the 1st? |
| 20:53.57 | syzygyBSD | or have a way for me to batch download all of them? |
| 20:54.47 | syzygyBSD | linuxtuxi1: so.. firewall action? |
| 20:55.37 | linuxtuxi1 | syzygyBSD: Yesterday you snet me some fw rules which we tried...but I was still not able to get a sip call placed |
| 20:56.01 | syzygyBSD | do you have tcpdump on the firewall box? |
| 20:56.08 | linuxtuxi1 | syzygyBSD: You suspected that it was probably related to the firewall |
| 20:56.23 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 20:56.24 | linuxtuxi1 | syzygyBSD: sure...but I'm not a tcpdump expert.... |
| 20:56.40 | syzygyBSD | oh, thats ok |
| 20:56.58 | linuxtuxi1 | syzygyBSD: which ethernet device would you like to see the output from ? |
| 20:57.01 | syzygyBSD | well, where to start, are you using the rules I gave you yesterday? |
| 20:57.18 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.247.Dial1.SanJose1.Level3.net) |
| 20:58.22 | syzygyBSD | pm me, I want to ask details about your setup |
| 20:58.52 | *** join/#asterisk DonX (i=don@the.lostserver.net) |
| 20:59.16 | DonX | does anyone have a sample SPA-941 diaplan that will let me send certain digits to another gw? |
| 20:59.33 | syzygyBSD | what do you mean another GW? |
| 21:00.10 | DonX | like, (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) is my current (and the default) dialplan |
| 21:00.56 | DonX | When I dealt with Sipura (now linksys) devices before, I had a dialplan that could send certain strings/digits to other gateways |
| 21:01.07 | DonX | IE: FWD, etc. |
| 21:01.29 | syzygyBSD | I don't know what you mean by other gateways |
| 21:01.46 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 21:01.46 | DonX | like, a different sip gateway than the one you're registered with |
| 21:02.27 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:02.31 | syzygyBSD | hmm, I have never seen that, especially not in a dial plan, but there might be a way |
| 21:03.13 | *** join/#asterisk linuxtuxie (n=kku@30.59-201-80.adsl-dyn.isp.belgacom.be) |
| 21:03.17 | heh_v_water | just installed destar to try out on Debian.. the docs say just log in then set manager stuff.. it wont let me log in as anything |
| 21:03.27 | heh_v_water | anyone mess with this? |
| 21:03.40 | linuxtuxie | syzygyBSD: ..ok I cut myself of...fixed now |
| 21:03.55 | syzygyBSD | it works? |
| 21:03.55 | linuxtuxie | syzygyBSD: I am now indeed using your provided fw rules |
| 21:03.59 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:05.22 | linuxtuxie | syzygyBSD: No I am still in the situation that sip registration works, but when I place a call...it only retransmits the invite |
| 21:06.33 | [Wiebel] | hmmz |
| 21:06.36 | [Wiebel] | if app_meetme doesnt build |
| 21:06.39 | [Wiebel] | what can that be? |
| 21:07.38 | Qwell[] | [Wiebel]: install zaptel |
| 21:07.45 | [Wiebel] | ah ok |
| 21:08.09 | [Wiebel] | thanks |
| 21:08.12 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 21:10.56 | syzygyBSD | Qwell[]: do you work for digium? |
| 21:11.16 | [Wiebel] | Qwell[]: how is the development of chan_skinny going? :P |
| 21:11.19 | Qwell[] | yes |
| 21:11.37 | [Wiebel] | any news on 7970 support? |
| 21:11.43 | syzygyBSD | are the files for thevoice.digium.com going to be hosted after tomorrow? |
| 21:11.54 | Qwell[] | [Wiebel]: Just give me an infinite amount of time to do everything else I'm doing :D |
| 21:11.56 | syzygyBSD | if not, is there a batch download available |
| 21:12.07 | Qwell[] | syzygyBSD: dunno |
| 21:12.16 | [Wiebel] | while true; do echo $time > Qwell[] ; done |
| 21:12.17 | [Wiebel] | :P |
| 21:13.32 | *** join/#asterisk linuxtuxi1 (n=kku@27.62-200-80.adsl-dyn.isp.belgacom.be) |
| 21:14.12 | *** join/#asterisk AllanKardec (n=root@201.45.22.130) |
| 21:15.08 | *** join/#asterisk allankardec (n=root@201.45.22.130) |
| 21:15.35 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 21:15.49 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 21:17.03 | DrAk0 | Why when I press # for call transfer it works for the internal phones (LAN) but with a call from the outside (PSTN / Sipura-3k) when I press # for call transfer does not work? |
| 21:18.21 | De_Mon | can I chanspy on 2 groups? |
| 21:18.47 | diablopico | anyone know how to get the codecs 723 to work with aserisk |
| 21:19.02 | diablopico | g723.1 in perticular |
| 21:19.50 | diablopico | with asterisk even ? |
| 21:20.23 | Corydon-w | diablopico: first, you pay the patent holder $300,000 |
| 21:21.34 | syzygyBSD | DrAk0: do you have the correct dtmf setup for that connection? |
| 21:22.34 | DrAk0 | syzygyBSD, hhmm not sure |
| 21:22.41 | DrAk0 | syzygyBSD, what it should be? |
| 21:23.02 | DrAk0 | dtmfmode=rfc2833 |
| 21:23.32 | syzygyBSD | uhh, i don't know, I found that one works for most, do any other key presses work? |
| 21:23.34 | *** join/#asterisk Un1x (i=Un1x@CPE000c419d026c-CM001225402bae.cpe.net.cable.rogers.com) |
| 21:23.36 | syzygyBSD | try inband |
| 21:24.38 | Corydon-w | diablopico: or you could buy the codec card from Digium for a fraction of that price |
| 21:25.26 | Strom_C | Corydon-w: is that thing out yet? |
| 21:25.39 | Corydon-w | Strom_C: probably not |
| 21:25.46 | Strom_C | yeah, i didnt think so |
| 21:27.06 | DrAk0 | syzygyBSD, no |
| 21:27.26 | syzygyBSD | I don't know what no means |
| 21:27.41 | DrAk0 | <syzygyBSD> uhh, i don't know, I found that one works for most, do any other key presses work? |
| 21:27.43 | DrAk0 | no |
| 21:27.51 | *** join/#asterisk doc9 (n=doc9@201.238.226.125) |
| 21:27.55 | syzygyBSD | have you tried inband? |
| 21:28.03 | doc9 | hi everyone |
| 21:28.08 | Strom_C | syzygyBSD: yes, maybe, possibly, and/or no :) |
| 21:28.09 | doc9 | i have a question |
| 21:28.10 | DrAk0 | i don't know whats inband |
| 21:28.20 | syzygyBSD | dtmfmode=inband |
| 21:28.33 | DrAk0 | got it |
| 21:28.34 | DrAk0 | let me try |
| 21:28.56 | syzygyBSD | doc9: what is your question, and thank you for giving us time to give you permission to ask it |
| 21:29.02 | doc9 | the function disa with file passwd it work ok? |
| 21:30.40 | *** join/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 21:30.42 | linuxtuxi1 | syzygyBSD: Should the the transmit of the invite be visible in the udp output of the tcpdump performed on ppp0 ? |
| 21:31.05 | doc9 | i have a DISA in extension.conf |
| 21:31.18 | doc9 | but not work with file passwd and context |
| 21:31.24 | DrAk0 | syzygyBSD, ok now when i call to another operator that needs key (like my ISP) the keys works |
| 21:31.25 | syzygyBSD | linuxtuxi1: I think so |
| 21:31.31 | waverly360 | Goosefrabba |
| 21:31.35 | DrAk0 | syzygyBSD, but call transfer does not work yet |
| 21:32.18 | linuxtuxi1 | syzygyBSD: Hmmm, I think I see that traffic during registration....but not when * is (re)transmitting the invite |
| 21:33.14 | waverly360 | Query: Is there a way to tell if a particular phone number is long distance to you? I know that generally a number outside of your area code is considered long distance..but that's not always the case. |
| 21:33.16 | syzygyBSD | k, that makes sense, |
| 21:33.26 | syzygyBSD | DrAk0: what is your codec? |
| 21:33.50 | DrAk0 | g711u |
| 21:33.55 | *** join/#asterisk AbuSer (n=polx@84-50-137-131-dsl.rkv.estpak.ee) |
| 21:34.02 | syzygyBSD | k, inband should work for that... |
| 21:34.35 | syzygyBSD | before you press the #, in asterisk do you do an Answer()? |
| 21:35.06 | DrAk0 | yes |
| 21:35.35 | *** join/#asterisk Skip2PBX (n=asd@ip-223-92.sn1.eutelia.it) |
| 21:36.02 | diablopico | hello.... is there any other possible solution for 723.1 ? |
| 21:36.14 | syzygyBSD | DrAk0: do you get audio in both directions? |
| 21:36.25 | DrAk0 | yes |
| 21:36.44 | Skip2PBX | Hi to everybody, i'd like to ask a question regarding ISDN cards ... and not exctly related to Asterisk... |
| 21:36.49 | syzygyBSD | can you hear the keypress you are trying to get? |
| 21:37.04 | Skip2PBX | Can i access a ISDN Card like a virtual DSP device ? |
| 21:37.28 | *** join/#asterisk Dr-Linux|home (n=nono@202.59.73.131) |
| 21:37.38 | Dr-Linux|home | is asterisk2billing program is opensource? |
| 21:38.04 | syzygyBSD | Skip2PBX: have you looked at hdlc yet? |
| 21:38.21 | Skip2PBX | hdlc ? No i don't know it ! |
| 21:38.50 | syzygyBSD | I *think* that is what you are looking for, but I don't really know |
| 21:38.59 | Skip2PBX | any link ? |
| 21:39.07 | syzygyBSD | nope, sorry |
| 21:39.23 | Skip2PBX | because the hdlc i know means High-Level Data Link Control but is something different |
| 21:39.29 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 21:39.59 | Skip2PBX | i need to use the ISDN card like in asterisk, but i need to emulate a DSP device for Audio Streaming ! |
| 21:40.01 | syzygyBSD | something like that... I don't really know what a "virtual DSP device" is |
| 21:40.12 | dioedu | hello all, if i have a message of channel.c "Dropping duplicate answer", is it a problem ? |
| 21:40.43 | Skip2PBX | dsp i mean Linux Audio Device ... can i access an ISDN card like an Audio Device /dev/dsp0 ? |
| 21:41.27 | DrAk0 | syzygyBSD, yes |
| 21:41.37 | DrAk0 | syzygyBSD, i can hear |
| 21:42.31 | syzygyBSD | so... asterisk is just not processing the keypress, my only guess is inband |
| 21:42.41 | Skip2PBX | how, if i insert an ISDN card to my linux ox this doesn't create any /dev/dsp device ... i need to use some special driver instead of the standard isdn4linux ? |
| 21:43.04 | DrAk0 | syzygyBSD, yes is not being processed by asterisk, but it is if is not thrugh the pstn |
| 21:43.06 | AbuSer | How to correct this error: file.c:587 ast_readaudio_callback: Failed to write frame ... .? |
| 21:43.16 | syzygyBSD | ISDN is closer to a modem then it is to a sound card |
| 21:43.57 | Skip2PBX | so you mean that Asterisk use an isdn card exactly like an audio Device for voice streaming ? |
| 21:44.40 | syzygyBSD | Skip2PBX: what are you trying to accomplish? |
| 21:45.13 | Skip2PBX | We develop a Software that connect Skype to a traditional PBX !!! Actually we use some USB Box that emulate an audio device... because Skype need an audio device. |
| 21:45.39 | Skip2PBX | Now we want to use PCI Cards like Digium or Eicon and to have even ISDN, PRI, BRI and maybe even SIP and H323 support. |
| 21:45.51 | DrAk0 | syzygyBSD, any idea? |
| 21:46.47 | Skip2PBX | We can handle all the signaling on the ISDN line, but for the Voice we need and audio device amulation !!! That's my question |
| 21:47.14 | syzygyBSD | DrAk0: no ideas, it sounds like asterisk isn't processing inband DTMF tones from your pstn |
| 21:47.25 | *** part/#asterisk AbuSer (n=polx@84-50-137-131-dsl.rkv.estpak.ee) |
| 21:47.31 | Skip2PBX | can we do it ? There are some libs that already implement this features ? |
| 21:49.33 | syzygyBSD | well, ISDN, PRI, and BRI cards are NOT audio cards, so they won't do what you are trying to do |
| 21:49.38 | Dr-Linux|home | is asterisk2billing program is opensource? |
| 21:50.45 | syzygyBSD | Dr-Linux|home: http://www.asterisk2billing.org/index.php?s=3 a quick google search tells us that, yes it is open source |
| 21:51.13 | Skip2PBX | syzygyBSD: So how asterisk access this cards for voice streaming ? |
| 21:51.42 | [Wiebel] | anyone with a asterisk box with moh wich is public reachable via sip? |
| 21:51.53 | Dr-Linux|home | syzygyBSD: thank you, actually GPL license was not clear to me |
| 21:51.58 | syzygyBSD | Skip2PBX: using zaptel drivers |
| 21:52.31 | Skip2PBX | syzygyBSD: ok this is a good start..... |
| 21:53.04 | syzygyBSD | Dr-Linux, well, it just means that any changes you make you have to give back, and any copies of the program you give to anyone need to have the source code with it, as well as the GPL license |
| 21:54.11 | *** join/#asterisk mroberto (n=root@S0106004063d8e527.ed.shawcable.net) |
| 21:54.27 | mroberto | How to I configure exten to play a gsm file |
| 21:54.28 | *** join/#asterisk linuxtuxie (n=kku@79.54-241-81.adsl-dyn.isp.belgacom.be) |
| 21:54.47 | Strom_C | mroberto: show application Playback |
| 21:55.02 | mroberto | where do I store the gsm file |
| 21:55.23 | Strom_C | /var/lib/asterisk/sounds/ |
| 21:56.07 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 21:58.19 | syzygyBSD | mroberto: remember, don't include the extension in the playback() command |
| 21:59.32 | mroberto | Well I am a newbie so can you point me to an example |
| 21:59.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:00.33 | syzygyBSD | exten 1,1,playback(asterisk-installation) |
| 22:00.40 | syzygyBSD | exten => 1,1,playback(asterisk-installation) |
| 22:00.52 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 22:00.59 | Skip2PBX | syzygyBSD: I've read that Zaptel Driver is needed only for Digium Cards... so if i use another card like Eicon that is supported by isdn4linux how Asterisk use the Voice Stream ? |
| 22:02.07 | syzygyBSD | Skip2PBX: http://www.voip-info.org/wiki/view/Asterisk+ISDN+overview |
| 22:03.02 | Skip2PBX | thanks... |
| 22:03.57 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 22:04.23 | mroberto | synygyBSD: what's the asterisk-installation refering to ? |
| 22:06.40 | *** join/#asterisk Shaun (n=ndci@ip68-5-62-187.oc.oc.cox.net) |
| 22:07.16 | *** join/#asterisk Shaun (n=ndci@ip68-5-62-187.oc.oc.cox.net) |
| 22:09.16 | DrkShdw | interface.c: Junk at the beginning of frame <-- anyone know what might be causing that? it seems to have locked my machine up about an hour ago |
| 22:10.24 | *** join/#asterisk stelios_ (n=stelios@noname-213.5.161.72.acn.gr) |
| 22:12.22 | JT | spontaneous dtmf :o |
| 22:13.10 | JT | inexplicable |
| 22:13.32 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 22:14.37 | Strom_C | JT: I think that's called "talk-off" :) |
| 22:14.40 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 22:15.28 | Primer | Is there no way to set/get event ids in the asterisk manager interface? Basically I want to ensure that the response I'm getting applies to the command I sent |
| 22:15.34 | JT | Strom_C: hmm? |
| 22:15.41 | *** part/#asterisk mroberto (n=root@S0106004063d8e527.ed.shawcable.net) |
| 22:15.59 | Primer | I'm using an asynchronous event driven programming framework to talk to the asterisk manager |
| 22:17.15 | *** join/#asterisk RoyK (n=roy@ti211310a080-3078.bb.online.no) |
| 22:17.42 | linuxtuxie | ok..I git something different now. I have renamed my friend section from voipstunt to sip.voipstunt.com...and registration still is ok...but if I now try to place call...I see udp traffic going through my fw |
| 22:17.57 | syzygyBSD | Primer: yes, you set the actionid when you sent the command |
| 22:18.06 | Primer | awesome |
| 22:18.07 | Primer | thanks |
| 22:18.16 | linuxtuxie | and instead of getting invite retransmits..I get SIP/2.0 401 Unauthorized |
| 22:18.51 | syzygyBSD | do you have the right username/password? |
| 22:19.03 | diclophis-work | howdy all |
| 22:19.10 | diclophis-work | is there a way to kick a user out of a conference? |
| 22:19.15 | linuxtuxie | syzygyBSD: yes |
| 22:19.28 | syzygyBSD | diclophis-work: where do you want them to go? |
| 22:19.35 | linuxtuxie | syzygyBSD: I used the same in the custom app from voipstunt |
| 22:19.46 | linuxtuxie | syzygyBSD: and also registry works... |
| 22:19.59 | diclophis-work | syzygyBSD: well in theory i want them to leave the MeetMe command, and continue executing stuff in my AGI script |
| 22:20.07 | diclophis-work | so i guess further down the dialplan |
| 22:20.08 | syzygyBSD | what does "sip show registry" from the asterisk console show? |
| 22:20.18 | diclophis-work | syzygyBSD: sorta like auto press " |
| 22:20.21 | *** join/#asterisk lorinc (n=ang@caracas-4779.adsl.interware.hu) |
| 22:20.26 | diclophis-work | er auto press "#" for them, with the " |
| 22:20.32 | diclophis-work | er with the "p" option set |
| 22:20.53 | linuxtuxie | syzygyBSD: -> sip.voipstunt.com:5060 linuxtuxie 105 Registered |
| 22:22.34 | [TK]D-Fender | linuxtuxie : 401 means the user/pass is bad. end of story. Just because you registered OK, doesn't mean your user/peer will work. |
| 22:22.47 | *** join/#asterisk stelios__ (n=stelios@noname-213.5.161.72.acn.gr) |
| 22:22.58 | *** part/#asterisk andresmujica (n=andresmu@201.244.196.229) |
| 22:23.08 | syzygyBSD | diclophis-work: I have never had meetme work well with agi, that doesn't mean it is not possible though |
| 22:24.05 | syzygyBSD | how do you want to trigger them to leave the conference? |
| 22:25.03 | linuxtuxie | <PROTECTED> |
| 22:26.36 | ManxPower | linuxtuxie: the [stuffinhere] is what the incoming username is. |
| 22:26.51 | ManxPower | username= is for OUTGOING username |
| 22:28.02 | *** join/#asterisk SuXz (n=msohail_@58.65.193.77) |
| 22:28.25 | SuXz | can any one help me to configure asterfax or please tell me where can i have its manual |
| 22:30.28 | SuXz | any asterfax buddy there |
| 22:31.40 | ManxPower | I don't even know what AsterFax is. |
| 22:31.47 | *** join/#asterisk toggy (n=toggy@host-81-191-169-198.bluecom.no) |
| 22:32.03 | SuXz | ok :) |
| 22:32.39 | *** join/#asterisk itguru (n=guru@host86-147-2-67.range86-147.btcentralplus.com) |
| 22:32.40 | Supaplex | it's just some lame email gateway? *sigh* |
| 22:32.48 | toggy | anyone know how to fix the: channel.c: Dropping duplicate answer! error on incoming capi calls? |
| 22:32.55 | *** join/#asterisk LordBacon (n=frb@unaffiliated/frb) |
| 22:33.01 | LordBacon | oink oink |
| 22:33.05 | LordBacon | What's a zaptel? |
| 22:33.48 | [TK]D-Fender | "Good grief Charlie Brown" |
| 22:33.54 | [TK]D-Fender | LordBacon : ... |
| 22:33.55 | [TK]D-Fender | ~book |
| 22:33.58 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:34.00 | ManxPower | LordBacon: Zaptel is a framework for telephony applications |
| 22:34.37 | [TK]D-Fender | LordBacon : In other words (short version) : The module(s) that support most of the hardware cards often used with * (and a few other oddities) |
| 22:35.15 | ManxPower | Now if we had a context for your question we might be able to be more helpful. |
| 22:35.19 | LordBacon | I don't have a hardware card, which is why I get confused when all the howtos mention using zaptel |
| 22:35.57 | [TK]D-Fender | LordBacon : Its also required for IAX2 in trunk mode, and for MeetMe (both for reasons that its a timing source) |
| 22:36.19 | [TK]D-Fender | LordBacon : Which if you don't need, well, kudos, you can do without installing it |
| 22:37.22 | toggy | anyone know how to fix the: channel.c: Dropping duplicate answer! ?? |
| 22:37.49 | Supaplex | I don't |
| 22:38.12 | *** part/#asterisk SuXz (n=msohail_@58.65.193.77) |
| 22:38.20 | *** part/#asterisk linuxtuxie (n=kku@79.54-241-81.adsl-dyn.isp.belgacom.be) |
| 22:40.35 | *** join/#asterisk SuXz (n=msohail_@58.65.193.77) |
| 22:40.43 | SuXz | asterfax buddy there |
| 22:42.30 | Supaplex | they went home |
| 22:46.33 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 22:47.07 | waverly360 | Hey yall. |
| 22:47.44 | waverly360 | Does anyone know where I can see an example extensions.conf where someone converted their dialplan fully to AGI? |
| 22:48.34 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 22:50.54 | Un1x | *CLI> show g729 |
| 22:50.54 | Un1x | 1/1 encoders/decoders of 4 licensed channels are currently in use |
| 22:50.55 | Un1x | *CLI> |
| 22:51.04 | [TK]D-Fender | waverly360 : I've only heard of one nut in here that did that, and I can't remember exactly who it was... |
| 22:51.08 | Un1x | w---t :D |
| 22:51.14 | Un1x | errr im happy lol |
| 22:51.15 | Un1x | :D |
| 22:51.44 | Un1x | heh at least it works g729 that is :D |
| 22:51.45 | syzygyBSD | waverly360: I did that before, but you don't want to take a look at it |
| 22:51.45 | Un1x | now |
| 22:52.00 | *** join/#asterisk g0tw00d (n=rchace@68-113-159-200.dhcp.nplt.ne.charter.com) |
| 22:52.24 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:53.24 | syzygyBSD | and the extensions.conf wont' really tell you anything if it is all AGI |
| 22:55.50 | *** join/#asterisk a2lti (n=alti@c.32.169.a475.sta.adsl.cyfra.net) |
| 23:01.27 | *** join/#asterisk itguru (n=guru@host86-147-4-154.range86-147.btcentralplus.com) |
| 23:08.06 | LordBacon | I was reading through teh book again, but it says nothing about disk space |
| 23:08.24 | LordBacon | how much space should I expect to be needed if I have say 10 phone lines with voice mail? |
| 23:08.41 | De_Mon | I want to send an email to someones voicemail address, how can I extract it? |
| 23:10.09 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 23:14.55 | JT | Strom_C: what is talk off? |
| 23:15.32 | De_Mon | jt the name of the sound file |
| 23:16.36 | JT | huh? |
| 23:16.51 | *** join/#asterisk AdmoIRC (n=Miranda@CPE-65-27-25-141.kc.res.rr.com) |
| 23:18.52 | syzygyBSD | LordBacon: depends on how much voicemail they are going to keep, figure 10MB per minute of voicemail |
| 23:19.04 | JT | De_Mon: ? |
| 23:19.12 | De_Mon | what? |
| 23:19.25 | syzygyBSD | jt look at festival |
| 23:19.38 | JT | syzygyBSD: what are you answering? |
| 23:19.52 | JT | De_Mon: what were you trying to tell me? |
| 23:19.54 | De_Mon | I was just looking at Strom's replies guess that was a different coversation |
| 23:20.10 | Un1x | Hey ive i want to USe Voicemail with asterisk |
| 23:20.14 | Un1x | do i have to install mpg123 |
| 23:20.23 | Un1x | or cabn the default slackware mpg321 |
| 23:20.23 | Un1x | be used |
| 23:20.36 | LordBacon | ok, after I figure out wht I'm doing, I'll get a bigger disk on there |
| 23:22.41 | *** join/#asterisk |Vulture| (n=Vulture@101.222.121.70.cfl.res.rr.com) |
| 23:25.01 | anthonyK | anyone here wanting to help a newb |
| 23:25.04 | anthonyK | my server is already up |
| 23:25.09 | anthonyK | I just need some phone number configuration |
| 23:27.27 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 23:29.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:30.11 | *** join/#asterisk guru_ (n=guru@host81-153-59-88.range81-153.btcentralplus.com) |
| 23:30.33 | Un1x | anthonyK what is it you need? |
| 23:30.36 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 23:31.41 | *** part/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 23:31.42 | anthonyK | im lost :( |
| 23:32.13 | *** join/#asterisk anthonyc (n=ac@adsl-75-10-114-228.dsl.frs2ca.sbcglobal.net) |
| 23:36.38 | *** join/#asterisk Galeras (n=Galeras@litigaractivos1.att.net.co) |
| 23:38.55 | Galeras | someone using ring-inuse app_queue patch? |
| 23:40.09 | Galeras | it didn't work for 2 or more queues, someone can help us with this improvement? |
| 23:40.28 | Galeras | i mean help u$ |
| 23:40.57 | *** join/#asterisk ltdwk (n=z@61.29.127.162) |
| 23:41.54 | |Vulture| | Anyone use any hardware or software to limit bandwidth to ensure the VoIP connections have priority? |
| 23:42.06 | ltdwk | Question on queues/agents.... Is there any way to have an agent always logged in, rather than them having to log in? |
| 23:43.01 | *** part/#asterisk nalioth (i=nalioth@freenode/staff/ubuntu.member.nalioth) |
| 23:43.03 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:43.28 | TripleFFFF | ahmm can u fit a digium or sangoma in a PCI-X slot ? |
| 23:43.46 | ltdwk | yes PCI-X is backwards compatible |
| 23:44.04 | ltdwk | make sure you get the right voltage card though |
| 23:44.37 | orlock | |Vulture|: no, buti am looking at QoS and performance metrics currently |
| 23:44.58 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1128748818.dsl.bell.ca) |
| 23:45.11 | momelod | hello people |
| 23:45.15 | momelod | anyone around? |
| 23:46.33 | momelod | i have a question about digium's te207p card: i put this card in and on bootup i get these strange errors; "wct4xxp: disagrees about version of symbol zt_ec_span", "wct4xxp: Unknown symbol zt_ec_span" |
| 23:46.49 | momelod | any ideas/pointers? |
| 23:47.20 | |Vulture| | orlock: are you looking at QoS over the local network or out to the internet? |
| 23:47.22 | syzygyBSD | when was the last time you recompiled zaptel? |
| 23:49.08 | momelod | syzygyBSD since before i installed this card |
| 23:49.13 | momelod | probably 4 months ago |
| 23:49.24 | momelod | ive been using an analog card up until now |
| 23:50.05 | syzygyBSD | why are you using wct4xxp? |
| 23:50.38 | momelod | auto detected |
| 23:50.48 | momelod | and thats what it says i should use on the difium site |
| 23:51.05 | momelod | http://www.digium.com/elqNow/elqRedir.htm?ref=http%3A%2F%2Fwww.digium.com%2Fen%2Fdocs/misc/quick_install_zaptel_asterisk.pdf |
| 23:51.21 | syzygyBSD | hmm, ok, thought it would be another driver, not the 4 port one |
| 23:51.58 | momelod | the card has 2 ports |
| 23:52.20 | syzygyBSD | and that page you linked says to use wct2xxp |
| 23:52.21 | momelod | actually your right |
| 23:52.27 | momelod | shoot |
| 23:52.35 | momelod | so why is my kernel loading the wrong one? |
| 23:52.43 | syzygyBSD | because 2xxp doesn't exist |
| 23:52.45 | syzygyBSD | :) |
| 23:52.58 | momelod | brilliant! |
| 23:53.04 | momelod | so i should recompile zaptel? |
| 23:53.21 | syzygyBSD | of course, the version I have is a year and a half old, maybe the newer version of zaptel does |
| 23:53.55 | momelod | 1.2.x has em according to the site |
| 23:54.15 | momelod | im going to try to recompile now |
| 23:54.17 | momelod | thanx for the tips |
| 23:54.21 | syzygyBSD | I see wctdm24xxp |
| 23:54.25 | momelod | hey, one last question |
| 23:54.53 | momelod | will i have to configure this card somewhere or will it magically detect my pri line and auto configure? |
| 23:55.21 | momelod | not in the dialplan stuff, just making all the proper connections between the card and the line |
| 23:55.23 | syzygyBSD | yes, you have to configure both /etc/zapata.conf and /etc/asterisk/zaptel.conf |
| 23:55.34 | momelod | rats :) |
| 23:55.43 | momelod | okay thanx alote for your help |
| 23:55.47 | momelod | g'night |
| 23:55.50 | syzygyBSD | they are 1 line configureation, very standard |
| 23:56.07 | syzygyBSD | well, maybe not quite that easy |
| 23:56.15 | *** join/#asterisk bluregard (n=bluregar@c-67-163-72-68.hsd1.il.comcast.net) |
| 23:57.06 | bluregard | linlin: you around? |
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