| 00:00.42 | seanbright | sorry, don't think i am going to be helpful on this one |
| 00:02.33 | jblack | whoah. wine 1.0 out. |
| 00:02.38 | jblack | what's next? Duke nukem forever? |
| 00:03.50 | harryv | jeez! 'set callerid' always returns 1, 'set context' always returns 0 and so on. that seems messy.. http://gundy.org/asterisk/agi.html |
| 00:04.10 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 00:04.27 | seanbright | harryv: its one of those things once released is hard to change |
| 00:10.27 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 00:16.14 | *** join/#asterisk sack (n=sack@237.Red-79-148-191.staticIP.rima-tde.net) |
| 00:28.24 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:33.46 | *** join/#asterisk supa_disko (n=bleh@iinet.guard.com.au) |
| 00:33.56 | l0verb0y | can I place concurrent calls with orginate? |
| 00:37.57 | *** join/#asterisk oilinki (n=oil@ppp-124-120-4-61.revip2.asianet.co.th) |
| 00:39.37 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 00:47.07 | jblack | l0verb0y: Can you originate more than one at a time? You should be able to. |
| 00:47.37 | l0verb0y | does each extention have to be empty? |
| 00:47.47 | l0verb0y | for example, if i have a conference, can i orginate like 10 calls to it |
| 00:48.56 | jblack | you should be able to in some way, yes |
| 00:49.29 | l0verb0y | thanks |
| 00:49.33 | jblack | certainly you could generate 10 callfiles that dump 10 different people into the same conference. |
| 00:50.04 | jblack | I don't have my * book at the moment, otherwise, I'd look deeper |
| 00:50.30 | lmadsen | don't use callfiles |
| 00:50.42 | lmadsen | not for originating lots of calls at once |
| 00:53.40 | oilinki | morning |
| 00:55.07 | jblack | Oh? |
| 00:55.58 | ManxPower | jblack: somewhere in eastern pacific rim |
| 00:56.45 | ManxPower | that was stupid. Somewhere in the WEST pacific rim |
| 00:57.21 | jblack | Why are you screwing with me this time? Did I kill two of your puppies in a previous life? |
| 01:05.20 | oilinki | do you have recommendations for text-to-speach application for asterisk? |
| 01:07.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7dd146fdfb0368e7) |
| 01:09.55 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 01:10.50 | *** part/#asterisk infinity1 (i=brendon@saleen.netcal.com) |
| 01:11.58 | *** join/#asterisk moy (n=moyhu@189.169.82.208) |
| 01:12.24 | jblack | oilinki: Asterisk has app_festival, which uses festival |
| 01:13.17 | jblack | You can read up on it on pages 303-305. There's more help for it on pages 395 and 472 |
| 01:14.43 | l0verb0y | hmmm |
| 01:15.12 | l0verb0y | anything special I have to do to a conf so I can dump calls into it? |
| 01:15.42 | oilinki | jblack: thanks. I'll check that one out. |
| 01:21.24 | oilinki | one night I was listening an podcast which was 'computer read'. it took me a while before I did notice that it was not an real person reading. |
| 01:21.59 | oilinki | I guess the text-to-speech technique start to be ready for usage |
| 01:22.25 | ManxPower | oilinki: only the non-free ones |
| 01:22.48 | ManxPower | Cepstral is about as good as you get for under $1,000, and Cepstral is under $100, IIRC |
| 01:23.41 | oilinki | ManxPower: does it has support for other languages as well? |
| 01:23.58 | ManxPower | I do not know. cepstral.com, I believe |
| 01:30.53 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:32.13 | oilinki | english, frensh, italian and german seems to be the languages. |
| 01:35.07 | jeev | lol, datacenter tech told me asterisk core dumped |
| 01:35.07 | jeev | lol |
| 01:35.10 | jeev | on console |
| 01:35.43 | *** join/#asterisk philippel (n=p_lindhe@pool-71-164-18-224.sttlwa.fios.verizon.net) |
| 01:37.47 | jeev | that's all my fault for building it and forgetting |
| 01:38.59 | oilinki | this was the podcast text-to-speech engine I was listening. sounds pretty good. http://www.odiogo.com/Gina_Hughes-From_Blog_to_Podcast_with_Odiogo.mp3 |
| 01:42.37 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-84.dslextreme.com) |
| 01:46.17 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 02:00.46 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:10.44 | *** join/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 02:11.06 | unpaidbill | pri uses the bchan/dchan configuration, right? |
| 02:12.57 | Strom_C | yes |
| 02:13.22 | drdrain | Is it normal for ztmonitor -vv to show a power reading on idle channels? |
| 02:15.19 | unpaidbill | so this is a dumb question, but wtf does bchan stand for, and dchan |
| 02:15.39 | unpaidbill | i need me a telephony book |
| 02:15.44 | unpaidbill | ~astbook |
| 02:15.50 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:15.51 | unpaidbill | ~book |
| 02:15.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:15.56 | _ShrikE | ~101 |
| 02:15.56 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:16.00 | drdrain | D = Data (siganlling) B = Bearer |
| 02:16.02 | unpaidbill | oh sweet, thanks |
| 02:16.16 | drdrain | The bearer channels carry the voice |
| 02:16.17 | _ShrikE | thank strom |
| 02:16.26 | unpaidbill | is bchan/dchan a standard terminology |
| 02:16.37 | drdrain | Well yeah |
| 02:16.43 | unpaidbill | well i mean, that abbreviation |
| 02:16.55 | drdrain | Well sure I guess |
| 02:17.05 | unpaidbill | or would i sounds like a better bullshitter if i said signalling chan and bearer channels |
| 02:17.12 | drdrain | Any telephony person whould know what that meant |
| 02:17.18 | unpaidbill | haha ok, thanks |
| 02:17.34 | drdrain | No D Channel or B channel is the correct way to reference them |
| 02:18.55 | Strom_C | I seriously need to grep my logs and see how many copies of "telephony 101" i serve per week |
| 02:19.04 | *** join/#asterisk juanjoc (n=juanjoc@host190.190-225-197.telecom.net.ar) |
| 02:19.07 | unpaidbill | im reading it now |
| 02:19.15 | unpaidbill | and i must say, it's making me excited |
| 02:19.17 | unpaidbill | in confusing ways |
| 02:19.44 | MooingLemur | -[~/.xchat2/xchatlogs:$]- fgrep -i 'telephony 101' *asterisk* | wc -l |
| 02:19.45 | MooingLemur | 43 |
| 02:20.29 | unpaidbill | now tell us how many times strom has said wack off |
| 02:20.39 | JT | to make it simpler, "channels" on a PRI are just moments in time, aka. timeslots |
| 02:20.43 | JT | ;) |
| 02:20.51 | unpaidbill | yeah |
| 02:20.54 | drmessano | ~102 |
| 02:20.54 | jbot | hmm... 102 is #asterisk |
| 02:21.00 | unpaidbill | we're switching our system over from e&m wink to a PRI |
| 02:21.08 | unpaidbill | with dynamic data/voice channels |
| 02:21.36 | unpaidbill | which somehow equates to: more features and $300 less per month |
| 02:21.48 | unpaidbill | i dont know how, or why, but i like it |
| 02:22.03 | ManxPower | unpaidbill: there is NO logic to telecom tariffs. |
| 02:22.19 | LiNeTuX|Home | unpaidbill: what's the base pri w/local loop costing you? |
| 02:22.22 | MooingLemur | zero |
| 02:22.40 | unpaidbill | the only thing i can figure is that our current system uses some shitty big old hardware on their side, and the new hardware they'll put us on allows for higher density of users |
| 02:22.42 | MooingLemur | actually 1 :P |
| 02:23.02 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:23.09 | ManxPower | unpaidbill: in many locations PRIs are much more expensive than E&M. Are you using a CLEC or an ILEC? |
| 02:23.28 | unpaidbill | 119/month for data, 35/month for each voice channel (we have 8), down from 398 data and 45 voice |
| 02:23.44 | ManxPower | unpaidbill: What's the local loop? |
| 02:23.58 | unpaidbill | i dont know, i just got a basic quote |
| 02:24.09 | ManxPower | chances are, that price is on top of local loop. |
| 02:24.13 | LiNeTuX|Home | wishes he could find a provider who'd charge per voice channel |
| 02:24.32 | ManxPower | Your sales rep will hate you, but ask for an actual monthly cost INCLUDING taxes and fees. |
| 02:24.39 | unpaidbill | clec, also |
| 02:24.55 | ManxPower | CLECs seem to prefer CT1 or PRI |
| 02:24.56 | unpaidbill | she's sending me a quote tomorrow, so hopefully that's all listed |
| 02:25.33 | *** join/#asterisk rpr_rpr (n=chatzill@107.154.218.87.dynamic.jazztel.es) |
| 02:25.50 | voxter | Huh. I am almost positive that this used to work: exten => 123,hint,SIP/123@otherhost |
| 02:25.53 | voxter | But now, it doesnt. |
| 02:26.02 | rpr_rpr | One question for the community I've created my own app for asterisk 1.4, but i can't find how to specify the dinamic libraries for compilation |
| 02:28.18 | rpr_rpr | anyone can help me? |
| 02:29.03 | ManxPower | rpr_rpr: also try #asterisk-dev |
| 02:29.05 | Strom_C | rpr_rpr: you'll probably have better luck in #asterisk-dev |
| 02:29.18 | rpr_rpr | oks. |
| 02:29.25 | rpr_rpr | Thanks. |
| 02:35.58 | *** part/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 02:44.05 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 02:44.28 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.236) |
| 02:44.31 | smach | good evening guys |
| 02:44.45 | BeeBuu | is asterisk support h.248ï¼ |
| 02:45.05 | ManxPower | BeeBuu: Is that a codec or a protocol? |
| 02:45.29 | smach | I was wondering if there was a possibility to change the FROM header ASterisk sends to a sip proxy ? |
| 02:46.18 | smach | I don't want Asterisk to send the FROM header with it's IP address |
| 02:49.16 | BeeBuu | ManxPower: that's a protocol |
| 02:51.13 | file | Megaco. |
| 02:51.45 | BeeBuu | file: yeah,that's it! |
| 02:51.59 | file | We do not support it, but I have heard rumblings that someone is working on it. |
| 02:52.06 | file | I do not recall their IRC nickname though. |
| 02:52.10 | BeeBuu | done? |
| 02:52.32 | BeeBuu | are they finished? |
| 02:52.37 | file | hasn't been submitted for inclusion yet... so can't tell you |
| 02:53.03 | BeeBuu | thanks ,ManxPower & file. |
| 02:57.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c336a429caff6eeb) |
| 02:58.06 | *** join/#asterisk paci`` (n=nessuno@cpe-071-065-236-211.nc.res.rr.com) |
| 02:58.08 | paci`` | hey |
| 02:58.10 | paci`` | how would i ban a user |
| 02:58.11 | paci`` | with if() |
| 02:58.19 | file | huh? |
| 02:58.29 | paci`` | like |
| 02:58.31 | paci`` | i want to make it hang up |
| 02:58.36 | paci`` | if the CID is a certain one |
| 02:59.08 | LiNeTuX|Home | why not blacklist? |
| 02:59.14 | paci`` | or that |
| 02:59.16 | paci`` | how do i do that |
| 03:00.09 | LiNeTuX|Home | are you on * or one of the other distros? |
| 03:00.09 | lmadsen | exten => _1NXXNXXXXXX/5195915119,1,Hangup() |
| 03:00.13 | paci`` | asterisk |
| 03:00.20 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 03:00.21 | lmadsen | paci``: see above |
| 03:00.21 | paci`` | so if the extension was 5 |
| 03:00.30 | paci`` | exten => 5/CALLERID,1,hangup |
| 03:00.31 | paci`` | right? |
| 03:00.36 | lmadsen | yes |
| 03:00.43 | lmadsen | where CALLERID = an actual number |
| 03:00.53 | d-k-t-2 | or pattern |
| 03:01.05 | LiNeTuX|Home | but why hang up? there's so much more fun stuff you can do with people you don't want to talk to :) |
| 03:01.11 | smach | no idea guys, if there is a way of modifying the ip address in the from header ? |
| 03:02.15 | d-k-t-2 | smach, externip? |
| 03:02.51 | smach | d-k-t-2; didn't get you |
| 03:03.08 | d-k-t-2 | smach, /etc/asterisk/sip.conf setting, externip |
| 03:03.26 | smach | ok I check what id does, thx |
| 03:03.45 | paci`` | hmm |
| 03:03.45 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 03:04.07 | paci`` | is there a way if the offending user is blocking their cller id, LiNeTuX|Home |
| 03:04.48 | LiNeTuX|Home | paci``: depends. you can do custom IVR's to make them say their name or something, then take the call. |
| 03:04.51 | smach | d-k-t-2: does it only change the ip in the from header ? |
| 03:05.00 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:05.08 | d-k-t-2 | smach, don't know... don't use it myself |
| 03:05.11 | jaytee | or you could just block any number without callerid |
| 03:05.11 | paci`` | LiNeTuX, but how would i parse that? |
| 03:05.19 | *** join/#asterisk zippytech (n=ron@244.zippytech.com) |
| 03:05.22 | paci`` | jaytee, we have alot of callers who block with that though |
| 03:05.32 | jaytee | so you don't want to block all of them |
| 03:05.38 | smach | d-k-t-2: I'll give it a try right now, let you know |
| 03:06.31 | LiNeTuX|Home | my first thought is to throw folks w/no callerid into a custom IVR, make them announce their name, have the call come to someone with that announcement, then have * give you the option of taking the call |
| 03:06.48 | paci`` | jaytee, yeah, just this one |
| 03:07.10 | LiNeTuX|Home | paci``: I don't know who you'd pick 'just one' of many |
| 03:07.16 | LiNeTuX|Home | who / how |
| 03:07.45 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-05237bc32d92edfa) |
| 03:09.54 | zippytech | does trix box support video with xlite? |
| 03:10.12 | JT | doesn't trixbox have a channel? |
| 03:10.29 | zippytech | or asterisk |
| 03:11.03 | zippytech | sorry i think of them as the same thing |
| 03:11.08 | zippytech | for the most part |
| 03:11.53 | zippytech | and some times i get better responses with trix because more play with it |
| 03:12.03 | jaytee | funny, I think of apples and oranges as the same thing sometimes |
| 03:12.48 | zippytech | they are all food |
| 03:12.50 | zippytech | lol |
| 03:12.58 | JT | zippytech: then ask in the trix channel if you get better responses there |
| 03:13.07 | zippytech | no one there |
| 03:13.15 | LiNeTuX|Home | i wonder why that is? |
| 03:14.16 | digitalirony | zippytech: yes asterisk supports video with x-lite i got it to work just the other day |
| 03:14.29 | digitalirony | zippytech: get rid of trixbox it sucks |
| 03:14.46 | zippytech | i run ubuntu with asterisk and freepbx |
| 03:14.53 | jaytee | videosupport=yes in general section of sip.conf and allow=h.323 |
| 03:14.58 | zippytech | with myth |
| 03:15.07 | digitalirony | zipptech: well run debian with asterisk and no gui |
| 03:15.19 | jaytee | and that should get some of the earlier versions of X-lite to work fine with * |
| 03:15.26 | digitalirony | zippytech: you could put myth on it if you wanted |
| 03:16.02 | zippytech | i have a small box that does phone, power controller (x10) and tv plus file server on in one plus network monitors |
| 03:16.13 | jaytee | that would make a swell home-entertainment/home-pbx system. |
| 03:16.22 | zippytech | thats the goal |
| 03:16.23 | digitalirony | zippytech: i'm just poking at you. do it how you want, but don't call digium asking to help with trixbox :P |
| 03:16.38 | zippytech | lol i don't |
| 03:16.56 | jaytee | just wouldn't try running 50 to a 100 sip phones and using meetme and voicemail while using it as a DVR at the same time :-) |
| 03:17.04 | digitalirony | heh yeah |
| 03:17.12 | zippytech | no i only have 5 to 10 users |
| 03:17.29 | digitalirony | jaytee: even then....get rid of freepbx |
| 03:17.39 | jaytee | I'm not running it |
| 03:17.42 | digitalirony | heh |
| 03:17.48 | digitalirony | @zippytech |
| 03:17.50 | digitalirony | sorry |
| 03:17.59 | [TK]D-Fender | Yup, I had my home running on * + X-10 (CM11A). Fun stuff |
| 03:18.16 | digitalirony | zippytech if you want a gui atleast get asterisknow....it has less bugs |
| 03:18.32 | jaytee | I run pure unadulterated Asterisk 1.4.15 with no GUI |
| 03:18.33 | digitalirony | so you won't have to go and ask questions to the people in the trix channel |
| 03:18.57 | digitalirony | jaytee: thats cool. I wish all my customers were like you |
| 03:19.03 | drmessano | thinks trixbox is green vomit |
| 03:19.33 | digitalirony | jaytee: its much easier for people to get support when they don't use other peoples software |
| 03:19.34 | LiNeTuX|Home | thinks Trixbox is a shiny green wrapper for FreePBX |
| 03:20.06 | jaytee | I bet if I edited the sip.conf in trixbox and got video working on an X-Lite by just doing a sip reload at the CLI trixbox would wipe it out the next time I restarted or did a refresh in the dialplan from the gui. |
| 03:20.06 | drmessano | They say that everytime a bell tolls an angel gets it wings.. what they don't tell you is that everytime someone installs Trixbox, an angel gets set on fire |
| 03:20.13 | drmessano | So, don't use Trixbox folks |
| 03:20.22 | digitalirony | yes please don't |
| 03:20.34 | LiNeTuX|Home | Friends don't let friends install Trixbox |
| 03:20.35 | jaytee | everytime you use trixbox, God rapes a kitten |
| 03:20.40 | digitalirony | 90% of the people that call digium support use it, and 90% of the time is the problem |
| 03:21.13 | digitalirony | well not that many i really made that number up |
| 03:21.15 | digitalirony | lol |
| 03:21.32 | LiNeTuX|Home | I do believe the Celtics are going to be the next NBA champs. For those who care. |
| 03:21.33 | file | 'tsk 'tsk Eric |
| 03:21.57 | digitalirony | file: ? |
| 03:22.08 | jaytee | yay!!!! |
| 03:22.11 | file | making up numbers is soooooo last week |
| 03:22.12 | digitalirony | im aloud to make up numbers if i say they are made up |
| 03:22.27 | drmessano | 90% of all statistics are made up |
| 03:22.32 | jaytee | I'm a diehard Celtics fan |
| 03:22.45 | digitalirony | is that basketball or football? |
| 03:22.59 | jaytee | being of Irish ancestry and growing up 10 miles south of Boston helps :-) |
| 03:23.03 | jaytee | basketball |
| 03:23.03 | LiNeTuX|Home | NBA = Basketball |
| 03:23.07 | digitalirony | ahh |
| 03:23.18 | file | digitalirony: are you still at the office? O.o |
| 03:23.19 | digitalirony | Well im a diehard AMD fan |
| 03:23.22 | jaytee | digitirony, don't get much freetime at Digium I take it? |
| 03:23.29 | digitalirony | file: yep i work late shift |
| 03:23.35 | file | ah yes |
| 03:23.37 | LiNeTuX|Home | jaytee: 30 pt lead with 10mins to go |
| 03:23.39 | drmessano | digitalirony works for digium? |
| 03:23.41 | file | I forgot |
| 03:24.04 | digitalirony | jaytee: yeah i get plenty of freetime....when im not working |
| 03:24.11 | jaytee | LiNeTuX|Home, I'd say it's sewn up. |
| 03:24.12 | digitalirony | drmessano: yes i do |
| 03:24.28 | drmessano | Hmmm |
| 03:24.41 | file | he is one of those fancy support people |
| 03:24.42 | jaytee | digitalirony, are you in Huntsville? |
| 03:24.46 | *** join/#asterisk xenonex (n=xenonex@89.218.236.233) |
| 03:24.47 | digitalirony | aye |
| 03:25.02 | digitalirony | jaytee: yes in huntsville |
| 03:25.19 | digitalirony | file: and what is it you do |
| 03:25.31 | file | digitalirony: I am in swdev |
| 03:25.33 | digitalirony | file: you just sit in jabber and op all day |
| 03:25.37 | drmessano | Remind me not to buy a digium card |
| 03:26.04 | digitalirony | file: drmessano: doesn't like me :P |
| 03:26.17 | file | you can not please everyone |
| 03:26.25 | drmessano | It's nothing personal.... wait, yes it is |
| 03:26.41 | digitalirony | file: i know...but see. he thinks because he doesn't like me we might have bad hardware |
| 03:26.47 | digitalirony | file: so thats bad for business |
| 03:26.53 | tzanger | heh |
| 03:27.03 | file | digitalirony: perhaps we should fire you then... |
| 03:27.06 | drmessano | I don't think digium hardware is bad.. and your statement is why I wont be calling |
| 03:27.10 | digitalirony | file: maybe |
| 03:27.20 | file | out of a canon! |
| 03:27.32 | digitalirony | drmessano: sorry to loose you |
| 03:27.43 | drmessano | Frankly, your deductive reasoning frightens me to no end |
| 03:27.58 | digitalirony | drmessano: i really don't care |
| 03:28.01 | drmessano | You did say you work late shift.. |
| 03:28.09 | zippytech | thanks guys that worked |
| 03:28.24 | drmessano | So maybe I will buy Digium.. just need to call before 6 :) |
| 03:29.07 | digitalirony | drmessano: actually im quite good at fixing peoples hardware, and astierks installs, but just because i don't know as much about 911 service as the DR. that makes me bad at my job? |
| 03:29.18 | jaytee | drmessano, he just started with them recently I think he said, cut him some slack, everyone is new at least once and even a broken clock is right twice a day :-) |
| 03:29.35 | d-k-t-2 | counts calls I've made to digium support about digium hardware |
| 03:29.37 | zippytech | i have a tdm 400 |
| 03:29.39 | d-k-t-2 | hmm, zero |
| 03:29.58 | d-k-t-2 | must be quiet over there digitalirony |
| 03:29.59 | drmessano | I'll cut him some slack.. when he's about halfway down the cliff |
| 03:29.59 | zippytech | and every time the phone guys test the lines it looks to be shorted out |
| 03:30.01 | digitalirony | zippytech: cool, hows it working for you |
| 03:30.07 | zippytech | good |
| 03:30.12 | drmessano | :) |
| 03:30.20 | digitalirony | d-j-t-2: we get calls all day long....but we don't mostly e-mails at night |
| 03:30.24 | jaytee | zero? .....hmmm, yeah I heard of that! Didn't the Mayans invent it? |
| 03:30.29 | digitalirony | *d-k-t-2: |
| 03:30.38 | zippytech | everything works but not sure why the card make the lines look that way |
| 03:30.49 | digitalirony | zippytech: look what way? |
| 03:31.01 | zippytech | like there is a short in the line |
| 03:31.12 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:31.14 | drmessano | Don't use that word ever again |
| 03:31.31 | zippytech | i have had this happen at 3 locations when he put a meter on the line to test |
| 03:31.31 | jaytee | what word? |
| 03:31.31 | digitalirony | zippytech: what do you mean? whats the output you see |
| 03:31.41 | LiNeTuX|Home | zippytech: I believe the word you are looking for is a "vertically challenged" line. |
| 03:31.52 | d-k-t-2 | zippytech, so it draws current when it's connected? |
| 03:31.52 | LiNeTuX|Home | oh wait... |
| 03:31.54 | zippytech | like the wires are touching |
| 03:32.01 | drmessano | I hate when someone calls something "a short"... Even a basic tech knows better than to use that term.. |
| 03:32.15 | digitalirony | zippytech: this is with FXO's right? |
| 03:32.21 | zippytech | right |
| 03:32.39 | digitalirony | drmessano: well i don't guess you noticed...but he isn't a tech |
| 03:32.43 | digitalirony | he is a customer |
| 03:32.48 | drmessano | "zippytech" |
| 03:32.53 | LiNeTuX|Home | heh |
| 03:33.16 | digitalirony | drmessano: not from my point of view |
| 03:33.47 | drmessano | digitalirony: You start calling people "customers" in here, and people's patience will wear very short |
| 03:33.52 | drmessano | lol |
| 03:34.02 | digitalirony | drmessano: sorry consumer then |
| 03:34.16 | zippytech | ok the point is the tdm400 makes the connections to the pots look line they are twisted to gether, how that |
| 03:34.17 | digitalirony | drmessano: or user |
| 03:34.18 | zippytech | lol |
| 03:34.22 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 03:34.25 | drmessano | Ahh |
| 03:34.28 | d-k-t-2 | duser |
| 03:34.32 | drmessano | "user" is an even better term |
| 03:34.32 | d-k-t-2 | digium user |
| 03:34.48 | zippytech | why would this be and other pbx's normaly don't show this |
| 03:34.52 | jaytee | I use Digium products. |
| 03:34.56 | LiNeTuX|Home | zippytech: just a guess, but are you sure your local stuff is wired/punched down right? |
| 03:35.02 | drmessano | I thought this was a peer supported chat channel, not a tech queue.. Do I need a number? |
| 03:35.14 | LiNeTuX|Home | drmessano: 153346 |
| 03:35.19 | jaytee | 2 TDM400 cards, 1 TE212PRI card and a baseball cap. |
| 03:35.30 | drmessano | jaytee: please, wait in line.. your call will be answered shortly |
| 03:35.31 | zippytech | yes , we test without the tdm plugged in |
| 03:35.38 | zippytech | the lines are clean to the card |
| 03:35.47 | LiNeTuX|Home | zippytech: that doesn't mean anything |
| 03:35.51 | digitalirony | drmessano: no but im pretty sure im allowed to help users in my own time if i choose as long as i don't get paid for it |
| 03:36.03 | drmessano | jaytee: Our support specialists are waiting to assist you, please be patient |
| 03:36.18 | d-k-t-2 | zippytech, so, the card draws current... does it look like the lines are twisted together harder if it goes off hook? |
| 03:36.28 | LiNeTuX|Home | heh |
| 03:36.33 | drmessano | hums Hungry Like The Wolf - Muzak version |
| 03:36.34 | jaytee | so far I haven't had to call either Digium or Polycom and I didnt' want to waste my time calling Grandstream because I don't speak Mandarin |
| 03:36.53 | drmessano | hums blah blah blah, blah blah blah blah, and i'm hungry like the wolf...... |
| 03:36.55 | jaytee | hahahha, Muzak version |
| 03:37.02 | LiNeTuX|Home | jaytee: I believe it's "Stupideese" over at GS |
| 03:37.38 | zippytech | unsure i will have to test, i just had a customer call the other day when att was there and he had told me tha same thing a year ago and never looked into it |
| 03:37.39 | d-k-t-2 | jaytee, 'Ni hao. Wo you yi ge hen da de wenti. Ni ke yi ban wo ma?' - useful phrase for next time you call |
| 03:37.45 | digitalirony | zippytech: i don't understand still...are you seeing odd out put, what is the problem ? |
| 03:38.09 | LiNeTuX|Home | De bo chi. Wa li la lu. Pung now le lah. |
| 03:38.17 | drmessano | jaytee: We appreciate your business.. please visit our website at doubleyou-doubleyou-doubleyou-dot-digium-dot-com |
| 03:38.28 | jaytee | converting some popular tune to Muzak is usually like beating some poor animal to death, with Hungy Like the Wolf it's more like mutilating the carcass. |
| 03:38.29 | zippytech | the cards work find but if you put a meter on the line from the co it looks like the wires are twisted to gether |
| 03:38.31 | drmessano | jaytee: Your call is very important to us |
| 03:38.47 | zippytech | when plugged into the tmd |
| 03:38.50 | zippytech | tdm |
| 03:39.04 | jaytee | "Press or say One now!" |
| 03:39.14 | digitalirony | zippytech: so there is no problem? |
| 03:39.14 | zippytech | he and i have never see any pbx or phone system show this |
| 03:39.18 | jaytee | "One!" |
| 03:39.23 | d-k-t-2 | zippytech, if the line looked like it was twisted together, it wouldn't work |
| 03:39.29 | zippytech | right , they work fine |
| 03:39.40 | jaytee | "Thank you! Please hold while we direct your call to "Two" |
| 03:39.40 | LiNeTuX|Home | so the problem is that there is no problem. |
| 03:39.53 | digitalirony | zippytech: it would have to be a line problem then....but if it works its just a fluke somewhere don't fix it if its not broken. |
| 03:39.56 | zippytech | it does , if you have one put a meter on it and see , i have 3 that do it |
| 03:40.01 | LiNeTuX|Home | Your approximate wait time is.... twelve ... hundreded... minutes... |
| 03:40.15 | drmessano | hums the next fine selection: "Neverending Story" - Muzak edition |
| 03:40.18 | digitalirony | zippytech a developer might can help you better understand that....but its odd |
| 03:40.34 | jaytee | vomits |
| 03:40.39 | zippytech | my thoughts for sure, just a question that i have run into in the 5 years i been using the cards |
| 03:41.05 | drmessano | jaytee: Sorry, we are experiencing higher than average call volumes.. your patience is greatly appreciated.. estimated hold time - 3 hours, 11 minutes |
| 03:41.13 | digitalirony | zippytech: next time i see some one from dev, i will ask them why |
| 03:41.25 | zippytech | i beleive the word is continuity test |
| 03:41.26 | zippytech | ? |
| 03:41.55 | zippytech | cool thanks for the help |
| 03:41.55 | LiNeTuX|Home | jaytee: Thank you for holding. We will now connect you call. <beep> <click> ::dialtone:: |
| 03:42.07 | JT | zippytech: can you be more specific? what resistance reading are you getting? |
| 03:42.32 | drmessano | jaytee: If you would like to leave a number for callback, smash the keypad with your palm now.. otherwise, please continue to hold, and someone will assist you shortly.. your estimated wait time is 3 hours, 10 minutes |
| 03:42.33 | LiNeTuX|Home | DAMN 38 point lead... ouch LA. |
| 03:43.58 | drmessano | digitalirony: I have a tech support question.. I just got a TDM410P... I took it out of the box and installed it in a free slot in the PBX I am building |
| 03:44.07 | drmessano | digitalirony: What is "linux"? |
| 03:44.22 | jaytee | it's always nice for me when the Celtics win, when they pound the Lakers it's just pure ecstasy. |
| 03:44.37 | digitalirony | drmessano: and open-source operating system based on unix |
| 03:44.47 | digitalirony | created by linus torvolds |
| 03:44.55 | digitalirony | its under the GNU license |
| 03:44.59 | jaytee | "What does baffled mean?" |
| 03:44.59 | LiNeTuX|Home | jaytee: useless trivia: between the lakers and celtics, they've won HALF of all NBA championships |
| 03:45.05 | drmessano | digitalirony: Do I need that to install a TDM410P? Is that in control panel? |
| 03:45.16 | LiNeTuX|Home | What's a gah-noo? |
| 03:45.32 | digitalirony | drmessano: nope, and your an asshat |
| 03:45.50 | drmessano | Who is Richard Stallman, and why does he keep smashing me in the back of the head when I mispronounce "GNU"? |
| 03:45.55 | digitalirony | drmessano: you already know the answer to those questions. i have seen you answer stuff that implies you know it |
| 03:46.07 | d-k-t-2 | drmessano, this is #asterisk, not #poke-the-tech-support-guy |
| 03:46.13 | jaytee | It's like because I grew up south of Boston I have this genetic encoding that no matter what I have no concious choice but to hate and loathe the Yankees. :-) |
| 03:46.14 | digitalirony | drmessano: if he is smashing you in the back of the head ask him |
| 03:46.15 | drmessano | digitalirony: You are a quick learner... |
| 03:46.27 | drmessano | digitalirony: I mean.. super quick |
| 03:46.56 | digitalirony | drmessano: well if it helps i already knew that stuff |
| 03:47.17 | digitalirony | drmessano: and im honestly not that quick at learning, just good at remembering |
| 03:47.25 | drmessano | d-k-t-2: I don't think you can refer to me by nickname here, you're just a user.. I think digitalirony needs to moderate your comment and submit it for approval |
| 03:47.38 | LiNeTuX|Home | t o o l |
| 03:47.43 | drmessano | d-k-t-2: Did you submit you comment to the queue? |
| 03:47.54 | digitalirony | dremessano: im just a user here too. this isn't my channel |
| 03:48.10 | digitalirony | drmessano: i come here on my own free will to help people so back off |
| 03:48.20 | d-k-t-2 | drmessano, queue, what queue, I am the queue |
| 03:48.43 | drmessano | digitalirony: You clearly referred to everyone "non-digium" in here as "customers" and "users", so therefore, we should all be moderated, no? |
| 03:48.53 | jaytee | Richard Stallman is just a female pubic hair's width closer to normal than Ted Kaczynski |
| 03:49.03 | LiNeTuX|Home | d-k-t-2: the queue starts behind that building, in the dark alley |
| 03:49.09 | drmessano | jaytee: bite your tongue, RMS is GOD |
| 03:49.12 | digitalirony | drmessano: i actually referred to the customer with the TDM card that was asking questions as a customer |
| 03:49.38 | jaytee | drmessano, how'd the Kool-Aid taste? was it the grape flavor? |
| 03:49.39 | drmessano | digitalirony: Does he have a valid support contract? Did you verify that first? |
| 03:49.57 | digitalirony | drmessano: doesn't matter if he does....this is free support |
| 03:50.02 | drmessano | jaytee: The NIKE's are what sold me |
| 03:50.26 | d-k-t-2 | drmessano, digium is too supporting, even without valid support contracts they appear to give unlimited support to anyone who's ever bought stuff from them |
| 03:50.31 | digitalirony | drmessano: me working for digium has nothing to do with my support here. this is me offering help to people for free |
| 03:50.41 | d-k-t-2 | drmessano, it's nice |
| 03:51.09 | digitalirony | d-k-t-2: we are an opensource company and we act like one |
| 03:51.11 | drmessano | digitalirony: So if I call for support on a device I have no contractual obligation to, can we pretend we're on IRC and you can help me for free? |
| 03:51.17 | d-k-t-2 | drmessano, then their employees even come here and give more free unlimited support, ace! |
| 03:51.24 | drmessano | Because I have a nice Digium X100P I need help with |
| 03:51.34 | digitalirony | drmessano: nope because we are not on IRC and i DON't have to help anyone here |
| 03:51.35 | jaytee | drmessano is so mean he has no MySpace friends |
| 03:51.42 | digitalirony | drmessano if you call me i will tell you no |
| 03:52.01 | drmessano | digitalirony: We can use nicknames, and I can prefix thoughts with "slash me" |
| 03:52.03 | [TK]D-Fender | drmessano: Got it working on your DEC Alpha under OS/2 yet? They oughtta help you with that! |
| 03:52.06 | drmessano | thinks |
| 03:52.17 | [TK]D-Fender | ~emo |
| 03:52.18 | jbot | /wrists |
| 03:52.25 | digitalirony | drmessano: heh good luck....you can try it if you want |
| 03:52.31 | jaytee | [TK]D-Fender! |
| 03:52.47 | jaytee | was just wondering if you were here and just lurking or off somewhere |
| 03:53.12 | [TK]D-Fender | jaytee: got in late from martial arts & visiting an old friend |
| 03:53.13 | drmessano | digitalirony: Dude, thats just totally cold.. you just told me "If its on IRC, no probs.. but if you call, screw you" |
| 03:53.18 | drmessano | digitalirony: SAD FACE |
| 03:53.41 | jaytee | cool, which martial art specifically? |
| 03:54.12 | digitalirony | drmessano: yep, because if you call me you are not using IRC which is free, if you call me you are using digiums lines, which cost them money that is paid for by their customers, if your not a customer your not getting help for free on a paid for line |
| 03:54.26 | [TK]D-Fender | jaytee: http://en.wikipedia.org/wiki/Tenshin_Shoden_Katori_Shinto-ryu |
| 03:55.05 | drmessano | digitalirony: What if I make a toll call and don't use the 800 number? Can we split the difference? |
| 03:55.17 | drmessano | digitalirony: I'll paypal you like $3.50 |
| 03:55.40 | LiNeTuX|Home | Celtics win... 131 to 92. |
| 03:55.48 | digitalirony | drmessano: sure you pay my sallary and the consulting time that we charge customers who call without hardware or contracts and well call it even |
| 03:56.35 | drmessano | digitalirony: That's not a very friendly attitude..Maybe I should stick to X100P clones on eBay |
| 03:56.36 | jaytee | [TK]D-Fender, very interesting mix of all the arts. I always wanted to do kendo and shuriken |
| 03:57.18 | jaytee | [TK]D-Fender, how long have you been doing it? |
| 03:57.26 | digitalirony | drmessano: well your not a friendly person. so i ask you to please do so |
| 03:57.29 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 03:57.33 | [TK]D-Fender | jaytee: not so much of a mix. Shuriken.... meh. Kendo is a sport.... no time for "do" :p |
| 03:57.48 | drmessano | niiice |
| 03:57.53 | [TK]D-Fender | jaytee: a little over 2 years now. |
| 03:58.22 | drmessano | digitalirony: Are the other support guys just like you? |
| 03:58.36 | digitalirony | drmessano: yes we are all clones |
| 03:58.44 | jaytee | some of it can give you a good workout and the rest can be boring and repetitive but that's as necessary as the rest. |
| 03:58.48 | [TK]D-Fender | jaytee: http://video.google.ca/videoplay?docid=-3592341485993959661&q=katori+shinto&ei=34dYSPDkN5-i-wHX0czlDg&hl=en |
| 03:59.30 | [TK]D-Fender | jaytee: Skip to 3:00 in |
| 04:00.26 | drmessano | digitalirony: Funny thing is, for the most part I was giving you a hard time, but man.. you're a real keeper.. Hope not all new techs are like you. |
| 04:00.46 | jaytee | what's that Japanese stringed instrument they play that's kinda like a ukelele? |
| 04:01.03 | [TK]D-Fender | jaytee: not a clue :) |
| 04:01.16 | digitalirony | drmessano: on the phone i will kiss the customers ass all i have too, but when your being rude to me i will be rude right back, and i can do that |
| 04:01.23 | drmessano | jaytee: Watch Karate Kid Part II.. it's probably in there |
| 04:01.28 | jaytee | I've got a major flareup of CRS at the moment but I used to know. |
| 04:01.38 | [TK]D-Fender | jaytee: I know the first 4 ken (sword on sword), 2mins), and 4 of the Bo (later on) |
| 04:02.10 | jaytee | I have a bokken |
| 04:02.23 | jaytee | but I only use it to threaten my cats |
| 04:03.07 | drmessano | digitalirony: Nice attitude.. If you look at providing quality support as simply "kissing the customers ass", I hope we never cross paths |
| 04:03.19 | [TK]D-Fender | jaytee: And of course 4 kneeing iai, 5 standing. |
| 04:04.26 | jaytee | I'm so out of shape that after 10 minutes of that my feet would be killing me and my knees would be buckling. |
| 04:07.01 | jaytee | I need to get off my ass and out of chat more and start bicycling before this old carcass goes into full meltdown. |
| 04:07.01 | [TK]D-Fender | digitalirony: You should never be rude with the customer, its not good business. If you're on the receiving end you should be able to say "I'm sorry but I will not be able to continue helping you so long as you maintain your belligerent attitude." And then offer to transfer them to your superior along with the recording of the call. |
| 04:08.27 | drmessano | jaytee: My bicycling days ended when I moved into a second floor apt and had to deal with lugging a cheap, heavy bike up and down steps |
| 04:08.46 | drmessano | jaytee: Talk about motivation to upgrade |
| 04:09.30 | digitalirony | [TK]D-Fender: you misunderstand me, I am NOT rude to any customers. |
| 04:09.35 | jblack | [TK]D-Fender: Never say never. :) |
| 04:10.21 | jblack | Bailbondman's would not be very effective if they said "Hey, I know you missed your appearance, so can you please come in so that I can take you to jail, I'd apreciate it" |
| 04:10.27 | [TK]D-Fender | digitalirony: "but when your being rude to me i will be rude right back, and i can do that" <- Sorry, I kinda read that "as-is" and seemed pretty clear to me... |
| 04:10.43 | digitalirony | [TK]D-Fender: I ment in irc |
| 04:10.46 | digitalirony | sorry |
| 04:11.15 | [TK]D-Fender | jblack: You missed an important detail.. a bailbondsman's target isn't heis CUSTOMER ;) |
| 04:11.42 | jblack | Yes, he is. :) |
| 04:11.49 | digitalirony | [TK]D-Fender: and i was refrring to him being rude to me |
| 04:11.55 | jaytee | drmessano, what a coincidence! I live in a 2nd floor apartment and my bike is in the bedroom at the moment. |
| 04:11.58 | jblack | digitalirony: You smell funny. |
| 04:12.40 | [TK]D-Fender | I'm ont he 4th floor of my building and I bike to work whenever the weather is clear (which has been shit-on-a-stick lately) |
| 04:12.56 | jblack | [TK]D-Fender: But another example... A man borrows money from the Mafia, and fails to pay it back. The mafia aren't supposed to be polite. They're supposed to break fingers. =) |
| 04:13.22 | jaytee | [TK]D-Fender, you're in Montreal, aren't you? |
| 04:13.45 | digitalirony | I don't want anyone here to get the wrong impression about tech support, we are not rude, and we do our job. end of discussion |
| 04:13.54 | [TK]D-Fender | jblack: Again... the "collector" is an outsourced agent who does whatever gets his boss (your shark) his money back. Again, YOU are not their "customer". |
| 04:13.58 | [TK]D-Fender | jaytee: yup |
| 04:14.21 | [TK]D-Fender | jblack: You keep picking poor samples to back up your failing point :p |
| 04:14.24 | jblack | Bah. You're nit picking. |
| 04:14.25 | Corydon76-dig | [TK]D-Fender: not always |
| 04:14.45 | *** join/#asterisk javawizard2539_ (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 04:15.03 | jblack | borrow something from the mafia and not give it back, and someone in the mafia pays you a visit. |
| 04:15.06 | *** join/#asterisk Gregabyte (n=gregabyt@c-68-62-173-134.hsd1.al.comcast.net) |
| 04:15.11 | [TK]D-Fender | Corydon76-dig: I suppose there are the times where the shark himself gets his hands dirty, but then again... thats bad for business, he's taking work away from (dis)honest thugs! |
| 04:15.37 | Corydon76-dig | [TK]D-Fender: eh? |
| 04:15.41 | jblack | The mafia doesn't usually outsource it's muscle. |
| 04:15.55 | smach | hey guys, I've been reading through * documentation and I can't find a way to change the ip addrees I send in the from field in an INVITE |
| 04:15.58 | smach | any ideas ? |
| 04:16.00 | [TK]D-Fender | jblack: and its rarely the guy who gave you the cash that breaks your legs :) |
| 04:16.04 | Corydon76-dig | BB's don't have to go after their customers 99% of the time |
| 04:16.13 | [TK]D-Fender | smach: From what, to what? |
| 04:16.23 | jblack | [TK]D-Fender: It's rarely the guy that takes my check that provided the service. |
| 04:16.39 | [TK]D-Fender | jblack: See, there you have it. |
| 04:16.46 | jblack | If I go to Denny's, one person collects my order, another cooks it, and a third collects the payment. |
| 04:16.47 | smach | [TK]D-Fender: From * ip address to anothe pbx ip |
| 04:16.48 | digitalirony | Gregabyt_: welcome |
| 04:16.56 | Gregabyte | waves |
| 04:17.03 | Qwell | jblack: be fair, they share jobs there |
| 04:17.11 | Qwell | the cook also cleans the restrooms |
| 04:17.12 | jblack | I didn't stop becoming Denny's customer just because the company hired someone to cook, or someone else to ring up checks. |
| 04:17.16 | [TK]D-Fender | smach: Highly irregular. Thats like calling the cops & saying its really your neighbour calling. |
| 04:17.48 | jaytee | wow, that was a long video but the spear against sword part at the end was worth it. |
| 04:18.09 | smach | [TK]D-Fender: sure but I won't change the contact addree, it's just that it's the only way that I can make calls using line registration # sip trunks |
| 04:18.13 | [TK]D-Fender | jaytee: That was Naginata, not spear actually... |
| 04:18.48 | [TK]D-Fender | jaytee: Spear is a little wierd, but naginata is just plain scary |
| 04:19.41 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 04:19.43 | [TK]D-Fender | jaytee: An awesome weapon to be sure. |
| 04:19.47 | jaytee | but even though you might think at first that it has an advantage if the swordsman has good technique he can overcome the longer reach |
| 04:20.12 | jaytee | so it makes it more interesting with the parrying and the feints |
| 04:20.27 | [TK]D-Fender | jaytee: reach plus parrying.... |
| 04:21.51 | jaytee | well, the guy with the naginata might win most of the time but a midget with fast reflexes might be able to duck and run in low and stab him in the balls :-) |
| 04:21.54 | jblack | I want my stimulis payment. |
| 04:22.21 | jaytee | I already spent mine on asian manufactured electronics at WalMart. |
| 04:22.34 | jblack | I'm going to spend mine on asian grown food. |
| 04:22.40 | jaytee | cuz I'm a NASCAR patriot! |
| 04:23.25 | jblack | food could double or triple in the next year, so I want to use my stimulus to hoard food. |
| 04:23.31 | mchou | I'm a bit confused regarding STUN, NAT and SIP. I have two PAP2s behind one NAT. Each PAP2 device is registered to different voip providers (although they point to the same STUN server). Somehow the 2nd PAP device always renders the first "unregistered." Any clue to what's going on? |
| 04:23.36 | *** part/#asterisk Gregabyte (n=gregabyt@c-68-62-173-134.hsd1.al.comcast.net) |
| 04:23.59 | jblack | mchou: Yeah. They're both using the same set of ports. |
| 04:24.37 | [TK]D-Fender | jaytee: http://uk.youtube.com/watch?v=IreQsNHSoK8 @ 5:10 |
| 04:24.42 | [TK]D-Fender | jaytee: Spear |
| 04:24.47 | mchou | jblack: yup. indeedy. but shouldn't the NAT keep track of src addr as well as src prot? |
| 04:24.52 | jblack | move one from port 5060 to 5061 or somesuch. |
| 04:24.57 | mchou | port* |
| 04:25.27 | jblack | I don't know the rules for your nat. |
| 04:25.34 | mchou | jblack: like how different web browser clients always use port 80 :) |
| 04:26.18 | *** join/#asterisk imcdona (i=imcdona@c-24-19-100-65.hsd1.mn.comcast.net) |
| 04:26.22 | mchou | jblack: but no "wrong" info gets sent to the inccorect web browser :) |
| 04:26.27 | mchou | incorrect* |
| 04:26.37 | jblack | I'm not in the mood for a debate. Forget I said anything |
| 04:26.40 | *** join/#asterisk pputman (n=centrex@216.207.245.1) |
| 04:27.18 | mchou | jblack: just tring to understand what's going on is all |
| 04:27.39 | drmessano | mchou: No port forwarding involved? |
| 04:27.55 | mchou | jblack: I'm sure moving ports would work. just not sure why same port doesnt is all |
| 04:28.09 | [TK]D-Fender | mchou: they are fighting over the same incoming ports. CHANGE one fo the devices to a different set of ports |
| 04:28.17 | mchou | drmessano: no port forwading enabled on NAT |
| 04:28.28 | drmessano | mchou: You should be able to put 100 devices behind the NAT without a problem |
| 04:28.28 | mchou | forwarding* |
| 04:28.37 | drmessano | 100 being some arbitrary number |
| 04:28.58 | mchou | drmessano: yeah, that's what I'm confused about (hence the web browser analogy) |
| 04:29.24 | smach | [TK]D-Fender: any idea how I can change the ip address in the FROM header ? |
| 04:29.26 | [TK]D-Fender | mchou: Anyways, I've already answered you, as has jblack. Go change the ports on one of them. |
| 04:29.37 | [TK]D-Fender | smach: No real way. |
| 04:29.37 | mchou | drmessano: somehow with different ports it all works but not if they are on same port |
| 04:29.44 | drmessano | mchou: If you change the order you power the devices, does it always bump the earlier connected one |
| 04:29.50 | [TK]D-Fender | smach: Anything you attempt will screw up return messages. |
| 04:30.03 | mchou | drmessano: no, it's non-deterministic |
| 04:30.11 | [TK]D-Fender | smach: Sounds like you want a hybrid B2BUA / Proxy effect, and that isn't anywhere in *'s scope. |
| 04:30.12 | jblack | "Dr, it hurts when I do this" |
| 04:30.19 | drmessano | mchou: Random.. only one device will work |
| 04:30.33 | mchou | drmessano: yeah, pretty much |
| 04:30.58 | drmessano | mchou: In theory, you should NOT be having a problem at all.. If changing ports fixes it, your NAT is screwy |
| 04:31.31 | *** join/#asterisk Sargun (n=Sargun@atarack/staff/sargun) |
| 04:31.35 | jblack | unless his router undrestands sip, and it's auto-forwarding 5060 to the most recent sip client. |
| 04:31.56 | mchou | drmessano: yeah, that's what I'm thinking....wonder if openwrt or something might be messed up |
| 04:31.59 | smach | which IDE do you guys use to code on asterisk ?? |
| 04:32.08 | [TK]D-Fender | smach: "copy con" <- |
| 04:32.11 | drmessano | jblack: That makes no sense |
| 04:33.15 | drmessano | It's a NAT... |
| 04:33.54 | smach | [TK]D-Fender: are you kidding ??? |
| 04:34.10 | mchou | jblack: doesnt NAT keep track of ports as well as src addrs? |
| 04:34.23 | drmessano | mchou: Indeed it does |
| 04:34.37 | mchou | drmessano: that's why I'm so confused |
| 04:34.39 | drmessano | mchou: That's why you shouldn't be having this issue |
| 04:34.56 | [TK]D-Fender | mchou: When an call comes IN it targets your dst port. that is always fixed. |
| 04:35.04 | digitalirony | drmessano: is right, sounds like something wrong with your NAT |
| 04:35.05 | drmessano | dst port |
| 04:35.17 | mchou | drmessano: I've even taken a look at my /proc/net/ip_conntrack :) |
| 04:35.22 | oilinki | some of the linksys adsl-routers have own, not working sip alg on the device. |
| 04:35.23 | smach | seriously guys, I'm fed up with coding on the terminal, do you use eclipse or another ide ? |
| 04:35.26 | digitalirony | mchou: do you know how to use tcpdumb ? |
| 04:35.30 | [TK]D-Fender | mchou: I may have a source of 6789, but I target your 5060 to call YOU. |
| 04:35.44 | digitalirony | *tcpdump |
| 04:35.52 | oilinki | which makes registration to some of the services impossible |
| 04:36.00 | mchou | digitalirony: sure I know how to tcpdump :) |
| 04:36.24 | drmessano | If that were the case, then that means all my web server connections are forwarded via Port 80 through my NAT to my web browser? |
| 04:36.31 | drmessano | Someone tell that to my apache server.. its gonna be pissed |
| 04:36.37 | mchou | drmessano: bingo! |
| 04:36.38 | drmessano | Oh, and the other users |
| 04:38.06 | mchou | in any case, I'm very puzzled..... |
| 04:38.10 | drmessano | mchou: I'm calling it NAT shenanigans.. I've had those sort of issues with OpenWRT before |
| 04:38.33 | [TK]D-Fender | mchou: drmessano its just multiple devices behind NAT fighting for 5060 inbound. |
| 04:38.36 | smach | I have a dial plan that routes all the calls to another pbx, is there a way to set the ip address in the from header statically and let all the other header generated normally |
| 04:38.47 | smach | I believe this wont mess up anything |
| 04:38.54 | drmessano | [TK]D-Fender: Its not using 5060 inbound.. |
| 04:39.14 | [TK]D-Fender | smach: No. You are trying to "proxy" it in a manner of speaking and I just told you * doesn't do that. |
| 04:39.30 | drmessano | [TK]D-Fender: This is where the 5 hour NAT-source-and-destination-ports argument comes in |
| 04:39.34 | [TK]D-Fender | drmessano: I know. but using 2 of those devices will have each fighting for the same 2 ports. |
| 04:39.50 | drmessano | [TK]D-Fender: If that were the case, tell that to my 3 PAPs over my friends house that connect back to my * box |
| 04:39.55 | mchou | drmessano: you mean this argument happens all the time? |
| 04:39.56 | [TK]D-Fender | drmessano: its the inbound that'll freak |
| 04:39.57 | drmessano | [TK]D-Fender: That's a non-issue |
| 04:40.04 | drmessano | mchou: Sadly, yes |
| 04:40.22 | smach | [TK]D-Fender: sorry to insist, was thinking actually about modifying * code |
| 04:40.27 | drmessano | facepalms |
| 04:40.43 | [TK]D-Fender | smach: that falls firmly into : |
| 04:40.46 | [TK]D-Fender | ~wglwat |
| 04:40.47 | jbot | methinks wglwat is well, good luck with all that |
| 04:40.53 | [TK]D-Fender | smach:... territory |
| 04:43.14 | smach | [TK]D-Fender: I'm not sure I got what ~wglwat means, sorry for my ignorance |
| 04:43.28 | [TK]D-Fender | smach: look directly below it... |
| 04:44.40 | TrentCreek | darn..where is the #800 Mexican guy? |
| 04:45.25 | pputman | what? lol |
| 04:46.19 | drmessano | picking lettuce? |
| 04:46.22 | mchou | lol |
| 04:46.26 | mchou | now now |
| 04:46.57 | mchou | not all mexicans work on the farm :) |
| 04:47.15 | drmessano | shut up and push that lawnmower |
| 04:47.22 | drmessano | j/k |
| 04:47.35 | [TK]D-Fender | drmessano: Nope.. the rest are assembling Volswagens for North America :) |
| 04:47.45 | [TK]D-Fender | Volkswagons* |
| 04:47.51 | mchou | I thought those are made in Canada |
| 04:47.55 | smach | [TK]D-Fender: sorry buddy, I've been googling wglwat for 10 min with success... |
| 04:48.11 | mchou | or is it Hondas I'm thinking of? |
| 04:48.13 | [TK]D-Fender | [00:40]<[TK]D-Fender>~wglwat |
| 04:48.14 | [TK]D-Fender | [00:40]<jbot_>methinks wglwat is well, good luck with all that |
| 04:48.16 | drmessano | holy shit |
| 04:48.21 | drmessano | ~wglwat |
| 04:48.21 | jbot | wglwat is, like, well, good luck with all that |
| 04:48.25 | [TK]D-Fender | smach: I said.... RIGHT BELOW IT. |
| 04:48.30 | drmessano | RIGHT THERE DUDE |
| 04:48.31 | drmessano | READ |
| 04:48.44 | jaytee | the section of Indianapolis that I live in is referred to as "Little Mexico" |
| 04:49.05 | jaytee | I like it. |
| 04:49.09 | drmessano | Thats my neighbors |
| 04:49.15 | smach | [TK]D-Fender: oh sorry, I'm slow after midnight |
| 04:49.25 | mchou | jaytee: can you get a decent burrito in lil' mexico? |
| 04:49.47 | mchou | jaytee: not talking about taco bell |
| 04:49.50 | pputman | mchou, you can find a good taco truck in just about every city =) |
| 04:49.55 | jaytee | the supermercado down the street has REAL Coca-Cola with cane sugar instead of that high-fructose corn syrup shit they forced down our throats since the 70's. |
| 04:49.59 | drmessano | Every night they get a rotisserie chicken and a pack of tortillas for dinner |
| 04:50.10 | jaytee | mchou, yes |
| 04:50.15 | mchou | pputman: nah, try NYC, you'll be sadly disappointed |
| 04:50.19 | pputman | :/ |
| 04:50.28 | jaytee | very good taquerias, several within a mile of me. |
| 04:51.16 | mchou | pputman: NYC still doesn't know mexican food is all about |
| 04:51.41 | mchou | pputman: even though the situation is improving |
| 04:51.43 | drmessano | One of our threats at work: Dont make me have to go to the salvation army and replace you |
| 04:52.02 | jaytee | hahaha |
| 04:52.26 | lowlevel | go download firefox3 |
| 04:52.29 | lowlevel | ;) |
| 04:52.30 | mchou | drmessano: forget salvation army. It's India now. |
| 04:52.35 | lowlevel | you don't haev to intsall it, just download it |
| 04:52.43 | pputman | mchou, that's sad, usually you can find a good truck parked at the side of the road with good mexican food. |
| 04:52.57 | drmessano | mchou: India is a far walk for onsite support :) |
| 04:53.11 | jaytee | lowlevel, why? do I get a prize? |
| 04:53.23 | jblack | speaking of india... Hear McDonald's latest? They're now off-siting "can I take your order". |
| 04:53.25 | drmessano | oh god |
| 04:53.37 | lowlevel | jay: yeah, firefox3. |
| 04:53.49 | drmessano | I wonder how much of the Download Day for FF3 has been ruined by fucking Digg trolls insisting people download it |
| 04:53.55 | jaytee | still buggy, just new bugs |
| 04:53.57 | jblack | Before long, you'll have to tell walid that you don't like pickles on your double quarter pounder. |
| 04:54.06 | drmessano | I beta tested it from alpha 1 and I am just embaressed at the n00bs trolling |
| 04:54.13 | drmessano | jaytee: FF3 is MUCH MUCH better |
| 04:54.43 | drmessano | jaytee: I've had my wife on it since RC1.. and she bitches less about her PC.. It's like night and day |
| 04:54.48 | [TK]D-Fender | I got my FF3 at about 2:30 EST. Not bad... |
| 04:54.54 | drmessano | jaytee: Now THAT is a beta test |
| 04:55.05 | drmessano | I had FF3 yesterday lol |
| 04:55.09 | [TK]D-Fender | MUCh faster, zooming is beautiful, I LOVE the new address bar... |
| 04:55.30 | drmessano | RC3 > Release last night via auto-update |
| 04:55.33 | [TK]D-Fender | address completion, etc.... auto search through favourites.... |
| 04:55.36 | jblack | [TK]D-Fender: Zooming? |
| 04:55.40 | jaytee | it's better than two and I've been running the last beta but I just don't feel that "ZOMG, IT'S OUT!!! LET'S GET IT!!!" feeling and want to jump in with all the lemmings. |
| 04:55.49 | [TK]D-Fender | jblack: Yup. |
| 04:56.07 | jblack | I have ff3 something. What's zooming? |
| 04:56.15 | drmessano | jaytee: I'm not normally that way, but having dealt with the memory issues of prior releases, I am just shy of the Digg trolls on this one.. |
| 04:56.21 | [TK]D-Fender | jblack: Firefox 3 <- |
| 04:56.37 | [TK]D-Fender | jblack: Ctrl + (+/-), or Crtl + scroll |
| 04:56.44 | jaytee | oh, cripes yeah, FF2 was a seive for memory. |
| 04:56.50 | jaytee | and forget flash |
| 04:57.07 | drmessano | jaytee: Leave FF2 on a flash page for 24 hours, and you'll run out of 3 GB RAM |
| 04:57.10 | jaytee | I couldn't watch 3 YouTube vids in a row without it locking. |
| 04:57.13 | jblack | huh. didn't that used to change font size? |
| 04:57.47 | jblack | Neat! |
| 04:58.35 | *** join/#asterisk zn3t (n=zn3t@pool-70-105-157-214.char.east.verizon.net) |
| 04:58.38 | drmessano | I have 8 tabs open... using the session I have been working with for a week or so now, restarted last night for the update.. right at 200MB used |
| 04:59.15 | drmessano | I also have an ass of plugins |
| 04:59.22 | jaytee | yep, definitely an improvement over two and I'll probably upgrade this week but I like to pace myself. |
| 04:59.33 | TrentCreek | I wonder why my FF on Linux keeps telling me I need to install JRE plugin when it already is |
| 04:59.52 | jblack | I wonder if that works with java |
| 05:00.06 | drmessano | TrentCreek: You need to symlink your java install |
| 05:00.17 | TrentCreek | oooohh..okay thanks |
| 05:00.41 | jblack | damn. Doesn't work with java. |
| 05:00.42 | pputman | way too lazy to compile things from open source, i'll wait til ff3 has a debian package =) |
| 05:01.02 | drmessano | FF3 doesn't work with Java? |
| 05:01.02 | jblack | pputman: It's been in ubuntu for ages, and I'm sure it's been in debian as well. |
| 05:01.14 | jblack | drmessano: the new graphical zooming thing doesn't. |
| 05:01.18 | jblack | Java works here just fine. |
| 05:01.18 | drmessano | oh |
| 05:01.27 | drmessano | I was gonna say.. lol |
| 05:01.32 | pputman | jblack, I don't believe so. I know it's been in ubuntu but debian is a little more strict, even for their unstable. Takes a good bit of time for them to make a package. |
| 05:01.40 | drmessano | Probably works with flash, tho |
| 05:01.52 | jblack | pputman: I bet it's been in sid for a long time. Probably even testing. |
| 05:02.06 | pputman | jblack, hrm I didn't see it in a search, I'll look closer. |
| 05:02.50 | jblack | Yeah. "iceweasel_3.0~b5" |
| 05:03.27 | jblack | Why it's called iceweasel, rather than firefox, is both boring and technical. |
| 05:03.37 | drmessano | Yeah |
| 05:03.38 | jaytee | purity |
| 05:03.45 | drmessano | Trademark BS |
| 05:03.48 | jaytee | yep |
| 05:04.27 | drmessano | RMS-ish nazism |
| 05:04.27 | drmessano | Call it GNU Iceweasel, damnit |
| 05:04.37 | drmessano | GNU waterbeaver |
| 05:04.47 | jaytee | Zap-DADHI overly litigious society |
| 05:05.45 | drmessano | ha |
| 05:06.02 | drmessano | Wine 1.0 |
| 05:06.07 | drmessano | You know what that means |
| 05:06.17 | drmessano | I'm so running photoshop on my asterisk box now |
| 05:06.20 | jaytee | Armageddon? |
| 05:07.06 | jaytee | I'd like Gimp better if they came up with classier name for it. Reminds me too much of the guy in Pulp Fiction |
| 05:07.22 | drmessano | ..and Digg stories of horrible feats of virtualization, like running MacOS X inside a Windows VMware session on top of Wine |
| 05:08.09 | drmessano | Or Ubuntu inside of a VMware session running in Wine on Ubuntu |
| 05:08.24 | drmessano | It's DUBUNTU your pleasure, DUBUNTU your fun! |
| 05:09.36 | jaytee | I'm waiting for someone to come in here claiming they're running a VMWare server with a CentOS/Asterisk VM inside a Xen VM and running Skype -> SIP and Flash -> SIP out of it. |
| 05:09.48 | Qwell | again? |
| 05:09.52 | drmessano | Dude, that was so last week |
| 05:10.07 | JT | SIP FLASH 3GP, whateva u want |
| 05:11.19 | drmessano | Flash inside of Silverlight inside of Adobe AIR, on top of Vista, inside Ubuntu, running IAX3, connected to to Vonage via Skype |
| 05:11.24 | drmessano | I WIN THE INTARWEB |
| 05:11.43 | jaytee | hands down, no contest! |
| 05:12.14 | drmessano | With a MAGICJACK for e911, bitches |
| 05:12.15 | jaytee | does IAX3 do color or is it monocrhome? |
| 05:12.32 | drmessano | IAX3 is full VGA |
| 05:13.31 | jaytee | so it probably won't run on this old 386 with a Hercules card and Red Hat 3, darn. |
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| 05:16.10 | [TK]D-Fender | ok, bed time. Later all... |
| 05:16.35 | jaytee | I found out today Lumenvox only does 32 bit RPMS for RHEL 5 or CentOS. I knew running * on the stock Dell install of RHEL5 64 bit was gonna come back to bite me in the ass someday. |
| 05:17.42 | TrentCreek | then dont do lemonvox |
| 05:17.56 | TrentCreek | ;-) |
| 05:22.09 | jaytee | probably having a dedicated box for VR IVR and use IAX to route calls to it might be a better idea anyways |
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| 05:42.23 | drmessano | YAAAY |
| 05:42.24 | drmessano | Police say someone is taking aim at cars with large bricks. |
| 05:42.25 | drmessano | It has happened at least 18 times in the past few week along Route 22 in Hillside, Union County. |
| 05:42.37 | drmessano | That's my old stomping grounds |
| 05:42.41 | drmessano | WTG Hillside! |
| 05:55.42 | jblack | huh. I thought we made a war for those kids to go break stuff in? |
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| 06:48.55 | jblack | lol. put windows on a coffee maker, and the coffee maker gets hacked. |
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| 06:50.24 | ^shark_ | any system can get hacked into |
| 06:50.37 | nick125 | Some just make it easier. |
| 06:55.21 | ^shark_ | at my present job we are acquiring analog lines from the telco & since i know how good asterisk is in the enterprise, i am looking for info i can deliver to my boss. |
| 06:55.47 | ^shark_ | i want to encourage him to go for asterisk, making him realise the benefits and how it works. |
| 06:56.28 | jblack | Tell him he can try it for free. |
| 06:56.36 | nick125 | ^shark_: You can do what you do with a normal PBX for a lot cheaper. |
| 06:57.24 | jblack | You can do a lot more with it too. |
| 06:57.34 | nick125 | Yeah. |
| 06:58.08 | jblack | You can implement a new feature, tuned to your specific needs, in a weekend of hacking, that someone like nortel will try and charge $10k for. |
| 06:58.23 | nick125 | jblack: 10k? That's cheap. |
| 06:58.28 | ^shark_ | nick125: I am looking for info how i can convice him to take it on. Any URL would be a gr8 idea |
| 06:58.48 | jblack | nick125: Pardon? How do you know which features I'm thinking of? |
| 06:58.50 | pputman | nick125, $10k isn't that cheap for some =) |
| 06:59.12 | nick125 | haha |
| 06:59.42 | jblack | Seriously. I'm all for being corrected when I'm wrong. But be ready to back it up |
| 06:59.51 | pputman | ^shark_, you can have him read the introduction of the asterisk book. |
| 07:00.06 | pputman | www.asteriskdocs.org |
| 07:00.30 | nick125 | jblack: It's a joke ;-) |
| 07:00.57 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:01.05 | nick125 | Basically, I'm saying that you can't get much for $10k from Nortel or one of the "others". |
| 07:02.49 | mort_gib | Morning |
| 07:03.02 | ^shark_ | pputman: ok thanks -- |
| 07:03.14 | jblack | I think you're confused. |
| 07:03.20 | jblack | We're talking about features, not entire pbx's. |
| 07:03.38 | mort_gib | Quick question, is there ANY way of sending a SMS from * |
| 07:03.58 | jblack | If you can do it from a script, you can do it from asterisk. =) |
| 07:04.22 | mort_gib | How so?? :-) I can't do it from a script.... |
| 07:04.33 | jblack | I see there's a package called smsclient. Surely you can use system() or agi to call it. |
| 07:04.49 | jblack | smsclient - A program for sending short messages (SM / |
| 07:04.58 | jblack | There's also smssend and smstools. |
| 07:05.27 | jblack | I have no idea how they work. If you get one of them working, you're 90% of the way there. |
| 07:06.00 | pputman | in trunk there is a doc/sms.txt |
| 07:06.51 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 07:11.10 | mort_gib | pputman: ??trunk |
| 07:13.25 | pputman | mort_gib, trunk is the development branch that I had right in front of me, but looking closer, the sms app is also in asterisk 1.4 as well. go to your source directory and open doc/sms.txt |
| 07:13.39 | mort_gib | Hang on :-) |
| 07:16.37 | mort_gib | So this is specific to the operator.... |
| 07:17.07 | mort_gib | Damn |
| 07:17.55 | mort_gib | Okay, I have to install a * system for a client, and they want to go the full way, so that I'm to use * as the network monitoring notifier too.... |
| 07:18.05 | mort_gib | Sometime I wonder why we do this ;-) |
| 07:22.44 | JT | mort_gib: you can't use a script? |
| 07:28.53 | mort_gib | I Suppose so, but my main problem is the local telco.... |
| 07:30.42 | JT | ? |
| 07:31.46 | mort_gib | My client is in Gibraltar, they offer a VERY limited range of services... |
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| 07:33.56 | LuisTorres | Howdy |
| 07:35.27 | jblack | uh, hi. |
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| 08:50.15 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 08:53.40 | dandre | hello, |
| 08:54.38 | dandre | if I make changes to zapata.conf file, must I do a reload or restart? |
| 08:55.10 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 08:55.18 | krdian | dandre: zap restart |
| 08:56.45 | dandre | ok krdian, but that hangs up all zap channels, and there is no 'when convenient' option |
| 08:57.23 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 08:57.52 | creativx | hehe ja |
| 09:00.20 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:00.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 09:02.32 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-5-72.revip2.asianet.co.th) |
| 09:02.36 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:04.51 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:07.14 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:07.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:09.06 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:09.45 | *** join/#asterisk anthooooooooo (n=AnthonyC@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 09:09.50 | anthooooooooo | hello |
| 09:10.01 | anthooooooooo | I work wuth asterisk 1.4.19.1 |
| 09:10.10 | anthooooooooo | In my asterisk log, I have this: |
| 09:10.38 | anthooooooooo | WARNING[2348] chan_sip.c: Remote host can't match request NOTIFY to call |
| 09:10.48 | *** join/#asterisk shinao1 (n=shinao1@41.221.175.10) |
| 09:11.03 | anthooooooooo | What does it mean this warning and why this warning appears? |
| 09:11.09 | anthooooooooo | thanks for your help |
| 09:18.00 | ThoMe | is chan_capi in asterisk 1.4X? |
| 09:18.24 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 09:22.54 | *** join/#asterisk makkksimal (n=makkksim@e177223037.adsl.alicedsl.de) |
| 09:26.41 | tzafrir_laptop | ThoMe, yes (an external one, as usual) |
| 09:28.21 | ThoMe | tzafrir_laptop: not includet by asterisk? |
| 09:28.28 | tzafrir_laptop | no |
| 09:28.36 | ThoMe | tzafrir_laptop: you mean with "external, this: http://www.melware.org/ChanCapi ? |
| 09:28.43 | tzafrir_laptop | yes |
| 09:28.49 | ThoMe | tzafrir_laptop: Okay, thank you! |
| 09:30.33 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:32.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 09:32.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 09:34.09 | tzafrir_laptop | Every time I see that name I'm trying to decide if it's "malware" or http://c2.com/cgi/wiki?StoryOfMel |
| 09:35.38 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 09:44.15 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 09:48.54 | marc7 | is there any way to invoke FollowMe at the same time as a Dial? |
| 09:50.40 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 09:50.42 | loompek | morning... |
| 09:50.44 | russellb | sure, Dial(Local/123@foo&SIP/myphone) |
| 09:50.52 | russellb | and have the 123@foo extension call FollowMe |
| 09:51.00 | loompek | did you know you need ztdummy module for musiconhold to work properly? |
| 09:51.12 | marc7 | loompek: yep... got nailed by that one awhile ago |
| 09:51.46 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 09:51.51 | marc7 | russellb: that was shockingly simple! |
| 09:51.56 | russellb | nods |
| 09:51.56 | marc7 | *gives it a try* |
| 09:51.57 | russellb | :) |
| 09:59.38 | *** join/#asterisk friendly12345 (n=friendly@ppp59-167-89-42.lns2.mel6.internode.on.net) |
| 10:04.12 | *** join/#asterisk InformatiQ (n=user@unaffiliated/ramezhanna) |
| 10:04.27 | InformatiQ | hi, how do i connect two asterisk servers together ? |
| 10:06.50 | marc7 | InformatiQ: have you looked into IAX at all? |
| 10:07.22 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:07.38 | InformatiQ | this is my first time with asterisk at all and i am only using gui so far, but seems that i should check the conf files |
| 10:08.30 | marc7 | InformatiQ: yeah... if you're just trying to play in a sandbox and pass calls back and forth between two asterisk servers... learning about channels would be a good start. be sure to flip through the book if you haven't already. |
| 10:08.31 | marc7 | ~book |
| 10:08.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 10:09.21 | InformatiQ | thanks a mil marc7 |
| 10:09.51 | marc7 | InformatiQ: no worries. google will also give you a hundred dozen tutorials that are also great places to start |
| 10:10.16 | InformatiQ | i have the book now |
| 10:10.30 | InformatiQ | you mentioned i should look at the "channels" |
| 10:12.30 | marc7 | russellb: on the subject of channels... the app_followme documentation says followme.conf needs a "number => family/key"... however when I try keying in something like number => SIP/2015551234@peer, it immediately defaults to chan_local instead (unable to allocate a channel for Local/SIP/2015551234@peer@Internal cause: Unknown) |
| 10:13.06 | marc7 | so I try to change the line to number => 2015551234 and then add context => peer |
| 10:13.33 | marc7 | but I'm also guessing that's because peer is less of a context... |
| 10:13.37 | marc7 | *fixes* |
| 10:14.08 | marc7 | oh that's awesome |
| 10:15.39 | marc7 | InformatiQ: you said you're currently using a graphical frontend to asterisk -- which one? |
| 10:16.00 | InformatiQ | marc7: the default that comes with asterisknow |
| 10:18.38 | *** join/#asterisk awk (n=awk@security.web.za) |
| 10:20.05 | marc7 | InformatiQ: you'll have to forgive me, I'm starting to become better versed in helping people get up to speed, I'm still not familiar with all the different projects to be able to best suggest what action to take given your existing setup. |
| 10:20.51 | marc7 | I'd suggest at the very least, start with chapter 4 of the book, particularly page 67 onwards (but as much of the background material as you think you could use) |
| 10:21.33 | InformatiQ | marc7: no problem i just needed you to pint me to the right section of info |
| 10:21.35 | InformatiQ | thanks |
| 10:21.58 | marc7 | you can connect two asterisk servers together with SIP or IAX |
| 10:22.40 | marc7 | as a result, you'll need to make changes in sip.conf or iax.conf in your asterisk configuration directory |
| 10:22.52 | marc7 | you'll then also need to make changes to your dialplan so you can actually initiate calls |
| 10:23.04 | kaldemar | or whatever other protocol asterisk and your setup happens to support, really. |
| 10:23.25 | marc7 | right, I'm just trying to stick to the easy ones. |
| 10:23.34 | marc7 | if you can find any of what I'm talking about in the asterisknow GUI, you can try adding filling out those pages and saving the changes |
| 10:23.42 | InformatiQ | i am a bit familiar with the dialplan thing and i have used it to use service providers |
| 10:23.48 | marc7 | then, poke around in your asterisk configuration directory and see what's changed |
| 10:23.53 | InformatiQ | i'll be using IAX |
| 10:26.16 | marc7 | based on where calls will be going to/from, you may want to familiarize yourself with the notion of users and peers |
| 10:26.17 | marc7 | http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer |
| 10:27.14 | marc7 | oh, maybe not. |
| 10:27.39 | marc7 | i'm dating myself, apparently that's not a big deal anymore. carry on. |
| 11:08.54 | dominic1 | whe will the first 1.6 release marked as stable? |
| 11:10.02 | pputman | dominic1, i'm not sure anyone would be able to give you a specific date on that, because I doubt anyone knows for sure. |
| 11:10.41 | marc7 | does the dev team do release candidates? |
| 11:11.33 | marc7 | that'd be the only hint you'd have that it's nearing completion... |
| 11:12.03 | loompek | i've got a practical question... |
| 11:12.46 | loompek | in case i have a tcpdump file with sip&rtp and everything... would it be possible for me to dump the rtp ringback tone in a file and use it? |
| 11:13.04 | loompek | i tried it with wireshark.. but it seems i don't use the correct codec or something... |
| 11:14.29 | loompek | g771a |
| 11:14.35 | loompek | err |
| 11:14.36 | loompek | g711A |
| 11:14.39 | loompek | so alaw |
| 11:15.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:20.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:24.37 | *** join/#asterisk teletouch (n=teletouc@213.8.90.92) |
| 11:30.08 | teletouch | hi, is there any way to catch digit presses during music on hold? |
| 11:34.41 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:54.22 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
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| 11:55.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:12.45 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 12:12.49 | artz22 | hi everybody |
| 12:13.19 | artz22 | i was wondering, can I force asterisk to translate dtmf signals from a SIP phone to dtmf sounds on the other side? |
| 12:16.29 | kaldemar | if one phone is using rfc2833 and the other doesn't support it, you can set dtmfmode=auto and asterisk should send them inband to the other phone. |
| 12:17.56 | *** join/#asterisk ManxPower (n=manxpowe@88.sub-75-248-233.myvzw.com) |
| 12:18.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.48 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 12:20.52 | artz22 | kaldemar: that would be inside a config file right? |
| 12:21.29 | artz22 | sip.conf I see |
| 12:22.10 | kaldemar | yes. no guarantees if it will work but i'd give it a try. |
| 12:22.19 | *** join/#asterisk InformatiQ (n=user@unaffiliated/ramezhanna) |
| 12:23.41 | *** part/#asterisk InformatiQ (n=user@unaffiliated/ramezhanna) |
| 12:23.42 | artz22 | it doesn't work so far.. i'm still trying |
| 12:25.03 | *** join/#asterisk af_ (n=getsmart@88-149-230-105.dynamic.ngi.it) |
| 12:25.32 | kaldemar | explicit definition for the clients could also be worth trying. dtmfmode=inband in the client context for the one that doesn't support rfc2833 and dtmfmode=rfc2833 for the one that does. |
| 12:30.03 | *** join/#asterisk hmmhesays (n=hmmhesay@71-217-182-197.farg.qwest.net) |
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| 12:31.46 | artz22 | kaldemar: thanks, it's already done.. I was developing a channel driver and i wanted asterisk to generate the digits on my channel, I just return -1 in digit_end and digit_begin |
| 12:33.06 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:34.40 | jaytee | [TK]D-Fender, good morning |
| 12:34.54 | [TK]D-Fender | jaytee: mornin' |
| 12:35.17 | [TK]D-Fender | artz22: What are you interfacing? |
| 12:35.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:36.51 | ManxPower | what kind of tinker-toy phone does not support RFC2833? |
| 12:37.01 | artz22 | [TK]D-Fender: it's a custom usb-phone, a prototype |
| 12:37.21 | artz22 | it has a soundcard-like audio interface and a control one to handle ring and such |
| 12:37.34 | [TK]D-Fender | artz22: And you want this USB phone to plug directly to your * SERVER? |
| 12:37.50 | artz22 | that's the idea, what would be wrong? |
| 12:38.18 | [TK]D-Fender | artz22: Only useful for 1 person in the office and thats assuming the server is sitting right next to them.... kinda crazy. |
| 12:38.35 | artz22 | no, what I mean is that it can provide a line to asterisk |
| 12:38.48 | [TK]D-Fender | artz22: Your usual business has their servers in a rack far away from users. |
| 12:38.57 | artz22 | yes, I know. |
| 12:39.02 | ManxPower | "Thank you for buying SumoATA, bucking the trend in lightweight ATAs since 2008!" |
| 12:39.23 | [TK]D-Fender | artz22: Ok, so if its plugged into a PC, what is the PC doing in the equation? |
| 12:39.28 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 12:39.29 | ManxPower | artz22: a USB phone cannot provide a line to Asterisk. |
| 12:39.46 | [TK]D-Fender | artz22: because there are plenty of USB phones out there which interface with soft-phones. |
| 12:40.00 | ManxPower | ~fxofxs |
| 12:40.00 | jbot | i guess fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 12:40.05 | [TK]D-Fender | artz22: and do be very careful on your use of the term "line" |
| 12:40.18 | artz22 | yes, that's right |
| 12:41.22 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:41.22 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:41.37 | ManxPower | Everyone knows a line is something you inhale thru a straw. |
| 12:41.46 | [TK]D-Fender | artz22: So can you describe how your phone and the connected PC talk to * that requires you to write a channel driver? |
| 12:42.06 | [TK]D-Fender | ManxPower: .... Snorting lines os snow... on a 1 hor.... oh nevermind... |
| 12:42.12 | [TK]D-Fender | of* |
| 12:42.24 | [TK]D-Fender | goes back to caroling |
| 12:42.55 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:43.24 | tzafrir_laptop | ManxPower, a USB phone connected to the system running Asteisk will work fine with chan_{oss,alsa,console} |
| 12:43.30 | tzafrir_laptop | normally |
| 12:44.07 | ManxPower | tzafrir_laptop: yeah, but it doesn't provide a FXO to Asterisk |
| 12:44.15 | tzafrir_laptop | sure |
| 12:45.02 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 12:45.42 | artz22 | [TK]D-Fender: it's a FXO which linux detects in part as a soundcard and it also creates a /dev device to control on/off hook and ringing. dtmf detection too |
| 12:46.13 | artz22 | so now I have a channel driver based on chan_phone and chan_oss which receives the calls |
| 12:46.27 | [TK]D-Fender | artz22: Ok, so its a USB FXO to plug directly into your * server. Please don't use the term "USB phone". It is not a phone, it is a line interface. |
| 12:46.51 | [TK]D-Fender | artz22: And what you might want to consider is making your driver a front end to zaptel instead. |
| 12:47.16 | artz22 | [TK]D-Fender: yes, you're right |
| 12:47.19 | [TK]D-Fender | artz22: this way you don't have to add core code to *. |
| 12:47.37 | ManxPower | The only USB device that I have any respect for with regards to Asterisk is the Astribank. |
| 12:47.49 | [TK]D-Fender | artz22: Why invent a new channel interface when we already deal with analog FXO channels. |
| 12:48.01 | [TK]D-Fender | ManxPower: Yup, Astribank does all this already... |
| 12:52.03 | artz22 | artz22: it's just another way to do it, since I"m getting started with usb |
| 12:52.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 12:54.07 | [TK]D-Fender | artz22: Keep scalability in mind. Who wants a new channel driver module to handle jsut 1 device? |
| 12:54.31 | artz22 | yes, it's expandable to 4 probably.. still don't know |
| 12:54.37 | artz22 | or maybe more |
| 12:54.45 | artz22 | so far it's just a test |
| 12:55.04 | [TK]D-Fender | artz22: that single device being yours. |
| 12:56.16 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 12:56.48 | [TK]D-Fender | artz22: Heck, you've be better off simulating a network interface and talking SIP to *. |
| 12:57.03 | [TK]D-Fender | you'd* |
| 12:57.31 | x86 | TK and his SIP.... |
| 12:57.37 | x86 | sighs |
| 12:57.39 | x86 | ;) |
| 12:58.08 | artz22 | [TK]D-Fender: what do you mean with '*' ? |
| 12:58.18 | [TK]D-Fender | artz22: ... what channel are you in? |
| 12:58.26 | x86 | LOL! |
| 12:58.27 | artz22 | ohh hehe |
| 12:58.33 | artz22 | too much abbreviation |
| 12:58.34 | x86 | nub ;) |
| 12:59.20 | *** join/#asterisk bigdc (n=bigdc1@c-67-177-239-158.hsd1.co.comcast.net) |
| 12:59.39 | [TK]D-Fender | artz22: Fine... I insist that you cal it a "Universal Serial Bus Foreign eXchange Office interface" in FULL from now on! |
| 13:00.02 | artz22 | nice name! |
| 13:00.13 | artz22 | and nice initials :P |
| 13:01.08 | x86 | UFxO is a better name ;) |
| 13:01.26 | [TK]D-Fender | sends x86 into orbit |
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| 13:25.24 | hsv-al | hello fellow internet addicts |
| 13:25.30 | hsv-al | are we looking forward to another long & glorious day of irc? :) |
| 13:25.57 | ManxPower | hsv-al: I'm still in therapy from the last long and glorious day of IRC |
| 13:26.31 | hsv-al | got up at 4:03 this morning, ran 5 miles, came back showered, bought 3 bottles of sugar free red bull, large coffee, and a banana |
| 13:26.40 | hsv-al | and now im "fueled" for another day of internet addiction |
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| 13:28.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:29.05 | mercestes | Hey, anyone got some time to help me troubleshoot a zaptel issue real quick? I have a red alarm and the provider says that they are showing all my channels as "locked." I show sending a yellow alarm, and I've restarted zaptel a few times, but they are still "locked." Unfortunately I do not know where to get more specific information |
| 13:30.00 | *** join/#asterisk badcfe (i=christia@nothing.beats.deathporn.com) |
| 13:30.37 | *** join/#asterisk falco_toadfoot (n=torgeir@1.80-202-243.nextgentel.com) |
| 13:30.55 | falco_toadfoot | is there any way to control the iax2 packet sizes (frame size) ? |
| 13:31.55 | falco_toadfoot | right now all packets are reported to be 56 bytes total (iptraf) |
| 13:34.30 | mercestes | no love for me? :( |
| 13:38.31 | *** join/#asterisk uTx (n=unix@modemcable074.229-82-70.mc.videotron.ca) |
| 13:39.27 | s0ck | anyone understand fxorxgain/tx ? |
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| 13:48.04 | *** join/#asterisk remycouture (n=spid3r@host2.avdl.com) |
| 13:48.35 | remycouture | is there a software you would recommend to tunnel iax2 over tcp ? |
| 13:48.51 | Sargun_screen | remycouture: that's a bad idea, but openvpn. |
| 13:48.53 | mercestes | remycouture, openvpn |
| 13:49.03 | Sargun_screen | falco_toadfoot: licensing? |
| 13:49.11 | remycouture | allright guys thank you |
| 13:50.26 | *** join/#asterisk ESCulapio_ (n=elvyn@mail.abaca.com.do) |
| 13:52.29 | tzafrir_laptop | falco_toadfoot, any idea if the RTP packetization control applies to IAX2 as well? |
| 13:53.03 | russellb | they do not. |
| 13:53.17 | russellb | since ... IAX2 doesn't use RTP ... |
| 13:53.57 | tzafrir_laptop | OK. I always had the thout of IAX2 "embedding" RTP streams |
| 13:56.37 | mercestes | any love for my issue? |
| 13:59.05 | mercestes | got locked PRI channels on an E1. Resetting zaptel had no effect. |
| 13:59.20 | russellb | mercestes: what card? |
| 13:59.29 | [TK]D-Fender | mercestes: pastebi everything.... you should know better... |
| 13:59.31 | russellb | regardless of the answer, why don't you contact your vendor for support |
| 14:00.21 | russellb | unless you got your card from one of the companies that will say it's a "software issue", and do not support both the hardware and software you use |
| 14:00.23 | russellb | that would be a shame. |
| 14:00.30 | mercestes | yessir |
| 14:00.37 | mercestes | I did, vendor said the channels are locked |
| 14:00.55 | russellb | that's a shame. |
| 14:01.03 | russellb | goes back to work |
| 14:02.24 | mercestes | Pastebin: http://pastebin.ca/1050137 |
| 14:02.48 | viperdude | hi guys, do you know of anyway of doing a ENUM lookup on a port other than 53? |
| 14:02.50 | mercestes | Yea, zaptel restart didn't "unlock" the channels |
| 14:02.57 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:02.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.25 | russellb | viperdude: set up a DNS server to listen on a port other than 53 i guess |
| 14:03.42 | russellb | but i don't think you can configure asterisk that way |
| 14:03.48 | russellb | to connect via a different port |
| 14:03.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 14:03.57 | viperdude | russellb: no i want to query a DNS that is listening on a port other than 53 |
| 14:04.52 | russellb | viperdude: gotcha ... don't think we have an option for that |
| 14:05.14 | viperdude | hmmmmm |
| 14:05.26 | viperdude | x-connect want me to use port 9053 |
| 14:05.32 | ManxPower | Asterisk does not connect to DNS using port 53 or any port. The DNS resolving library (libresolv?) and the OS do that. |
| 14:05.59 | viperdude | ManxPower: so how do i get the resolver to use another port? |
| 14:06.23 | russellb | ManxPower: touche!!! |
| 14:06.28 | ManxPower | viperdude: heck if I know. I doubt you can. But regardless it's really a DNS/resolver/OS issue, not an Asterisk issue. |
| 14:06.35 | russellb | i obviously should have known that |
| 14:06.38 | russellb | stupid ... mornings ... |
| 14:06.59 | ManxPower | You could, of course, use iptables/ipchains to change the destination port number, then you don't have to screw with everything else. |
| 14:07.00 | *** join/#asterisk ^shark_ (n=^shark_@41.222.2.65) |
| 14:07.01 | Sargun_screen | ManxPower: I though libc resolvs |
| 14:07.11 | ManxPower | russellb: you correct me all the time, don't feel bad. |
| 14:07.21 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:07.23 | ManxPower | Sargun_screen: it could, hence the ? at the end of the libname I have. |
| 14:07.25 | ManxPower | gave. |
| 14:07.32 | viperdude | ManxPower: ok thanks for the hint |
| 14:07.54 | ManxPower | viperdude: unlike SIP/RTP/etc, DNS does not do weird stuff that breaks NAT, etc. |
| 14:08.00 | russellb | ManxPower: thanks :) |
| 14:08.13 | russellb | SIP </3 |
| 14:08.22 | russellb | an IETF disaster |
| 14:08.35 | ManxPower | Most of my consulting income does not come from Asterisk, it comes from network/WAN/router consulting. |
| 14:09.08 | *** join/#asterisk tapic (n=tap@88.255.77.200) |
| 14:09.19 | mercestes | [TK]D-Fender, You get pastebin? |
| 14:09.29 | coppice | DNS has plenty of dumbness. Upper and lower case being equivalent should have resulted in the death penalty |
| 14:09.29 | Sargun_screen | russellb: true that, IAX FTW! |
| 14:09.38 | russellb | IAX2 <3 |
| 14:09.40 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-1253d3475695bc0f) |
| 14:09.40 | Sargun_screen | Network consulting if fun. |
| 14:09.43 | *** join/#asterisk john_fbac (n=johnfbac@216.186.221.211) |
| 14:09.50 | Sargun_screen | DNS does have plenty of issues, but less than SIP |
| 14:09.52 | ManxPower | viperdude: you could also set up a local DNS server on the Asterisk server. named allows you to control the source port of the query, at least, I would assume it would allow you to control the destination port as well. |
| 14:10.21 | [TK]D-Fender | mercestes: Yes, and see that you didn't bother showing your configs, proc/interrupts, telling us the model of card you're using, or pretty much anything useful at all. |
| 14:11.01 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:12.21 | coppice | SIP a dee doo dah |
| 14:12.23 | coppice | SIP a dee yah! |
| 14:13.03 | mercestes | http://pastebin.ca/1050145 updated pastebin |
| 14:13.15 | tapic | hi all, my company operates a big IVR system on asterisk and we are trying to restructure our asterisk platform using socket server and I need some consultancy about setups, is this the right place to ask such questions or could you please direct me to the right directions? thx |
| 14:13.35 | mercestes | it is a wildcard TE1222 |
| 14:13.39 | mercestes | * TE122 |
| 14:13.55 | ^shark_ | ok -- i have an analog phone with an rj11 port & i am wondering what i shld use to connect it to my asterisk box |
| 14:14.11 | *** part/#asterisk mintee (n=mintone@75.150.132.150) |
| 14:15.44 | ^shark_ | i want a shared connection to my fxs port for the analog phones. |
| 14:15.47 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:15.56 | ^shark_ | wat sort of hardware do need for this? |
| 14:18.04 | tapic | we have installed asterisk-now and directed all calls to a socket server (asterisk.net c#) and I am testing the system using sipp and when I have about 20 or more concurrent connections and I dial the line from my desk, the call waits for long times until the test wav file is played. |
| 14:18.28 | [TK]D-Fender | ^shark_: Linsys SPA-2102 or PAP2 |
| 14:19.58 | tapic | any similar experiences with asterisk & external socket server? |
| 14:20.12 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 14:20.32 | [TK]D-Fender | tapic: What is a "socket server"? This is a very generic sounding term, perhaps you should clarify this a bit... |
| 14:20.33 | hwt | hey. does asterisk support the REFER method? all i get now is the 603 Declined (no dialog). |
| 14:20.51 | hwt | if anyone can point me in the direction of docs, it would be highly appreciated. |
| 14:22.19 | russellb | hwt: what asterisk version |
| 14:22.32 | tapic | I actually mean a seperate computer on the network running windows OS and a .NET application which uses asterisk.net API and communicates with asterisk using TCP sockets. |
| 14:22.40 | russellb | hwt: 1.2 _might_ not support it ... 1.4 absolutely does |
| 14:22.59 | hwt | russellb: 1.2. |
| 14:23.17 | russellb | a lot of SIP transfer work went into 1.4. |
| 14:23.18 | hwt | russellb: 1.2.14 that is |
| 14:23.26 | *** join/#asterisk underguiz (n=undergui@unaffiliated/underguiz) |
| 14:23.27 | russellb | chokes |
| 14:23.34 | russellb | at least try the latest version of 1.2 :) |
| 14:23.43 | [TK]D-Fender | tapic: .. "asterisk.net API"? Huh? |
| 14:24.03 | ^shark_ | [TK]D-Fender: i guess if i go with the SPA-2102 i connect it to a LAN switch and the rest of the analog phones connect to it, isnt it? |
| 14:24.06 | russellb | hwt: 64 changes to chan_sip in 1.4 since that version |
| 14:24.13 | hwt | russellb: yeah.. but this is on a prod box. |
| 14:24.15 | tapic | http://www.voip-info.org/wiki/view/Asterisk+.NET |
| 14:24.20 | [TK]D-Fender | ^shark_: Yes |
| 14:24.37 | hwt | basically what i want is that when it gets a REFER, generate an INVITE to the Refer-To value. |
| 14:24.55 | [TK]D-Fender | tapic: Next time, just say its an AGI. |
| 14:25.09 | russellb | if only the code could modify itself while running. |
| 14:25.21 | [TK]D-Fender | tapic: Usually its not that bad, so I'd question your windows side for speed issues |
| 14:25.47 | [TK]D-Fender | russellb: remember chan_skynet.so? You don't want that happening AGAIN, do you? |
| 14:25.58 | russellb | no, i don't :( |
| 14:26.26 | russellb | (want that to happen again, that is) |
| 14:26.29 | [TK]D-Fender | sends chan_skynet.so back in time to kill russellb's inner child. |
| 14:27.07 | ManxPower | hwt: It sounds like you need a SIP proxy. |
| 14:27.32 | hwt | ManxPower: we have that, but i am trying to work around a problem on our SS7 gw. |
| 14:27.34 | tapic | ok its an AGI:) I am new to the world of asterisk, thanks for guidance. Is there any limit or configuration for external socket connections on the asterisk side? |
| 14:27.48 | ManxPower | hwt: What SIP proxy are you using? |
| 14:29.20 | [TK]D-Fender | tapic: AGI can load a system down, how many simultaneous channels? What server hardware? |
| 14:32.53 | hwt | ManxPower: SER |
| 14:33.46 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:33.48 | ManxPower | hwt: SER should easily be able to do the custom modifications to the SIP packets/transaction. I strongly doubt Asterisk can do it without significant coding on your part. |
| 14:35.47 | *** join/#asterisk peterpen (n=fantasti@212.57.232.254) |
| 14:36.03 | peterpen | Lo, I'm trying to follow the guide at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 14:36.16 | peterpen | (Auto dial out by putting call files in /var/spool/asterisk/outgoing) |
| 14:36.44 | peterpen | I put my file in and it seems to get deleted straight away, the call isn't made and I've got 'asterisk -vvvvr' running and it doesn't react when I copy it in |
| 14:36.50 | peterpen | Any ideas where I might be going wrong? |
| 14:37.17 | tapic | hardware is p4 2.4 1gb ram, when I send over 20 concurrent SIP calls I starts to make following calls wait |
| 14:38.13 | tapic | no such case when used with an AGI file on the server. |
| 14:38.25 | ManxPower | peterpen: /path/to/src/asterisk/doc/callfiles.txt was not helpful? |
| 14:38.35 | tapic | exten => _2005,1,AGI(agi://10.10.1.41/customivr) |
| 14:39.42 | ManxPower | peterpen: just don't tell anyone about the "doc" directory. It's the best kept Asterisk secret. |
| 14:40.00 | *** join/#asterisk nauticalthinker (n=mratliff@cust-baileys-90-146.mounet.com) |
| 14:40.01 | [TK]D-Fender | tapic: I am failing to understand why you are mixing SIP & AGI into this issue. |
| 14:40.01 | tapic | is the extension when I use the external application server |
| 14:40.52 | peterpen | ManxPower, I won't :P |
| 14:40.57 | tapic | I am using AGI and test it with SIP calls using SIPP |
| 14:41.37 | ManxPower | peterpen: also you should create the .call files in a different directory on the SAME partition, then mv the file, that way there is no race condition that might cause Asterisk to delete a partially created .call file because it thinks it's a corrupted file. |
| 14:41.55 | tapic | exten => _2003,1,AGI(agi-test.agi) is the extension when I use an internal agi file. |
| 14:41.55 | peterpen | ah ok |
| 14:42.18 | ManxPower | tapic: What is your QUESTION? |
| 14:42.42 | peterpen | ManxPower, no that didn't help - I'll keep reading the doc for things I might have missed, but to be honest I've read alot of this paraphrased on voip-info |
| 14:43.02 | ManxPower | peterpen: voip-info is full of wrong information |
| 14:43.16 | peterpen | :D |
| 14:43.16 | [TK]D-Fender | peterpen: You might want to consider using an AMI Originate instead of call-files. |
| 14:43.32 | peterpen | probably but I've been asked to get call files to work |
| 14:43.47 | peterpen | AMI Originate is easier I know, I wrote an operators panel in java and a CTI client built in :P |
| 14:43.52 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 14:44.09 | [TK]D-Fender | peterpen: Either way, ManxPower has told you what you should do to deal with *'s locking quirks |
| 14:44.41 | peterpen | well I tried it, but to no avail - ignore me for a bit while I read up a little more |
| 14:44.48 | ManxPower | peterpen: if you want to create a file to be processed in the future, set the timestamp of the file to the future. |
| 14:44.51 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:44.53 | tapic | my question is that if there is a configuration in asterisk for the maximum number of connections with an AGI |
| 14:44.58 | peterpen | ManxPower, yes I know that |
| 14:45.11 | seanbright | tapic: not that i am aware of. |
| 14:45.27 | ManxPower | tapic: you would have to check the source code to be sure, but I don't think there is. |
| 14:45.44 | ManxPower | There is, of course, an OS limit of about 65,000 sockets |
| 14:45.45 | seanbright | tapic: the limit on the number of open files |
| 14:46.02 | ManxPower | Then there is the OS enforced limit for open files (sockets are files) using ulimit |
| 14:46.37 | seanbright | if you need more than 2^16 agi calls, you might need a second box |
| 14:46.37 | seanbright | :) |
| 14:46.54 | *** join/#asterisk Greek-Boy (n=email@41.221.58.14) |
| 14:48.02 | tapic | thank you for your comments, I am experiencing a huge performance difference between the two extensions I have mentioned above |
| 14:48.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:48.44 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:49.06 | Greek-Boy | Re Astricon, is everyone excited yet? :) |
| 14:49.08 | ManxPower | tapic: AGI does fork/exec every time it is called, whereas FastAGI does not have to. |
| 14:52.42 | seanbright | tapic: FastAGI is the way to go when you have high volume |
| 14:54.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:56.34 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 15:07.16 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-77.unitz.ca) |
| 15:07.35 | eric2 | is there a way to turn down the volume of the music on hold? |
| 15:09.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:09.59 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:13.54 | tapic | yes I was thinking this way, but experience a kind of bottleneck with fastagi and can not find out why.. |
| 15:14.58 | tapic | suspect that the underlying .net library (which handles the socket connections with asterisk) has some issues |
| 15:18.38 | *** join/#asterisk pputman- (n=centrex@c-68-62-214-146.hsd1.al.comcast.net) |
| 15:22.08 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 15:22.32 | l0verb0y | hey does anyone know the format of caller id in a call file? |
| 15:23.07 | pputman- | callerid:3843838343 ? |
| 15:23.17 | l0verb0y | what about for leaving a name with the caller id? |
| 15:23.22 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:23.26 | pputman- | not sure |
| 15:23.37 | russellb | it's documented in the sample call file ... |
| 15:23.40 | russellb | sample.call |
| 15:24.01 | l0verb0y | thanks |
| 15:25.26 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 15:26.40 | *** join/#asterisk ar3dam (n=ar3dam@189.156.243.80) |
| 15:27.23 | *** join/#asterisk nauticalthinker (n=mratliff@cust-baileys-90-146.mounet.com) |
| 15:28.18 | nauticalthinker | what is the best approach to providing asterisk load balacing on multiple servers? |
| 15:28.41 | _ShrikE | openser |
| 15:29.27 | ar3dam | hi there, some can guideme how to make call across the fxo? |
| 15:29.31 | nauticalthinker | I've not tried openser... is it pretty simple as far as configuration |
| 15:29.55 | _ShrikE | nauticalthinker: its not that bad. |
| 15:29.56 | ar3dam | i ve installed, and running, because, i cant find what is the problem. |
| 15:30.10 | nauticalthinker | okay...I'll check it out...thanks for the tip |
| 15:31.58 | BCS-Satori | I have several systems that experience a delay on audio files, for example when users check their voicemail, they here "comedian mail, mailbox?" where 95% of the time "comedian mail" is distorted or only half spoken. It appears to happen on xlite, linksys, and polycom phones. Any idea? |
| 15:32.56 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:33.25 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 15:33.28 | r0land | hello all :) |
| 15:33.32 | r0land | hi [TK]D-Fender |
| 15:33.50 | r0land | i was wondering if some1 could help me to add an option to my incoming operator menu |
| 15:34.21 | r0land | currently, when some1 calls in, he gets a recorded msg asking for him to punch in an extension or dial 100 for operator assistance |
| 15:34.55 | r0land | wht i want is to add 2 other things, firstly, if in a period of time the person didnt punch in an extension i want him to b directed atomaticly to the operator |
| 15:35.11 | r0land | 2ndly, to add an option of lets say, press 2 to listen to availabe extensions |
| 15:36.05 | r0land | http://www.pastebin.ca/1050215 |
| 15:36.09 | r0land | this is my current extensions.conf |
| 15:37.04 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:37.05 | *** join/#asterisk Schreiber1337 (n=Schreibe@spectrumcontrol.com) |
| 15:39.43 | BCS-Satori | r0land: This is what I made for my attendant, take a look. http://rafb.net/p/7UlReP60.html It should help you. |
| 15:40.18 | r0land | BCS-Satori lemme see jus a sec |
| 15:40.56 | Schreiber1337 | Anyone upgrade from Ubuntu 6.10 to 8.04 recently, any problems with Asterisk after the upgrade? |
| 15:41.54 | BCS-Satori | r0land: exten = s,n,WaitExten(15) |
| 15:42.29 | r0land | BCS-Satori tht would wait asterisk wait for 15 seconds to accept the digits entered |
| 15:42.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:42.39 | BCS-Satori | r0land: That is my period of 15 seconds of no response. to get sent to the next step of a ringroup |
| 15:42.50 | r0land | though i want something to b added right after this, that if the caller didnt punch in any digit to direct him to say 100 |
| 15:43.41 | BCS-Satori | r0land: so the WaitExten(15) followed by a Dial(SIP/100) ,if you are using SIP |
| 15:43.52 | r0land | exten => _100,1,Dial(SIP/100,15) |
| 15:43.57 | ThoMe | hello |
| 15:44.07 | r0land | BCS-Satori would this do! exten => _100,1,Dial(SIP/100,15) |
| 15:44.22 | ThoMe | I would like over a other server dial |
| 15:44.30 | ThoMe | switch => IAX2/systemimpuls/eingehend <<is this correct? |
| 15:44.43 | ThoMe | but the other server dial my calling number from server1 |
| 15:44.56 | ThoMe | how i can set my dialnummer, which I want |
| 15:45.14 | BCS-Satori | r0land: that would work as long as your attendant is operating udner _100 |
| 15:45.24 | r0land | BCS-Satori ya she is |
| 15:46.00 | BCS-Satori | r0land: not from your paste it isnt. |
| 15:46.30 | r0land | BCS-Satori 201 is my sipura PSTN extension |
| 15:46.55 | r0land | BCS-Satori tht means if some1 called, sipura directs the phone to extension 201 to asterisk |
| 15:47.01 | BCS-Satori | r0land: exten => 201,4,WaitExten(8) exten => 201,5,Dial(SIP/100,15) |
| 15:47.19 | r0land | hmm ok |
| 15:47.20 | r0land | thanks :) |
| 15:47.21 | BCS-Satori | r0land: correct and you are making 201 act as the attendant |
| 15:47.27 | r0land | ah ok ok |
| 15:47.41 | r0land | sorry i guess i missunderstood the attendant issue i thought it would b the actual operator |
| 15:48.15 | r0land | BCS-Satori ok so wht about a 2nd option in the menu |
| 15:48.57 | r0land | like while playing the msg "welcome in .... if u know the extension of ther person u are calling dial it now or press 2 for available extensions" |
| 15:49.00 | *** join/#asterisk markgreene (n=markgree@209.12.142.2) |
| 15:49.13 | r0land | how can i tell asterisk tht if some1 pressed "2" to play him a certain msg |
| 15:49.59 | russellb | set up custom features in features.conf |
| 15:50.01 | russellb | DYNAMIC_FEATURES |
| 15:50.28 | BCS-Satori | r0land: look at my example, for Sales, 1,1 the first 1 means thats what they entered, for tech support 2,1 means they hit 2 and so on |
| 15:50.38 | markgreene | Hey guys. Quick question. I want to set the outgoing callerID depending on what extension is making the call. For me it seems I am have a misunderstanding of how to use variables. I was trying something like, exten => _1XXXXXXXXXX,1,SET(CALLERID(all) = ${OUT_CID${CALLERID(num)}}) |
| 15:50.46 | r0land | ok got it |
| 15:50.48 | r0land | thank you BCS-Satori |
| 15:50.51 | BCS-Satori | r0land: its all under the same [header] |
| 15:50.51 | r0land | BCS-Satori appreciate it |
| 15:50.59 | r0land | BCS-Satori yep saw it thank you |
| 15:51.05 | BCS-Satori | r0land: no problem |
| 15:51.08 | markgreene | Where I have variables set above it along the lines of OUT_CID0449 = Mark Greene <12051234567> |
| 15:51.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:51.53 | markgreene | But I can't seem to pull the OUT_CID0449 variable. I tried to output it with a NoOp statement and it's blank... |
| 15:53.45 | badcfe | on a sip channel, does asterisk _always_ send a 100 Trying when receiving a INVITE request? |
| 15:56.12 | russellb | looks like it |
| 15:57.35 | markgreene | Can someone tell me if i am allowed to set a variable just by doing something like "INT_CID0449 = Mark Greene <1234567891>" ? |
| 16:01.49 | *** join/#asterisk Corazu (n=Corazu@bas3-toronto12-1128688788.dsl.bell.ca) |
| 16:01.59 | Corazu | Hi chan |
| 16:02.34 | Corazu | I'm having some problems getting a card to work. |
| 16:02.42 | russellb | what card? |
| 16:03.36 | Corazu | TDM400 (I think..let me check the exact model) - but I'm getting a zt_chanconfig no such device or address )6) error..and the searches I've done on it don't seem to give me any information to help me |
| 16:03.48 | russellb | please contact digium technical support |
| 16:03.54 | russellb | they provide free installation assistance |
| 16:04.08 | Corazu | Alright thanks |
| 16:04.22 | russellb | np |
| 16:04.41 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:05.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:05.23 | [TK]D-Fender | markgreene: exten => _1XXXXXXXXXX,1,SET(CALLERID(all) = ${EVAL(${OUT_CID${CALLERID(num)}})}) |
| 16:08.12 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 16:08.20 | s0ck | how do i tweak gain on a tdm/aex card? |
| 16:08.35 | s0ck | it sounds too quiet outbound, so i guess txgain |
| 16:08.40 | Strom | s0ck: rxgain and txgain in zapata.conf |
| 16:09.08 | s0ck | Strom: ya, but what values do what? i cant find a guide to tweaking it |
| 16:09.49 | pputman- | s0ck, it's in decibels, but I would generally up it to anywhere between 1-5 and see if that improves |
| 16:10.27 | s0ck | is 5 max? |
| 16:10.43 | s0ck | what do minus values achieve? :s |
| 16:10.50 | *** join/#asterisk deeperror (n=deeperro@76.226.176.21) |
| 16:10.53 | outtolunc | = putting a sock in it <G> |
| 16:10.56 | pputman- | no 100 is max but after a certain amount you start causing more problems like echo. and the minus values lower it... |
| 16:11.32 | markgreene | [TK]D-Fender, EVAL function is not working for me. |
| 16:11.41 | pputman- | s0ck, actually scratch that, i dont know if the max is 100 or not |
| 16:11.46 | [TK]D-Fender | markgreene: pastebin your new dialplan and the CLI output. |
| 16:11.47 | markgreene | [TK]D-Fender, I am getting the same results. A blank string |
| 16:12.00 | Strom | s0ck: ideally, you'll adjust the gains in conjunction with a milliwatt test to measure and compensate for the exact attenuation on your circuit |
| 16:12.06 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:12.15 | deeperror | have a queue, agent answers phone and is talking with caller. I would like for the agent to have the ability to press #0 and Playback the called id...any clues? |
| 16:12.53 | pputman- | Strom, only you have the equipment laying around in your home to do such a thing though :P |
| 16:13.06 | Strom | no I don't |
| 16:13.14 | Strom | milliwatt test is located at the telco |
| 16:13.22 | Strom | you just need to know the number |
| 16:13.35 | Qwell | why is that number so difficult to get? |
| 16:13.53 | Strom | is it? I just use the telco's CLEC site |
| 16:14.02 | Qwell | in some localities |
| 16:14.15 | Strom | *shrug* telcos have a tradition of secrecy? |
| 16:14.20 | Qwell | speaking of random numbers... |
| 16:14.25 | Qwell | Strom: is popcorn now dead? |
| 16:14.58 | Qwell | I know they officially stopped supporting it, but is it gone now? |
| 16:15.12 | s0ck | so what is the option for me who has no special equipment |
| 16:15.17 | s0ck | mess around until it sounds right? ;/ |
| 16:15.25 | Strom | Qwell: well, popcorn was the northern california number |
| 16:15.36 | Strom | in southern california, it was 853-1212 |
| 16:15.42 | Strom | but...yeah, it's gone |
| 16:15.48 | Qwell | shame |
| 16:15.58 | Strom | oh well |
| 16:17.40 | pputman- | s0ck, pretty much |
| 16:17.41 | Strom | Qwell: did I show you my photos from yesterday morning? |
| 16:17.47 | Qwell | no.. |
| 16:17.50 | Qwell | should I be scared? |
| 16:18.00 | Strom | um |
| 16:18.01 | Strom | I don't know |
| 16:18.04 | s0ck | isn't there some debug output for checking gain levels... |
| 16:18.05 | Strom | do weddings terrify you? :) |
| 16:18.07 | Strom | http://www.flickr.com/photos/stromcarlson/sets/72157605672090051/ |
| 16:18.29 | dandre | hello |
| 16:18.35 | Qwell | if it was your wedding...maybe |
| 16:18.46 | s0ck | pputman: can you elaborate? :) |
| 16:19.50 | ThoMe | hm, is it not posible? exten => _X.,n,${nummer}@eingehend |
| 16:19.55 | ThoMe | have: Jun 18 18:19:11 WARNING[16846]: pbx.c:1720 pbx_extension_helper: No application '${nummer}@eingehend' for extension (freeline, 3866767, 6) |
| 16:19.58 | dandre | I have a problem with one phone and conference : the user that enter the meetme room is announced twice. How could I fix it? |
| 16:19.58 | ThoMe | nummer = number |
| 16:20.02 | ThoMe | eingehend = incomming |
| 16:20.17 | [TK]D-Fender | ThoMe: and no APPLICATION <--- |
| 16:20.41 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 16:20.43 | ThoMe | [TK]D-Fender: hm. i dont know what you mean. :-( |
| 16:20.43 | [TK]D-Fender | ThoMe: You don't jsut shove some kind of number there, you call an APPLICATIOn to do something. To place a call does DIAL ring a bell? |
| 16:20.49 | ThoMe | hm |
| 16:20.53 | ThoMe | i need |
| 16:20.57 | ThoMe | DIAL(nummer@bla) ? |
| 16:20.59 | [TK]D-Fender | DIAL <-- |
| 16:21.05 | pputman- | s0ck, I dunno, I usually set it to 5 and see if it sounds better. if it needs to go higher I up it, lower I lower it.... |
| 16:21.12 | [TK]D-Fender | ThoMe: Go read the book, you seem to have completely regressed |
| 16:22.06 | ThoMe | [TK]D-Fender: hm. I would like only jump to "eingehend" with my insert number. |
| 16:22.10 | markgreene | [TK]D-Fender, here is my pastebin output. http://asterisk.pastebin.ca/1050267 |
| 16:22.18 | deeperror | when an extension is dialed from a queue is there a way to prevent dtmf from being heard by the caller when the callee is pressing to execute a feature? |
| 16:22.40 | [TK]D-Fender | ThoMe: you don't "jump" anywhere. this is no way to link servers together, and you need to reference a technology. |
| 16:23.26 | s0ck | pputman: alreet |
| 16:23.30 | harryv | is there a variable specifying the sound lib? |
| 16:23.33 | *** join/#asterisk [tasty]freeze (n=yamahabr@204-181-48-126.skybest.com) |
| 16:23.35 | [tasty]freeze | I have been looking for a while now, but I cannot seem to find any information on how to dial multiple extension at once, for say incoming calls; the only thing I can find are tutorials using FreePBX etc, and I want to learn to do it by hand. Any help? |
| 16:23.44 | s0ck | i guess i would do something like this... |
| 16:23.54 | ThoMe | [TK]D-Fender: is it posible? exten => _X.,n,DIAL(eingehend/${nummer}) |
| 16:24.00 | Strom | [tasty]freeze: Dial(SIP/100&SIP101&SIP/102) |
| 16:24.02 | Strom | etc etc etc |
| 16:24.04 | s0ck | install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp opermode=UK fxotxgain=5.0 && /sbin/ztcfg |
| 16:24.06 | s0ck | what ya reckon? |
| 16:24.14 | [tasty]freeze | Strom: Thank you! |
| 16:24.15 | errr | the sound quality on my pri just went to hell in a hand basket. I have a sangoma a101d. What can I do to rule out the sangoma card as being the problem? |
| 16:24.18 | Qwell | s0ck: set the gains in the config |
| 16:24.28 | [TK]D-Fender | markgreene: ... |
| 16:24.32 | *** join/#asterisk Schreiber1337 (n=Schreibe@spectrumcontrol.com) |
| 16:24.33 | s0ck | Qwell: it's done in software then... |
| 16:24.33 | [TK]D-Fender | markgreene: -- Executing [8500@office:1] NoOp("SIP/0229-09740ff8", "CID is 0229") in new stack |
| 16:24.41 | [TK]D-Fender | markgreene: You don't HAVE a variable for that CID! |
| 16:24.43 | anonymouz666 | Strom: oh my... I shouldn't click on that link |
| 16:24.49 | Strom | anonymouz666: ? |
| 16:25.23 | s0ck | pputman: would you try increments of 5? |
| 16:25.32 | [TK]D-Fender | markgreene: == Setting global variable 'OUT_CID0449' to 'Mark Greene <2051234567>' == Setting global variable 'OUT_CID8994' to 'Party-Extras <18661234567>' |
| 16:25.36 | Qwell | s0ck: 1-2 at best |
| 16:25.39 | [TK]D-Fender | markgreene: no 0229! |
| 16:25.44 | s0ck | ok |
| 16:25.46 | Qwell | it can also take a decimal |
| 16:25.50 | s0ck | ic |
| 16:25.56 | Qwell | (though, more/less than .5 isn't useful) |
| 16:26.15 | *** join/#asterisk john_fbac (i=johnfbac@88.sub-75-202-94.myvzw.com) |
| 16:26.40 | dandre | I have a problem with one phone and conference : the user that enter the meetme room is announced twice. How could I fix it? |
| 16:26.52 | [TK]D-Fender | dandre: SHOW US |
| 16:27.34 | deeperror | when an extension is dialed from a queue is there a way to prevent dtmf from being heard by the caller when the callee is pressing to execute a feature? |
| 16:29.06 | ManxPower | deeperror: that should be the default |
| 16:29.23 | ManxPower | unless you did something silly like configure the phone for inband dtmf and Asterisk for rfc2833 drmf. |
| 16:29.30 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:29.45 | markgreene | [TK]D-Fender, HA! Wow. One of those things that you need a second pair of eyes for. Thanks so much |
| 16:29.51 | deeperror | ManxPower, the phone is analog |
| 16:30.05 | [TK]D-Fender | markgreene: Another reason to always pastebin EVERYTHING. |
| 16:30.06 | deeperror | ManxPower, but i hear short blips on the caller side |
| 16:30.07 | ManxPower | deeperror: connected to a TDM4xxP? |
| 16:30.24 | deeperror | ManxPower, A104 |
| 16:30.25 | dandre | [TK]D-Fender: http://pastebin.org/44491 |
| 16:30.32 | [TK]D-Fender | markgreene: People tend to say "But I did everything right, and it doesn't work...". Sort answer = BULLSHIT :p |
| 16:30.37 | [TK]D-Fender | short* |
| 16:30.47 | harryv | i'm writing some agi-stuff that executes a Record() -- i need to check if the file exists afterwards, but then i need to know where asterisk sounds is located. default is /var/lib/asterisk/sounds -- but i suppose you can change that compilation time or something. can't i extract which dir is used? |
| 16:30.55 | ManxPower | Ah, so you are only hearing blips, not the full DTMF. tones. I doubt there is much you can do about that. |
| 16:31.23 | ManxPower | harryv: Record(/path/where/you/want/files/soundfile.wav) |
| 16:31.38 | deeperror | ManxPower, correct...but wouldn't asterisk be involved in shortening this? as if caller dials dtmf i hear the full tones |
| 16:31.51 | harryv | ManxPower: sure, but if the user chooses to do it relative to asterisk sound path .. |
| 16:32.11 | ManxPower | OK, do you hear full tones or blips? |
| 16:32.22 | [TK]D-Fender | dandre: you hear it twice because you hear the same announcement everyone else hear about your entering the conference. |
| 16:32.22 | ManxPower | (11:30:06 AM) deeperror: ManxPower, but i hear short blips on the caller side |
| 16:32.32 | ManxPower | There is a big difference between a blip and a full DTMF tone. |
| 16:32.45 | deeperror | ManxPower, caller hears blips when callee presses buttons, callee hears full tones when caller presses buttons |
| 16:33.04 | [TK]D-Fender | dandre: this is not a bug. If you didn't announce your callers, it would jsut thank you privately, and then have you enter. |
| 16:33.10 | Strom | deeperror: "called party" and "calling party" -- not "callee" and "caller" |
| 16:33.23 | ManxPower | deeperror: don't say "caller" and "callee" as EITHER one could initate the transfer. say transferer and transferee or something like that. |
| 16:33.29 | [TK]D-Fender | harryv: asterisk.conf <- spool folder, under sounds. |
| 16:33.43 | [TK]D-Fender | harryv: and you can set the precise file YOURSELF in Record. |
| 16:33.44 | ManxPower | deeperror: is there SIP involved anywhere in the call path? |
| 16:33.45 | dandre | No the conf members hear the announcment twice |
| 16:33.55 | deeperror | ManxPower, sip termination |
| 16:34.08 | deeperror | rfc2833 |
| 16:34.10 | [TK]D-Fender | dandre: they hear both? That would be unusal. |
| 16:34.15 | seanbright | ~rfc2833 |
| 16:34.23 | seanbright | we need one of those |
| 16:34.34 | [TK]D-Fender | seanbright: Go make it. |
| 16:34.36 | ManxPower | [TK]D-Fender: I didn't see the announcement being played twice in dandre's pastebin. Did I miss it? |
| 16:34.42 | dandre | the members that are already in the conf |
| 16:34.43 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 16:35.04 | seanbright | [TK]D-Fender: nah |
| 16:35.05 | seanbright | :) |
| 16:35.08 | [TK]D-Fender | ManxPower: http://pastebin.org/44491 <- 21 /25. He says people in the conf hear both. |
| 16:35.21 | ManxPower | deeperror: now, restate the problem using the terms transferer and transferee so I make sure I understand it correctly |
| 16:35.21 | dandre | Â Â -- <Zap/pseudo-1524281664> Playing '/var/spool/asterisk/meetme/meetme-username-31-2' (language 'fr') |
| 16:35.21 | dandre | appears twice |
| 16:35.37 | deeperror | but there is no transfer occuring |
| 16:35.40 | ManxPower | [TK]D-Fender: I see it now. |
| 16:35.57 | ManxPower | deeperror: then why are you pressing DTMF? |
| 16:36.01 | deeperror | i place a call to a number and this rings into a queue and an agent answers |
| 16:36.26 | ManxPower | Golly, Beave, a foll description of the problem is helpful? |
| 16:36.27 | deeperror | the agent presses #9 to replay an id number of the call to them if it doesn't show up on cid |
| 16:36.33 | dandre | can this be related to it?: |
| 16:36.33 | dandre | [Jun 18 18:29:07] NOTICE[22547]: app_meetme.c:1918 conf_run: Audio bytes: 160Â Buffer size: 320 |
| 16:36.34 | *** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net) |
| 16:36.36 | *** join/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 16:36.45 | anebi |  i want to ask how can i disable hardware support for asterisknow and asterisk |
| 16:36.55 | anebi | because we use only sip |
| 16:36.58 | deeperror | ManxPower, which is setup in features.conf as an applicationmap |
| 16:37.01 | UnixDog | rm the card |
| 16:37.03 | ManxPower | deeperror: OK, so you have analog phone -> analog card/Asterisk -> Internet/SIP -> SIP provider -> PSTN? |
| 16:37.04 | anebi | and in the logs we get a lot of errors for zap and so on. i would like to disable these errors |
| 16:37.13 | [TK]D-Fender | dandre: What ver of *? |
| 16:37.35 | dandre | Asterisk 1.4.18.1 |
| 16:37.37 | UnixDog | well zaptel is used for timing |
| 16:37.50 | ManxPower | anebi: pastebin the errors |
| 16:37.55 | deeperror | analog phone -> fxs bank -> t1 card -> asterisk -> sip provider |
| 16:38.13 | ManxPower | deeperror: and the SIP provider is running the Queue app? |
| 16:38.19 | deeperror | asterisk is |
| 16:38.33 | ManxPower | deeperror: so SIP is not involved at all? |
| 16:39.17 | ManxPower | If one leg of the call is SIP then we trouble shoot in a totally different direction than if SIP is not involved. |
| 16:39.29 | ManxPower | I have to get back to work. Best of luck, deeperror |
| 16:39.33 | deeperror | all inbound and outbound calls from my asterisk box are sip to our provider. everything on the inside is analog |
| 16:39.58 | ManxPower | deeperror: Perhaps if you write up an e-mail and ask on the mailing list? |
| 16:39.58 | deeperror | but it's more the dtmf that is being heard over the line that asterisk is passing back to one leg of the call |
| 16:40.06 | deeperror | ManxPower, ok thanks for your time |
| 16:40.46 | anebi | actually they are warnings, but we would like to disable these modules, because we don't use them (hardware support) |
| 16:40.46 | anebi | http://pastebin.com/d4802fe9f |
| 16:40.53 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.mn.comcast.net) |
| 16:41.18 | dandre | [TK]D-Fender: Asterisk 1.4.18.1 |
| 16:42.43 | [TK]D-Fender | dandre: might want to try upgrading... |
| 16:43.12 | dandre | ok I'll try |
| 16:43.46 | [TK]D-Fender | anebi: "noload [modulename]" in modules.conf |
| 16:44.32 | anebi | [TK]D-Fender: this way sip will work, but hardware support will be disable ? |
| 16:44.37 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 16:44.50 | anebi | aa, we disable with modulename |
| 16:44.53 | *** part/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 16:44.53 | anebi | ok, thanks |
| 16:46.50 | dandre | I have tried with another phone and add no announcement problem |
| 16:50.42 | artz22 | I'm trying a sip phone and i hear a big latency, is it phone dependant only or can i modify some parameters in asterisk |
| 16:52.00 | [TK]D-Fender | artz22: Its either the networking or the phone, not * |
| 16:52.19 | [TK]D-Fender | artz22: I've noticed inexplicable lag with Ekiga in the past. |
| 16:53.04 | artz22 | I imagined that.. latency is about 1 sec sometimes :S |
| 16:57.42 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 17:00.00 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 17:09.18 | ThoMe | [TK]D-Fender: hm. |
| 17:09.47 | ManxPower | Softphones are what give VoIP a bad reputation |
| 17:10.49 | Qwell | and Vonage |
| 17:10.55 | coppice | yeah, like it couldn't get a bad name without their help :-) |
| 17:11.14 | tzanger | :-) |
| 17:11.23 | tzanger | coppice: do you do much fpga work? |
| 17:11.48 | coppice | not these days |
| 17:12.46 | tzanger | coppice: what kinds of stuff did you do 'back in the day' |
| 17:12.59 | coppice | DSP mostly |
| 17:13.20 | coppice | and some TDM manipulation |
| 17:13.33 | tzanger | hmm |
| 17:15.08 | tzanger | I'm looking at putting twenty-one independent tdm controllers (common clock though) on an fpga and implementing a basic crossbar switch (tdm controller x channel y tx/rx connected to tdm controller a channel b) type of thing |
| 17:15.28 | tzanger | not really fancy, just lots of bits flying around |
| 17:16.38 | coppice | that sort of thing should work out really well with these cheap modern FPGAs |
| 17:17.29 | tzanger | yeah, I'm trying to figure out a nice way of getting it on a PCI card, but I think I might be out of room (physical) to get twenty of those tdm controller's signals "out" of the card |
| 17:17.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:18.01 | coppice | well, I hope you mean PCI-E, or its obsolete before it starts :-) |
| 17:18.16 | tzanger | coppice: yes |
| 17:18.28 | tzanger | PCIe has more bandwidth than I need but totally agreed |
| 17:19.11 | MrNaz | tzanger how do you plan on getting that many line sockets on the outside of the PC ? |
| 17:19.18 | tzanger | MrNaz: that's the point |
| 17:19.35 | MrNaz | 21 controllers can handle 84 lines, right? |
| 17:19.36 | tzanger | I am thinking a simple PC interface with one TDM port set going to an external board that has the other 20 |
| 17:19.48 | tzanger | MrNaz: ? I'm looking at 128 timeslots per TDM port |
| 17:20.11 | MrNaz | why dont you just use digital circuits? |
| 17:20.33 | MrNaz | or is this just a hobby |
| 17:21.01 | tzanger | MrNaz: I have to interface to legacy systems, and that requires timeslot manipulation anyway, so I'll let the FPGA handle most of that and map in 160 timeslots to the PC |
| 17:21.06 | tzanger | hard to describe |
| 17:21.27 | coppice | why 21? its an odd number |
| 17:21.45 | tzanger | coppice: 20 shelves on the legacy PBX, 1 to the PC |
| 17:21.50 | [TK]D-Fender | coppice: You and your sill base-10! :p |
| 17:21.53 | [TK]D-Fender | <PROTECTED> |
| 17:22.15 | artz22 | [TK]D-Fender: could latency increase when talking from a sip phone to a fxo line (mine in this case, might be my fault ) |
| 17:22.33 | artz22 | I mean, is it normal, or should it be the same with a fxo line? |
| 17:22.46 | [TK]D-Fender | artz22: I can't comment on your device. try Echo in the dialplan and testing with something else. |
| 17:22.51 | MrNaz | [TK]D-Fender 21 makes sense if you lost 3 fingers in the war :P |
| 17:22.55 | tzanger | haha |
| 17:23.03 | artz22 | [TK]D-Fender: thanks, that's useful |
| 17:24.06 | coppice | tzanger: so you want to build a TST switch to cross connect any of 21x128 circuits? |
| 17:26.20 | tzanger | MrNaz: each shelf has 32 timeslots, 20x32 = 640 voice channels total |
| 17:26.53 | coppice | you said 128 slots per port |
| 17:26.55 | tzanger | coppice: eventually, yes. At this point though it'll just cross connect any of the 20 to any in the 1 |
| 17:27.39 | tzanger | coppice: yes, there are 128 timeslots per shelf, but 32 of the 128 are payload, 16 are sync/command and the rest are shelf card configuration |
| 17:27.51 | tzanger | onhook/offhook and mapping state |
| 17:28.19 | artz22 | [TK]D-Fender: echo does really fine with my channel, thanks! it might be the ulaw thing, don't know |
| 17:28.25 | tzanger | oh and the best part, coppice... only the payload timeslots are interleaved |
| 17:28.28 | coppice | strange setup. most PBX backplanes look like an E1. event the american ones |
| 17:28.47 | tzanger | but they're not interleaved in timeslots, they're interleaved 7 bits in the timeslot, and 1 bit in the next timeslot |
| 17:28.51 | tzanger | 1970's chassis technology :-) |
| 17:28.52 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 17:29.19 | tzanger | coppice: yes; basically it is E1-like, but the shelf controller takes the remaining timeslot data and derives card commands from it... it's a little weird |
| 17:29.31 | coppice | most backplanes date from the 70s. its when everything went PCM |
| 17:29.31 | tzanger | I swear the guys who figure this out use alien technology |
| 17:29.37 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 17:29.41 | chandoo | hi |
| 17:29.42 | tzanger | yep, this is a nortel SL1 backplane |
| 17:29.52 | tzanger | nortel's first TDM backplane, IIRC |
| 17:29.58 | chandoo | need help in configuring asterisk |
| 17:30.05 | Qwell | ~help |
| 17:30.09 | chandoo | i installed asterisk and asterisk-gui |
| 17:30.21 | coppice | I thought the SL1 was like a pure E1. the DMS switches are. I did line card work for those |
| 17:30.21 | tzanger | I've spent hte last while getting the channels into tdmoe, but I'm having issues with tdmoe stablity |
| 17:30.24 | russellb | Qwell: ~ask |
| 17:30.29 | Qwell | yeah, wrong bot |
| 17:30.45 | tzanger | it's not the network cards, I think it's the zaptel ztd-eth code or ztdynamic code |
| 17:31.00 | tzanger | coppice: they use "loops" I think to communicate between the buffer card and the main CPU |
| 17:31.09 | tzanger | before superloop and such |
| 17:33.24 | *** join/#asterisk hohum (n=dcorbe@apollo.wavelen.net) |
| 17:33.28 | unpaidbill | is there a recommended version of imap for use with asterisk 1.4 ? |
| 17:33.34 | unpaidbill | for imap voicemail |
| 17:34.12 | unpaidbill | oh nm i found the doc |
| 17:34.21 | chandoo | http://localhost:8088/asterisk/static/config/cfgadvanced.html |
| 17:34.26 | chandoo | this link is not working |
| 17:35.01 | chandoo | i noticed asterisk has its own httpd server |
| 17:35.08 | chandoo | how to restart that server |
| 17:35.24 | russellb | chandoo: run "make checkconfig" from the asterisk-gui svn checkout |
| 17:36.13 | chandoo | got Good Luck at the end |
| 17:36.16 | chandoo | russellb: |
| 17:37.02 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:37.05 | russellb | i don't know if cfgadvanced exists anymore ... |
| 17:37.15 | russellb | check /var/lib/asterisk/static-http |
| 17:37.18 | russellb | i think that's the dir... |
| 17:37.22 | grandpapadot | Hi all. Is there a reliable way to check for the existence of a mailbox in extensions.conf? |
| 17:37.27 | jaytee | chandoo, on Red Hat and CentOS distros type service httpd restart, not sure for debian distros |
| 17:37.37 | Qwell | jaytee: umm..no |
| 17:37.43 | grandpapadot | debian: /etc/init.d apache2 restart |
| 17:37.43 | chandoo | jaytee: i use fedora9 |
| 17:37.44 | russellb | grandpapadot: *> core show application MailboxExists |
| 17:38.03 | grandpapadot | Thanks, russelb. |
| 17:38.07 | chandoo | jaytee: asterisk has its own httpd server i belive |
| 17:38.11 | lmadsen | 'service httpd restart' is for apache |
| 17:38.18 | lmadsen | not the asterisk http server |
| 17:38.25 | jaytee | ok, my mistake |
| 17:38.40 | *** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:38.55 | russellb | chandoo: cfgbasic.html |
| 17:39.53 | russellb | or maybe it's home.html |
| 17:39.55 | russellb | i have no idea. |
| 17:40.13 | chandoo | russellb: i restarted http server |
| 17:40.22 | russellb | ok. |
| 17:40.31 | chandoo | but i didnt make any changes in httpd.conf of httpd server |
| 17:40.48 | chandoo | all entries i did are in /etc/asterisk/httpd.conf |
| 17:40.49 | Qwell | where did make checkconfig say to go? |
| 17:41.06 | chandoo | the path i mentioned |
| 17:41.43 | Qwell | define "not working" |
| 17:43.00 | chandoo | apache is working on my localhost |
| 17:43.09 | chandoo | i can see fedora Test page |
| 17:43.27 | russellb | apache != asterisk http server |
| 17:43.29 | chandoo | but i cant see Asterisk link which it gave me during make checkconfig |
| 17:43.51 | Qwell | what does it say? |
| 17:43.53 | chandoo | russellb: that is what i am asking, how to start http server on asterisk |
| 17:44.13 | chandoo | The requested URL was not found on this server. |
| 17:44.13 | chandoo | Asterisk Server |
| 17:44.20 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:44.20 | russellb | that means it _is_ working |
| 17:44.23 | russellb | and that you have the wrong URL |
| 17:44.23 | Qwell | so then you aren't going to the right URL |
| 17:44.31 | russellb | try cfgbasic.html instead of cfgadvanced |
| 17:44.57 | chandoo | tried both basic and advanced, both same error |
| 17:45.10 | Qwell | what is the *exact* URL that make checkconfig gives you? |
| 17:45.24 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 17:45.24 | chandoo | http://localhost.localdomain:8088/asterisk/static/config/cfgbasic.html |
| 17:46.13 | chandoo | [root@localhost ~]# netstat -na|grep 8088 |
| 17:46.13 | chandoo | tcp 0 0 127.0.0.1:8088 0.0.0.0:* LISTEN |
| 17:47.17 | Qwell | remove the asterisk |
| 17:48.13 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 17:49.35 | chandoo | Qwell: same error |
| 17:50.27 | russellb | did you ... install the GUI |
| 17:50.36 | chandoo | yes |
| 17:50.39 | Qwell | how? |
| 17:50.40 | russellb | ls /var/lib/asterisk/static-http |
| 17:51.02 | chandoo | russellb: got SVN trunk and compiled |
| 17:51.15 | Qwell | how did you install it? |
| 17:51.34 | chandoo | [root@localhost ~]# ls /var/lib/asterisk/static-http |
| 17:51.34 | chandoo | ajamdemo.html astman.css astman.js config docs index.html prototype.js |
| 17:51.48 | chandoo | ./configure && make && make install |
| 17:51.56 | chandoo | make samples |
| 17:52.04 | chandoo | make checkconfig |
| 17:52.46 | russellb | pastebin "http show status" from the asterisk CLI |
| 17:54.16 | chandoo | http://pastebin.com/me4408eb |
| 17:54.29 | *** join/#asterisk kannan (n=kannan@123.201.60.114) |
| 17:54.33 | kannan | hello all |
| 17:54.55 | russellb | chandoo: did you try what Qwell said earlier by removing "asterisk" from the URL? |
| 17:55.05 | chandoo | yes |
| 17:55.06 | russellb | http://localhost:8088/static/config/cfgbasic.html |
| 17:55.37 | chandoo | oops i got it now |
| 17:55.45 | chandoo | i was tirying for advanced |
| 17:55.49 | Qwell | so, by yes, you mean no |
| 17:55.49 | russellb | heh |
| 17:55.52 | russellb | :-p |
| 17:56.23 | chandoo | thanks for tip Qwell |
| 17:56.29 | Qwell | ~tips |
| 17:56.30 | jbot | from memory, tips is (Trillion Instructions Per Second) This is a rating of a REALLY FAST computer. 1 TIPS is 1,000,000,000 instructions per seccond |
| 17:56.34 | Qwell | damn |
| 17:56.56 | russellb | i was looking forward to something witty |
| 17:57.40 | russellb | ~thwack Qwell |
| 17:57.41 | jbot | ACTION bludgeons Qwell on the eblow with a Cisco Manual |
| 17:57.59 | keith4 | what's an eblow? |
| 17:58.04 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 17:58.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:58.04 | Qwell | looks out his window |
| 17:58.06 | Qwell | I want a vehicle that tears up lawns... |
| 17:58.14 | russellb | heh |
| 17:58.23 | Qwell | I wouldn't ever have to mow again |
| 17:58.43 | Qwell | speaking of.. anybody want to come mow my lawn? |
| 17:59.15 | russellb | for $100 |
| 17:59.28 | Qwell | You clearly haven't seen my yard. |
| 18:02.07 | *** join/#asterisk frieze (n=frieze@pool-71-251-13-242.nycmny.fios.verizon.net) |
| 18:02.26 | chandoo | guyz what next |
| 18:02.35 | chandoo | i am brand new here :-) |
| 18:02.46 | chandoo | with asterisk |
| 18:03.55 | frieze | okay, so the patch on chan-mobile.org does not appear to work with 1.4.20.1. Anyone know if it works with 1.4.21? Or do I need an old version of asterisk-addons? |
| 18:04.01 | Deeewayne | ~nowhat |
| 18:04.37 | hwt | hm, i am still having problems with REFERs: get 604 Delined (no dialog) back. |
| 18:04.46 | hwt | and i am struggeling to find docs. |
| 18:06.13 | marc7 | just confirming that it isn't possible to add more than one e-mail address in the user_email_address field of voicemail.conf (needing to send to three addresses in total, so I can't just use the e-mail and pager fields. |
| 18:07.04 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 18:07.45 | frieze | anyone have experience using chan_mobile? I can't seem to build it for 1.4 |
| 18:08.14 | jets | marc7: i always create departmental mailing groups for this need aka sales@pvtinternet.com |
| 18:08.24 | jets | which is a group of all 3 sales people, etc. |
| 18:08.33 | AlexTO | There is someone familiar sending using * to send SMS? |
| 18:09.05 | [TK]D-Fender | Deeewayne: ... |
| 18:09.08 | [TK]D-Fender | ~nowwhat |
| 18:09.08 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 18:09.38 | marc7 | jets: yep, that and invoking Voicemail(u101&102&103) are my two options.... the problem is that we're hosting asterisk for someone whose e-mail we don't manage, so I suppose I could ask them to create their own e-mail address. |
| 18:11.50 | jets | *nod* ya i know what you mean, although if you have a local MTA -- postfix, etc you could make a localhost alias |
| 18:11.51 | hwt | should it be possible to do REFERs outside dialogs? |
| 18:11.55 | jets | group@localhost and put them in your aliases |
| 18:12.13 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 18:12.16 | Deeewayne | [TK]D-Fender: thanks! |
| 18:14.17 | iratik | I'm reading the asterisk bible ... and I can't figure out where it talks about how to carry out a command after a dial command ... I need the dialplan to execute a command after the one who is dialing and is making the call hangs up. Its getting to priority 4 ... and apparently doesn't execute the System(curl command... but if i switch priorities 4,5.. then the curl command executes... but the file doesn't exist yet so thats pointless) |
| 18:14.17 | iratik | <PROTECTED> |
| 18:14.36 | kannan | could anyone provide recommendations for wireless SIP phones suited for use with Asterisk? |
| 18:14.40 | iratik | oh yeah: http://www.pastie.org/217426 |
| 18:14.46 | frieze | anyone have chan_mobile running? |
| 18:17.13 | iratik | basically ... how do i make something happen after dial? |
| 18:17.46 | [TK]D-Fender | iratik: "core show application dial" |
| 18:18.04 | [TK]D-Fender | iratik: and read up on your "Asterisk Standard Extensions" on the WIKI |
| 18:18.55 | [TK]D-Fender | kannan: Most suck outright. Hitachi's seem better than most, but YMMV. If possible you're better off with a cordless phone + ATA, etc |
| 18:19.28 | kannan | [TK]D-Fender, thanks, (whats YMMV?) |
| 18:19.44 | jjshoe | kannan the aastra dect phone is pretty nice, haven't tried snom's dect, I try to avoid s nom, |
| 18:19.47 | seanbright | your mileage may vary |
| 18:19.50 | _ShrikE | kannan: I have been using the Snom M3. It's DECT and not that bad. |
| 18:19.54 | kannan | jjshoe , thanks |
| 18:20.04 | seanbright | kannan: YMMV = your mileage may vary |
| 18:20.10 | [TK]D-Fender | kannan: Other reviews just recently reported the M3 as crap. |
| 18:20.45 | [TK]D-Fender | kannan: Do you NEED the phone handset itself to be WiFi? |
| 18:21.46 | kannan | [TK]D-Fender, yes thats what i was thinking, there is a company that wants to be totally wireless, they are dong an Aruba networks set up. They say the polycom models as too pricey, so i was looking at alternatives |
| 18:22.13 | [TK]D-Fender | kannan: ... yuck. This is not going to be easy / good.. |
| 18:22.18 | kannan | they'll probably accept a audiocdes fxs _ analog phones tho |
| 18:22.25 | fogo | kannan: we've been using linksys WIP330s and they're not bad - sound quality is excellent |
| 18:22.45 | kannan | fogo, thats exactly what i been considering actually |
| 18:23.13 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 18:23.39 | kannan | [TK]D-Fender, thanks, i'll surely pas on your advice |
| 18:23.40 | fogo | kannan: they're easy to setup, and have auto-provisioning, but I haven't used it since we only have a handful of them |
| 18:23.55 | [TK]D-Fender | kannan: Last serious warning : AVOID LINKSYS WIFI <---- |
| 18:24.09 | tzanger | [TK]D-Fender: why? |
| 18:24.11 | chandoo | i have this error when i run ekiga |
| 18:24.14 | chandoo | You will not be able to receive incoming SIP calls. Please check that no other program is already running on the port used by Ekiga. |
| 18:24.16 | tzanger | my linksys wrt54 is working ifne |
| 18:24.25 | [TK]D-Fender | kannan: Above and beyond all else. The WIP's = GARBAGE. Thats personal experience and those of people whom I've warned and didn't listen. |
| 18:24.26 | tzanger | oh you mean wifi PHONES |
| 18:24.36 | tzanger | no way DECT or bust |
| 18:25.00 | kannan | [TK]D-Fender, thanksagain, lol , but i 'll take your advice, you always sorted out problems accurately so fay as I have seen |
| 18:25.53 | kannan | so its either a TDM card or audiocodes + analog phones OR decent IP phones |
| 18:26.33 | kannan | i myself was wondering how the totally wireless setup will funstion, i may be sitting there forever sorting out things i fear |
| 18:27.03 | hwt | bah. "REFER can be sent outside or inside of a dialog. Asterisk only accepts REFER inside of a dialog." |
| 18:27.15 | hwt | is this valid for 2.6 too? |
| 18:27.23 | hwt | uhm. 1.6 |
| 18:27.53 | [TK]D-Fender | kannan: this sums it up : |
| 18:27.57 | [TK]D-Fender | ~wifivoip |
| 18:27.58 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 18:28.35 | [TK]D-Fender | kannan: the idea isn't so bad, only the current product options VS their price. |
| 18:28.52 | [TK]D-Fender | kannan: "acceptable" quality does not align with "acceptable price" |
| 18:29.33 | kannan | [TK]D-Fender, yes . the best things come with a price tag |
| 18:29.49 | kannan | what i said, if yoyu can buy Aruba , you can buy polycom |
| 18:30.04 | [TK]D-Fender | kannan: Sometimes good things come in very competitive (Poloycom's lineup in North America for example) |
| 18:30.56 | kannan | [TK]D-Fender, hmm, perhaps becoz of all the real cheap Chinese makes, polycom look more expensive |
| 18:32.47 | *** join/#asterisk francogwapito (n=chatzill@203.82.47.128) |
| 18:33.20 | [TK]D-Fender | kannan: Compare to the Hitachi's for sense of range. |
| 18:33.26 | francogwapito | is there any way or any application i can use to record video call? |
| 18:37.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:40.34 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 18:40.34 | _MrSeb_ | Hi to all |
| 18:40.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:42.20 | *** join/#asterisk neverblue (n=profx@unaffiliated/neverblue) |
| 18:42.39 | _MrSeb_ | someone is able to explain to me because after 20 seconds the call hangup? |
| 18:42.47 | neverblue | can someone recommend a decent IP phone ( under $200 ) ? |
| 18:43.21 | hwt | anyone with knowledge about asterisk + REFER here+ |
| 18:43.21 | hwt | ? |
| 18:43.53 | UnixDog | Polycom 320/3320 430/440 |
| 18:44.14 | UnixDog | you can find polycom 500/501 on ebay for around 120 |
| 18:45.19 | [TK]D-Fender | neverblue: Where are you located? |
| 18:46.17 | neverblue | Canada |
| 18:46.51 | [TK]D-Fender | neverblue: www.telephonydepot.com <- Polycom IP 320/330 is very inexpensive. |
| 18:47.02 | [TK]D-Fender | neverblue: and shipping up here is fast |
| 18:47.47 | neverblue | cool, taking a look |
| 18:49.03 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 18:50.59 | UnixDog | they have a 501 for 144 |
| 18:51.06 | UnixDog | thast a good starter phone |
| 18:51.22 | frieze | anyone have chan_mobile running on 1.4.21? |
| 18:51.50 | Qwell | 501 is a dead end |
| 18:52.08 | UnixDog | why |
| 18:52.16 | UnixDog | the 501 work great |
| 18:52.16 | [TK]D-Fender | IP 501 is a tricky phone to validate the purchase of. |
| 18:52.17 | Qwell | because there are better phones for cheaper |
| 18:52.28 | [TK]D-Fender | Qwell: Debateably "better". |
| 18:52.36 | Qwell | reasonably similar |
| 18:53.03 | [TK]D-Fender | Qwell: IP 501 has a bigger screen (more readable), good feel, better speakerphone, etc than its lower competitors, and 1 more line-key. |
| 18:53.22 | [TK]D-Fender | Qwell: Depending on usage yes a lower model can be more "cost effective", but not necessarily "better" |
| 18:53.53 | [TK]D-Fender | I have a 501 I might be willing to part with at a discount. |
| 18:55.15 | UnixDog | if your looking for a starter then the 320/330 if you looking for a stable office phone well tested then the 501 |
| 18:57.36 | jets | I love the 501's, we usually opt for the 601 to have a browser for the services button |
| 18:57.58 | [TK]D-Fender | jets: 501 has the MicroBrowser... |
| 18:58.37 | UnixDog | the 501 has the micro browser button |
| 18:59.54 | UnixDog | called services/applications |
| 19:01.16 | jets | oh then a IT gent lied to me he claims the 501 didn't have a microbrowser even though it has the services button |
| 19:01.25 | jets | (i'm not surprised) |
| 19:02.21 | [TK]D-Fender | jets: Not necessairy "lied" so much as not having known that since SIP 2.1.0 the 501 & 430 gained the MicroBrowser. |
| 19:02.37 | [TK]D-Fender | jets: Its a question about knowing your firmware featuresets. |
| 19:02.50 | jets | Awwww |
| 19:02.51 | [TK]D-Fender | jets: 2.2.0 added tables support, etc... |
| 19:02.52 | jets | mkay |
| 19:03.04 | UnixDog | and knowing how too write the php pages needed |
| 19:03.09 | UnixDog | for diff apps |
| 19:04.15 | [TK]D-Fender | UnixDog: If you want that kind of "dynamic", sure. |
| 19:05.47 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 19:06.24 | *** join/#asterisk implicit (n=implicit@ip68-4-97-211.oc.oc.cox.net) |
| 19:06.41 | UnixDog | what else are you going to use it for ? |
| 19:06.47 | UnixDog | what other apps |
| 19:06.52 | _MrSeb_ | someone is able to explain to me because after 20 seconds the call hangup (asterisk server and asterisk clients are behind a nat)? |
| 19:07.30 | [TK]D-Fender | _MrSeb_: READ : |
| 19:07.32 | [TK]D-Fender | ~sipnat |
| 19:07.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:07.39 | UnixDog | you migh thave your dial timeout set to 20 sec |
| 19:09.50 | [TK]D-Fender | UnixDog: I have a script that generates static pages for my phones here at work to retrieve for instance. |
| 19:10.02 | _MrSeb_ | UnixDog: before I put a router all was working, I've changed the configuration abuot nat and now the call hangup... |
| 19:12.28 | [TK]D-Fender | _MrSeb_: pastebin your SIP.conf masking only passwords, and SIP debug of a cal that fails from beginning to end, with SIP DEBUG enabled. |
| 19:12.30 | [TK]D-Fender | ~pb |
| 19:12.30 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:12.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:14.26 | UnixDog | TK nice |
| 19:18.05 | _MrSeb_ | [TK]D-Fender: http://rafb.net/p/C2N94078.html |
| 19:20.13 | hsv-al | heh |
| 19:20.16 | hsv-al | iphone2 is already hacked |
| 19:20.21 | hsv-al | w/ a working sip client |
| 19:20.30 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:21.21 | jaytee | life seems to be full of tough choices like, to write documentation or drive my car a 100mph into a bridge abuttment. Right now the car choice is looking better and better. |
| 19:24.38 | [TK]D-Fender | _MrSeb_: [Eutelia] and [Messagenet] should be "nat=no". and you di not put "qualify=yes" for your remote phones behind NAT. |
| 19:24.59 | [TK]D-Fender | _MrSeb_: Fix those. Then confirm what you have forwarded from your router to *. |
| 19:26.03 | Kobaz | jaytee: take pictures |
| 19:26.19 | _MrSeb_ | [TK]D-Fender: I've forwarded only 5060 udp |
| 19:26.31 | jaytee | Kobaz, I'll have to have someone else do it if I'm driving :-) |
| 19:26.54 | Kobaz | could always rig up something |
| 19:26.57 | [TK]D-Fender | _MrSeb_: then you have not followed the guide at all. You should have 5060, and 10000-20000 all UDP forwarded to *. |
| 19:27.02 | jaytee | but I'll leave instructions for them to upload to photobucket or someplace and then post the link in here for all to enjoy. |
| 19:28.29 | _MrSeb_ | [TK]D-Fender: I've not done this because audio is ok |
| 19:29.11 | [TK]D-Fender | _MrSeb_: .... follow the instructions. These are not typically "optional". |
| 19:30.33 | _MrSeb_ | [TK]D-Fender: thanks, I do this during the night, now I can't modify router settings... I've tried without forwarding and the call has hangup after 11 seconds |
| 19:31.33 | [TK]D-Fender | _MrSeb_: Stop with the useless description. If doesn't tell us ANYTHING of value, and you did not show me your failed call at verbose 10 or apparently follow the guide of make any of the changes I've just told you to. ALL of these things can be done while leaving your server up. |
| 19:32.25 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-142-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:33.01 | *** join/#asterisk kowalma (n=kowalma@5357043E.cable.casema.nl) |
| 19:33.48 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 19:33.58 | _MrSeb_ | [TK]D-Fender: I've not the router password and I must wait my friend |
| 19:34.59 | _MrSeb_ | [TK]D-Fender: thanks anyway and good evening |
| 19:39.08 | kowalma | Hi All, |
| 19:39.30 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 19:40.54 | kowalma | gotta question - anybody knows how to list active zap channels (with active calls)? zap show channels show only all channels but without information if it's busy |
| 19:42.12 | [TK]D-Fender | kowalma: "core show channels concise |
| 19:43.07 | kowalma | nope |
| 19:43.15 | neverblue | so after all that, the 501 good, or no good ? |
| 19:44.00 | kowalma | I need information that timeslot no 102 is busy by number XXX |
| 19:44.32 | [TK]D-Fender | neverblue: All the Polycom's are "good", just depends on your needs, PoE, price you can find it for, etc. |
| 19:44.39 | [TK]D-Fender | neverblue: the 501 is a fine phone. |
| 19:44.44 | neverblue | ah |
| 19:44.51 | neverblue | they a US company ? |
| 19:45.01 | Strom | pleasanton, california |
| 19:45.02 | [TK]D-Fender | neverblue: primarily, yes |
| 19:45.19 | neverblue | ah, US companies never put USD after prices |
| 19:45.32 | neverblue | its just expected that others do it :D |
| 19:45.53 | [TK]D-Fender | neverblue: they have a worldwide presence |
| 19:46.10 | neverblue | i bet they dont have a .ca :) |
| 19:46.12 | [TK]D-Fender | never Don't forget they are huge in the videoconferencing world. |
| 19:46.44 | neverblue | these Grandstreams just arent cutting it any more |
| 19:47.30 | [TK]D-Fender | ~gs |
| 19:47.31 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:47.34 | [TK]D-Fender | ^ says it all... |
| 19:47.46 | neverblue | hehe |
| 19:47.54 | neverblue | ill have to second that motion |
| 19:48.09 | Corydon76-dig | I have to disagree with that |
| 19:48.15 | neverblue | really? |
| 19:48.24 | neverblue | u a sales rep with them ? |
| 19:48.47 | Corydon76-dig | Cisco, on the other hand, makes extremely expensive junk, which should be avoided with extreme prejudice |
| 19:48.54 | Corydon76-dig | No, I'm not in sales |
| 19:49.04 | Corydon76-dig | nor do I work for GS at all |
| 19:49.23 | Corydon76-dig | but I think their phones get a bad rap |
| 19:49.25 | neverblue | by bashing another phone manu. doesnt justify ur disagreement :D |
| 19:49.53 | Corydon76-dig | neverblue: the price of Cisco phones is such that they should be able to fix them |
| 19:50.35 | eric2 | is a fan of the snom lineup of phones |
| 19:50.50 | Corydon76-dig | I like the SNOM series, too, and the Polycom |
| 19:51.12 | Corydon76-dig | But I'd put the Cisco dead last, well after Grandstream phones, in terms of preference |
| 19:51.24 | eric2 | cisco is over rated! |
| 19:51.36 | neverblue | the polycom it is then |
| 19:51.48 | kowalma | I my company we had Cisco BTS platform, it had outage everyweek :D |
| 19:52.12 | [TK]D-Fender | neverblue: Very few have had viable complaints about Polycom. Can't be said for jst about any other manufacturer. |
| 19:52.31 | Corydon76-dig | The only reason people think they want the Cisco phones is that Cisco has managed to get product placement on TV shows like 24 and The Office |
| 19:53.25 | Corydon76-dig | and once you think powerful and exciting government agents use those phones... well, the marketing was obviously well thought-out |
| 19:53.40 | neverblue | yeah, the 4 line Polycom seem to jump up in price |
| 19:53.58 | neverblue | but i think its probably better to just get a good phone, and pay the cost now |
| 19:54.19 | eric2 | quality, price or service .... pick two! |
| 19:56.17 | neverblue | im getting about 50 |
| 19:56.18 | neverblue | :D |
| 19:57.00 | *** join/#asterisk mikehime (i=mikehime@207-5.97-97.tampabay.res.rr.com) |
| 19:57.11 | mikehime | waves |
| 19:57.48 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:58.00 | NovceGuru | The cisco phones are a pita |
| 19:58.04 | NovceGuru | with the sip firmware |
| 19:58.06 | NovceGuru | my $.02 |
| 19:59.22 | unpaidbill | haha yep. |
| 19:59.29 | [TK]D-Fender | neverblue: looking to equip a whole company? |
| 19:59.31 | unpaidbill | i have a bunch of them and they're the bane of my existence |
| 19:59.56 | unpaidbill | the a-holes changed the UI on the damn things between sip and sccp firmware, and then dont allow you to arrange the soft buttons as you please |
| 20:02.59 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 20:04.30 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:04.38 | unpaidbill | they let you change the logo and ringtone though! yay |
| 20:05.52 | mikehime | speaking of hardphones, I'm about to buy a couple |
| 20:05.59 | mikehime | any recommendations for under $100 ? |
| 20:06.07 | unpaidbill | i hear snom are OK |
| 20:06.13 | unpaidbill | never used any myself though |
| 20:06.47 | mikehime | cool i'll check them out, too |
| 20:06.54 | unpaidbill | yeah they're right around 100 bucks i think |
| 20:07.00 | unpaidbill | you can probably finda deal |
| 20:07.28 | unpaidbill | polycom have some that are pretty close too, you may be able to find some on ebay used for sub 100 |
| 20:07.44 | mikehime | I have had polycom recommended to me before so they're already on the list |
| 20:08.02 | mikehime | we (a friend and i) figured we'd just buy several from different manufacturers and see which one's we prefer |
| 20:08.17 | mikehime | suggestions are always nice, though =) appreaciate it |
| 20:08.20 | unpaidbill | sounds like a good idea |
| 20:11.01 | jjshoe | under a hundred? 9112 perhaps by aastra |
| 20:13.23 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 20:13.29 | [TK]D-Fender | mikehime: If you're in North America, forget everything except Polycom. |
| 20:14.42 | *** join/#asterisk pelaofeliz (n=PelaoFel@67.108.236.230) |
| 20:15.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:17.00 | alrs | mikehime: " fender |
| 20:19.11 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 20:21.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:21.16 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-75bef1505f0dd418) |
| 20:21.55 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 20:22.14 | pots_line | devstate working for anyone |
| 20:22.41 | mikehime | thanks for advice [TK]D-Fender & alrs |
| 20:23.09 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 20:24.32 | pots_line | any idea why (${DEVSTATE(Custom:9999)}) always kicks back UNKNOWN |
| 20:25.01 | [TK]D-Fender | bbiab |
| 20:27.08 | pots_line | anyone know how to get originating BLF working? |
| 20:28.32 | *** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 20:30.05 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 20:30.43 | pots_line | nevermind figured it out |
| 20:33.05 | *** join/#asterisk kowalma (n=kowalma@5357043E.cable.casema.nl) |
| 20:33.34 | *** join/#asterisk docelm0 (n=chatzill@206.248.239.194) |
| 20:33.44 | docelm0 | Can anyone recommend a good ATA that supports t.38? |
| 20:37.22 | mvanbaak | grandstream |
| 20:37.24 | mvanbaak | hides |
| 20:38.08 | Nugget | heh |
| 20:38.47 | unpaidbill | most of them do from what i've seen |
| 20:38.56 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 20:39.00 | drfreeze | Hello |
| 20:39.08 | unpaidbill | and grandstreams is actually pretty cheap and it works from what i hear, as far as the ata is concerned |
| 20:39.23 | *** join/#asterisk cnielsen (n=Cole@71-223-115-148.phnx.qwest.net) |
| 20:39.24 | unpaidbill | there was one in particular i think that was like 30-60 bucks with t.38 support |
| 20:40.48 | unpaidbill | hmm, handytone 286.. but it is showing as t.38 pending |
| 20:41.23 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 20:41.43 | unpaidbill | although the manual for it says it supports t.38, so i'd say spend 34 bucks and hope it works. http://www.grandstream.com/user_manuals/HandyTone.pdf |
| 20:41.43 | cnielsen | Hey all - I'm putting together some specs for some hardware to run Asterisk on - I haven't found much by way of documentation. Can anyone point me in the right direction? |
| 20:42.25 | docelm0 | mvanbaak: next suggestion.. |
| 20:45.59 | mikehime | cnielsen: have you read the asterisk book yet? |
| 20:46.11 | mikehime | already becoming dated but there's good advice there |
| 20:46.43 | cnielsen | LOL a while ago - I suppose there's a chapter in there on hardware that I skipped over. |
| 20:46.53 | mikehime | yes, chapter 2 ;) |
| 20:47.02 | cnielsen | Thanks, thanks :) |
| 20:47.07 | mikehime | np |
| 20:47.16 | ManxPower | cnielsen: There are SO many variables that significantly impact performance, there is no easy to size your server. |
| 20:47.36 | mikehime | agree w/ manx, really depends on how you want to use it |
| 20:47.37 | ManxPower | I think there is a page on voip-info.org with general info and guidelines. |
| 20:48.01 | cnielsen | ManxPower: Definitely agree -- the more info I can find the more empowered I am to make good decisions. |
| 20:48.16 | cnielsen | ManxPower: I'll check there and see what I find. Thanks :) |
| 20:49.55 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:50.34 | cnielsen | mikehime: 25-30 users, in/out calling with voicemail |
| 20:50.47 | cnielsen | Probably all SIP but haven't got that far yet. |
| 20:51.10 | mikehime | i think more importabtly you need to be aware of the codecs the exts will use |
| 20:51.19 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 20:51.27 | mikehime | g729 will require an order of magnitude more cpu with that many extensions |
| 20:51.30 | mikehime | over gsm |
| 20:52.06 | *** join/#asterisk makkksimal (n=makkksim@e177223037.adsl.alicedsl.de) |
| 20:53.13 | lanning | also, need to know the expected concurrent call level. |
| 20:53.51 | cnielsen | I was just going to comment that Chapter 2 says that it's not a matter of users, it's a matter of concurrent calls. |
| 20:54.43 | mikehime | oh well yeah, i usally do worst case estimates and assume all extensions will be in use |
| 20:54.46 | lanning | ya, a user is just a small database entry. concurrent calls, are the actual activity. |
| 20:55.02 | mikehime | personally i have a 5 year old test machine here that can do 20 concurrent calls |
| 20:55.12 | cnielsen | mikehime: I'm fairly new to PBX, IPv6, etc... Do you have any recommended reading (aside from the asterisk book) that I could pick up? |
| 20:55.13 | mikehime | so if that's your only load you can almost use anything off-the-shelf |
| 20:55.52 | mikehime | honestly i'm rather new myself, and since i'm only doing SIP the asterisk book and voip-info.org have been enough resources to learn my way through it ;) |
| 20:56.10 | cnielsen | there you have it then :) I'll get busy. |
| 20:56.28 | cnielsen | LOL |
| 20:56.35 | mikehime | :D good luck |
| 20:57.12 | cnielsen | Much appreciation for everyone's help. I'm sure I'll pop back in from time to time. |
| 20:57.31 | *** join/#asterisk GleepGlop (n=derek@199.227.66.250) |
| 20:57.31 | lanning | if you have matching codecs and "canreinvite=yes" then the media will go phone to phone and bypass your PBX, taking even more load off. |
| 20:57.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.43 | pelaofeliz | I must be missing something obvious -- how do you tell what channel a given resource is on, via the AMI? e.g. I'm SIP/Pelaofeliz, and I originate a call, but the channel is 'SIP/Pelaofeliz-123abc'. |
| 20:57.50 | *** join/#asterisk peterpan746 (n=wopulnt@dsl-242-47-184.telkomadsl.co.za) |
| 20:58.05 | mvanbaak | cnielsen: the best reference is the source ;) |
| 20:58.06 | mikehime | lanning: good advice. no transcoding = win |
| 20:58.08 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 20:58.15 | [TK]D-Fender | pelaofeliz: "given resource".... like what exactly? |
| 20:59.06 | pelaofeliz | [TK]D-Fender: say I'm SIP/Pelaofeliz, and I originate a call. The channel name always ends up being something like 'SIP/Pelaofeliz-123abc' |
| 20:59.15 | lanning | in most voip protocols, the "channel" is dynamic creation. |
| 20:59.19 | pelaofeliz | right |
| 20:59.26 | *** join/#asterisk greek_user (i=f@adsl104-134.kln.forthnet.gr) |
| 20:59.40 | pelaofeliz | in order to do anything with the call via the AMI, it always requires the channel (obviously) |
| 20:59.46 | [TK]D-Fender | pelaofeliz: Indeed... that is a channel name. SIP/Pelaofeliz is a DEVICE name. You can have several calls against that same account. |
| 20:59.55 | *** part/#asterisk GleepGlop (n=derek@199.227.66.250) |
| 21:00.22 | pelaofeliz | so is there a way to get a list of channels for a device? |
| 21:00.40 | pelaofeliz | or even better, the channel name when the call is originated? |
| 21:00.56 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 21:01.00 | [TK]D-Fender | pelaofeliz: for a device no, but a general list, yes |
| 21:01.04 | kamh | hi all |
| 21:01.08 | [TK]D-Fender | pelaofeliz: parse it out yourself |
| 21:01.12 | pelaofeliz | I'm originating the call via AMI |
| 21:01.21 | ManxPower | pelaofeliz: AMI should tell you the channel name. |
| 21:01.30 | ManxPower | Your issue is not unique nor is it uncommon |
| 21:02.07 | pelaofeliz | ok -- so I just do a SHOW CHANNELS, then parse out the device that I'm looking for? |
| 21:02.42 | ManxPower | pelaofeliz: have you looked at /path/to/src/asterisk/doc/manager.txt ? |
| 21:02.52 | [TK]D-Fender | pelaofeliz: yes |
| 21:03.31 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:03.37 | pelaofeliz | ManxPower: it's been a while |
| 21:03.44 | pelaofeliz | I'll take a look |
| 21:03.47 | pelaofeliz | thanks for the help |
| 21:03.48 | ThoMe | hm |
| 21:03.51 | ThoMe | what is on my line wrong? exten=> _${KOPFNUMMER}751.,1,Playback(tt-monkeys) |
| 21:04.03 | ThoMe | _${KOPFNUMMER} = 123456 |
| 21:04.05 | ManxPower | pelaofeliz: Might be helpful to look at it again. Also, look on voip-info.org, but if manager.txt and voip-info disagree, trust manager.txt |
| 21:04.14 | ManxPower | ThoMe: You cannot do that |
| 21:04.20 | ThoMe | and i would LIKE as result: 123456751 |
| 21:04.25 | ThoMe | ManxPower: hm. |
| 21:04.50 | ThoMe | ManxPower: how is better? |
| 21:04.54 | ManxPower | i.e. Variables are not supported anywhere on the line before the first ( in the application parameters. |
| 21:05.43 | ManxPower | i.e. exten =>12345,1,Dial(VARIABLES WORK STARTING HERE |
| 21:05.43 | ThoMe | exten=> _${KOPFNUMMER}751,1,Playback(tt-monkeys) |
| 21:05.43 | ThoMe | better? |
| 21:05.43 | ManxPower | ThoMe: What did I just say? |
| 21:05.50 | ThoMe | ManxPower: hm. ok. sorry. |
| 21:05.51 | [TK]D-Fender | ThoMe: You can do that if its a gloabl variable defined under [globals] |
| 21:05.59 | ThoMe | [TK]D-Fender: i have it. |
| 21:06.00 | ManxPower | What is the value of ${KOPFNUMMER} |
| 21:06.14 | ThoMe | ManxPower: the master-number from my phone-lines |
| 21:06.22 | ThoMe | [general] |
| 21:06.22 | ThoMe | KOPFNUMMER=666682 |
| 21:06.29 | ManxPower | ThoMe: then put that in, it's not like it changes |
| 21:07.36 | ManxPower | ThoMe: You *MIGHT* be able to do what you want to do using global variables if you force some module to load early in /etc/asterisk/modules.conf, but I don't know what module you would load. I vaguely remember something similar talked about on the mailing list sometime in the past 5 years. |
| 21:07.39 | ThoMe | exten=> 666682751,1,Playback(tt-monkeys) |
| 21:07.40 | ThoMe | <PROTECTED> |
| 21:07.52 | ThoMe | ManxPower: oh. hm. ok. |
| 21:07.53 | pelaofeliz | ManxPower: What did you mean by this: "AMI should tell you the channel name." |
| 21:09.48 | [TK]D-Fender | pelaofeliz: Fine.. so you've started your Originate... what is your issue following that? |
| 21:10.09 | greek_user | [q] what does matchexterniplocally do? (it's in sip.conf) |
| 21:10.39 | ThoMe | ManxPower: hm. this works exten=> _${KOPFNUMMER}0,1,Dial(misdn/g:isdn-int/${EXTEN}) |
| 21:11.30 | lanning | exten => _666682.,1,GoTo(blah,${EXTEN:6}) |
| 21:11.45 | [TK]D-Fender | lanning: Highly unlikely... |
| 21:12.45 | ThoMe | [TK]D-Fender: and this is not unlikeyl? _${KOPFNUMMER}7,1,Playback(tt-monkeys) ? |
| 21:12.48 | ThoMe | ;) |
| 21:12.57 | [TK]D-Fender | ThoMe: wasn't talking to you. |
| 21:13.02 | ThoMe | [TK]D-Fender: sorry. |
| 21:13.09 | [TK]D-Fender | ThoMe: If done the way I advised it should be fine |
| 21:13.20 | ThoMe | [TK]D-Fender: ok. excuse me. |
| 21:14.40 | pelaofeliz | [TK]D-Fender: So I start my originate, the call connects, etc. After that, I want to be able to interact with the channel, e.g. hang it up. |
| 21:15.08 | [TK]D-Fender | pelaofeliz: And what will trigger that decision? |
| 21:15.32 | pelaofeliz | [TK]D-Fender: In this case, the user clicking a button |
| 21:16.21 | [TK]D-Fender | pelaofeliz: When you create the Originate, set a channel variable in it to a unique value that you can track by doing a channel dump. |
| 21:20.32 | pelaofeliz | [TK]D-Fender: K, I've added the channel variable -- how do I do a channel dump that includes that variable? |
| 21:21.07 | [TK]D-Fender | pelaofeliz: Search for the channel containing it and look at the AMI function listing and use some imagination. |
| 21:21.15 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 21:21.19 | talntid | hi all |
| 21:24.11 | pelaofeliz | [TK]D-Fender: That's my problem -- how do I do a search for the channel containing it? I added a 'Variable: unique_id: [id]', but it doesn't show up when I SHOW CHANNELS... |
| 21:24.36 | [TK]D-Fender | pelaofeliz: correct because you'll have to dump EACH ONE in detail to scan for it. |
| 21:26.06 | pelaofeliz | [TK]D-Fender: I see. I'll see what I can work up. |
| 21:26.09 | pelaofeliz | Thanks for the help |
| 21:29.42 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-b0b9eecab8d14564) |
| 21:33.15 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 21:34.56 | pta200 | Has anyone been able to get hinting to work across two asterisk box connected via sip? |
| 21:35.26 | drumkilla | that is not supported (yet) |
| 21:35.41 | pta200 | figured |
| 21:36.01 | drumkilla | same network, or across the internet? |
| 21:36.02 | pta200 | I thought I might be able to a phone to subscribe to hints on another server but that's not working either |
| 21:36.07 | pta200 | across MPLS |
| 21:36.24 | pta200 | so private network |
| 21:36.35 | drumkilla | what kind of latency? |
| 21:37.21 | pta200 | average 13 ms between both servers |
| 21:37.48 | drumkilla | hm, ok, that's not too bad ... then I might have something that will work, but it's pretty bleeding edge as far as support for it in asterisk releases ... |
| 21:38.04 | drumkilla | it's in the main development tree, scheduled for inclusion in 1.6.1 |
| 21:38.13 | pta200 | cool |
| 21:38.21 | drumkilla | http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/ |
| 21:38.39 | drumkilla | do you know if you can you do multicast over that network? |
| 21:38.51 | pta200 | I think so |
| 21:38.55 | pta200 | I would need to check |
| 21:38.57 | Qwell | over MPLS? that'd be scary |
| 21:39.19 | trapa | I have a question about multiple sip lines. We have a bunch of inbound-only sip lines. I have made these work by having multiple register => commands in the [general] section of the sip.conf file ... How would i be able to detect which line it is that is rining? |
| 21:39.21 | pta200 | I think it depends on the provider |
| 21:39.23 | drumkilla | well, the implementation of the framework I tested with uses multicast, which is why i asked |
| 21:39.29 | pta200 | right |
| 21:39.33 | pta200 | I'll check it out |
| 21:39.40 | Qwell | funky, there's an RFC for it |
| 21:40.00 | drumkilla | cool, aside from that, there are some other things people have built to try to do it, synchronizing state using the manager interface |
| 21:40.07 | Qwell | RFC 3353, Overview of IP Multicast in a[n] MPLS Environment |
| 21:40.15 | pta200 | nice |
| 21:40.31 | drumkilla | let me find the link to that other thing ... |
| 21:41.19 | drumkilla | ah yes ... the devstate thing on this page ... http://allan.cassaro.googlepages.com/asterisk |
| 21:41.25 | drumkilla | never tried it, but it's something else I came across |
| 21:41.42 | pta200 | awesome thanks a lot |
| 21:41.50 | drumkilla | you're welcome |
| 21:41.57 | drumkilla | that will cost you $5 and a chocolate chip cookie |
| 21:42.18 | pta200 | I'll send through the cdrom tray |
| 21:42.20 | Qwell | pta200: Send the cookie "c/o Qwell" |
| 21:42.31 | drumkilla | heh |
| 21:42.33 | drumkilla | sillyness |
| 21:42.33 | nick125 | pta200: dcc it! |
| 21:42.49 | drumkilla | i'm afraid what kind of "cookie" i'll get via dcc |
| 21:42.57 | drumkilla | Magic-Cookie.mpg |
| 21:42.58 | pta200 | true |
| 21:42.59 | drumkilla | yeah, i'll pass |
| 21:43.03 | Qwell | ... |
| 21:43.08 | file | drumkilla: don't worry, the firewall is acting as protection - wouldn't let it through anyway |
| 21:43.13 | drumkilla | true |
| 21:49.13 | putnopvut | speaking of magic cookies: z9hG4bK |
| 21:49.22 | putnopvut | Yeah, I just made a SIP joke. What of it? |
| 21:49.43 | file | haha... |
| 21:50.04 | drumkilla | putnopvut: you ... look like a ... nonce ... |
| 21:50.34 | mikehime | i'm having trouble getting unixODBC to compile/install on slack 12 |
| 21:50.34 | mikehime | any advice? |
| 21:50.35 | drumkilla | i suggest a hammer |
| 21:50.37 | mikehime | LOL |
| 21:50.41 | putnopvut | mikehime: sorry, I've only used the debian package. I've never compiled it from source. |
| 21:50.48 | drumkilla | same here |
| 21:50.52 | *** join/#asterisk JHilgeman (n=jh@209.48.241.194) |
| 21:50.54 | mikehime | i'll ask in the slack chan, ty anyway |
| 21:51.01 | JHilgeman | quick question - proabbly easy, too |
| 21:51.13 | drumkilla | quick answer |
| 21:51.38 | JHilgeman | i'm redirecting an extension to a context called custom-screen (call screening) |
| 21:51.54 | JHilgeman | i need that context to dial the extension |
| 21:52.04 | JHilgeman | but {EXTEN} evaluates to "s" |
| 21:52.15 | JHilgeman | and i can't find any sort of variable that contains the destination extension |
| 21:52.18 | drumkilla | you probably need to pastebin your config of what you're trying to do |
| 21:52.28 | drumkilla | ~pb |
| 21:52.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:52.45 | JHilgeman | this is the exten line: |
| 21:52.46 | JHilgeman | exten = 1109,1,Goto(custom-screen|s|1) |
| 21:52.52 | JHilgeman | k |
| 21:52.53 | seanbright | heh |
| 21:53.01 | seanbright | exten = 1109,1,Goto(custom-screen|${EXTEN}|1) |
| 21:53.04 | seanbright | done |
| 21:53.27 | drumkilla | and then in custom-screen, have an exten => _X.,1,NoOp(my catch all thingy) |
| 21:53.41 | drumkilla | or use a Macro that takes arguments ... |
| 21:53.43 | drumkilla | lots of options |
| 21:53.52 | *** join/#asterisk asdx (n=diego@adsl-131-5.click.com.py) |
| 21:54.18 | JHilgeman | k let me try that - just a sec |
| 21:54.22 | JHilgeman | thx |
| 21:54.41 | drumkilla | ~whatnow |
| 21:54.42 | jbot | it has been said that now is a good time to tell you that I have 6 gigabytes of data |
| 21:54.50 | drumkilla | o.O |
| 21:54.52 | seanbright | that's it? |
| 21:54.58 | drumkilla | ~whatnext |
| 21:54.58 | jbot | methinks next is NEXT! |
| 21:55.03 | drumkilla | dangit |
| 21:55.05 | drumkilla | what was it ... |
| 21:55.08 | [TK]D-Fender | ~nowwhat |
| 21:55.09 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 21:56.41 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 21:56.42 | Hadi- | hell |
| 21:56.43 | Hadi- | o |
| 21:56.50 | seanbright | h |
| 21:56.50 | seanbright | i |
| 21:56.58 | drumkilla | oh snap |
| 21:57.13 | [TK]D-Fender | snaps |
| 21:57.17 | Hadi- | is there a document online on setting up asterisk to work with a Radius voip billing software |
| 21:57.41 | mikehime | oh wait looks like it compiled and installed... should be able to tell by the presence of libodbc.so.1 right? maybe i just need to ad it to the path |
| 21:57.49 | Hadi- | we are planning on testing asterisk to work in a |
| 21:57.59 | mikehime | excuse my n00bness |
| 21:58.12 | drumkilla | mikehime: make sure you re-run the configure script after installing it |
| 21:58.42 | mikehime | trying that now.. surprised though, it doesn't suggest to do that in the installation notes |
| 21:59.14 | Hadi- | anyone have any idea? |
| 21:59.43 | seanbright | Hadi-: i would just search google and spew what i found into the channel |
| 21:59.52 | seanbright | Hadi-: so i doubt i can be of any help |
| 22:00.08 | drumkilla | mikehime: i mean re-run the configure script of asterisk |
| 22:00.13 | drumkilla | mikehime: so that asterisk finds it |
| 22:00.16 | seanbright | ~google asterisk radius voip |
| 22:00.21 | seanbright | weak |
| 22:00.59 | trapa | I have a question about multiple sip lines. We have a bunch of inbound-only sip lines. I have made these work by having multiple register => commands in the [general] section of the sip.conf file ... How would i be able to detect which line it is that is rining? |
| 22:01.14 | mikehime | @drumkilla: ah right :) I haven't gotten that far yet. just trying to get the mysql odbc connector working first |
| 22:01.46 | drumkilla | ah. |
| 22:02.00 | drumkilla | mikehime: use debian/ubuntu instead ... fixed! |
| 22:02.01 | drumkilla | :-p |
| 22:02.53 | [TK]D-Fender | mikehime: Don't forget to ldconfig, and you may have to account for a slightly different .so version and symlink, etc. |
| 22:03.02 | chandoo | i have error while using ekiga |
| 22:03.12 | chandoo | unable to open sound devices |
| 22:03.14 | [TK]D-Fender | mikehime: make sure to set up your dsn and test with OOo or something first |
| 22:03.22 | chandoo | ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave |
| 22:03.39 | [TK]D-Fender | chandoo: Go ask in their channel then. |
| 22:03.48 | mikehime | lol i love kubuntu but i'm trying learn as much as i can from a barebones OS |
| 22:03.53 | mikehime | good experience |
| 22:04.05 | drumkilla | ah, yeah |
| 22:04.14 | drumkilla | i got that experience from using gentoo for a while |
| 22:04.19 | drumkilla | that was masochism |
| 22:04.24 | [TK]D-Fender | mikehime: Slackware is solid, but a very different experience. You might be better serverd by RH or Debian based |
| 22:05.32 | chandoo | [TK]D-Fender: looks like no one is alive in ekiga |
| 22:05.34 | mikehime | [TK]D-Fender: seems to be the common consensus ;) |
| 22:05.56 | [TK]D-Fender | chandoo: Go try in ##linux or soemthing then. |
| 22:06.07 | [TK]D-Fender | chandoo: Either way nothing remotely to do with *. |
| 22:07.05 | mwalling | drumkilla: *slap* |
| 22:07.12 | drumkilla | :-p |
| 22:07.23 | drumkilla | i learned a lot! |
| 22:07.24 | mwalling | has asterisk purring along nicely on his slack box |
| 22:07.35 | mwalling | (i'm also slow to read the buffer) |
| 22:07.35 | drumkilla | oh, i thought the slap was for gentoo trolling |
| 22:07.46 | mwalling | no, gentoo doesnt get trolled enough |
| 22:07.47 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 22:07.49 | mwalling | ;) |
| 22:07.54 | mikehime | mwalling: as do i :) but getting everything else setup is the painful part for newbies |
| 22:08.26 | drumkilla | i know nothing about slackware, i can't intelligently troll on it |
| 22:08.32 | mwalling | heh |
| 22:08.34 | drumkilla | but i can ignorantly troll on it |
| 22:08.37 | drumkilla | ... it sucks ... |
| 22:08.45 | mikehime | lol @drumkilla |
| 22:08.53 | unpaidbill | slackware was my first linux dist! you be nice! |
| 22:09.03 | drumkilla | goes back to work ... |
| 22:09.15 | unpaidbill | yea that's what i thought |
| 22:13.10 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:15.58 | mikehime | [TK]D-Fender: you rocked my socks, i had to run ldconfig to update links to shared libs |
| 22:16.06 | mikehime | exactly the kind of practical experience i need |
| 22:18.09 | [TK]D-Fender | Slack was my first distro too, and I only switched to CentOS a year ago +/- |
| 22:19.41 | drumkilla | i'm not sure if that's an upgrade or downgrade |
| 22:19.43 | drumkilla | ducks |
| 22:21.01 | mwalling | down |
| 22:21.28 | mwalling | actually, i'm running ubuntu on my desktop and big laptop... my eee runs slack though |
| 22:22.01 | [TK]D-Fender | Well only recently had Slack gone with a 2.6 kernel stock.... Slack = behind the times. Stable as hell, but just a bit far behind. |
| 22:22.10 | [TK]D-Fender | Sorta like Debian schmucks ;) |
| 22:22.39 | Qwell | My gentoo box already runs 2.8 |
| 22:22.45 | mwalling | 2.6 kernels were in testing/ and extra/ for a while before 12 |
| 22:22.52 | alrs | I switched from Slack to Debian after a nasty rooting, ca. 1998 |
| 22:22.53 | mwalling | Qwell: -OMGOPTOMIZE |
| 22:23.02 | [TK]D-Fender | mwalling: Yes... how "current". More work for a newb. |
| 22:23.03 | Qwell | mwalling: Does it have pam yet though? |
| 22:23.06 | Qwell | That's the real test |
| 22:23.13 | mwalling | no |
| 22:23.17 | mwalling | thusly it passes |
| 22:23.20 | mwalling | ;) |
| 22:23.36 | [TK]D-Fender | "Gentoo : When you positively MUST get that extra 0.05% out of your system at the cost of your manhood" |
| 22:23.56 | drumkilla | that should be ~gentoo |
| 22:24.04 | Qwell | ~gentoo |
| 22:24.04 | jbot | hmm... gentoo is foo |
| 22:24.13 | drumkilla | jbot_: forget gentoo |
| 22:24.25 | mwalling | dropline gnome (an "aftermarket" gnome addon for slack) provides pam... their first release for 12.0 they screwed up one of the configs and it stopped checking passwords in gdm |
| 22:24.32 | mwalling | nothing should be that easy to f-up |
| 22:24.50 | pelaofeliz | [TK]D-Fender: Thanks for the help -- that worked perfectly. |
| 22:24.54 | mwalling | (and the developer should be shot) |
| 22:24.54 | drumkilla | jbot_: gentoo is <reply> Gentoo : When you positively MUST get that extra 0.05% out of your system at the cost of your manhood |
| 22:25.00 | alrs | goes looking for the November 1995 infomagic 5-disc set |
| 22:25.09 | drumkilla | kicks jbot_ |
| 22:25.21 | Qwell | lol |
| 22:25.24 | mwalling | ~gentoo |
| 22:25.25 | jbot | [gentoo] foo |
| 22:25.33 | Qwell | mwalling: I didn't realize you had actually tried to add pam stuff to slack |
| 22:25.34 | mwalling | fail |
| 22:25.57 | drumkilla | ~gentoo |
| 22:25.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:26.04 | [TK]D-Fender | You know its a bad sign when you have to get Gnome through 3rd party sources because its maintainer can't be bothered |
| 22:26.15 | mwalling | [TK]D-Fender: it is one guy... |
| 22:26.22 | mikehime | pam is for auth in the gui? |
| 22:26.36 | [TK]D-Fender | mwalling: Yay.... the only way he can acheive concensus! |
| 22:26.38 | mwalling | mikehime: gdm can use pam for login |
| 22:26.47 | drumkilla | ~gentoo |
| 22:26.52 | drumkilla | do it now! |
| 22:26.52 | outtolunc | ~genfoo |
| 22:26.55 | drumkilla | bah. |
| 22:27.12 | mikehime | yeah that's what I thought you were getting at |
| 22:27.13 | [TK]D-Fender | Genpoo! |
| 22:27.25 | mikehime | I'm using the cli though so bash is fine imo |
| 22:27.47 | mwalling | Qwell: i have a vm with a pamified slack set up as a server, but its not in production... |
| 22:28.04 | mwalling | some of what pam can do is nice though (for SSO applications) |
| 22:28.05 | [TK]D-Fender | Strangely I did well on Slack because it came with "mc" right from the CD :) |
| 22:28.16 | mwalling | hehe |
| 22:28.21 | Qwell | [TK]D-Fender: because they haven't been able to package vim yet |
| 22:28.23 | Qwell | ducks |
| 22:28.33 | mwalling | swings low |
| 22:28.45 | alrs | Qwell: Not "they", "he" |
| 22:29.13 | Qwell | alrs: is Patrick the only one that does any of the work at all? |
| 22:29.27 | mwalling | alrs: pat has minions |
| 22:29.44 | mwalling | alienBOB, rworkman, etc |
| 22:29.50 | alrs | mwalling: where did he get them? |
| 22:29.50 | [TK]D-Fender | mwalling: Sounds like a netherworld despot ;) |
| 22:29.55 | mwalling | heh |
| 22:30.03 | mikehime | fascinating, never considered sso |
| 22:30.05 | mikehime | i'll keep that in mind |
| 22:30.32 | mwalling | yeah, i looked into pam_mysql when ldap was too hard for me |
| 22:30.37 | mwalling | :) |
| 22:30.41 | [TK]D-Fender | mikehime: It IS a learning experience, and it is solid. Feel free to stick with it and switch when you hit the last wall you can tolerate :) |
| 22:31.04 | [TK]D-Fender | mwalling: I suck at linux and compiling a kernel & dealing with a boot loader scared me :) |
| 22:31.20 | mwalling | i sucked at linux too! |
| 22:31.23 | [TK]D-Fender | mwalling: Which is where CentOS's stock 2.6 for ztdummy support came in :) |
| 22:31.25 | *** join/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 22:31.45 | [TK]D-Fender | Helped with a few other things too. |
| 22:32.00 | [TK]D-Fender | mikehime: I did unixODBC from source as well as the rest of the LAMP stack. |
| 22:32.00 | ipstatic | Anyone using the UNISTIM channel here? |
| 22:32.05 | Qwell | mwalling: what about grub? is there grub yet? |
| 22:32.08 | [TK]D-Fender | ipstatic: ....LOL |
| 22:32.11 | mwalling | in extra/ |
| 22:32.13 | [TK]D-Fender | Qwell: NOPE! |
| 22:32.17 | ipstatic | :( I know, dont laugh |
| 22:32.20 | [TK]D-Fender | Qwell: Stock LILO FTW! |
| 22:32.25 | mikehime | thanks for the support everyone |
| 22:32.30 | ipstatic | I got a snom 360 beside me as well! |
| 22:32.39 | mikehime | I have confidence in myself that's not the issue :) no boundary i can't scale provided the resources |
| 22:32.44 | mwalling | [TK]D-Fender: whats wrong with that? |
| 22:32.45 | [TK]D-Fender | ipstatic: when people mention Nortel, we typically just point & laugh. |
| 22:32.47 | mikehime | but sometimes people don't put up with newbs- that's the hard part |
| 22:33.00 | mikehime | and yeah I did my LAMP from source, too |
| 22:33.02 | mwalling | i eat newbs |
| 22:33.06 | [TK]D-Fender | mwalling: LILO? ... not much actually, but just so far behind the times... |
| 22:33.08 | drumkilla | ipstatic: the people that have used it have all reported success from what I've seen |
| 22:33.09 | mikehime | mwalling LOL |
| 22:33.19 | drumkilla | ipstatic: if it doesn't work, you're screwed based on how it's written, heh |
| 22:33.28 | ipstatic | well I got it running |
| 22:33.36 | mwalling | [TK]D-Fender: ... whats wrong with being behind the times when it still works? |
| 22:33.38 | [TK]D-Fender | mwalling: Slack IS a great basic distro. Not particularly good at anything, but solid. A good minimalist distro I guess. |
| 22:33.49 | ipstatic | its just caller id doesn't seem to work if you call the device from a zap channel |
| 22:34.08 | [TK]D-Fender | mwalling: But is more work than the vast majority have any need to take into consideration. |
| 22:34.09 | mwalling | [TK]D-Fender: how old is your car |
| 22:34.23 | [TK]D-Fender | ipstatic: perhaps your Zap channel is the problem. |
| 22:35.02 | ipstatic | caller ID does get passed to my snom phone though |
| 22:35.11 | ipstatic | and I can NoOp the callerid to the console |
| 22:36.13 | ipstatic | and the zap device is a Digium TE122 connected to a PRI circuit |
| 22:36.44 | ipstatic | so I am coming in from POTS to the Nortel |
| 22:44.14 | mcab | Slackware is great. I learned a lot about linux when I set it up. |
| 22:44.48 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:44.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:44.57 | [TK]D-Fender | ouch |
| 22:45.01 | [TK]D-Fender | Stupid peer! |
| 22:45.04 | *** part/#asterisk JHilgeman (n=jh@209.48.241.194) |
| 22:45.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:52.08 | x86 | peer is a sunnofabitch |
| 22:52.54 | x86 | [TK]D-Fender: hey are you into guns at all? |
| 22:54.10 | [TK]D-Fender | x86: Used to be. Father collected, and I put on displays for gun shows |
| 22:54.26 | [TK]D-Fender | x86: And terhew as that year I spent working at the firearms importer/exporter |
| 22:54.29 | ManxPower | mumbles something that sounds like "Make Love, not War" |
| 22:55.03 | [TK]D-Fender | ManxPower: I gave up guns a long time ago though... onto SWORDS now :p |
| 22:55.23 | ManxPower | On the other hand, there's a homophobe redneck that lives in the area I'd not mind shooting in the knee. |
| 22:56.03 | ManxPower | [TK]D-Fender: I'm only one of two perm residents at the campground that don't own a gun 8-) |
| 22:57.46 | jaytee | somehow the mental image of a homophobe redneck limping along on one leg with the other one trailing blood makes me giggle. |
| 22:58.06 | jaytee | I don't own a gun |
| 22:58.20 | jaytee | I have a katana |
| 22:59.11 | [TK]D-Fender | jaytee: Got a gallery up for it? |
| 22:59.21 | jaytee | pics? |
| 22:59.24 | ManxPower | The only time I seriously considered getting a gun of some sort was Nov 1999. If there were going to be major y2k problems they would have started happening more and more during Dec 1999, as there were not major issues, I decided to just stock up on a little food and water. |
| 22:59.31 | jaytee | lemme check my bucket |
| 23:00.59 | jaytee | I'd worked in IS since the late 80's so I didn't stock up on anything for Y2K. I knew most of the majors had done their homework. |
| 23:01.23 | [TK]D-Fender | jaytee: My latest acquisition : http://www.roninswords.com/ |
| 23:01.37 | [TK]D-Fender | oops |
| 23:01.39 | [TK]D-Fender | http://www.roninswords.com/custom_kiku_in_tea.htm |
| 23:02.45 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:04.15 | frieze | anyone know if there's an updated version of chan_mobile somewhere other than in trunk? The one I have does not seem to be built using the same data structures as in either 1.4.2 or 1.6b9 |
| 23:04.21 | jaytee | scuse me while I drool |
| 23:07.02 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 23:07.14 | jaytee | [TK]D-Fender, very very nice. I like the hilt, mine is the cheesy fake ivory carved dragon style and my blade is only 440 staintless although it's a folded blade. |
| 23:07.37 | mwalling | [TK]D-Fender: re: behind the times: Wed Jun 18 14:42:48 CDT 2008 |
| 23:07.38 | [TK]D-Fender | jaytee: Ah... lemme guess the Marto Highlander modle, right? |
| 23:07.38 | mwalling | xap/mozilla-firefox-3.0-i686-1.tgz: Upgraded to firefox-3.0. |
| 23:07.43 | jaytee | that must have cost a pretty penny |
| 23:07.57 | [TK]D-Fender | mwalling: And who built that package? |
| 23:08.12 | mwalling | donno... prolly pat |
| 23:08.13 | jaytee | [TK]D-Fender, yup :-) |
| 23:08.18 | [TK]D-Fender | jaytee: Only $800 after taxes and shipping? |
| 23:08.31 | [TK]D-Fender | mwalling: really... COMES on the CD? |
| 23:08.32 | jaytee | yours? |
| 23:08.48 | [TK]D-Fender | jaytee: Yup, thats the makers gallery page for my custom order |
| 23:08.50 | mwalling | [TK]D-Fender: when -current is released, yes |
| 23:08.52 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 23:09.23 | jaytee | wow, I figure like that wouldn't go for less than 1500. |
| 23:09.26 | [TK]D-Fender | mwalling: Well as long as you're using chan_fluxcapaciter.so .... lemme look at CentOS 10.0 ;) |
| 23:09.34 | jaytee | something like that one |
| 23:09.46 | jaytee | awesome price |
| 23:09.51 | [TK]D-Fender | jaytee: 440.... dear god don't swing it around... |
| 23:09.58 | jaytee | hahaha |
| 23:10.03 | jaytee | it's a nice ornament |
| 23:10.45 | jaytee | I wouldn't consider it a true professional grade but you could impale someone nicely enough. |
| 23:10.48 | [TK]D-Fender | jaytee: correct term is "wall hanger" : http://www.youtube.com/watch?v=YPDL4eiKhVc |
| 23:11.18 | jaytee | hahahaha, yeah I saw that before |
| 23:11.45 | jaytee | <PROTECTED> |
| 23:12.14 | jaytee | couple of those guys are lucky they've made it to adulthood they're such clumsy idiots. |
| 23:12.42 | [TK]D-Fender | jaytee: yeah, his boss looks like a total turd, and he's just a douchebag |
| 23:15.01 | jaytee | when HSN first came on cable I remember seeing one segment that featured some 286 AT clone and the way the salesman described it was hilarious. |
| 23:15.29 | alrs | hsn was on broadcast tv in Los Angeles back in the 80s |
| 23:15.31 | alrs | may still be |
| 23:16.01 | jaytee | people who think the word frequency is just another word for often using words like megahertz really amuse me. |
| 23:16.47 | pelaofeliz | Is it possible for an agent to not have to answer the phone every time a call is programmatically originated? |
| 23:16.51 | [TK]D-Fender | jaytee: not entirely inaccurate, |
| 23:17.08 | [TK]D-Fender | pelaofeliz: have you considered not picking it up? |
| 23:17.36 | jaytee | well, I got it on cable but I don't think it was available via broadcast in the early 80's in the Boston area but it could have been simulcast on a local UHF channel. |
| 23:18.26 | alrs | I wonder if Tom Vu is out of jail |
| 23:18.28 | pelaofeliz | [TK]D-Fender: I'm not sure what you mean... or are you being facetious? |
| 23:18.53 | [TK]D-Fender | pelaofeliz: What forces you to pick up the receiver? |
| 23:19.13 | jaytee | The Will of Landru |
| 23:19.21 | [TK]D-Fender | must.... OBEY!!! |
| 23:21.23 | pelaofeliz | [TK]D-Fender: well, currently whenever I generate a call, the agent phone rings, and the number is dialed. It'd be nice if I could leave the agent hands free between, and just have asterisk dial out and connect to the agent channel |
| 23:21.27 | pelaofeliz | ...if that makes any sense |
| 23:22.08 | [TK]D-Fender | pelaofeliz: by defaul, * calls whatever you tell it to call for as long as you tell it to allow ringing to go on for. |
| 23:22.23 | [TK]D-Fender | pelaofeliz: but by default there is nothing to force them to answer. |
| 23:22.40 | jaytee | someone like me who's done some time in the helpdesk trenches would love it if corporate america would fund IT to roll out robotic arms with RDP telepresence controls so you can reach and slap the guy on the other end of the phone and yell, "Hello!!! McFly!!!!" |
| 23:22.59 | [TK]D-Fender | pelaofeliz: Now depending what phone they use you might be able to set an auto-answer header to instruct the phone to pick up on hands-free, etc like you would for pagin. |
| 23:23.31 | jaytee | one in every PEBKAC type employee's cube. |
| 23:23.48 | [TK]D-Fender | jaytee: ... I tt-monkey paged a good chunk of my office last week :) |
| 23:24.17 | [TK]D-Fender | jaytee: Next time it'll be a bolt being pulled back and a machine gun firing with glass breaking, etc :) |
| 23:25.14 | jaytee | "Hi!! and thanks for calling! I noticed your CALLERID says "Unavailable" and coincidentally.....SO AM I!!!!" Hangup() |
| 23:25.25 | pelaofeliz | [TK]D-Fender: okay -- so with a predictive dialer type setup, when * dials 3 calls per agent, how does that work, since the agent isn't actually on any of the 3 calls until later? |
| 23:26.25 | [TK]D-Fender | pelaofeliz: Dpends on what logic this "dilaler" of yours is running, now doesn't it? Why would any system dial 3 *'s at a time for 1 person? |
| 23:26.32 | [TK]D-Fender | #'s? |
| 23:26.45 | *** join/#asterisk Drunktard (n=sebas@201.198.239.167) |
| 23:27.18 | Drunktard | this might be a little off topic but what tool can i use to transcode from a G.711 mu-law file to a normal PCM? |
| 23:27.29 | [TK]D-Fender | Drunktard: sox |
| 23:27.50 | [TK]D-Fender | Drunktard: and * CLI can convert as well to any format it can read |
| 23:27.55 | [TK]D-Fender | (write) |
| 23:28.29 | Drunktard | [TK]D-Fender: thanks will look for it, do you know offhand of specific flags i may need to use? i'll read the man but just need to see how's the file first |
| 23:28.47 | [TK]D-Fender | drunk look it up on the WIKI |
| 23:29.09 | *** join/#asterisk nick125 (i=nick@pdpc/supporter/student/nick125) |
| 23:29.12 | pelaofeliz | [TK]D-Fender: standard predictive dialer logic -- assuming that only 1 in 3 calls is answered, and I have 3 agents available, I'll dial 9 lines at once, to statistically get 3 people on the phone with 3 agents |
| 23:31.23 | mikehime | pelao: you writing predictive algos? |
| 23:31.43 | pelaofeliz | nothing fancy |
| 23:32.01 | mikehime | was gunna say you could probably just borrow someone else's ;) |
| 23:32.22 | pelaofeliz | yeah, basically just want to dial 1.5 or 2x the numbers as agents |
| 23:32.46 | pelaofeliz | we don't have enough agents to really benefit from a smart algorithm |
| 23:33.07 | cnielsen | mikehime: Thanks for your help this afternoon - I was able to get a spec put together that will meet our needs :) |
| 23:33.22 | mikehime | cnielsen: awesome glad to hear :D |
| 23:37.28 | *** join/#asterisk LoOoD (n=gman@64.201.247.2) |
| 23:39.59 | mikehime | mysql doesn't create a my.cnf by default? |
| 23:40.18 | mikehime | myodbc is looking in /tmp for the mysql.sock file |
| 23:40.45 | mikehime | but it's in /var/run/mysql by default |
| 23:42.39 | LoOoD | asterisk says there is a active call and it been running for 24hrs. But the user it says started the call isn't even on the call, he hung up yesterday. I can soft hang the call easy, but how do figure whats causing it not to hangup in the first place ? |
| 23:48.10 | *** join/#asterisk s0lid (n=s0lid@124.106.140.114) |
| 23:52.10 | mikehime | nm there's an option in the DSN setup for socket location |
| 23:52.13 | mikehime | i had a moment of r-tism |
| 23:53.58 | unpaidbill | awww my 7960 has tux on it |
| 23:54.15 | unpaidbill | i feel so ... something the opposite of manly |
| 23:54.42 | seanbright | ~seanbright |
| 23:54.43 | jbot | i guess seanbright is a girl with standards |
| 23:55.07 | unpaidbill | nah i'd be considered a moralless slut |
| 23:55.16 | unpaidbill | immoral i guess. haha |
| 23:56.00 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 23:56.30 | znoG | to compile zaptel with, say, gcc-4.1 ('gcc' defaults to gcc-4.2), should it be as simple as export CC=gcc-4.1; ./configure && make clean && make ? |
| 23:56.43 | unpaidbill | should be |
| 23:57.04 | unpaidbill | at worst you could link /usr/bin/gcc to the 4.1 binary for the compile |