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| 00:39.06 | silvano-v | Having an issue where blind/attended transfers drop within 5 sec when rpid=yes on trunk. I'm getting timeout /failed to respond to critical packet in asterisk console. Same trunk, settings, no problem on older PIAF install. Same issue on asterisk 10,11,12. |
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| 01:00.52 | silvano-v | https://issues.asterisk.org/jira/browse/ASTERISK-18999 having this issue with a proximiti trunk and asterisk 10,11 and 12. |
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| 02:56.02 | MarkSX | Hey guys is there a way to disable simple bridge and force native_rtp |
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| 03:12.38 | MarkSX | Or perhaps someone can explain why with this exampple as User 102 dialing 111, I hear ringing until the end of the macro. I'd prefer to hear the other party straight away. |
| 03:12.39 | MarkSX | http://pastebin.com/tWaX8ebc |
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| 03:18.52 | anhvt | hello |
| 03:19.28 | anhvt | i have an issue with asterisk conference module performance, does anyone can help me? |
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| 04:07.44 | WIMPy | anhvt: Tht is a pretty vague question. |
| 04:11.03 | anhvt | yes |
| 04:44.26 | ChannelZ | Confbridge? MeetMe? |
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| 06:33.32 | anhvt | we have asterisk 11.10.2 install in a server with 16 core CPU, 32GB |
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| 06:33.54 | anhvt | the maximum conference room can create is about 40~50 room |
| 06:34.17 | anhvt | each room have only 1 participant |
| 06:34.36 | anhvt | and the CPU usage reach 99~100% |
| 06:35.05 | anhvt | any idea to reduce the CPU usage while conferencing? |
| 06:38.06 | anhvt | the application we use is ConfBridge |
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| 07:15.53 | ChannelZ | There's a lot of variables. What codec are the calls? |
| 07:16.06 | anhvt | the codec is ulaw |
| 07:17.26 | ChannelZ | OK at least you don't have tons of transcoding going on. You might want to tweak on (like, disable) the DSP settings, disabling noise reduction, silence detection, etc. which takes CPU |
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| 07:17.45 | anhvt | thanks ChannelZ |
| 07:17.55 | anhvt | i've disabled all of that |
| 07:18.23 | ChannelZ | You can increase the mixing interval so it's working on fewer larger chunks for less overhead, however that will add latency |
| 07:19.48 | anhvt | how can I do that ChannelZ? |
| 07:20.57 | ChannelZ | I've personally not done any tests with confbridge but other than that I'm not sure what to suggest.. perhaps asterisk's mix engine is just not terribly optimized |
| 07:20.58 | anhvt | in confbridge userprofile, i've set following setttings: |
| 07:21.01 | anhvt | Drop_silence: disabled |
| 07:21.01 | anhvt | Silence Threshold: 2500ms |
| 07:21.01 | anhvt | Talking Threshold: 160ms |
| 07:21.01 | anhvt | Denoise: disabled |
| 07:21.01 | anhvt | Jitterbuffer: disabled |
| 07:21.02 | anhvt | Talk Detect Events: disabled |
| 07:21.02 | anhvt | DTMF Pass Through: disabled |
| 07:21.11 | ChannelZ | Please don't flood |
| 07:21.14 | ChannelZ | ~pb |
| 07:21.16 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 07:21.38 | anhvt | sorry |
| 07:21.45 | ChannelZ | the mixing_interval is a bridge option |
| 07:23.03 | anhvt | ChannelZ, is there any other idea to solve this issue? |
| 07:23.13 | ChannelZ | Actually I guess dsp_drop_silence being on can actually help performance as I guess silence detection is cheaper than mixing. |
| 07:24.02 | ChannelZ | Since in a conference of 10 people only one of them is going to be talking (minus some crossover as people interrupt each other) so the other 9 streams don't need to be mixed. |
| 07:24.48 | ChannelZ | (and they're all individual mixes, mind you) |
| 07:24.55 | anhvt | we expect that asterisk can create hundreds of room |
| 07:26.19 | anhvt | Let me try to test with dsp_drop_silence=yes |
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| 07:28.58 | XATRIX | Hi guys, i'd like you to advise me on, is it a good idea to setup OpenVZ container for PBX or it's better to use a hardware server for ? |
| 07:30.32 | ChannelZ | I don't really have enough ways to call into a conf and benchmark more than a few calls.. |
| 07:33.24 | anhvt | ChannelZ, I use sipp, to dial in an extension which create a room with random number, and add an musiconhold to that room as member |
| 07:34.19 | anhvt | so total rooms can be created is about 40 rooms |
| 07:37.27 | XATRIX | Did anyone try GSM-gate to connect cell-network into the PBX ? |
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| 08:35.28 | tparcina | igcewieling: Yes, I should learn yum and rpm. |
| 08:36.46 | tparcina | BakaKuna: Thank you. And centos-asterisk.repo has the DAHDI dependencies and sounds. "hence, it is always recommended to add the centos-asterisk repository definition" |
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| 09:19.40 | MaliutaLap | anyone know if you can automatically add a SIP peer to a queue? |
| 09:20.32 | bulkorok | MaliutaLap: you can put the peer in queue.conf permanently |
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| 09:25.06 | MaliutaLap | bulkorok: thanks, just found that :) |
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| 09:36.18 | MaliutaLap | Tonight is my night - beating * into submission :) |
| 09:36.48 | MaliutaLap | of course that's just me avoiding writing several LDAP schema :) |
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| 09:49.20 | izbushka_ | hi |
| 09:51.07 | izbushka_ | is it possible to call AGI script from CLI? |
| 09:52.02 | izbushka_ | Usage: agi exec <channel name> <app and arguments>. What is "channel name"? |
| 09:57.35 | MaliutaLap | izbushka_: channel name is the peer/device you want to use (so SIP/mypeer or DAHDI/1) |
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| 09:58.11 | izbushka_ | what if my AGI script does not need one? |
| 09:59.00 | izbushka_ | MaliutaLap, is there some default channel or so? |
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| 10:01.12 | MaliutaLap | izbushka_: not as far as I know |
| 10:01.30 | MaliutaLap | izbushka_: the options in <> aren |
| 10:01.34 | MaliutaLap | 't compulsory |
| 10:02.13 | MaliutaLap | izbushka_: so you should be able to call "agi exec SCRIPT SCRIPT_ARGS" |
| 10:02.27 | MaliutaLap | or without the arguments if you don't need them |
| 10:03.32 | izbushka_ | Channel SCRIPT does not exist. :( |
| 10:04.35 | izbushka_ | MaliutaLap, thanks anyway |
| 10:06.45 | MaliutaLap | hmm |
| 10:07.04 | MaliutaLap | izbushka_: what is the agi script doing? |
| 10:09.12 | izbushka_ | MaliutaLap, It sends SMS by exec DongleSendSMS(). But it also chooses right dongle |
| 10:13.29 | MaliutaLap | izbushka_: if the AGI is going to overide the channel then why not send it to the first dongle on the list |
| 10:13.42 | MaliutaLap | izbushka_: if you need to exec it from within the CLI |
| 10:15.18 | izbushka_ | MaliutaLap, "core show channels" says "0 active channels" Which should i use? |
| 10:15.43 | izbushka_ | how could I get channel name for the dongle? |
| 10:17.49 | MaliutaLap | that's active channels - how are the dongles registered? |
| 10:19.59 | izbushka_ | dongles are local devices (chan_dongle.so) |
| 10:41.14 | izbushka_ | MaliutaLap, core show channeltype Dongle: http://bpaste.net/show/453535/ But channel "Dongle/dongle_id" also does not exist. Dongle_id is from "dongle show devices" |
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| 10:53.07 | izbushka_ | looks like agi exec needs only active channel.. |
| 10:53.55 | izbushka_ | so there is no way to call agi script from CLI without channel? |
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| 11:09.24 | AbuAbu | Hi all, I have two sip clients behind nat. I have nat=force_rport,comedia |
| 11:09.46 | AbuAbu | however during call I see RTP packets going to IP behind nat :( |
| 11:09.47 | AbuAbu | Got RTP packet from 78.159.32.36:57840 (type 18, seq 018540, ts 3364240, len 000020) |
| 11:09.48 | AbuAbu | Sent RTP packet to 10.50.7.213:4000 (type 18, seq 024502, ts 3364240, len 000020) |
| 11:09.55 | AbuAbu | what's wrong there? |
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| 11:14.13 | AbuAbu | sip show peer 100 show me |
| 11:14.14 | AbuAbu | <PROTECTED> |
| 11:16.31 | file | until media is received from it Asterisk doesn't know the source IP address+port |
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| 12:44.31 | RadJackson | Hi , On a context , i answer the customer call , and link it to another call. is there a way to wait for customer exten during the call ? i mean when he types 9 i want to end the call and send him to another context. |
| 12:44.34 | RadJackson | is it possible? |
| 12:51.03 | [TK]D-Fender | "link it" how? |
| 12:51.13 | MaliutaLap | yeah, that was my thought |
| 12:52.22 | MaliutaLap | wants to bitchslap one one my providers - haven't been able to call out of them since I got the account. Force the CID to that account number and calls flow |
| 12:52.25 | RadJackson | well my question is precisely is it possible to make a WaitExten during a Dial. |
| 12:52.28 | MaliutaLap | stoopid |
| 12:53.00 | MaliutaLap | RadJackson: what options are you passing to Dial()? Normally you'd need Dial() to return |
| 12:53.04 | MaliutaLap | then do something |
| 12:53.07 | [TK]D-Fender | RadJackson: "core show application dial" <- |
| 12:53.33 | RadJackson | like during the conversation , when the customer type 9 i want to control that |
| 12:53.40 | RadJackson | [TK]D-Fender very cool |
| 12:54.17 | [TK]D-Fender | RadJackson: features.conf <- |
| 12:55.52 | MaliutaLap | [TK]D-Fender: should have seen my candidates face when I showed him all the phones in the office ringing because they were in a queue - probably also because the indial wasn't working until today |
| 12:55.59 | MaliutaLap | that provider was no help either |
| 12:57.07 | [TK]D-Fender | MaliutaLap: how long did it take to figure that out? |
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| 12:57.30 | [TK]D-Fender | MaliutaLap: many don't let you set CID, so setting fromuser is usually the first thing you do. |
| 12:57.37 | MaliutaLap | [TK]D-Fender: the indials? probably about 1/2 an hour of fiddling |
| 12:58.11 | MaliutaLap | [TK]D-Fender: the one that hadn't been working for years was a personal account, and it had a fromuser in it |
| 12:58.54 | MaliutaLap | [TK]D-Fender: the one for the office was a different matter |
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| 12:59.22 | MaliutaLap | [TK]D-Fender: and the queue took a while - mainly bacause I had to re-conf a line on each of the phones |
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| 13:17.06 | [TK]D-Fender | MaliutaLap: Why does the phone matter in that? |
| 13:19.14 | RadJackson | [TK]D-Fender Thanks but that file contains several commands , Nothing wrong seeking help , all i ask if it is possible to wait for an extension during a Dial. ? or Do i have to wait until the Dial is hung up and then do my thing ? |
| 13:19.38 | [TK]D-Fender | RadJackson: WHEN during a dial? |
| 13:19.43 | MaliutaLap | [TK]D-Fender: well the sip peers. the phones are 6 line cisco units, I have one sip peer per line |
| 13:20.04 | MaliutaLap | [TK]D-Fender: so it was more the sip peers I had to re-conf |
| 13:20.24 | [TK]D-Fender | MaliutaLap: Wondering why a phone config impacts your peer to your ITSP. |
| 13:21.44 | RadJackson | Dial is for establishing an outgoing connection on a channel , [TK]D-Fender for example a customer calls one extension, my context calls a sip account , so the customer is on a conversation with an agent , so i want to control when the customer type 9 for example , i would love to hang up the call and put him online with another agent. |
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| 13:22.36 | RadJackson | SO is it possible to detect when he types 9 ? |
| 13:22.38 | [TK]D-Fender | RadJackson: features.conf <- |
| 13:23.35 | RadJackson | very helpful |
| 13:24.06 | RadJackson | I didnt even ask for HOW , but just if it is possible to detect an extension dtmf type while a Dial . Anyway thanks |
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| 13:25.23 | [TK]D-Fender | RadJackson: Yes, that's what features.conf lets you do... while IN a bridged call. |
| 13:25.47 | [TK]D-Fender | RadJackson: That's wh I asked "when". Pre-answer is one thing. In a bridged call is another. |
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| 13:27.15 | MaliutaLap | [TK]D-Fender: 2 different issues on 2 different machines and 2 different providers |
| 13:27.39 | [TK]D-Fender | MaliutaLap: ok/fine/sure :) Glad it's working now! |
| 13:27.50 | MaliutaLap | [TK]D-Fender: the problem with indials not working is a "business grade" ITSP for a campaign I'm working on |
| 13:28.13 | MaliutaLap | [TK]D-Fender: the one that wasn't allow me to dial out is the free account from my ISP |
| 13:28.53 | [TK]D-Fender | MaliutaLap: Ask for your money back! |
| 13:29.02 | MaliutaLap | [TK]D-Fender: the candidate was impressed that in just under an hour I got the indials working and that when the office number is dialed all the phones ring |
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| 13:29.24 | MaliutaLap | [TK]D-Fender: I should hit up internode - they officially say they don't support * |
| 13:29.49 | MaliutaLap | and no config example I could find worked |
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| 16:17.23 | pjensen00 | I've just upgraded to asterisk 12.4 from 12.3.2 and now I can't connect with the AMI. I'm troubleshooting and I'm not seeing any messages in the asterisk log. I have debug turned on and still no response. Any recommendations? |
| 16:17.47 | pjensen00 | er, debug turned on and no messages show up when I try to connect |
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| 16:20.30 | *** join/#asterisk rkessler (ad4cb8a2@gateway/web/freenode/ip.173.76.184.162) |
| 16:21.53 | rkessler | Hey all. Wondering if anyone could point me in the right direction. I have 2 asterisk servers. Server 1 sends calls to server 2 (for transcoding) then out to our SIP provider. When I try to send calls, Server 2 is giving me the following error "rejected because extension not found in context 'default'" I checked via sip.conf and the context IS set in both the global settings and the individual trunk |
| 16:21.56 | rkessler | any suggestions? |
| 16:22.42 | file | was the extension found in context default? |
| 16:22.50 | file | or rather, is it? |
| 16:23.28 | rkessler | On server 2, that is the only error that is showing |
| 16:23.43 | rkessler | I do have the extension setup, reloaded asterisk couple times no change |
| 16:24.50 | file | does dialplan show <number dialed>@default |
| 16:24.55 | file | in the Asterisk CLI elicit a response |
| 16:24.55 | rkessler | On server 2, here is how I have it setup |
| 16:25.07 | rkessler | [6005] description=xxxx callerid=<6005> canreinvite=no context=shipsat dmtfmode=rfc2833 host=dynamic language=en call-limit = 4 ;nat=no nat=force_rport,comedia qualify=yes type=friend secret= trustrpid=no sendrpid=yes username=6005 accountcode=user1 disallow=all allow=g729 allow=silk8 |
| 16:26.12 | rkessler | If I use the context shipsat, it does show a response, but if I use default it does not |
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| 16:28.43 | rkessler | shouldn't the code context=shipsat force all calls on that trunk to use context shipsat? |
| 16:28.47 | rkessler | rather then trying default? |
| 16:32.58 | rkessler | If I run a sip show peers, under Def.Context it does show the proper context, but when calls comes in it is still trying to divery them to default |
| 16:33.22 | raspberrypifan | is freeswitch better then asteris |
| 16:34.07 | rkessler | Raspberrypifan - I have used both - I find asterisk has more functionality and is better supported. But that might just be my opnion. If your using an RPI check out raspbx |
| 16:34.30 | raspberrypifan | yea i know ive already tried it |
| 16:34.33 | raspberrypifan | i like freepbx |
| 16:34.34 | raspberrypifan | it jsut works |
| 16:34.43 | raspberrypifan | but ive been toying with freeswitch and i cant really get it together |
| 16:34.55 | rkessler | i set it up once, but didnlt like it so I switched back |
| 16:35.16 | raspberrypifan | idk the freeswitch book is much clearer then the asterisk book |
| 16:35.29 | raspberrypifan | i really feel like goign back, but i also feel like i might be missing out if i dont liek freeswitch |
| 16:35.31 | rkessler | k |
| 16:35.37 | raspberrypifan | i mean can asterisk do billing nad call termiantion |
| 16:35.40 | raspberrypifan | and handle lots of minutes |
| 16:35.41 | raspberrypifan | and what not |
| 16:35.45 | rkessler | billing sure. |
| 16:35.50 | rkessler | check out a2billing |
| 16:35.54 | rkessler | or write your own |
| 16:36.13 | raspberrypifan | is a2billing free? |
| 16:36.23 | rkessler | http://www.asterisk2billing.org/ |
| 16:37.14 | raspberrypifan | yea not free |
| 16:37.20 | raspberrypifan | im pretty sure freeswitch has it for free |
| 16:37.22 | raspberrypifan | in a mod somewehre |
| 16:37.25 | rkessler | k |
| 16:37.57 | raspberrypifan | idk |
| 16:38.03 | raspberrypifan | why cant it be free for asterisk |
| 16:38.04 | rkessler | "2Billing is free and open source software for Asterisk, providing telecoms customer management including admin, agent, customer and online signup pages, with flexible inline rating and billing of calls and services in realtime." |
| 16:38.17 | rkessler | you pay for support |
| 16:38.20 | raspberrypifan | ohhh |
| 16:38.41 | rkessler | @file - Any suggestions how I can force the trunk to use the proper context? |
| 16:39.00 | raspberrypifan | can someone then please rewrite the asterisk book to make it more topical |
| 16:39.04 | raspberrypifan | like the freeswitch book |
| 16:40.29 | rkessler | Anyone else have any suggestions for forcing a call through a specific context, rather then having it try default? |
| 16:43.02 | raspberrypifan | seriously who wrote the asterisk book |
| 16:43.04 | raspberrypifan | anyone in here? |
| 16:43.11 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 16:43.15 | [TK]D-Fender | [12:36]raspberrypifanis a2billing free? "A2Billing is free and open source software for Asterisk," |
| 16:43.24 | [TK]D-Fender | raspberrypifan: It is free. |
| 16:43.35 | raspberrypifan | [TK]D-Fender: yea |
| 16:45.09 | raspberrypifan | i might return to asterisk after much pain with freeswitch |
| 16:45.13 | raspberrypifan | but i reall want a better book |
| 16:49.37 | *** join/#asterisk rkessler_ (ad4cb8a2@gateway/web/freenode/ip.173.76.184.162) |
| 16:50.26 | rkessler_ | Lost connection - so lost history - Anyone currently in here? Have a question/issue currently with Contexts |
| 16:51.45 | raspberrypifan | im here |
| 16:51.47 | raspberrypifan | but no real isuses |
| 16:52.35 | rkessler_ | Are you familiar with Asterisk Commandline / Contexts? |
| 16:53.32 | raspberrypifan | apparentyl |
| 16:54.14 | rkessler_ | ok. The issue im having is incoming calls are defaulting to the "default" context rather then the one I have specified in sip.conf |
| 16:54.25 | raspberrypifan | ah |
| 16:55.07 | rkessler_ | any ideas? |
| 16:55.12 | mjordan | raspberrypifan: the authors of the book are on the front cover... |
| 16:56.13 | raspberrypifan | but are they on irc |
| 16:56.28 | mjordan | yes |
| 16:56.38 | raspberrypifan | where i must hunt them down |
| 16:57.21 | mjordan | so you can thank them for spending their free time writing a book that attempts to make a complex topic approachable for a large variety of people with different backgrounds? |
| 16:57.26 | mjordan | Because I'm *sure* that's what you mean to do. |
| 16:57.42 | raspberrypifan | im gonna refer them to the freeswitch book |
| 16:57.45 | raspberrypifan | which is more sensibile |
| 16:58.07 | mjordan | heh |
| 16:58.09 | mjordan | ok |
| 16:58.23 | mjordan | I'm sure they'll respond to you if they feel like it. I know what my response to you would be :-) |
| 16:58.31 | raspberrypifan | its just more approachable |
| 16:58.36 | raspberrypifan | the asterisk book is so long and complex |
| 17:04.55 | [TK]D-Fender | [12:54]rkessler_ok. The issue im having is incoming calls are defaulting to the "default" context rather then the one I have specified in sip.conf <- look at the call and you'll see why. "sip set debug on" |
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| 17:43.25 | ageis | is it possible to run MixMonitor on exten 's' instead of specific extens, so that everything in that context gets recorded? |
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| 17:48.09 | mjordan | ageis: MixMonitor is a dialplan application. A channel that executes that application gets recorded |
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| 17:48.28 | mjordan | ageis: if you want a channel to always get recorded, write a subroutine and call it from your extensions |
| 17:48.55 | mjordan | ageis: but the 's' extension is just another extension, used as a default. You can call MixMonitor from there, but it is not executed any differently than any other extension |
| 17:49.23 | ageis | mjordan: do you have info on subroutines? I'm trying to avoid repeatedly including the MixMonitor() line |
| 17:50.18 | mjordan | It's GoSub |
| 17:50.35 | mjordan | You'd still have to call GoSub from your extensions, but you can put shared logic into the subroutine |
| 17:51.02 | ageis | that's cool |
| 17:51.05 | ageis | thanks for the tip |
| 17:53.03 | mjordan | something like this http://pastebin.com/gDdtxtPD |
| 18:01.07 | sruffell | Soâ¦I know there aren't any "official" packages for Ubuntu trusty listed at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages, but is there a consensus about the best ones to use from the community are? |
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| 18:17.55 | pabelanger | sruffell, ya, those are dead. tzafrir maintains the debian version, so you should just be able to use the ones shipped in ubuntu |
| 18:18.09 | pabelanger | otherwise, use upstream debian version as you see if |
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| 18:18.39 | sruffell | kk, thanks |
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| 18:57.15 | dw5304 | after noon guys |
| 18:57.39 | dw5304 | so my t1 card showed uptoday question should i go with astersk 11 or 12 |
| 18:58.48 | [TK]D-Fender | If you're OOK with upgrading to 13 when it EOL's at the end of the year and want to try that functionality, then 12 |
| 18:58.54 | [TK]D-Fender | Otherwise 11 for the longer term right now |
| 18:59.19 | dw5304 | ok |
| 18:59.58 | dw5304 | first time user |
| 19:00.02 | dw5304 | so assuming 11 it should be :) |
| 19:01.12 | [TK]D-Fender | 12 rouks fine as well.. |
| 19:01.26 | [TK]D-Fender | just a question of how long you might want to ignore the branch you're on for. |
| 19:01.42 | dw5304 | ok |
| 19:01.49 | dw5304 | well what has the best t1 support both? |
| 19:03.44 | [TK]D-Fender | equal |
| 19:04.10 | [TK]D-Fender | It's the other goodies taht 12 comes with that might matter like the PJSIP driver stack, etc |
| 19:04.31 | dw5304 | ah |
| 19:04.36 | dw5304 | so i should go with 12 then? |
| 19:04.37 | dw5304 | lol |
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| 19:09.06 | [TK]D-Fender | dwif you're fine with the thought of upgrading when it EOL's |
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| 19:15.15 | *** join/#asterisk Jacoby6000 (~Jacoby600@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:15.41 | Jacoby6000 | Is there any way to make voicemail to email send emails to multiple addresses? |
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| 19:17.57 | [TK]D-Fender | Jacoby6000: that's up to your MTA to do delivery groups |
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| 19:20.50 | Jacoby6000 | Alright, that's what I thought. I've written a small patch for asterisk that allows support for a semicolon seperated list of emails. Do you know if that is something worth opening an issue/request for, or will everybody basically tell me to use delivery groups? |
| 19:23.04 | dw5304 | so decided to go with 12 |
| 19:23.18 | [TK]D-Fender | If you made a patch to make it directly possible I'm sure there arr a lot of people who'd want it. Submit it to jira |
| 19:23.48 | dw5304 | getting a /usr/bin/ld: /usr/local/lib/libsrtp.a(srtp.o): relocation R_X86_64_32 against `.rodata.str1.1' can not be used when making a shared object; recompile with -fPIC |
| 19:24.08 | dw5304 | was fallowing http://astrecipes.net/index.php?q=astrecipes/compiling+asterisk+12+on+centos+6.5 |
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| 19:32.00 | [TK]D-Fender | it just gave you the parameter to use... |
| 19:32.49 | ipengineer | Is there a way to set the hostname in pjsip.conf or similiar? Asterisk is picking up pbx.s1 instead of the fqdn pbx.s1.company.com |
| 19:33.06 | ipengineer | It is setting the wrong hostname on sip INVITEs causing no media |
| 19:34.56 | mjordan | ipengineer: ASTERISK-23994 |
| 19:35.08 | mjordan | I'm guessing that's your issue |
| 19:35.44 | mjordan | Jacoby6000: you can certainly propose it on an issue, but honestly, I think most people just use e-mail groups |
| 19:35.49 | ipengineer | mjordan: That is the exact issue |
| 19:40.26 | ipengineer | mjordan: Does asterisk get fqdn at startup and store it from there on until reload? |
| 19:40.49 | mjordan | ipengineer: it's getting just the hostname - that's the problem |
| 19:40.49 | file | it currently pulls the hostname via a PJSIP function and stores it |
| 19:41.38 | [TK]D-Fender | dw5304: ./configure CFLAGS=-fPIC ..... |
| 19:42.09 | dw5304 | k |
| 19:43.29 | ipengineer | file: would a res_pjsip reload clear that and cause it to request it again or would asterisk have to be restarted |
| 19:43.43 | file | restart |
| 19:45.32 | ipengineer | K thanks. |
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| 20:03.42 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:03.59 | pa | hi |
| 20:04.27 | pa | what does this mean? Rejected connect attempt from 127.0.0.1, requested/capability '0x40 (slin)'/'0x4c (ulaw|alaw|slin)' incompatible with our capability '0x0 (nothing)'. |
| 20:04.42 | pa | i am trying to send a fax using iaxmodem |
| 20:05.28 | mbowie | Jacoby6000: That was actually mentioned in #asterisk-dev yesterday... referencing http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/ |
| 20:10.32 | Jacoby6000 | mbowie: yeah, that was actually me. His solution doesn't work with a direct copy/paste in to the app_voicemail.c file, so it's taken some modification. |
| 20:11.27 | pa | by the way, why does iaxmodem get to 100% cpu as soon as i run it, without even using it? |
| 20:11.52 | mbowie | lol... I was doing well just to remember the context; let alone the poster. :-) |
| 20:12.04 | [TK]D-Fender | pa you allowed NO codecs |
| 20:12.12 | [TK]D-Fender | pa: fix your peer |
| 20:12.34 | pa | [TK]D-Fender, meaning iaxmodem or the asterisk server? |
| 20:12.55 | pa | in iaxmodem i have: codec slinear |
| 20:13.07 | [TK]D-Fender | sip.conf <--------- |
| 20:13.14 | [TK]D-Fender | iax,conf rather |
| 20:13.27 | pa | in asterisk iax.conf : allow slin disallow all |
| 20:13.33 | Qwell | http://pastebin.com/8EmBZsdR |
| 20:13.37 | Qwell | looks around |
| 20:13.48 | Qwell | mbowie, Jacoby6000: ^^^ |
| 20:14.34 | Qwell | oh, you want to contribute it |
| 20:14.34 | [TK]D-Fender | pa: pastebin it |
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| 20:15.19 | pa | [TK]D-Fender, ok. btw i still have 100% cpu as soon as i run iaxmodem |
| 20:15.36 | [TK]D-Fender | pa: 1 problem at a time. |
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| 20:17.31 | pa | [TK]D-Fender, http://codepad.org/sK4HNSjQ |
| 20:18.06 | [TK]D-Fender | show the full call |
| 20:18.18 | pa | full what? |
| 20:18.20 | pa | file? |
| 20:18.35 | [TK]D-Fender | * CLI with IAX2 debug |
| 20:18.59 | pa | ah |
| 20:19.32 | pa | how do i turn on the debug? |
| 20:19.41 | [TK]D-Fender | iax set debug on |
| 20:19.46 | [TK]D-Fender | or iax2 I forget which |
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| 20:21.16 | pa | [TK]D-Fender, if i run iaxmodem, this is what i get: http://codepad.org/oNCS2Zo5 |
| 20:21.20 | pa | and i get 100%cpu |
| 20:21.24 | pa | without doing anything |
| 20:21.42 | [TK]D-Fender | I don't see that capabilities warning... |
| 20:21.52 | pa | that appear when i try to call |
| 20:21.55 | pa | moment |
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| 20:29.05 | [TK]D-Fender | Times up over hear, BBL |
| 20:30.21 | pa | after call, i get: http://codepad.org/8xXhEh5t |
| 20:30.26 | pa | lots of blabla |
| 20:30.30 | pa | donno what's wrong |
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| 20:37.33 | pa | i suspect the problem might be efax |
| 20:37.42 | pa | what else can i use that is not hylafax? |
| 20:37.58 | pa | (i dont really want to set up the whole beast. just need a fax client to send faxes) |
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| 21:09.39 | Jacoby6000 | Qwell: What was that paste? I missed it. |
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| 22:09.00 | TazzNZ | hey all |
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