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| 03:36.08 | volga629 | What codec recommended through IAX trunk between two boxes ? |
| 03:36.48 | volga629 | I am getting -- Got SIP INFO response 415 "Unsupported Media Type" back from host '10.34.22.33:5060' |
| 03:36.51 | volga629 | [2013-08-12 05:33:22] WARNING[20074][C-0000000a]: res_rtp_asterisk.c:2640 ast_rtp_write: Don't know how to send format unknown packets with RTP |
| 03:36.58 | volga629 | this when G729 in use |
| 03:37.33 | volga629 | on another end bria for asterisk soft client |
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| 04:52.16 | igcewieling | stop using g729 |
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| 05:01.52 | volga629 | yes, removed |
| 05:02.56 | volga629 | yes, switched to ulaw look like iax2 not working very well with g729 |
| 05:03.11 | volga629 | and is iax godd with h264 ? |
| 05:03.14 | volga629 | good |
| 05:04.07 | volga629 | the problem that remove office is on limited internet speed |
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| 06:33.31 | _omer | Hello, |
| 06:33.58 | _omer | Is there a way to hangup all calls from ConfBridge once Admin or Marked user leaves the conference? |
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| 07:54.27 | ik_5 | hello |
| 07:56.24 | ik_5 | asterisk 11.5.0, changemonitor fails with error of permission denied (no matter what user asterisk is running, and that the user do have write permissions to the folder), what else should I be looking for ? |
| 07:57.22 | ChannelZ | What are you calling it with? |
| 07:57.31 | ik_5 | manager |
| 07:57.45 | ChannelZ | No I mean arguments |
| 07:58.20 | ik_5 | I'll use gist, a sec |
| 07:59.54 | ik_5 | https://gist.github.com/ik5/794dc244332a850b7172 I'm using adhearsion as the framework (that does the actual code), here are what I'm doing and getting back |
| 08:11.28 | ik_5 | ChannelZ, does it helps ? |
| 08:16.19 | ChannelZ | well it's interesting that it's only classified as a warning. |
| 08:22.00 | ChannelZ | But, the 'permission denied' is pretty damning, seems straightforward enough.. are you sure the user asterisk runs as has access to the entire path? |
| 08:25.10 | ik_5 | yes, and i was able to create a file as the user |
| 08:25.30 | ik_5 | i even tried to execute asterisk as root, same result |
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| 08:29.01 | ChannelZ | I've no other explanation then |
| 08:29.36 | ik_5 | thanks |
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| 08:37.20 | tokozedg | Anyone can suggest class 5 softswitch, not only opensource. I checked Kazoo and sipwise, but I none is actully ready to go. |
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| 08:44.30 | angryuser | Good day, i am searching a gateway for ~46 FXS, any suggestions ? (i can't use the internal cards) |
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| 12:01.31 | *** join/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 12:01.38 | StaRetji | Howdy folks |
| 12:03.00 | StaRetji | need little tip, when placing .call in spool. Is there a way to do Application: (silence of 4 seconds). I need this option to activate sim cards, but I need silence. Any help is highly appreciated. Thx |
| 12:03.30 | StaRetji | I tried with Application:Hangup |
| 12:04.02 | tuxx- | answer(), wait(4) ? |
| 12:04.03 | StaRetji | and with Playback: some file (this blocks sim card on gateways for some reason. So would like to try with pure silence |
| 12:04.09 | StaRetji | in .call? |
| 12:04.12 | tuxx- | hmye |
| 12:04.27 | tuxx- | make an extension in extension.conf that does that |
| 12:04.32 | tuxx- | and refer yuor call file to that extension |
| 12:04.54 | StaRetji | got you |
| 12:04.57 | StaRetji | thx tux |
| 12:05.11 | tuxx- | np |
| 12:06.08 | kaldemar | StaRetji: Playback(silence/4) |
| 12:06.31 | kaldemar | cd |
| 12:06.38 | StaRetji | thx kaldemar, so much |
| 12:06.41 | StaRetji | I will try at once |
| 12:21.52 | StaRetji | hm, it made a call |
| 12:21.58 | StaRetji | but length was 0 seconds |
| 12:22.17 | StaRetji | Application: Playback(silence/4) |
| 12:23.30 | igcewieling | Do it in the dialplan using Local/ |
| 12:23.45 | igcewieling | then you'll at least be able to see what might be going on. |
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| 12:24.03 | igcewieling | What sort of port are you dialing out of? |
| 12:24.24 | StaRetji | gsm gateway port |
| 12:24.45 | StaRetji | if I call with sip phone, I can call and hangup after few seconds |
| 12:24.49 | igcewieling | not an analog FXO port. |
| 12:24.51 | StaRetji | and call shows 4 seconds |
| 12:24.56 | StaRetji | no, it is sip trunk |
| 12:25.09 | StaRetji | then, if I do .call |
| 12:25.15 | StaRetji | it makes call, but 0 seconds |
| 12:25.21 | StaRetji | I will try what you said |
| 12:25.28 | igcewieling | Use Local/ so you can see what might be going on |
| 12:25.45 | StaRetji | in .call? |
| 12:26.00 | igcewieling | yes. This way BOTH legs of the call are in the dialplan |
| 12:26.15 | StaRetji | Application: Local(silence/4) |
| 12:26.20 | igcewieling | no. |
| 12:26.32 | StaRetji | oh, sorry mate |
| 12:26.34 | igcewieling | You would not use the Application format. |
| 12:27.46 | StaRetji | Channel: SIP/75524323@trunk1 Callerid: 1234 WaitTime: 30 Application: Playback(silence/7) |
| 12:27.55 | StaRetji | this is what I tried, gave 0 seconds |
| 12:28.07 | StaRetji | I got it all wrong? |
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| 12:30.44 | igcewieling | google will give you examples of using .call files with the Local/ channel |
| 12:31.15 | StaRetji | okay igcewieling, thanks |
| 12:31.18 | igcewieling | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 12:31.22 | igcewieling | as an example. |
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| 12:31.28 | igcewieling | you would NOT use Application: |
| 12:34.02 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 12:34.28 | igcewieling | Channel: Local/75524323@out_context and use Extension Context and Priority so that BOTH legs of the call are in the dialplan and you can debug it. |
| 12:35.03 | StaRetji | got it |
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| 15:23.04 | *** join/#asterisk monsterco (461b5fde@gateway/web/freenode/ip.70.27.95.222) |
| 15:23.36 | monsterco | Hi everyone - I have Dahdi + Sangoma timing device installed on Proxmox mother node - now, how can I give access to same timing device on OpenVZ container which has Asterisk installed? |
| 15:24.13 | igcewieling | monsterco: sounds like a question for your VM, not for Asterisk\ |
| 15:25.04 | igcewieling | I assumed you would need to do is install dahdi and sangoma in the VM and have the VM give the drivers access to the hardware |
| 15:25.06 | monsterco | igcewieling - you are right - I was hoping someone here has the experience. Do you know if I have to install DAHDI on vm as well or is there a symlink or something else I have to create to give it access? |
| 15:25.22 | igcewieling | monsterco: only crazy people try using DAHDI inside a VM |
| 15:25.30 | Greenlight | +1 |
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| 15:25.51 | monsterco | igcewieling - there are many people reporting success so why not |
| 15:26.18 | igcewieling | monsterco: there are many people who successfully run Asterisk in a VM. there are not many people who use Asterisk with DAHDI in a VM |
| 15:26.52 | Greenlight | I'd be more inclined to use timerfd for your timing on the VMs than DAHDI |
| 15:27.30 | monsterco | I have a Sangoma UT50 installed specifically for this and I know of production systems that work just fine with that. I am not sure if you are speaking of experience you have or just saying in general and without the use of a hardware timing device |
| 15:27.39 | igcewieling | You can't have multiple VMs accessing the DAHDI hardware anyway. |
| 15:28.02 | [TK]D-Fender | indeed |
| 15:28.07 | monsterco | I am open to using anything (DAHDI) or not - but I have the Sangoma UT50 installed and I know I have to make it available to VM somehow |
| 15:28.21 | Greenlight | UT50 isn't that great - my kernel has better timing. I speak from experience. Mind went in the bin. |
| 15:28.24 | Greenlight | *mine |
| 15:28.34 | igcewieling | monsterco: there is nothing special about the UT50 compared to any other device for which there are no VM support. |
| 15:28.40 | monsterco | igcewieling - No, I heard and seen just the opposite. That's the beautify of the Sangoma device. It allows multiple VMs to use it at the same time |
| 15:28.44 | [TK]D-Fender | monsterco: that device *IS* DAHDI hardware. |
| 15:29.05 | igcewieling | monsterco: I wish you the best of luck. |
| 15:29.11 | monsterco | Greenlight - so what do you use now? |
| 15:29.17 | Greenlight | The kernel timing |
| 15:29.29 | [TK]D-Fender | it will not get shared. This will not end well |
| 15:29.44 | igcewieling | I suggest you contact Sangoma for that magical unicorn ability. Asterisk doesn't support it. |
| 15:29.51 | monsterco | Is that available to VM by default or something I should do to make it available? |
| 15:30.09 | igcewieling | kernel timing has been available in Asterisk since v1.6 |
| 15:30.16 | Greenlight | It's very good these days |
| 15:30.35 | Greenlight | And i'd bet works a hell of a lot better inside a VM than DAHDI |
| 15:31.01 | monsterco | So, how can I test the timing accuracy without DAHDI loaded? I mean dahdi_test won't work |
| 15:31.07 | Greenlight | Just don't use crappy hardware, and use a modern os/kernel |
| 15:31.15 | Greenlight | asterisk -rx "timing test" |
| 15:31.19 | Greenlight | I guess |
| 15:31.56 | Greenlight | Or put a bunch of calls into a conference |
| 15:32.19 | monsterco | What does this mean? |
| 15:32.20 | monsterco | Attempting to test a timer with 50 ticks per second. Using the 'pthread' timing module for this test. It has been 1002 milliseconds, and we got 50 timer ticks |
| 15:32.31 | Greenlight | hmmm pthread |
| 15:32.36 | monsterco | Is my VM using the kernel timing? |
| 15:32.38 | Greenlight | You should have timerfd |
| 15:32.44 | Greenlight | What OS are you on ? |
| 15:32.58 | monsterco | CentOS and I used asterisk-current by yum intall |
| 15:33.01 | monsterco | CentOS 5.9 |
| 15:33.03 | Greenlight | yum... |
| 15:33.04 | Greenlight | eww |
| 15:33.11 | igcewieling | Wow |
| 15:33.32 | Greenlight | You might wanna jump to CentOS6 if you can, I think it has support for timerfd |
| 15:33.34 | igcewieling | I'd like "a list of thing you should not do with Asterisk for $500, Alex!" |
| 15:33.41 | Greenlight | lol |
| 15:34.09 | monsterco | timerfd is not supported in 5.9? |
| 15:34.19 | Greenlight | CentOS is based off Redhat and use older "stable" kernel, and sometimes misses support for these things |
| 15:34.29 | Greenlight | Yes, iirc |
| 15:34.44 | Greenlight | And, install Asterisk from source not from yum |
| 15:34.48 | monsterco | so what advantage is there to move to 6.0 in terms of timing? |
| 15:34.58 | igcewieling | I thought the requirements for the various timers were documented in the Asterisk book |
| 15:35.05 | igcewieling | ~book |
| 15:35.05 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:35.08 | Greenlight | Yes, they are |
| 15:35.19 | igcewieling | Greenlight: we could tell him to go read the book. |
| 15:35.21 | Greenlight | Although I'm not sure it mentions kernel versions specifically |
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| 15:36.52 | monsterco | what does 50 tricks and 1001 ms really mean from my readings? |
| 15:37.03 | Greenlight | Each "tick" is 20ms |
| 15:37.17 | Greenlight | So 50/1001 is fine. |
| 15:37.54 | Greenlight | 48/1000 or say 50/1090 and you'll start to get issues with timing dependant applications. |
| 15:38.29 | Greenlight | But, yea, you should read that section of the book for certain. |
| 15:40.03 | Greenlight | And rememeber, there's no point testing now when it's not in use - you want to test while "under load", as thats when things will slip. |
| 15:41.03 | *** join/#asterisk ibercom (4de0778b@gateway/web/freenode/ip.77.224.119.139) |
| 15:41.32 | monsterco | so, what does your server read under load? |
| 15:42.17 | Greenlight | Hmm...two secs... |
| 15:43.06 | igcewieling | Using the 'DAHDI' timing module for this test. |
| 15:43.06 | igcewieling | It has been 1016 milliseconds, and we got 51 timer ticks |
| 15:43.13 | Greenlight | Using the 'timerfd' timing module for this test. |
| 15:43.13 | Greenlight | It has been 1000 milliseconds, and we got 50 timer ticks |
| 15:43.21 | Greenlight | 390 active channels |
| 15:43.21 | Greenlight | 195 active calls |
| 15:43.21 | Greenlight | 7951306 calls processed |
| 15:43.36 | Greenlight | (That one is a VM) |
| 15:45.04 | malcolmd | and that's excellent :) |
| 15:46.35 | monsterco | k, so your test show better with timerfd vs dahdi |
| 15:46.53 | Greenlight | Both are acceptable and will not give any issues |
| 15:46.55 | monsterco | but timerfd doesn't use an FXO or a digital card? |
| 15:47.01 | Greenlight | Exactly |
| 15:47.24 | Greenlight | Beleive it or not, but servers can tell the time without addin cards :) |
| 15:47.31 | monsterco | so, both method can be used on the fly? or did you disable one and then enabled the other? or just run tests using and the other? |
| 15:47.48 | Greenlight | The book tells you :) |
| 15:47.55 | Greenlight | There's a priority |
| 15:48.18 | Greenlight | It'll use the "best" on available. Although (also in the book) it's recommended you disable the others. |
| 15:48.24 | monsterco | well, yeah. I know it's a software issue in older version but I bought this device 2 yrs ago and wanted to use it one day |
| 15:48.59 | Greenlight | Do you have a door that needs jammed open ? |
| 15:49.04 | monsterco | but since it's useless now, I may list it on Ebay - I know I wont' get the chance |
| 15:49.38 | monsterco | I have to look into timerfd and probably I need the mothernode to allow vm to use it... |
| 15:50.29 | Greenlight | I use ESXi, and it "just works" TM |
| 15:50.37 | monsterco | door jammer would be best use |
| 15:50.42 | Greenlight | :) |
| 15:51.01 | Greenlight | Or a "dust protector" for a USB socket... |
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| 18:31.33 | igcewieling | Wow! I found a system still running zaptel |
| 18:32.57 | malcolmd | upgrade it with reckless abandon! don't make backups of anything! |
| 18:35.22 | igcewieling | malcolmd: There is a different way? |
| 18:35.29 | malcolmd | hah |
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| 18:54.07 | bananapie | I think that asterisk is doing DTMF detection on the audio on my SIP channel despite the fact that I have it set to rfc2833. Is this possible? |
| 18:54.27 | ChannelZ | Yes because dtmf isn't in the audio with rfc2833. zing! |
| 18:55.03 | bananapie | I mean, can Asterisk detect a DTMF on my SIP channel if it is set to rfc2833 and no rfc2833 event is received by asterisk ? |
| 18:55.21 | [TK]D-Fender | no |
| 18:55.36 | bananapie | ok. Thanks |
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| 19:09.59 | raidghost | Trying to install asterisk 11.5.0 on debian 7.0 and for some reason it needs sqlite3, Why dont just use mysqlserver? and ive getting some annoying errors. |
| 19:10.02 | raidghost | make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. |
| 19:10.05 | raidghost | make: *** [main] Error 2 |
| 19:10.42 | [TK]D-Fender | ASTDB is in sqlite since v10 |
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| 19:12.14 | raidghost | Ok. then i guess sqlite is what i need to keep my eyes on from now. |
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| 19:36.20 | bananapie | is rfcmode=2833 the same as rfcmode = rfc2833? Does asterisk care about spaces? |
| 19:36.45 | [TK]D-Fender | dtmfmod, not "rfcmode" |
| 19:36.54 | [TK]D-Fender | dtmfmode* |
| 19:37.08 | [TK]D-Fender | get your parameter names right... |
| 19:38.26 | bananapie | sorry. dtmfmode=rfc2833 and dtmfmode = rfc2833, both are the same ? |
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| 19:39.18 | [TK]D-Fender | generally should be fine |
| 19:39.33 | bananapie | ok |
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| 19:50.13 | ChannelZ | Anyone need to build something with these: http://www.supermicro.com/products/chassis/4u/748/sc748tq-r1000.cfm |
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| 19:57.38 | drmessano | ChannelZ, those look pretty awesome |
| 19:58.06 | drmessano | I wonder if they ship with divorce papers, filled out with the buyers name and county/state of residence |
| 19:58.15 | drmessano | Because if I got one, my wife would find that handy |
| 19:58.49 | igcewieling | Your first mistake was getting married. |
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| 20:05.14 | navaismo | stupid question: the -C option is required when I generate a certificate to use with asterisk? If so can i use any string for domain, based on i dont know the IP or domain address |
| 20:07.00 | bananapie | how can I see if dsp.c dtmf detector is running on a given channel? |
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| 20:30.48 | xpot | Question: how do I fix this? "Dropping incompatible voice frame on SIP/100056-00000001 of format ulaw since our native format has changed to (alaw)" Asterisk 11.5 |
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| 20:33.36 | navaismo | set your preferred codec as alaw instead ulaw |
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| 20:36.58 | J4nus | hi i would like to have some info about the codec negotiation |
| 20:37.42 | J4nus | i have transcoding issue between 2 peers |
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| 20:38.19 | J4nus | is it possible to force asterisk to avoid transcoding |
| 20:39.16 | navaismo | using same codecs i suppose it will not transcode |
| 20:42.23 | Chainsaw | J4nus: There is no intelligent decision maker in Asterisk for this. |
| 20:42.39 | Chainsaw | J4nus: Reduce the set of codecs on both peers down until you get a codec path you approve of. |
| 20:43.05 | Chainsaw | J4nus: And I'm afraid navaismo has a point. The decision making can be that boneheaded that you end up with just a single codec on each peer. |
| 20:48.36 | J4nus | yes but i'm using voip over wifi/3G |
| 20:48.45 | J4nus | so i want to use light codecs |
| 20:48.58 | J4nus | i have good result with ilbc and gsm codecs |
| 20:49.08 | J4nus | but in some cases ilbc is not supported |
| 20:49.12 | J4nus | no it use gsm |
| 20:49.19 | J4nus | but with the other peers ilbc is supported |
| 20:49.24 | J4nus | so i have gsm to ilbc |
| 20:49.48 | J4nus | if it uses gsm, then i would like he chose gsm on the other side also |
| 20:49.52 | J4nus | and not ilbc |
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| 20:50.54 | [TK]D-Fender | just set it per peer |
| 20:51.12 | [TK]D-Fender | it shoulnd't require guessing |
| 20:51.24 | navaismo | is there a voip providers with tls/srtp support? |
| 20:52.11 | J4nus | yes but i do disallow=all; allow=ilbc,gsm on both peers |
| 20:52.16 | J4nus | then one peer has only gsm |
| 20:52.30 | J4nus | the other peer have both (gsm & ilbc) |
| 20:52.36 | J4nus | and i got transcoding.. |
| 20:52.51 | J4nus | why not selecting gsm because both support it.. |
| 20:54.32 | navaismo | you can check the sdp transaction to see it |
| 20:54.57 | J4nus | i will look, is it via the debug cmd ? |
| 20:55.40 | navaismo | yes sip set debug on |
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| 23:07.44 | pensmit | Does anyone know how to enable ssh or telnet on a snom phone? |
| 23:07.58 | pensmit | I can't believe you can't ping from a 720. |
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| 23:39.58 | mitchrodrigues | Im having the oddest issues, im getting unauthorized using insecure=port ;/ |
| 23:40.03 | tonsofpcs | can you set up a group of extensions (or perhaps even a single extension) that has its own MOH? Can you make an extension automatically put incoming calls on hold? |
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| 23:49.52 | mitchrodrigues | anyone got any ideas? |
| 23:51.15 | ChannelZ | are you expecting to not have authorization at all, or..? |
| 23:52.07 | ChannelZ | tonsofpcs: you can specify the MOH context in sip.conf for peers individually if you like. |
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| 23:52.21 | paulc | tonsofpcs: Use the MusicOnHold() application to just play music for people |
| 23:52.29 | tonsofpcs | awesome, thanks. |
| 23:53.20 | ChannelZ | except you have no way to get them back off hold (it's not really hold) |
| 23:53.29 | ChannelZ | probably more useful to park them |
| 23:53.50 | mitchrodrigues | Well ya |
| 23:53.58 | mitchrodrigues | we use opensips and firewalls to controll that stuff |
| 23:54.34 | ChannelZ | mitchrodrigues: well if you're asking why insecure=port fails because Asterisk requests auth but doesn't get it, you want insecure=invite |
| 23:54.50 | mitchrodrigues | http://pastebin.com/8SZtW663 |
| 23:54.54 | ChannelZ | Or if you ARE expecting it to auth correctly, then your peer isn't matching. |
| 23:55.09 | mitchrodrigues | thats what i got |
| 23:55.11 | mitchrodrigues | lol |
| 23:55.41 | ChannelZ | you'll have to look at the INVITE and see if it's coming from "openser5" because otherwise it won't match anything |
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| 23:56.04 | mitchrodrigues | the ip looks correct in debug mode |
| 23:57.00 | ChannelZ | pastebin a sip debug of the attempted call. It's probably giving you a clue in the middle about peer matching |
| 23:58.02 | mitchrodrigues | two seconds |
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| 23:59.36 | mitchrodrigues | http://pastebin.com/nVBZNDnY |