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| 04:14.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:41.04 | jonasschneider | newbie question: i want to set up a voice connection between WebRTC browser clients and traditional telephones with asterisk in between. |
| 08:41.25 | jonasschneider | what is the kind of service that provides me the connection between IP and the voice network? would that be "voice termination"? |
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| 10:50.31 | WIMPy | ~itsp |
| 10:50.31 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 10:50.43 | WIMPy | Or a piece of hardware to connect to a line. |
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| 11:13.09 | dan_j | Asterisk keeps stalling for about 30 seconds and doesn't respond to SIP packets. Happens about once a day at random times. I'm thinking of switching from realtime to a static sip.conf file and adding credentials when needed, performing a 'sip reload'. |
| 11:13.32 | dan_j | Are there any disadvantages with a static sip.conf file compared to realtime? |
| 11:13.45 | WIMPy | DNS issues? |
| 11:13.54 | dan_j | I use the IP |
| 11:14.04 | dan_j | I think.... 1sec |
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| 11:15.15 | dan_j | ODBC has the hostname but the hosts file contains the IP |
| 11:15.46 | WIMPy | Not the database connection. Anywhere in Asterisk. |
| 11:16.29 | dan_j | As far as I'm aware, the only thing that causes asterisk to stop responding to SIP, is when it can't retrieve data from the realtime db. |
| 11:17.00 | WIMPy | Or when it's busy doing somethign else. |
| 11:17.20 | dan_j | I've enabled debug to see if I can spot anything next time it happens. |
| 11:17.22 | WIMPy | Like waiting for DNS. Are you using dnsmgr? |
| 11:18.15 | dan_j | enable=yes |
| 11:19.00 | WIMPy | You probably need full debug to spot what's going on. But you need a lot of space for that. |
| 11:19.24 | dan_j | The logs will rotate and delete if they get too big, so i can do that. |
| 11:19.41 | dan_j | I've set debug to 1 for now. |
| 11:20.25 | dan_j | However, from a system resources point of view, I dont think I'm getting much benefit from sip realtime. |
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| 11:22.26 | dan_j | When I perform 'sip reload' with realtime, all the registrations are lost. Does the same happen with static files? |
| 11:25.14 | WIMPy | No |
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| 11:31.24 | dan_j | Ok. Thanks for that. |
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| 18:13.43 | atotclic | <PROTECTED> |
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| 18:22.28 | atotclic | help my config aix.conf not audio |
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| 18:27.19 | WIMPy | You use Asterisk on AIX? |
| 18:28.15 | atotclic | I am configuring the Asterisk PBX to make connections with iax2, but I have problems with the audio in one side |
| 18:29.22 | WIMPy | That's almost certainly happening somewhere else. |
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| 18:37.03 | atotclic | WIMPy: Sorry, iax2 |
| 18:37.36 | WIMPy | Yes, but that's still not your issue. |
| 18:38.18 | atotclic | WIMPy: Of this form only I have to open a port in the router |
| 18:38.53 | WIMPy | Was that a question? |
| 18:41.30 | atotclic | WIMPy: I have the iax.conf formed and extension.conf, the two sofphones register and can realize calls but, I do not receive the audio in one of the ends, to which it can be due? |
| 18:42.34 | WIMPy | Do these softphones ue IAX? |
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| 18:43.29 | WIMPy | And it's best to try with only one phone for a start, like calling VoceMail or an extension just playing some file. |
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| 18:45.18 | atotclic | sflphone or zoiper |
| 18:46.24 | WIMPy | And how are they talking to Asterisk? |
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| 18:53.17 | atotclic | WIMPy: WIMPy http://pastebin.com/hcCEcELv |
| 18:57.03 | atotclic | WIMPy: http://pastebin.com/HfWN8Gnn |
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| 19:02.43 | vlt | Hello. I'm running Asterisk 1.8.10.1~dfsg-1ubuntu1 on Ubuntu 12.04.5 LTS with kernel 3.2.0-86-virtual. This machine runs as Xen domU. Until recently I used the Xen hypervisor 4.0.2 on a linux 3.0 dom0 and everything worked fine. |
| 19:03.48 | vlt | Now I changed the Xen dom0 to Xen 4.4.1 and kernel 3.16, and DAHDI doesn't work anymore. |
| 19:04.25 | vlt | When I try to start a MEETME call I get "[Jun 28 20:55:24] WARNING[782]: app_meetme.c:1228 build_conf: Unable to open DAHDI pseudo device" |
| 19:04.48 | vlt | Any idea where to look for the problem? |
| 19:05.03 | WIMPy | Well, you have to recompile dahdi after you upgrade your kernel. |
| 19:07.59 | vlt | Ubuntu has a dahdi-dkms package that I have installed. Shouldn't this take care for re-compiling? Can anyone running a similar setup tell me how to trigger that? |
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| 19:08.58 | WIMPy | No idea about what Ubunto would or wouldn't do. But if it no longer works, it looks like it doesn't. |
| 19:09.05 | WIMPy | What do you still use MeetMe at all? |
| 19:12.16 | vlt | WIMPy: ERrm ... Is there something new to replace metme? |
| 19:14.36 | WIMPy | ConfBridge |
| 19:18.15 | vlt | WIMPy: Thank you. |
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| 19:21.17 | vlt | WIMPy: How can I set a PIN for ConfBridge? |
| 19:24.05 | WIMPy | You set the pin option for the user. Either in the confbridge.conf or via dialplan. |
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| 19:34.18 | vlt | WIMPy: Ok, thank you. I'll try to find the docu for ConfBridge ;-) |
| 19:40.10 | WIMPy | ~wiki |
| 19:40.41 | WIMPy | D'oh |
| 19:40.49 | WIMPy | ~asterisk-wiki |
| 19:41.05 | WIMPy | Well, wiki.asterisk.org |
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| 20:01.36 | dan_j | WIMPy: Do you recall last nights conversation about asterisk not responding to SIP messages for about 30 seconds? |
| 20:03.59 | dan_j | It just happened, but i cant see anything obvious within the debug when it's set to 1. I am currently performing a copy of some large files on the drive as the mysql server. So I'm almost certain that it's the DB not responding quickly enough for asteirsk. |
| 20:04.37 | dan_j | Is there any way to increase the timeout that asterisk gives to ODBC calls? |
| 20:06.15 | WIMPy | I have no idea. |
| 20:06.32 | WIMPy | But is it always 30 seconds? |
| 20:09.43 | dan_j | Approx. not sure the exact figure, but it's less than a minute as its monitored every 60 seconds by nagios |
| 20:09.57 | dan_j | Any idea what this is? Found it in the messages log |
| 20:09.58 | dan_j | <PROTECTED> |
| 20:15.34 | WIMPy | I don't even know that module, but according to google it's caused by the radius module. |
| 20:19.42 | dan_j | Yeh, just checked and it's not loaded. |
| 20:20.14 | dan_j | Probably not relevant to my issue |
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| 23:20.32 | tompaw | How often does AMI prompt change? |
| 23:20.46 | tompaw | (i.e. this: Asterisk Call Manager/2.7.0) |
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