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| 01:42.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.0 (2016/11/23), 11.25.0 (2016/11/23), Standard: 14.2.0 (2016/11/23); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 04:46.02 | everestt | Which pri and dahdi version do I need for linux kernel 4.4.0-47-generic and asterisk-certified-13.8-cert3? |
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| 06:08.51 | FuriousGeorge | hey all. i used to be able to set up a sip client behind nat to register with an * server behind another nat, with no problems. i assume i still could. for some reason i have a hell of a time setting up a client behind nat with a server with public IP |
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| 06:10.23 | FuriousGeorge | everything looks as expected if i sip show peers/registry, but when e.g. dial a number from the remote client behind nat, the server responds, and i see dial(sip/provider/9095551212), but it just times out |
| 06:11.22 | FuriousGeorge | the endpoint never rings. im not sure if this is a nat problem, or a problem with my provider, which is closed for 36 hours due to the holiday, so i can't verify.... |
| 06:12.40 | FuriousGeorge | but i'm leaning toward it being a problem with my provider, because if the signalling is getting from client behind nat to asterisk on public IP -- as evidenced by the * console showing me it is trying to dial out via trunk peer -- then there is no reason the signalling would not get from * to provider |
| 06:13.56 | FuriousGeorge | i would expect the call to go with no media, i would not expect nothing to happen. |
| 06:26.38 | FuriousGeorge | erm, disregard, it was a 3rd factor... some problem with my dialplan |
| 06:45.00 | FuriousGeorge | so let's say i want to use "reinvite" in this scenario. since my sip clients are behind nat, it wouldn't be possible. however, if i set up a sip proxy on the gateway, then that should get around the issue, no? |
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| 07:13.12 | FuriousGeorge | if i wanted to do reinvite with the clients behind nat, is the best solution a sip proxy on the gateway's wan interface? that would give both endpoints a resolvable public IP somewhere along the line, so all should be fine, no? |
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| 10:53.42 | rah | is there a way to define the ports used for SRTP connections, like there are for RTP connections in rtp.conf? |
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| 11:09.56 | file | it does not use a different port range, uses the same as RTP |
| 11:11.12 | rah | file: so setting the RTP port range in rtp.conf will also modify the port range used for SRTP? |
| 11:11.19 | file | yes. |
| 11:11.26 | rah | I see, thanks |
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| 13:36.57 | rah | I'm trying to get MusicOnHold to work |
| 13:37.42 | rah | I've created a .sln file using "avconv -i The_Solid_Doctor_-_Punctus_Contra_Punctum.wav -ar 8000 -ac 1 -acodec pcm_s16le -f s16le The_Solid_Doctor_-_Punctus_Contra_Punctum2.sln" and I've previously done the same with sox, as described in the O'Reilly book |
| 13:38.13 | rah | unfortunately, when asterisk plays the file, all I hear is noise |
| 13:38.18 | rah | this is asterisk 13 |
| 13:38.29 | rah | am I doing something wrong? |
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| 14:34.02 | Kobaz | happy happy |
| 14:34.31 | Kobaz | i'm thankful for res_pgsql not connecting to an ssl enabled postgres |
| 14:34.36 | Kobaz | anyone run into that? |
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| 14:35.52 | Kobaz | res_config_pgsql rather |
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| 15:06.05 | ZX81 | Hi all, can I read Asterisk events via ARI without having an application in stasis? |
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| 16:14.30 | danielyk | Is there any mirror of downloads.asterisk.org in Europe? |
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| 19:57.40 | MintBerrryCruNCH | Hi guys, im calling a macro from another macro (via agi script), but as soon as the second nested macro is called, the first macro abruptly returns. Is this something expected? Nested macros normally work, but does calling from agi produce unexpected results? |
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