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| 02:27.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:17.05 | pchero | Hi guys. I'm looking for the voicemail welcome messages wav files from all over the world and any languages. Does anyone who knows this? :) |
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| 20:21.31 | P-NuT | Hi there. |
| 20:22.20 | P-NuT | I've installed asterisk 15 and am using it with sipgate. I have SIP signalling working and can take a call, but I just have no audio at all. |
| 20:22.25 | P-NuT | Any ideas? |
| 20:23.38 | robmal | Does it work on the same server with older versions? |
| 20:24.34 | P-NuT | I've not got an old server, this is new installation. |
| 20:26.12 | robmal | Ok. Do you have any working asterisk running right now? |
| 20:26.23 | robmal | Besides this server. |
| 20:26.47 | P-NuT | No I dont |
| 20:26.55 | robmal | Understood. |
| 20:27.02 | robmal | Give me a sec. |
| 20:27.03 | robmal | ~book |
| 20:27.04 | 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 |
| 20:27.08 | robmal | There it is. |
| 20:27.12 | robmal | Read this. |
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| 20:27.29 | robmal | You'll understand how the water flows. |
| 20:27.49 | robmal | Your issue is with RTP and/or NAT. Maybe SIP ALG. |
| 20:28.14 | P-NuT | Errrr, ok, but this does not cover Josip. |
| 20:28.20 | kuon | Hello. I just made an install of asterisk for my home PBX. I am using an external proxy for PSTN calls. I can call internal extension and make outgoing calls. But incoming calls are answered by a message "congratulation for setting up asterisk...". |
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| 20:28.38 | kuon | I have exten => s,1,Dial(SIP/1000) in my extension.conf |
| 20:29.36 | P-NuT | Sorry, does not cover pjsip |
| 20:29.38 | robmal | P-NuT: It does cover the basics, and your issue is so basic it's not worth investigating for the n-th time :-) It's one i've mentioned above for sure. sip set debug all the way, know how to read INVITE, figure out what's missing. |
| 20:30.21 | kuon | Could it be the incoming context that is not set properly? |
| 20:30.27 | robmal | kuon: Your trunk is in the wrong context. |
| 20:32.21 | kuon | robmal: How can I set the context of the peer? This is my sip.conf https://gist.github.com/kuon/3b03f4cfc08497c7a15a2b7109423194 |
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| 20:32.52 | kuon | My context should be INTERNO for calls comming from peoplefone. |
| 20:33.27 | robmal | So, maybe just read the docs? The variable seriously is called context ;-) |
| 20:34.11 | robmal | Just sip reload afterwards and you'll be fine. |
| 20:34.25 | kuon | I just figured out that the context is set up only in the template base-voip and not the peer |
| 20:34.29 | kuon | My bad. |
| 20:34.41 | robmal | :-) |
| 20:35.13 | kuon | Working fine. |
| 20:35.35 | robmal | Awesome. |
| 20:36.02 | kuon | Thanks for the help. |
| 20:36.27 | robmal | Glad i could :-) |
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