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| 02:12.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 02:16.37 | lembron | created trunk and outboundRoute - "all circuts are bussy now" - meanwhile i dont even see it "try" or anything looking at tcpdump on the line :/ |
| 02:22.41 | lembron | duh, didnt add the trunk in there |
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| 02:56.15 | igcewieling | mirror2.freepbx.org is unreachable from at least 2 ISPs. |
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| 12:42.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 17:52.57 | igcewieling | Using Sangoma Linux release 7.5.1805 (Core). Any idea what might cause this? [2020-06-04 13:51:22] WARNING[6638]: loader.c:581 load_dlopen: Error loading module 'chan_dahdi.so': /usr/lib64/asterisk/modules/chan_dahdi.so: undefined symbol: ast_variable_list_replace |
| 17:54.57 | file | what version of Asterisk? |
| 17:55.34 | igcewieling | using asterisk-version-switch to switch to the same version I'm already running appears to have resolved the issue. |
| 17:55.49 | igcewieling | file: 13.32.0, |
| 17:56.00 | file | I'd only expect the message to occur if Asterisk was one version and chan_dahdi was from another |
| 17:56.32 | igcewieling | file: *nod* That is what I thought too, but since it is supposed to all be managed by packages, I was a bit surprised. |
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| 18:53.06 | igcewieling | Is there any way to get the Reason header from a hangup in the dialplan Reason: SIP;cause=503;text="no-fraud-detected" |
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| 19:09.17 | rexwin | anybody know what is restcomm and how it works? |
| 19:10.39 | Samot | igcewieling: There's no way to pull BYE/CANCEL |
| 19:10.49 | Samot | You can get details from the INVITE |
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| 19:33.16 | anonymouz666 | hi all. someone know why asterisk escapes ":" with %3a when dialing ${SIPREFERTOHDR} ? |
| 19:33.44 | anonymouz666 | or encode |
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| 19:37.15 | anonymouz666 | when dialing pjsip/${siprefertohdr}/endpoint works when dialing pjsip/${siprefertohdr}@endpoint does not. |
| 19:46.09 | igcewieling | What is what I figured. Too bad PJSIP doesn't have a use_q850_reason option like chan_sip. |
| 19:48.38 | file | it does, it behaves the same way as chan_sip |
| 19:49.13 | file | the res_pjsip_rfc3326 module implements it, but neither allow access to the text portion |
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| 19:50.17 | file | and both only support the Q.850 type |
| 19:54.46 | Samot | Broken Record: Asterisk is not a softswitch. |
| 19:56.24 | anonymouz666 | Samot: that feature is useful |
| 19:56.44 | Samot | What feature? |
| 19:57.02 | anonymouz666 | q850_reason |
| 19:57.10 | Samot | Sure. |
| 19:57.45 | Samot | You know what can react to that BYE/CANCEL and get the Reason header(s)? Softswitch |
| 19:57.53 | Samot | Proxy/SBC |
| 19:59.53 | anonymouz666 | PJSIP/${SIPREFERTOHDR}/endpoint@IP.proxy) should work? the ideia is build the request with RURI that is in the var, but send to Proxy IP |
| 20:00.07 | anonymouz666 | it complains that does not find the endpoint |
| 20:02.22 | Samot | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 20:02.28 | Samot | Because that's not valid. |
| 20:03.04 | Samot | PJSIP/<endpoint>/<sipuri> |
| 20:09.10 | anonymouz666 | oh sorry I meant PJSIP/endpoint/${SIPREFERTOHDR}@IP.proxy |
| 20:09.14 | anonymouz666 | that should work |
| 20:09.31 | file | depends on what SIPREFERTOHDR contains... |
| 20:09.53 | anonymouz666 | that's right |
| 20:09.57 | anonymouz666 | it's a complete RURI |
| 20:10.04 | anonymouz666 | that is causing me trouble |
| 20:10.08 | file | then it would not work |
| 20:10.24 | file | the outbound proxy would need to be configured on the endpoint itself |
| 20:10.57 | anonymouz666 | i confirmed that. |
| 20:11.11 | anonymouz666 | oh |
| 20:11.12 | anonymouz666 | great |
| 20:11.45 | anonymouz666 | I tried SIP/${SIPREFERTOHDR}@endpoint, it dials but |
| 20:11.56 | anonymouz666 | INVITE sip:sip%3asip.pstnhub.microsoft.com%3a5061;transport=tls;x-m=8%3aorgid%3a209e507d-05f2-416a-b7b7-073ef794d50a@192.168.200.1:5083 SIP/2.0 |
| 20:12.13 | anonymouz666 | all the : were converted to %3a |
| 20:12.22 | file | you'd want PJSIP/endpoint/${SIPREFERTOHDR} |
| 20:12.35 | anonymouz666 | with outboundproxy |
| 20:12.40 | anonymouz666 | like you said |
| 20:12.43 | anonymouz666 | gonna try that |
| 20:25.44 | anonymouz666 | file: it is working now |
| 20:26.34 | anonymouz666 | thanks |
| 20:26.56 | anonymouz666 | strange why this dials is reached in "s" extension. |
| 20:27.17 | anonymouz666 | don't know why, put there is the Dial |
| 20:30.21 | anonymouz666 | I can tell to everybody in this channel that is almost impossible to use a B2BUA standlone to talk to MS Direct Routing |
| 20:30.38 | anonymouz666 | better stick with asterisk + kamailio or opensips |
| 20:30.52 | anonymouz666 | then you can custom the headers the way MS wants |
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| 23:13.33 | DannyA | hey all. is there a way to do pattern matching for a variable number of digits? like, a match for any extension that starts with 4 and has anywhere from 3 to 5 digits after that? |
| 23:17.31 | Samot | Yup |
| 23:17.55 | Samot | It's in the wiki |
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