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| 01:05.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.1.1, 16.15.1 (2020/12/22) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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| 03:26.41 | ckb | igcewieling: can I msg you? |
| 03:27.06 | igcewieling | no. |
| 03:28.29 | ckb | I need a tutor for how VoIP works. |
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| 08:45.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.1.1, 16.15.1 (2020/12/22) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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| 09:12.23 | ckb | If I'm looking at pjsip show contacts will it show me multiple entries for the same "extension"? |
| 10:03.00 | ckb | I figured that out. what's the easiest way for me to find active calls? |
| 10:07.08 | ckb | aka if a sip peer or pjsip contact is "in use" |
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| 10:58.10 | jkroon | core show channels concise? |
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| 13:08.35 | ruied | I'm forwarding all calls from my grandstream phone (ext 60) to an external mobile number. From ext 64 if I dial the external number it works fine, but if I dial to ext 60 (that is forwarding to the same number I can't complete the call. https://pastebin.com/eJJwCRJb |
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| 13:10.56 | ruied | Everyone is busy/congested at this time (1:0/0/1) CHANUNAVAIL |
| 13:10.57 | file | have you examined the SIP signaling to see the difference? It is entirely possible it is due to callerid, you may be passing it through which is not permitted |
| 13:11.14 | file | or something along those lines |
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| 13:17.34 | ruied | hmm... it might be that... going to check it out |
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| 14:41.16 | sibiria | are there any reasons why 486 Busy Here for an outbound call would result in channel variables REASON and HANGUPCAUSE being 0 instead of 4 or 17 (BUSY) |
| 14:41.39 | sibiria | (in the h extensions) |
| 14:41.46 | sibiria | and also, pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 14:44.55 | sibiria | handset is busy and the SIP response in the log confirms Asterisk and pjsip has a correct view of things. the call goes to the "failed" extension of the context specified in the call file, and cdr.c says "... dispo BUSY ...", but from that point on Asterisk seems to lose track of that and simply considers the call "failed" instead |
| 14:46.29 | sibiria | this is ast 16.15.0 |
| 14:46.59 | sibiria | could it have something to do with how cdr.conf is set-up, e.g. unanswered/congestion/endbeforehexten? |
| 14:47.05 | igcewieling | I use hangup handlers instead of 'h' exten. |
| 14:48.36 | igcewieling | I don't think they operate differently. Have you tried getting the variables using MASTER_CHANNEL? |
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| 14:49.25 | sibiria | no not yet |
| 14:53.23 | avb | hangup handlers are way to go indeed |
| 14:53.29 | avb | h extensions are a mess :) |
| 14:57.59 | sibiria | the handler in this case is an AGI script launched from the hangup extension, but modifying this is a different topic |
| 15:03.11 | igcewieling | try master_channel |
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| 15:04.29 | igcewieling | Also, running DunpChan in your AGI might provide useful info. |
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| 17:43.29 | igcewieling | Do they REALLY think this will work? "From: <sip:|'-'|@37.49.229.192:50277>;tag=X7A0729gXo5v" stupid hacker idiots |
| 18:17.28 | Samot | Yeah, I'm seeing those too. |
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| 19:23.20 | avb | haha |
| 19:23.30 | avb | we are lucky that most of those are idiots |
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