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| 05:37.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.0.1, 13.37.1, 16.14.1 (2020/11/05) Standard: 17.8.1 (2020/11/05); 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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| 09:46.29 | badrik | hi |
| 09:47.17 | badrik | i facing while register sip user through webrtc |
| 09:47.19 | badrik | res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"webrtc_client" <sip:webrtc_client@192.168.42.253>' failed for '192.168.42.253:46010' (callid: 8e31c2af-2186-a44d-cff9-d15596af6b92) - Failed to authenticate |
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| 14:53.45 | charrit69 | Do you know how to allow sending MESSAGE prior the channel answer? (with chan_sip works fine but with PJSIP the message is not transfered to the other endpoint) |
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| 15:03.38 | charrit69 | While 180 Ringing, Message sent from phone to asterisk is 202 accepted in the asterisk side but not forwarded to the other party |
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| 16:06.44 | sibiria | charrit69: it should work if you're registered |
| 16:09.50 | charrit69 | sibiria: not working, the message is accepted by asterisk but not forwarded to the other party. Instead, if the call is established... (after OK) it works |
| 16:09.52 | sibiria | you should be getting a channel type "Message" which you can verify as part of your incoming extension |
| 16:10.01 | sibiria | before doing something like Answer() or Ringing() |
| 16:13.02 | charrit69 | I think that the problem could be with the From header, my setup is with multiple network cards and PJSIP, for calling the other party asterisk uses the IP of another network card (I don't kow why) |
| 16:13.56 | charrit69 | Both parties are reachable from the same network card, old chan_sip use always the destination card ip address for contacting the second agent |
| 16:20.41 | charrit69 | I think that the problem could be with the From header, my setup is with multiple network cards and PJSIP, for calling the other party asterisk uses the IP of another network card (I don't kow why) |
| 16:20.53 | charrit69 | You can see here a kind of trace: https://pastebin.com/q3VSPtvz |
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| 16:28.31 | charrit69 | I upgraded asterisk to 18.0.1 and the behaviour is the same |
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| 16:31.17 | charrit69 | Why asterisk is not picking the IP address of the interface where is connected the phone? |
| 16:31.26 | charrit69 | transport and endpoint configurations: https://pastebin.com/pyBP0iUi |
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| 16:38.19 | charrit69 | sibiria: where can I see the channel type "Message"? (I tried core show channels, pjsip list channels. core show channel PJ... and pjsip show channel PJ... also don't show message info) |
| 16:40.01 | sibiria | it's a channel variable. you can access it in the dial plan via ${CHANNEL(channeltype)} |
| 16:40.58 | sibiria | you'll find the message details in MESSAGE(body), MESSAGE(from) etc. |
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| 16:47.02 | charrit69 | In variables shown in the core show channel PJSIP/240-....... is not pressent |
| 16:56.39 | sibiria | i could be wrong but i'm guessing the channel doesn't exist that long |
| 16:56.46 | sibiria | unless you do something to keep it alive in the dial plan |
| 16:58.32 | sibiria | a MESSAGE, just like an INVITE, can and will jump to whatever dial plan context you have configured for the endpoint |
| 17:02.32 | charrit69 | The channel it's still alive when I dump the channel from the console, but vars are not shown |
| 17:03.13 | charrit69 | I'm trying to see the message with the function MESSAGE without success for now |
| 17:05.45 | charrit69 | in this case, message is inside the call. It's linked with a CALL-ID. It's generated in the ringing phone and it should be passed to the origin |
| 17:12.37 | charrit69 | sibiria: a lot of thanks for your help, I'll continue the debugging tomorrow with your advise. |
| 17:13.28 | charrit69 | Bye |
| 17:13.32 | sibiria | good luck. i'm sorry i cannot help more, i only have experience passing messages outside calls |
| 17:13.38 | sibiria | i hope you'll sort it out |
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| 20:20.40 | theGoat | + |
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| 20:41.14 | beta2 | I've been subject of an unfortunate upgrade and now I can't get asterisk to open up port 5060 (pjsip). driving me nuts! |
| 20:41.33 | beta2 | ss -tulpn shows no 5060 on asterisk |
| 20:45.22 | sibiria | perhaps chan_sip is running and got there first |
| 20:46.19 | beta2 | chan_sip module is deactivated in modules.conf |
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| 21:29.48 | beta2 | if pjsip is used and not chan_sip is port 5060 supposed to be open when no phones are registered? |
| 21:30.07 | beta2 | There are no errors i can find so it's hard to diagnose this issue |
| 21:31.01 | file | if it's configured to be listening on it... then it would be... |
| 21:31.16 | file | any errors would show in the Asterisk console or log at startup |
| 21:45.22 | beta2 | checked /var/log/asterisk/messages and a logger file with high verbosity... nothing. I do see that my phones are trying to register (but fail since they are trying go on 5060) |
| 21:46.12 | beta2 | I see warnings in the logs but nothing about registration since presumably asterisk isn't even seeing the traffic (since it's not opening up a port to see the traffic) |
| 21:46.52 | file | is PJSIP configured to listen on 5060? is the module loaded? |
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| 21:47.37 | seanbright | is your computer plugged in? |
| 21:47.44 | beta2 | yes pjsip is loaded |
| 21:47.49 | beta2 | i see my endpoints |
| 21:48.13 | seanbright | *CLI> pjsip show transports |
| 21:50.13 | beta2 | winner winner - i didn't have a transport set |
| 21:50.19 | beta2 | thank you thank you thank you |
| 21:50.46 | beta2 | (thank you)^5 seriously |
| 22:01.28 | seanbright | prego |
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