Call Terminate after few seconds

Asterisk Asterisk WebRTC
1

Hi I am using asterisk version 22.3.

I have created one webrtc and one zenitel intercom configurations in pjsip.conf and extensions.conf, http.conf, rtp.conf

After I done the changes the PJSIP has been registered and call has been established between udp and webrtc transports.

The issue is the call has been terminated within few seconds and websocket connection also getting closed in react side. The front end is react to make calls

The error which I mentioned below.

:envelope_with_arrow: Received SIP Message: {headers: {
}, type: ‘request’, method: ‘BYE’, requestUri: ‘sip:36l6a4he@192.168.250.120:64097;transport=ws;ob’, sipVersion: ‘SIP/2.0’} index-Bbbv2bP9.js:252 :envelope_with_arrow: Received SIP Message: {headers: {
}, type: ‘request’, method: ‘BYE’, requestUri: ‘sip:36l6a4he@192.168.250.120:64097;transport=ws;ob’, sipVersion: ‘SIP/2.0’} index-Bbbv2bP9.js:222 Wed May 07 2025 12:53:33 GMT+0200 (Central European Summer Time) | sip.Transport | Received WebSocket text message: BYE sip:36l6a4he@192.168.250.120:64097;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 192.168.250.137:8089;rport;branch=z9hG4bKPjfae91f93-64f5-4729-a938-caba3cf997b1;alias From: sip:BUTESTNL@butestnl.dtap.local;tag=fc337b9f-518b-462a-9ff9-23282d72a90e To: sip:ATGLTMSPdtaplocal@otasip.dtap.local;tag=ladm6246qv Call-ID: 06gvk92g27tdc46c1vou CSeq: 12344 BYE Reason: Q.850;cause=16 Max-Forwards: 70 User-Agent: Asterisk PBX 22.3.0 Content-Length: 0

2

Cause 16 is normal clearing, not an error. It might be the result of an error, but it provides no information about the error.

Also it is much easier for people to understand the Asterisk logs.

Also, the exact number of seconds is often a clue to the underlying problem.

3

The log message is below. Please refer and assist

[May 7 14:49:18] Asterisk 22.3.0 built by root @ TPL-VISICS-UBUNTU24 on a x86_64 running Linux on 2025-05-06 13:44:16 UTC [May 7 14:49:18] NOTICE[89362] loader.c: 334 modules will be loaded. [May 7 14:49:18] NOTICE[89362] cdr.c: CDR simple logging enabled. [May 7 14:49:18] NOTICE[89362] indications.c: Default country for indication tones: us [May 7 14:49:18] NOTICE[89362] indications.c: Setting default indication country to ‘us’ [May 7 14:49:18] WARNING[89362] res_phoneprov.c: Unable to find a valid server address or name. [May 7 14:49:18] NOTICE[89362] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [May 7 14:49:18] NOTICE[89362] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge [May 7 14:49:18] NOTICE[89362] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [May 7 14:49:18] WARNING[89362] cel_pgsql.c: CEL pgsql config file missing global section. [May 7 14:49:18] NOTICE[89362] codec_siren14.c: ITU G.722.1 Annex C (Siren14, licensed from Polycom) transcoding module version 22.0_1.0.7 [May 7 14:49:18] NOTICE[89362] codec_siren14.c: Copyright (C) 1999-2009 Digium, Inc. [May 7 14:49:18] NOTICE[89362] codec_siren14.c: This module is supplied under a commercial license granted by Digium, Inc. [May 7 14:49:18] NOTICE[89362] codec_siren14.c: Please see the full license text supplied in the accompanying [May 7 14:49:18] NOTICE[89362] codec_siren14.c: “LICENSE” file, or ask for a copy from Digium. [May 7 14:49:18] NOTICE[89362] codec_siren14.c: This product includes software from the Speex library. Please see [May 7 14:49:18] NOTICE[89362] codec_siren14.c: the accompanying “SPEEX_LICENSE” file for license information. [May 7 14:49:18] NOTICE[89362] codec_siren7.c: ITU G.722.1 (Siren7, licensed from Polycom) transcoding module version 22.0_1.0.7 [May 7 14:49:18] NOTICE[89362] codec_siren7.c: Copyright (C) 1999-2009 Digium, Inc. [May 7 14:49:18] NOTICE[89362] codec_siren7.c: This module is supplied under a commercial license granted by Digium, Inc. [May 7 14:49:18] NOTICE[89362] codec_siren7.c: Please see the full license text supplied in the accompanying [May 7 14:49:18] NOTICE[89362] codec_siren7.c: “LICENSE” file, or ask for a copy from Digium. [May 7 14:49:18] WARNING[89362] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 196-196: macro call to ael-dundi-e164 cannot be found in the AEL code! [May 7 14:49:18] WARNING[89362] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 209-209: macro call to ael-dundi-e164 cannot be found in the AEL code! [May 7 14:49:18] WARNING[89362] ael/pval.c: Warning: file /etc/asterisk/extensions.ael, line 328-328: macro call to ael-std-exten-ael cannot be found in the AEL code! [May 7 14:49:18] WARNING[89362] loader.c: Module ‘res_adsi’ has been loaded but may be removed in a future release. [May 7 14:49:18] WARNING[89362] loader.c: Module ‘app_adsiprog’ has been loaded but may be removed in a future release. [May 7 14:49:18] WARNING[89362] loader.c: Module ‘app_getcpeid’ has been loaded but may be removed in a future release. [May 7 14:49:18] WARNING[89368] pbx_config.c: users.conf is deprecated and will be removed in a future version of Asterisk

4

There are no calls in this.

It looks like there are errors in your AEL code. You’ll need to supply thqat as that is part of your customisation of the system. Relatively few people are familiar with AEL. I am not.

You appear to have closed source codecs. Any problems with those need to be addressed to the supplier, as the community does not have the information necessary to fully support them especially ones i’ve never heard of being used, before,

I’m not going to re-instate the original formatting of logs given on one line, next time, for look at them.