WEBRTC - OUTBOUND CALLS NOT TRANSFERRING TO CONFERENCE AGENT

[dito]

I did almost successful WebRtC Config.
Agent Exten Sucessfuly logged into conference hearing the " you are the only one in the conference "
launched outbound calls.. i see them answered in the asterisk cli but not transferred to the conf agent room.
both exten channel and outbound calls are in ulaw .. all codecs allowed
i have this in asterisk cli:
-- Channel Local/xxxxxxxxxxx@default-00000095;2 joined 'simple_bridge' basic-bridge <59a01ea5-81f7-4970-94da-8dfa2acb2f29>
-- <Local/xxxxxxxxxx@default-00000095;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
-- Executing [138369@default:2] Wait("Local/xxxxxxxxxxx@default-00000095;1", "2") in new stack
-- Executing [138369@default:3] Hangup("Local/xxxxxxxxxxxx@default-00000095;1", "") in new stack
== Spawn extension (default, 138369, 3) exited non-zero on 'Local/xxxxxxxxxxxxxxxx@default-00000095;1'
[Jan 31 19:19:20] WARNING[6869][C-0000017e]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
_______________________________________________________________________________________________________________
I will try to debug more to see what's wrong ..
---------------------------
VICIDAL 8.0
VERSION: 2.14-650a
BUILD: 180111-1544
Asterisk 13
on a 4 Core 16Gb RAM
___________________

[wjohnson133]

Do you have sip/extension in your vicidial conference? I would recommend getting the manager manual.

[dito]

Do you have sip/extension in your vicidial conference? I would recommend getting the manager manual.

hello ! of course the extension webrtc is well logged in the conference room
the manual does not handle the web rtc issues.
vicidial "normal phone" mode work fine ..

[vkad]

Same issue here.

Did you fix the webrtc (not available on latest chrome) with asterisk 13.

Thanks

[williamconley]

Do you have sip/extension in your vicidial conference? I would recommend getting the manager manual.

hello ! of course the extension webrtc is well logged in the conference room
the manual does not handle the web rtc issues.
vicidial "normal phone" mode work fine ..

extension 138369 is not part of vicidial autodial. So this is not running the same way as it would with a normal outbound autodial.

once your agent in the webrtc is in the vicidial conference, the rest should run normally with extension 8368 when a call is answered.

so: provide the contents of the agent conference (meetme listing of the conference) and the full output from "dial" for the autodialed call at the asterisk cli. If it does not go through extension 8368, please explain why. cuz it should.

[dito]

Do you have sip/extension in your vicidial conference? I would recommend getting the manager manual.

hello ! of course the extension webrtc is well logged in the conference room
the manual does not handle the web rtc issues.
vicidial "normal phone" mode work fine ..

extension 138369 is not part of vicidial autodial. So this is not running the same way as it would with a normal outbound autodial.

once your agent in the webrtc is in the vicidial conference, the rest should run normally with extension 8368 when a call is answered.

so: provide the contents of the agent conference (meetme listing of the conference) and the full output from "dial" for the autodialed call at the asterisk cli. If it does not go through extension 8368, please explain why. cuz it should.

Hi Sir,
i left the vici for some period back today found some time finally !

first of all thank you for the time you take to analyse all users posts here and to make this effort to "debug" the drafted posts.
strangely that "13" in front of the 8369 amd ext i don't know from where it comes i switched on the vps today to go in and suddenly it disappeared! 8369 normal
when doing the webrtc ssl config i followed this topic:
http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=8&t=37686
it did the work https certified and asterisk keys too ..
today i used seperated certification for asterisk using this script:
https://github.com/asterisk/asterisk/blob/master/contrib/scripts/ast_tls_cert
and Allelyuah it worked ! YEAH !
so i think "may be ssl config" should be done separately web part with certbot and for asterisk with this ast_tls_cert script.
if you want we can prepare kind of complete how to ..
good thing both "normal" extension and webrtc extension are working fine
i used https://github.com/chornyitaras/PBXWebPhone working ok
don't know why it work the vici webrtc phone only worked dialing manual for me
thx all

[williamconley]

Good postback. And certainly everyone would appreciate it if you posted your step-by-step and scripts used. Also could be that someone will improve upon them if you post them.

[dito]

Good postback. And certainly everyone would appreciate it if you posted your step-by-step and scripts used. Also could be that someone will improve upon them if you post them.

of course started a mini how to draft i will finish it tonight, of course it will be great if it will be improved.
thanks a lot sir william!

[mflorell]

As a note, the "13" you saw appended to the front of the routing extension is required if you use Asterisk 13.

Also, with newer versions of Chrome, WebRTC is going to soon stop working on Asterisk 11 servers.

[williamconley]

Also, with newer versions of Chrome, WebRTC is going to soon stop working on Asterisk 11 servers.

Now that you've mentioned it a couple times: Does this not apply to Firefox/IE? Or is this WebRTC limited to Chrome right now?

[mflorell]

Honestly, we only seem to be doing testing on WebRTC with Chrome, it is the most reliable WebRTC platform, and the chromium project is the one setting the WebRTC standards, so even if Firefox doesn't have that requirement at the moment, it will eventually because Chrome calls the shots with the not-yet-finalized WebRTC standards.

[marveelou]

Hello All,
I'm currently having the same issue here. The call is not sending to the agent. Just installed the
latest vicibox 9 from their site.Does anybody have the same issue with the latest version?

error logs

[Sep 26 06:46:42] -- Executing [138368@default:3] Hangup("Local/63214085267209@default-00000021;1", "") in new stack
[Sep 26 06:46:42] == Spawn extension (default, 138368, 3) exited non-zero on 'Local/63214085267209@default-00000021;1'
[Sep 26 06:46:42] WARNING[10998][C-0000003c]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Sep 26 06:46:42] -- Executing [h@default:1] AGI("Local/63214085267209@default-00000021;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack

[carpenox]

I have helped several people on this forum with similar issues, its not always the same thing. I am willing to help you out but its too slow thru the forum to debug everything it could be, so hit me up on FB: Cyburity, LLC -:- Skype: carpenox_3 -:- whatsapp: +19549477572

-Nox