No one is in your session

[Cyberwolf2]

Hey, I got an issue with meetme/webrtc/viciphone. When the agent logs in I can't hear the recording "You are the only one in the conference". The screen then says "No one is in your session: 8600052"
I installed this server recently using callcentertech.net's vicifast script.

I've been banging on my keyboard for a few days. Any ideas?


AlmaLinux8
Asterisk 16.17.0-vici
astGUIclient: VERSION: 2.14-878a, BUILD: 230312-2212

screen -ls:

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There are screens on:
   2234.ASTconf3way   (Detached)
   2231.ASTemail   (Detached)
   2227.ASTVDadFILL   (Detached)
   2214.ASTfastlog   (Detached)
   2211.ASTVDadapt   (Detached)
   2208.ASTVDremote   (Detached)
   2205.ASTVDauto   (Detached)
   2202.ASTlisten   (Detached)
   2199.ASTsend   (Detached)
   2196.ASTupdate   (Detached)
   1854.asterisk   (Detached)
   1844.astshell20230319110836   (Detached)
12 Sockets in /run/screen/S-root.
[root@97CB80C ~]#



Asterisk CLI:

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  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 55558600051@default
    -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000002;2", "8600051,K") in new stack
[Mar 19 11:34:07] WARNING[16704][C-00000010]: app_meetme.c:5258 admin_exec: Conference number '8600051' not found!
    -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000002;2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000002;2'
[Mar 19 11:34:07] WARNING[16704][C-00000010]: func_hangupcause.c:138 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/55558600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/55558600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == WebSocket connection from '174.66.202.131:56249' for protocol 'sip' accepted using version '13'
    -- Registered SIP '5001' at 174.66.202.131:56249
  == Manager 'sendcron' logged on from 127.0.0.1
  == Using SIP RTP CoS mark 5
    -- Called 5001
  == Manager 'sendcron' logged off from 127.0.0.1
[Mar 19 11:34:39] ERROR[1976]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
  == WebSocket connection from '174.66.202.131:54579' forcefully closed due to fatal write error


[jamiemurray]

Have a look at the settings container for VICIPHONE_SETTINGS.

If you see the line:

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# determines if the reg_exten is called upon successful registration
dialRegExten : 1


Change it to:

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# determines if the reg_exten is called upon successful registration
dialRegExten : 0


When this feature was implemented, I noticed that unless the phone is in default context, it doesn't work (note the context of the phone is ignored if a template is used, only the context set in the template is used which if not specified will default to trunkinbound).

and try again.

[martinch]

Hey, sorry for the late post here.

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[Mar 19 11:34:07] WARNING[16704][C-00000010]: app_meetme.c:5258 admin_exec: Conference number '8600051' not found!


Asterisk is saying it can't find a conference number on the telephony server. You can check on the admin panel;

• Go to /vicidial/admin.php.
• Admin
• Conferences
• Add your MeetMe conferences in there if they are missing along with the telephony server IP you want to use.

Hope this helps.

[jamiemurray]

I think that's just because you have Agent Login Kick-all enabled on system settings you're seeing the line about the conference not found because there is an attempt to kick all participants from the conference upon login.

I think the issue is with your SSL certificate, is it self signed or is it from a CA? Are the paths to the certificate and key correct in http.conf and the phone template?

[carpenox]

What SSL cert did you use? Use let's encrypt and as martin said check if your conferences and vicidial conferences are there