vicidial.org

Hi,

Vicidial 8.0.1
VERSION: 2.14-761a
BUILD: 200708-1033

Have installed this server couple of weeks ago, everything was running smoothly. Suddenly we've started getting this weird issue. When we login to admin and Monitor a call, it accepts the session but if we click on any other user "Listen" button we get SIP 486. If we refresh the page and then again press "Listen" again first time it works fine.

Logs for first call

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[Jul 27 12:49:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul 27 12:49:01]     -- Executing [176*009*xxx*xxx*08600057@default:1] Goto("Local/176*009*xxx*xxx*08600057@default-00000280;2", "default,08600057,1") in new stack
[Jul 27 12:49:01]     -- Goto (default,08600057,1)
[Jul 27 12:49:01]     -- Executing [08600057@default:1] Dial("Local/176*009*xxx*xxx*08600057@default-00000280;2", "IAX2/ASTblind:sHMpuxoZs6jS7cU@127.0.0.1:41569/68600057,55,To") in new stack
[Jul 27 12:49:01]     -- Called IAX2/ASTblind:sHMpuxoZs6jS7cU@127.0.0.1:41569/68600057
[Jul 27 12:49:01]     -- Accepting AUTHENTICATED call from 127.0.0.1:
[Jul 27 12:49:01]     --        > requested format = gsm,
[Jul 27 12:49:01]     --        > requested prefs = (gsm|ulaw),
[Jul 27 12:49:01]     --        > actual format = ulaw,
[Jul 27 12:49:01]     --        > host prefs = (ulaw),
[Jul 27 12:49:01]     --        > priority = mine
[Jul 27 12:49:01]     -- Call accepted by 127.0.0.1 (format ulaw)
[Jul 27 12:49:01]     -- Format for call is (ulaw)
[Jul 27 12:49:01]     -- Executing [68600057@default:1] MeetMe("IAX2/ASTblind-14066", "8600057,Fmq") in new stack
[Jul 27 12:49:01]     -- IAX2/127.0.0.1:41569-9170 answered Local/176*009*xxx*xxx*08600057@default-00000280;2
[Jul 27 12:49:01]        > Channel Local/176*009*xxx*xxx*08600057@default-00000280;1 was answered
[Jul 27 12:49:01]     -- Executing [7001@default:1] Dial("IAX2/127.0.0.1:41569-9170", "SIP/7001,60,") in new stack
[Jul 27 12:49:01]     -- Executing [h@default:1] AGI("Local/176*009*xxx*xxx*08600057@default-00000280;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
[Jul 27 12:49:01]   == Using SIP RTP CoS mark 5
[Jul 27 12:49:01]     -- Called SIP/7001
[Jul 27 12:49:01]     -- <Local/176*009*xxx*xxx*08600057@default-00000280;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0 completed, returning 0


Logs when pressed "Listen" button for another user

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[Jul 27 12:49:43]     -- Executing [176*009*xxx*xxx*08600055@default:1] Goto("Local/176*009*xxx*xxx*08600055@default-00000284;2", "default,08600055,1") in new stack
[Jul 27 12:49:43]     -- Goto (default,08600055,1)
[Jul 27 12:49:43]     -- Executing [08600055@default:1] Dial("Local/176*009*xxx*xxx*08600055@default-00000284;2", "IAX2/ASTblind:sHMpuxoZs6jS7cU@127.0.0.1:41569/68600055,55,To") in new stack
[Jul 27 12:49:43]     -- Called IAX2/ASTblind:sHMpuxoZs6jS7cU@127.0.0.1:41569/68600055
[Jul 27 12:49:43]     -- Accepting AUTHENTICATED call from 127.0.0.1:
[Jul 27 12:49:43]     --        > requested format = gsm,
[Jul 27 12:49:43]     --        > requested prefs = (gsm|ulaw),
[Jul 27 12:49:43]     --        > actual format = ulaw,
[Jul 27 12:49:43]     --        > host prefs = (ulaw),
[Jul 27 12:49:43]     --        > priority = mine
[Jul 27 12:49:43]     -- Call accepted by 127.0.0.1 (format ulaw)
[Jul 27 12:49:43]     -- Format for call is (ulaw)
[Jul 27 12:49:43]     -- Executing [68600055@default:1] MeetMe("IAX2/ASTblind-1898", "8600055,Fmq") in new stack
[Jul 27 12:49:43]     -- IAX2/127.0.0.1:41569-876 answered Local/176*009*xxx*xxx*08600055@default-00000284;2
[Jul 27 12:49:43]        > Channel Local/176*009*xxx*xxx*08600055@default-00000284;1 was answered
[Jul 27 12:49:43]     -- Executing [7001@default:1] Dial("IAX2/127.0.0.1:41569-876", "SIP/7001,60,") in new stack
[Jul 27 12:49:43]     -- Executing [h@default:1] AGI("Local/176*009*xxx*xxx*08600055@default-00000284;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
[Jul 27 12:49:43]   == Using SIP RTP CoS mark 5
[Jul 27 12:49:43]     -- Called SIP/7001
[Jul 27 12:49:43]     -- <Local/176*009*xxx*xxx*08600055@default-00000284;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0 completed, returning 0
[Jul 27 12:49:43]   == Spawn extension (default, 08600055, 1) exited non-zero on 'Local/176*009*xxx*xxx*08600055@default-00000284;2'
[Jul 27 12:49:44]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul 27 12:49:45]     -- SIP/7001-000001b3 is ringing
[Jul 27 12:49:45]     -- Got SIP response 486 "Busy Here" back from 182.176.xxx.xxx:58821
[Jul 27 12:49:45]     -- SIP/7001-000001b3 is busy
[Jul 27 12:49:45]   == Everyone is busy/congested at this time (1:1/0/0)
[Jul 27 12:49:45]     -- Executing [7001@default:2] Goto("IAX2/127.0.0.1:41569-876", "default,850266666666667001,1") in new stack
[Jul 27 12:49:45]     -- Goto (default,850266666666667001,1)
[Jul 27 12:49:45]     -- Executing [850266666666667001@default:1] Wait("IAX2/127.0.0.1:41569-876", "1") in new stack
[Jul 27 12:49:46]     -- Executing [850266666666667001@default:2] VoiceMail("IAX2/127.0.0.1:41569-876", "7001,u") in new stack
[Jul 27 12:49:46]     -- <IAX2/127.0.0.1:41569-876> Playing 'vm-theperson.gsm' (language 'en')


are you using viciphone or softphone to monitor with?

Viciphone

Previously with vici 9.0.2 I had same issue but when I installed 8.0.1 I was able to listen to live calls by just pressing "Listen" on whomever extension I want to without refreshing the tab.

I think something broke when I restarted the server couple of days back because issue started after that

Maverick1 wrote:Previously with vici 9.0.2 I had same issue but when I installed 8.0.1 I was able to listen to live calls by just pressing "Listen" on whomever extension I want to without refreshing the tab.

I think something broke when I restarted the server couple of days back because issue started after that

My apologies, please ignore this, had misunderstanding.

Is there a way I can register extension on soft-phone with webrtc?

yea it sees the extension as already being used if you do it thru viciphone or webrtc, you have to go to report options, clear the extension and submit, then go back in and put it back, i setup 2 listening phones using viciphone template and i just change the phone im listening with as i switch between agents. this may be a bug in the new viciphone, ill report back to the designer for it and see if he knows about it already

Yes that is exactly what's happening. Is there a way we can register the extension on softphone in this scenario?

yes, you would have to change the phone settings to SIP or IAX2 and install xlite or zoiper or similar....

here is the old zoiper v3 that was free: https://server.dreamescapes.us/d0wnl0ads/Zoiper.exe

Great, Thank you...your a life saver

np, lemme know if it works for you

I was able to register but getting call is not going through with SIP template

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[Jul 28 14:19:31] ERROR[1965]: chan_sip.c:17171 register_verify: 'UDP' is not a valid transport for '7001'. we only use 'WSS'! ending call.
[Jul 28 14:19:31] NOTICE[1965]: chan_sip.c:28490 handle_request_register: Registration from '7001<sip:7001@176.9.xxx.xxx>' failed for '23.226.134.74:9602' - Device not configured to use this transport type


If I add same template as I used for web, extension does not register

After removing WSS from the template

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[Jul 28 14:22:38] WARNING[1965][C-00000a3b]: chan_sip.c:10129 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7982 RTP/AVP 0 3 101
[Jul 28 14:22:38] WARNING[1965][C-00000a3b]: chan_sip.c:10501 process_sdp: Failing due to no acceptable offer found
[Jul 28 14:22:38]   == Everyone is busy/congested at this time (1:0/0/1)

After removing AVPF and force AVP

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[Jul 28 14:26:44] WARNING[1965][C-00000a5c]: chan_sip.c:10519 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
[Jul 28 14:26:44]   == Everyone is busy/congested at this time (1:0/0/1)

select no template

Nope, doesn't work with no template as well

[Jul 29 17:18:48] WARNING[1965][C-0000128b]: chan_sip.c:10519 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
[Jul 29 17:18:48] == Everyone is busy/congested at this time (1:0/0/1)

do you have your certbot SSL certificate installed correctly?

Well I dont have any issue in making calls using web / viciphone, though I didn't used certbot to install certificate

you shouldnt be gonig through SRTP using zoiper. It seems you have something set wrong. Under the phone, do you have webphone set to N? are you sure the template is off? I can help you debug the issue but it would be hard through text. If you have facebook, hit me up under "Cyburity LLC" or i have skype and whatsapp as well

Yes webphone is set to N and yes Template is set to NONE, I'll PM you my skype.