Ratio and Adaptive Dialing calls and hangs up on an answered
Posted: Tue Nov 16, 2010 2:24 am
I'm hoping someone can see in this what I cannot. I am having an issue like many others it seems with the predictive dialling feature. Manual dial campaigns works 100% but when the campaign gets switched to either ratio or adaptive dialing the agent hears "you are currently the only person in the conference", but is never passed a call. I shouldn't say never, because out of 100 calls I did manage to get 2 calls passed to an agent. I've tried to replicate that so I can post the cli output but haven't managed. I used a list with personal numbers in it, the call would be placed but would hangup after about 2 seconds after being answered. I've played around with drop call times and other settings but nothing seems to have any effect on the outcome.
The Beast:
GoAutoDial 2.0 CE | Asterisk 1.4.27.1-1 | No Hardware | No Extra Software
The Set-Up:
IAX2 Trunk to trixbox server with dial plan:
All manual calling works fine either from agent interface or direct from registered sip phone.
CLI ->
Trixbox side shows call is hung up on vicidial side
Relevant extensions.conf:
Completely default from base install. ^
The only error I see is "file.c:1292 waitstream_core: Unexpected control subclass '-1'" which I have googled and can't seem to find anything that matches.
Doesn't seem to be a codec issue, I have both servers and all phones set to disallow=all allow=ulaw
Sip debug doesn't show anything abnormal that I can see anyway.
Any suggestions anyone might have would be greatly appreciated.
Thanks
************************************
Figured it out.
As I said in my post I have an IAX2 trunk from goautodial to a trixbox server as my provider is shared with an office pbx. On the trunk I have disallowed all but the ulaw codec. In the extensions.conf file it calls an audio file called sip-silence
sip-silence is a gsm encoded audio file... bad news for me. Funny how I didn't see anything in the logs about a codec mismatch but whatever.
silence on the other hand is not... its a wav file.
I know someone's probably going to tell me that changing this is a bad idea, but after doing it, I made 10 calls and all 10 were passed to the agent without a hang-up.
Just thought I'd post my fix for anyone that may have the same problem, and possibly in-case someone has some advice about a better way to fix the problem.
[/code]
The Beast:
GoAutoDial 2.0 CE | Asterisk 1.4.27.1-1 | No Hardware | No Extra Software
The Set-Up:
IAX2 Trunk to trixbox server with dial plan:
- Code: Select all
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRIXBOX}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup
All manual calling works fine either from agent interface or direct from registered sip phone.
CLI ->
- Code: Select all
[Nov 15 23:40:48] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 15 23:40:48] -- Executing [9***********@default:2] Dial("Local/***********@default-6d4d,2", "IAX2/trixbox-peer/***********||tTor") in new stack
[Nov 15 23:40:48] -- Called trixbox-peer/**********
[Nov 15 23:40:48] -- Call accepted by 10.10.11.2 (format ulaw)
[Nov 15 23:40:48] -- Format for call is ulaw
[Nov 15 23:40:51] -- IAX2/trixbox-peer-28043 is making progress passing it to Local/***********@default-6d4d,2
[Nov 15 23:40:58] -- IAX2/trixbox-peer-28043 answered Local/***********@default-6d4d,2
[Nov 15 23:40:58] > Channel Local/***********@default-6d4d,1 was answered.
[Nov 15 23:40:58] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 15 23:40:58] -- Executing [8368@default:1] Playback("Local/***********@default-6d4d,1", "sip-silence") in new stack
[Nov 15 23:40:58] -- <Local/***********@default-6d4d,1> Playing 'sip-silence' (language 'en')
[Nov 15 23:40:58] WARNING[5469]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Nov 15 23:40:58] WARNING[5469]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Nov 15 23:40:58] -- Executing [8368@default:2] AGI("Local/***********@default-6d4d,1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 15 23:40:58] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 15 23:40:58] -- Executing [8368@default:3] AGI("Local/***********@default-6d4d,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Nov 15 23:40:58] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 15 23:40:59] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 15 23:40:59] -- Executing [8368@default:4] AGI("Local/***********@default-6d4d,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Nov 15 23:40:59] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 15 23:41:00] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 15 23:41:00] -- Executing [8368@default:5] Hangup("Local/***********@default-6d4d,1", "") in new stack
[Nov 15 23:41:00] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/***********@default-6d4d,1'
[Nov 15 23:41:00] -- Executing [h@default:1] DeadAGI("Local/***********@default-6d4d,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Nov 15 23:41:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Nov 15 23:41:00] -- Executing [h@default:1] DeadAGI("Local/***********@default-6d4d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----2") in new stack
[Nov 15 23:41:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----2 completed, returning 0
[Nov 15 23:41:00] -- Hungup 'IAX2/trixbox-peer-28043'
[Nov 15 23:41:00] == Spawn extension (default, ***********, 2) exited non-zero on 'Local/***********@default-6d4d,2'
Trixbox side shows call is hung up on vicidial side
Relevant extensions.conf:
- Code: Select all
; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup
Completely default from base install. ^
The only error I see is "file.c:1292 waitstream_core: Unexpected control subclass '-1'" which I have googled and can't seem to find anything that matches.
Doesn't seem to be a codec issue, I have both servers and all phones set to disallow=all allow=ulaw
Sip debug doesn't show anything abnormal that I can see anyway.
Any suggestions anyone might have would be greatly appreciated.
Thanks
************************************
Figured it out.
As I said in my post I have an IAX2 trunk from goautodial to a trixbox server as my provider is shared with an office pbx. On the trunk I have disallowed all but the ulaw codec. In the extensions.conf file it calls an audio file called sip-silence
- Code: Select all
; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup
sip-silence is a gsm encoded audio file... bad news for me. Funny how I didn't see anything in the logs about a codec mismatch but whatever.
- Code: Select all
; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup
silence on the other hand is not... its a wav file.
I know someone's probably going to tell me that changing this is a bad idea, but after doing it, I made 10 calls and all 10 were passed to the agent without a hang-up.
Just thought I'd post my fix for anyone that may have the same problem, and possibly in-case someone has some advice about a better way to fix the problem.
[/code]