Autodial hangs up when customer answers, manual works fine
Posted: Tue Jun 11, 2013 7:04 am
Hi,
I've been working on our vicidial setup and it's almost working, but I've run into a problem and I can't figure it out. When using manual dialing (off a list, so Agent just clicks Dial Next Lead) it works perfectly fine, but when I switch to RATIO it immediately hangs up when the call is answered. The person called then just hears a busy tone (I think it's the busy tone, just goes beep beep beep with only a very very brief pause between). Initially I thought it might be due to a problem in the NAT/Routing or the sound hardware/drivers, but since it works in manual mode it can't be that.
Seeing the asterisk log I suspect it is something to do with /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi, but do note that in the webagent it does say the customer answered, but ofc (correctly) changes to hung up. Rebooting doesn't help.
I should mention that I did have it working at one point, but I'm not sure what I have changed since then. More importantly I'm not sure when that was, as I did much of my testing/developing with manual mode, or else I could get the then-current configs from my git
Here's the asterisk log (removed " == Manager 'sendcron' logged" lines) first for a manual call, logoff, and logon (changed campaign mode before logon):
and for an automatic call:
Vicidial Version: VERSION: 2.6-366c BUILD: 130402-2250 Server: 10.129.119.10
Asterisk: 1.8.19.0-vici-beta
Server Specs: 2*8core Opteron with vmware esxi 4.1. 8 CPUs and 6 GB RAM assigned to the VM in which this runs.
Load average on the server: below 0.2 (with 8 cores, so >>90% idle)
Codec used: ulaw
VoIP: pjsua from pjsip as softphone for the agents, then sipgate as carrier
OS: Zentyal 3.0 (a derivative of Ubuntu LTS 12.04), fully updated as of 11am or noon today (UTC+2)
Status: System has not gone into production yet, but outbound dialling did work at one point.
Well, thanks for making it this far, if I forgot anything please ask.
Steffen
Edit: the relevant snipped from extensions.conf:
Also, I know that support for Asterisk-1.8 is in beta, but it seemed to work fine for me, so I would very much like to stick to the newer version. Plus, it's less risky to test things on a new install like mine than to have people upgrade their production systems
Would be most grateful for any ideas, pointers or suspicions!
I've been working on our vicidial setup and it's almost working, but I've run into a problem and I can't figure it out. When using manual dialing (off a list, so Agent just clicks Dial Next Lead) it works perfectly fine, but when I switch to RATIO it immediately hangs up when the call is answered. The person called then just hears a busy tone (I think it's the busy tone, just goes beep beep beep with only a very very brief pause between). Initially I thought it might be due to a problem in the NAT/Routing or the sound hardware/drivers, but since it works in manual mode it can't be that.
Seeing the asterisk log I suspect it is something to do with /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi, but do note that in the webagent it does say the customer answered, but ofc (correctly) changes to hung up. Rebooting doesn't help.
I should mention that I did have it working at one point, but I'm not sure what I have changed since then. More importantly I'm not sure when that was, as I did much of my testing/developing with manual mode, or else I could get the then-current configs from my git
Here's the asterisk log (removed " == Manager 'sendcron' logged" lines) first for a manual call, logoff, and logon (changed campaign mode before logon):
- Code: Select all
-- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000003;2", "8600052,F") in new stack
> Channel Local/8600052@default-00000003;1 was answered.
-- Executing [03412228680@default:1] NoOp("Local/8600052@default-00000003;1", "nationalf") in new stack
-- Executing [03412228680@default:2] AGI("Local/8600052@default-00000003;1", "agi://127.0.0.1:4577/call_log") in new stack
-- <Local/8600052@default-00000003;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [03412228680@default:3] Set("Local/8600052@default-00000003;1", "SIPFROMUSER=1595970t0@sipconnect.sipgate.de") in new stack
-- Executing [03412228680@default:4] SIPAddHeader("Local/8600052@default-00000003;1", "P-Preferred-Identity: <sip:4930231882322@sipconnect.sipgate.de>") in new stack
-- Executing [03412228680@default:5] Dial("Local/8600052@default-00000003;1", "SIP/03412228680@testcarrier,,r") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/03412228680@testcarrier
-- Executing [58600052@default:1] MeetMe("Local/58600052@default-00000004;2", "8600052,Fmq") in new stack
> Channel Local/58600052@default-00000004;1 was answered.
-- Executing [8309@default:1] Answer("Local/58600052@default-00000004;1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/58600052@default-00000004;1", "wav,11472_20130611-132448_8002_3412228680") in new stack
-- Executing [8309@default:3] Wait("Local/58600052@default-00000004;1", "3600") in new stack
-- SIP/testcarrier-00000009 is ringing
-- SIP/testcarrier-00000009 is making progress passing it to Local/8600052@default-00000003;1
-- SIP/testcarrier-00000009 answered Local/8600052@default-00000003;1
-- Executing [h@default:1] AGI("Local/8600052@default-00000003;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----24-----20") in new stack
-- <Local/8600052@default-00000003;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----24-----20 completed, returning 0
== Spawn extension (default, 03412228680, 5) exited non-zero on 'Local/8600052@default-00000003;1'
== Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000003;2'
-- Executing [h@default:1] AGI("Local/8600052@default-00000003;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- <Local/8600052@default-00000003;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
== Spawn extension (default, 58600052, 1) exited non-zero on 'Local/58600052@default-00000004;2'
-- Executing [h@default:1] AGI("Local/58600052@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- <Local/58600052@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
== Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600052@default-00000004;1'
-- Executing [h@default:1] AGI("Local/58600052@default-00000004;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- <Local/58600052@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
-- Hungup 'DAHDI/pseudo-2137390417'
== Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/202-00000008'
-- Executing [h@default:1] AGI("SIP/202-00000008", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/202-00000008>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
-- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-00000005;2", "8600052,K") in new stack
[Jun 11 13:27:13] WARNING[22809]: app_meetme.c:4658 admin_exec: Conference number '8600052' not found!
-- Executing [55558600052@default:2] Hangup("Local/55558600052@default-00000005;2", "") in new stack
== Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-00000005;2'
-- Executing [h@default:1] AGI("Local/55558600052@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <Local/55558600052@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
== Using SIP RTP CoS mark 5
> Channel SIP/202-0000000a was answered.
-- Executing [8600052@default:1] MeetMe("SIP/202-0000000a", "8600052,F") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': == Found
-- Created MeetMe conference 1022 for conference '8600052'
-- <SIP/202-0000000a> Playing 'conf-onlyperson.ulaw' (language 'en')
and for an automatic call:
- Code: Select all
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [03412228680@default:1] NoOp("Local/03412228680@default-00000006;2", "nationalf") in new stack
-- Executing [03412228680@default:2] AGI("Local/03412228680@default-00000006;2", "agi://127.0.0.1:4577/call_log") in new stack
-- <Local/03412228680@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [03412228680@default:3] Set("Local/03412228680@default-00000006;2", "SIPFROMUSER=1595970t0@sipconnect.sipgate.de") in new stack
-- Executing [03412228680@default:4] SIPAddHeader("Local/03412228680@default-00000006;2", "P-Preferred-Identity: <sip:4930231882322@sipconnect.sipgate.de>") in new stack
-- Executing [03412228680@default:5] Dial("Local/03412228680@default-00000006;2", "SIP/03412228680@testcarrier,,r") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/03412228680@testcarrier
-- SIP/testcarrier-0000000b is ringing
-- SIP/testcarrier-0000000b is making progress passing it to Local/03412228680@default-00000006;2
-- SIP/testcarrier-0000000b answered Local/03412228680@default-00000006;2
> Channel Local/03412228680@default-00000006;1 was answered.
-- Executing [8368@default:1] Playback("Local/03412228680@default-00000006;1", "sip-silence") in new stack
-- <Local/03412228680@default-00000006;1> Playing 'sip-silence.gsm' (language 'en')
-- Executing [h@default:1] AGI("Local/03412228680@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0") in new stack
-- Executing [8368@default:2] NoOp("SIP/testcarrier-0000000b", "fooa") in new stack
-- Executing [8368@default:3] AGI("SIP/testcarrier-0000000b", "agi://127.0.0.1:4577/call_log") in new stack
-- <SIP/testcarrier-0000000b>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [8368@default:4] NoOp("SIP/testcarrier-0000000b", "foob") in new stack
-- Executing [8368@default:5] AGI("SIP/testcarrier-0000000b", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- <SIP/testcarrier-0000000b>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing [010*129*119*010*8600052@default:1] NoOp("SIP/testcarrier-0000000b", "nationalf") in new stack
-- Executing [010*129*119*010*8600052@default:2] Hangup("SIP/testcarrier-0000000b", "") in new stack
== Spawn extension (default, 010*129*119*010*8600052, 2) exited non-zero on 'SIP/testcarrier-0000000b'
-- Executing [h@default:1] AGI("SIP/testcarrier-0000000b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/testcarrier-0000000b>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
-- <Local/03412228680@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0 completed, returning 0
== Spawn extension (default, 03412228680, 5) exited non-zero on 'Local/03412228680@default-00000006;2'
-- Executing [58600052@default:1] MeetMe("Local/58600052@default-00000007;2", "8600052,Fmq") in new stack
> Channel Local/58600052@default-00000007;1 was answered.
-- Executing [8309@default:1] Answer("Local/58600052@default-00000007;1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/58600052@default-00000007;1", "wav,11473_20130611-132730_8002_3412228680") in new stack
-- Executing [8309@default:3] Wait("Local/58600052@default-00000007;1", "3600") in new stack
== Spawn extension (default, 58600052, 1) exited non-zero on 'Local/58600052@default-00000007;2'
-- Executing [h@default:1] AGI("Local/58600052@default-00000007;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- <Local/58600052@default-00000007;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
== Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600052@default-00000007;1'
-- Executing [h@default:1] AGI("Local/58600052@default-00000007;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- <Local/58600052@default-00000007;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
> Refreshing DNS lookups.
> ast_get_srv: SRV lookup for '_sip._udp.sipconnect.sipgate.de' mapped to host sipconnect.sipgate.de, port 5060
> ast_get_srv: SRV lookup for '_sip._udp.sipconnect.sipgate.de' mapped to host sipconnect.sipgate.de, port 5060
Vicidial Version: VERSION: 2.6-366c BUILD: 130402-2250 Server: 10.129.119.10
Asterisk: 1.8.19.0-vici-beta
Server Specs: 2*8core Opteron with vmware esxi 4.1. 8 CPUs and 6 GB RAM assigned to the VM in which this runs.
Load average on the server: below 0.2 (with 8 cores, so >>90% idle)
Codec used: ulaw
VoIP: pjsua from pjsip as softphone for the agents, then sipgate as carrier
OS: Zentyal 3.0 (a derivative of Ubuntu LTS 12.04), fully updated as of 11am or noon today (UTC+2)
Status: System has not gone into production yet, but outbound dialling did work at one point.
Well, thanks for making it this far, if I forgot anything please ask.
Steffen
Edit: the relevant snipped from extensions.conf:
- Code: Select all
; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,n,NoOp(fooa)
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,NoOp(foob)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,NoOp(fooc)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,NoOp(food)
exten => 8368,n,Hangup()
Also, I know that support for Asterisk-1.8 is in beta, but it seemed to work fine for me, so I would very much like to stick to the newer version. Plus, it's less risky to test things on a new install like mine than to have people upgrade their production systems
Would be most grateful for any ideas, pointers or suspicions!