vicidial.org

I enabled the on hook feature when phone rings and i received the phone call got disconnect cli is given below

any suggestion


[Apr 14 12:51:21] -- Executing [8884605739@trunkinbound:1] AGI("SIP/sale-00000466", "agi-DID_route.agi") in new stack
[Apr 14 12:51:21] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 14 12:51:21] ERROR[825]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Apr 14 12:51:21] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 14 12:51:21] -- Executing [99909*5***DID@default:1] Answer("SIP/sale-00000466", "") in new stack
[Apr 14 12:51:21] -- Executing [99909*5***DID@default:2] AGI("SIP/sale-00000466", "agi-VDAD_ALL_inbound.agi") in new stack
[Apr 14 12:51:21] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Apr 14 12:51:21] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:21] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:23] -- Started music on hold, class 'default', on SIP/sale-00000466
[Apr 14 12:51:25] == Parsing '/etc/asterisk/manager.conf': [Apr 14 12:51:25] Found
[Apr 14 12:51:25] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 14 12:51:25] -- Executing [172*017*001*006*1226@default:1] Goto("Local/172*017*001*006*1226@default-60fc,2", "default|1226|1") in new stack
[Apr 14 12:51:25] -- Goto (default,1226,1)
[Apr 14 12:51:25] -- Executing [1226@default:1] Dial("Local/172*017*001*006*1226@default-60fc,2", "SIP/1226|60|") in new stack
[Apr 14 12:51:25] -- Called 1226
[Apr 14 12:51:26] -- SIP/1226-00000467 is ringing
[Apr 14 12:51:27] -- Stopped music on hold on SIP/sale-00000466
[Apr 14 12:51:27] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:27] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:27] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:27] -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)
[Apr 14 12:51:27] -- SIP/1226-00000467 answered Local/172*017*001*006*1226@default-60fc,2
[Apr 14 12:51:27] > Channel Local/172*017*001*006*1226@default-60fc,1 was answered.
[Apr 14 12:51:27] -- Executing [8331*10806*Y4141251210001137096*navdeep*1226@default:1] AGI("Local/172*017*001*006*1226@default-60fc,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 14 12:51:27] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 14 12:51:27] -- Executing [8331*10806*Y4141251210001137096*navdeep*1226@default:2] Set("Local/172*017*001*006*1226@default-60fc,1", "CALLTIME=14-Apr-2012-22-21") in new stack
[Apr 14 12:51:27] -- Executing [8331*10806*Y4141251210001137096*navdeep*1226@default:3] MixMonitor("Local/172*017*001*006*1226@default-60fc,1", "/var/ftp/31*10806*Y4141251210001137096*navdeep*1226-14-Apr-2012-22-21.gsm") in new stack
[Apr 14 12:51:27] -- Executing [8331*10806*Y4141251210001137096*navdeep*1226@default:4] Dial("Local/172*017*001*006*1226@default-60fc,1", "SIP/8331*10806*Y4141251210001137096*navdeep*1226@globaltech|55|o") in new stack
[Apr 14 12:51:27] -- Called 8331*10806*Y4141251210001137096*navdeep*1226@globaltech
[Apr 14 12:51:27] == Begin MixMonitor Recording Local/172*017*001*006*1226@default-60fc,1
[Apr 14 12:51:28] -- SIP/globaltech-00000468 is circuit-busy
[Apr 14 12:51:28] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 14 12:51:28] -- Executing [8331*10806*Y4141251210001137096*navdeep*1226@default:5] Hangup("Local/172*017*001*006*1226@default-60fc,1", "") in new stack
[Apr 14 12:51:28] == Spawn extension (default, 8331*10806*Y4141251210001137096*navdeep*1226, 5) exited non-zero on 'Local/172*017*001*006*1226@default-60fc,1'
[Apr 14 12:51:28] -- Executing [h@default:1] DeadAGI("Local/172*017*001*006*1226@default-60fc,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
[Apr 14 12:51:28] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 14 12:51:28] == End MixMonitor Recording Local/172*017*001*006*1226@default-60fc,1
[Apr 14 12:51:28] -- Executing [h@default:1] DeadAGI("Local/172*017*001*006*1226@default-60fc,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----1") in new stack
[Apr 14 12:51:28] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---3-----1 completed, returning 0
[Apr 14 12:51:28] == Spawn extension (default, 1226, 1) exited non-zero on 'Local/172*017*001*006*1226@default-60fc,2'
[Apr 14 12:51:29] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 14 12:51:33] -- Started music on hold, class 'default', on SIP/sale-00000466
[Apr 14 12:51:47] -- Stopped music on hold on SIP/sale-00000466
[Apr 14 12:51:47] == Spawn extension (default, 99909*5***DID, 2) exited non-zero on 'SIP/sale-00000466'
[Apr 14 12:51:47] -- Executing [h@default:1] DeadAGI("SIP/sale-00000466", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 14 12:51:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 14 12:52:01] == Parsing '/etc/asterisk/manager.conf': [Apr 14 12:52:01] Found
[Apr 14 12:52:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 14 12:52:01] == Parsing '/etc/asterisk/manager.conf': [Apr 14 12:52:01] Found


___________________________________________________________
Cluster | GoAutodial 2.1 Installer | VERSION: 2.4-309a | BUILD: 110430-1642 | No other hardware

Interesting that you experience failure with this feature on. But please confirm that this agent can receive calls when the feature is turned off.

In other words: Indicate some reason to believe that the problem is in this feature and not simply in your basic configuration.

Next, describe the method/path of the call which is being sent to the phone in question, and provide related configuration information for the path (ie: if it's an inbound call ultimately being routed to an external extension ... please share the DID settings and the related configuration for each step all the way to the external dial plan used to reach said phone ...).

Welcome to the party!

yes its working fine without enabling on hook feature

DID setting

DID Extension: 8884605739
DID Route: IN_GROUP
Extension: 9998811112
Extension Context: trunkinbound

Inbound Carrier
[sale]
type=peer
port=5060
allow=g729
insecure=invite
context=trunkinbound
host=x.x.x.x
defaultexpirey=120

Outbound carrier
[globaltech]
type=friend
port=5060
host=x.x.x.x
username=globaltech
secret=xxxx
qualify=no
insecure=invite
context=trunkoutbound

exten => _XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXX.,2,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M)})
exten => _XXX.,3,Mixmonitor(/var/ftp/${EXTEN:2}-${CALLTIME}.gsm)
exten => _XXX.,4,Dial(SIP/${EXTEN}@globaltech,55,o)
exten => _XXX.,5,Hangup

This recently happened to me in an opener/closer environment where the closer had a physical on-hook phone. The opener would convince the respondant to a sale and transfer to a closer to take the credit card info. The call would hangup when sent to closer. The phone would ring, he'd pick up the call and bam! ...the call is gone and the potential customer gone..... Unfortunately I did not get any debug as I was pulling my hair out trying to fix other stuff but I moved the closer to a soft phone (like the rest of them) with an off-hook phone.....

In my acse call is not disconnecting it actually not tranfered to ip phone . call continue in queue and ip phone ring again -2 bt wenever i pick the reiever its engage tone