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Auto dial disconnects, when user attend the phone

PostPosted: Wed Jan 27, 2016 4:53 am
by saikumar.P
Hi Team,

I have checked manual dial is working perfectly, But auto dial call disconnected, when client attend the phone, Please find the below asterisk log details
Here XXXXXXXXXX is client phone no

-- Executing [8600051@default:1] MeetMe("SIP/8001-00000000", "8600051,F") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': == Manager 'sendcron' logge d off from 127.0.0.1
== Found
-- Created MeetMe conference 1023 for conference '8600051'
-- <SIP/8001-00000000> Playing 'conf-onlyperson.gsm' (language 'en')
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [XXXXXXXXXX@default:1] Dial("Local/XXXXXXXXXX@default-00000000; 2", "Dongle/dongle0/XXXXXXXXXX") in new stack
-- Called Dongle/dongle0/9841782122
-- Dongle/dongle0-0100000000 is making progress passing it to Local/XXXXXXXXXX@default-00000000;2
-- Dongle/dongle0-0100000000 answered Local/XXXXXXXXXX@default-00000000;2
-- Executing [8368@default:1] Playback("Local/9841782122@default-00000000;1" , "sip-silence") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- <Local/9841782122@default-00000000;1> Playing 'sip-silence.gsm' (language 'en')
-- Executing [h@default:1] AGI("Local/9841782122@default-00000000;2", "agi:/ /127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12---- -0") in new stack
-- Executing [8368@default:2] AGI("Dongle/dongle0-0100000000", "agi://127.0. 0.1:4577/call_log") in new stack
-- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=AGRIANEW ))


Thanks & Regards,
Sasikumar.P

Re: Auto dial disconnects, when user attend the phone

PostPosted: Sat Jul 23, 2016 7:21 pm
by williamconley
Did you change the IP of the server after installation?

Your dialplan entry for the carrier appears to be incorrect. The first line (XXXXXXXXXX@default:1) shows "Dial" as the command, but this line must always be the AGI script. I suspect you did not follow the manager's manual (available at eflo.net) when creating the carrier. The three lines for the carrier's dialplan entry are ALL necessary, the agi line, the dial line, and the hangup line. None can be skipped.