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Dialling problem in asterisk(Solved)

PostPosted: Mon Mar 30, 2015 4:01 am
by ssam
we have installed vicidial from scratch on ubuntu 12.04. After 2 months working properly there's a new issue.
When dialling manually via softphone the calls once go once don't. it's irregular. I guess the problem is in asterisk. what's wrong with it ?
VERSION: 2.10-16 BUILD: 141001-2200

Re: Dialling problem in asterisk

PostPosted: Mon Mar 30, 2015 10:58 pm
by ambiorixg12
Post your Asterisk CLI output while dialing.. and verbose level at least 5

Re: Dialling problem in asterisk

PostPosted: Tue Mar 31, 2015 4:53 am
by ssam
Output when no ring:
== Using SIP RTP CoS mark 5
    -- Executing [736620760@default:1] AGI("SIP/100-00000030", "agi://127.0.0.1:4577/call_log") in new stack
    -- <SIP/100-00000030>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [736620760@default:2] Dial("SIP/100-00000030", "SIP/36620760@8001,,tToR") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/36620760@8001
    -- SIP/8001-00000031 answered SIP/100-00000030
    -- Executing [h@default:1] AGI("SIP/100-00000030", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----1") in new stack
    -- <SIP/100-00000030>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---3-----1 completed, returning 0
  == Spawn extension (default, 736620760, 2) exited non-zero on 'SIP/100-00000030'

Another output of no ring:
== Using SIP RTP CoS mark 5
    -- Executing [736620760@default:1] AGI("SIP/100-00000034", "agi://127.0.0.1:4577/call_log") in new stack
    -- <SIP/100-00000034>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [736620760@default:2] Dial("SIP/100-00000034", "SIP/36620760@8001,,tToR") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/36620760@8001
    -- SIP/8001-00000035 answered SIP/100-00000034
    -- Executing [h@default:1] AGI("SIP/100-00000034", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----22-----20") in new stack
    -- <SIP/100-00000034>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -22-----20 completed, returning 0
  == Spawn extension (default, 736620760, 2) exited non-zero on 'SIP/100-00000034'


Output when ring:
== Using SIP RTP CoS mark 5
    -- Executing [736620760@default:1] AGI("SIP/100-00000032", "agi://127.0.0.1:4577/call_log") in new stack
    -- <SIP/100-00000032>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [736620760@default:2] Dial("SIP/100-00000032", "SIP/36620760@8001,,tToR") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/36620760@8001
    -- SIP/8001-00000033 answered SIP/100-00000032
    -- Executing [h@default:1] AGI("SIP/100-00000032", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----7") in new stack
    -- <SIP/100-00000032>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---9-----7 completed, returning 0
  == Spawn extension (default, 736620760, 2) exited non-zero on 'SIP/100-00000032'

Re: Dialling problem in asterisk(Solved)

PostPosted: Tue Mar 31, 2015 5:05 am
by ssam
The problem solved. It was from voip provider.