problem in autodialing for asteriskgui client 2.0

[asterisk_at_my_risk]

hello
we are having a problem in autodialing calls are getting disconnected
we are using asterisk version Asterisk 1.2.12.1
we tried all options from our side and realy dont know what to do now please help us

i am attaching the logs for auto calls getting disconnected
*******************************************************
Connected to Asterisk 1.2.12.1 currently running on sip (pid = 2238)
Verbosity is at least 21
-- SIP/SIPtrunk-09c516b8 answered Local/919911595825@default-5296,2
> Channel Local/919911595825@default-5296,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/919911595825@default-5296,1", "call_log.agi|8365") i n new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Spawn extension (default, 919911595825, 2) exited non-zero on 'Local/919911 595825@default-5296,2'
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c516b8", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c516b8", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c516b8", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/SIPtrunk-09c516b8", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on 'SIP/SIPtrunk-09c516b

[mflorell]

try this:
http://www.eflo.net/VICIDIALmantis/view.php?id=39

[asterisk_at_my_risk]

sir i spent some time on the link you gave and read other people issues similar to this on forum

what i am making my reading all those thing is
that in one of your blog you mentioned a issue .........
Local channel resolution delay fix
and issue in bug traker
0000039: SIP trunk/Local channel Masquerading

sir pleae give me a direction that wheter the issue of call getting disconneted in auto dialing is related to above things or not

[asterisk_at_my_risk]

hello

i introduced the silence.gsm file in my extension.conf but it seems to be still not working

exten => 8365,1,Playback(silence)
exten => 8365,2,AGI(call_log.agi,${EXTEN})
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,6,Hangup

there is one more silence.wav file in asterisk do i have to remove
that file or keep ot like that only

[cdoyle]

Your problem must be something other than the local channel masquerading thing, because each time you run the agi-VDADtransfer.agi, it's showing the SIP channel already. Though, I don't see the Playback(silence).. did you do an "extensions reload" ?

have you turned AGI output to BOTH in the servers table? that might provide more debug info in the asterisk console.

[asterisk_at_my_risk]

Hello

i have debugged the call again but the issue remain same

can i have some more hints to play around in my extensions.cong
will increasing of time for siience will help

Channel SIP/3003-09c5eb58 was answered.
Executing MeetMe("SIP/3003-09c5eb58", "8600051") in new stack
Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919811177880@default-d794,2", "call_log.agi|91981117 7880") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/919811412508@default-aec7,2", "call_log.agi|91981141 2508") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/919811177880@default-d794,2", "sip/919811177880@SIP trunk|55|0") in new stack
-- Called 919811177880@SIPtrunk
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/919811412508@default-aec7,2", "sip/919811412508@SIP trunk|55|0") in new stack


-- Called 919811412508@SIPtrunk
-- SIP/SIPtrunk-09c87190 is ringing
-- SIP/SIPtrunk-09c74978 is ringing
-- SIP/SIPtrunk-09c87190 answered Local/919811412508@default-aec7,2
> Channel Local/919811412508@default-aec7,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/919811412508@default-aec7,1", "silence") in new stack
== Spawn extension (default, 919811412508, 2) exited non-zero on 'Local/919811 412508@default-aec7,2'
-- Playing 'silence' (language 'en')
-- Executing AGI("SIP/SIPtrunk-09c87190", "call_log.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c87190", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c87190", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09c87190", "agi-VDADtransfer.agi|8365") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("SIP/SIPtrunk-09c87190", "") in new stack
== Spawn extension (default, 8365, 6) exited non-zero on 'SIP/SIPtrunk-09c8719 0'
-- Hungup 'Zap/pseudo-52266369'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/3003-09c5eb58

[asterisk_at_my_risk]

Thanks MATT our issue is resolve now we are thankful for you support
the issue were basicaly two
1asterisk PERL was module was not installed properly
2we were not using the o option in the DIAL command

other things were ok and our dialplan is working wihout silence file or with that

thanks to MATT and all other people who were working on that