[asterisk-users] Inbound Call Disconnect in 3 seconds

David @ULC ucoms2001 at gmail.com
Wed Jan 28 11:55:37 CST 2009


exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)



These above lines were missing from Extension.conf file.

So, now call is NOT getting disconnected but either party cant hear each
other.


On Wed, Jan 28, 2009 at 9:57 PM, David @ULC <ucoms2001 at gmail.com> wrote:

>
> My Inbound calls lands , but line get disconnect in exactly 3 secs.
>
> Here goes my extension.conf setting :
>
> [from-ipkall]
> exten => 901835,1,Ringing ; call ringing
> exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI
> exten => 901835,3,Answer ; Answer the line
> exten =>
> 901835,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-----LB-----SALESLINE-----901835-----Closer-----park----------999-----1-----TESTCAMP)
> exten => 901835,5,Hangup
>
> What could be wrong ?
>
> Can it be Provider Issue ?
>
> CLI Output :--------------------------
>
> > Channel SIP/cc101-0887f498 was answered.
> -- Executing MeetMe("SIP/cc101-0887f498", "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
> -- Executing Ringing("SIP/66.54.140.46-08874e68", "") in new stack
> -- Executing Wait("SIP/66.54.140.46-08874e68", "1") in new stack
> -- Executing Answer("SIP/66.54.140.46-08874e68", "") in new stack
> -- Executing AGI("SIP/66.54.140.46-08874e68",
> "agi-VDAD_ALL_inbound.agi|CIDLOOKUPRC-----LB-----SALESLINE-----901835-----Closer-----park----------999-----1-----TESTCAMP")
> in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
> == Parsing '/etc/asterisk/manager.conf': Found
> == Manager 'sendcron' logged on from 127.0.0.1
> -- Executing AGI("Local/192*168*000*002*78600051 at default-230b,2", "agi://
> 127.0.0.1:4577/call_log") in new stack
> -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
> -- Executing Dial("Local/192*168*000*002*78600051 at default-230b,2",
> "SIP/192*168*000*002*78600051 at sip64||tTor") in new stack
> -- Called 192*168*000*002*78600051 at sip64
> == Spawn extension (default, 192*168*000*002*78600051, 2) exited non-zero
> on 'Local/192*168*000*002*78600051 at default-230b,2'
> -- Executing DeadAGI("Local/192*168*000*002*78600051 at default-230b,2",
> "agi://127.0.0.1:4577/call_log") in new stack
> -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
> -- Executing DeadAGI("Local/192*168*000*002*78600051 at default-230b,2",
> "agi://
> 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)")
> in new stack
> -- AGI Script
> agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------)
> completed, returning 0
> -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
> -- Executing AGI("SIP/66.54.140.46-08874e68", "agi://
> 127.0.0.1:4577/call_log") in new stack
> -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
> -- Executing Dial("SIP/66.54.140.46-08874e68",
> "SIP/192*168*000*002*8600051 at sip64||tTor") in new stack
> -- Called 192*168*000*002*8600051 at sip64
> -- SIP/sip64-0888b218 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing Hangup("SIP/66.54.140.46-08874e68", "") in new stack
> == Spawn extension (default, 192*168*000*002*8600051, 3) exited non-zero on
> 'SIP/66.54.140.46-08874e68'
> -- Executing DeadAGI("SIP/66.54.140.46-08874e68", "agi://
> 127.0.0.1:4577/call_log") in new stack
> -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
> -- Executing DeadAGI("SIP/66.54.140.46-08874e68", "agi://
> 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)")
> in new stack
> -- AGI Script
> agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------)
> completed, returning 0
> == Manager 'sendcron' logged off from 127.0.0.1
>
>
>
>
> ______________________________________________
>
>
> Why is it that "Called 192*168*000*002*8600051 at sip64" ?
>
> Is it trying to call 8600051 as an USA number using sip64 and thats why is
> it hanging up in 3 secs ?
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090128/0a6c665c/attachment.htm 


More information about the asterisk-users mailing list