[asterisk-users] Inbound Call Disconnect in 3 seconds

David @ULC ucoms2001 at gmail.com
Wed Jan 28 11:06:30 CST 2009


But we are using Vicidial and we got that file with the CD.
Manual says so to write below lines in extension.conf file :

_________________________________________________________________________________________________________
Now we need to set up the dialplan to send the calls coming into this number
to go to VICIDIAL.

The following lines need to be added to your dialplan(the extensions.conf
file) by your System
Administrator:
exten => 8002277655,1,Ringing    ; call ringing
exten => 8002277655,2,Wait(1)    ; Wait 1 second for CID delivery from PRI
exten => 8002277655,3,Answer     ; Answer the line
exten => 8002277655,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-----LB-----
SALESLINE-----8002277655-----Closer-----park----------999-----1-----TESTCAMP)
exten => 8002277655,5,Hangup
th
NOTE: no line break on the 4  line.
    Then you need to reload Asterisk for this to go into effect.
_________________________________________________________________________________________________________


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/20090128/50372f2c/attachment.htm 


More information about the asterisk-users mailing list