[asterisk-users] call file challenge...
Positively Optimistic
positivelyoptimistic at gmail.com
Wed Jun 22 04:45:42 CDT 2011
Thanks for the response.. I just got the opportunity to try this with the
wait time adjusted to 15.. and got the same result...
[2011-06-22 04:42:47] NOTICE[19692]: pbx_spool.c:339 attempt_thread: Call
failed to go through, reason (3) Remote end Ringing
....so far I've been unable to identify what generates the "Remote end
Ringing" message...
On Wed, Jun 15, 2011 at 5:39 AM, DHAVAL INDRODIYA
<dhaval.it01034 at gmail.com>wrote:
> Hi,
>
> I think you need to update *waittime* parameter in .call file please put
> atleast 10 seconds.
> for more understanding please try to read
>
> *http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out*
>
> Regards
> Dhaval
>
> On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic <
> positivelyoptimistic at gmail.com> wrote:
>
>> Greetings!!
>>
>> We're getting some strange results using call files.. no matter the
>> technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason
>> (3) Remote end Ringing" message when attempting to originate a call from a
>> call file. Numbers changed to protect the innocent....
>>
>>
>>
>> using call file....
>> //------------CALL FILE------------//
>>
>> Channel: DAHDI/g1/918005551212
>> Callerid: 8002211212
>> WaitTime: 2
>> MaxRetries: 6
>> RetryTime: 8
>>
>> Context: xs-globx-ds3
>> Extension: 12564286000
>> Priority: 1
>>
>> //------------CALL FILE------------//
>>
>> //------------CLI SNIPPET------------//
>>
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 1)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 2)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:35:24] NOTICE[27177]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 3)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:35:34] NOTICE[27179]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 4)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:35:44] NOTICE[27182]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 5)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:35:54] NOTICE[27183]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 6)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:36:04] NOTICE[27185]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>> -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
>> xs-globx-ds3:1 (Retry 7)
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- PROGRESS with cause code 31 received
>> -- Hungup 'DAHDI/1-1'
>> [2011-06-15 01:36:14] NOTICE[27188]: pbx_spool.c:339 attempt_thread:
>> Call failed to go through, reason (3) Remote end Ringing
>>
>> //------------CLI SNIPPET------------//
>>
>> Software Version(s)
>>
>> Asterisk 1.6.2.16.1
>> DAHDI Version: 2.4.0
>> libpri version: 1.4.11.5
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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