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 <[email protected]>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 < > [email protected]> 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 >> >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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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