Hi All
I have a problem which I think is related to the codec but I'm not sure about that.
I have 2 SAIL PBX connected using a SAIL-to-SAIL trunk
Site B has the following at the console when a call is attempted.
pbx*CLI>
-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx:
> requested format = g729,
> requested prefs = (g729|alaw|ulaw|gsm),
> actual format = gsm,
> host prefs = (gsm|ulaw|alaw),
> priority = mine
-- Executing [202@internal:1] AGI("IAX2/sailpbx-8395", "selintra|OutCluster|202") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing [202@qrxvtmny:1] AGI("IAX2/sailpbx-8395", "selintra|Alias|/402 /403|202") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (/402&/403||)
[May 19 23:34:00] WARNING[9634]: channel.c:3025 ast_request: No channel type registered for ''
[May 19 23:34:00] WARNING[9634]: app_dial.c:1183 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
[May 19 23:34:00] WARNING[9634]: channel.c:3025 ast_request: No channel type registered for ''
[May 19 23:34:00] WARNING[9634]: app_dial.c:1183 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (2:0/0/2)
-- AGI Script selintra completed, returning 0
== Auto fallthrough, channel 'IAX2/sailpbx-8395' status is 'CHANUNAVAIL'
-- Executing [h@qrxvtmny:1] Hangup("IAX2/sailpbx-8395", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'IAX2/sailpbx-8395'
-- Hungup 'IAX2/sailpbx-8395'
pbx*CLI>
Site A has the following at the console when the same call is being attempted.
sail*CLI>
-- Executing [202@internal:1] AGI("SIP/401-b7e22848", "selintra|OutCluster|202") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing [202@qrxvtmny:1] AGI("SIP/401-b7e22848", "selintra|OutRoute|CCES PBX") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (IAX2/sailpbx@pbxsail/202||T)
-- Called sailpbx@pbxsail/202
-- Call accepted by xxx.xxx.xxx.xxx (format gsm)
-- Format for call is gsm
-- IAX2/pbxsail-11845 is circuit-busy
-- Hungup 'IAX2/pbxsail-11845'
== Everyone is busy/congested at this time (1:0/1/0)
-- AGI Script Executing Application: (Background) Options: (were-sorry)
-- <SIP/401-b7e22848> Playing 'were-sorry' (language 'en')
-- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
-- <SIP/401-b7e22848> Playing 'call-cannot-complete' (language 'en')
-- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
-- <SIP/401-b7e22848> Playing 'please-hang-up-and-try-again' (language 'en')
-- AGI Script selintra completed, returning 0
== Auto fallthrough, channel 'SIP/401-b7e22848' status is 'CONGESTION'
-- Executing [h@qrxvtmny:1] Hangup("SIP/401-b7e22848", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-b7e22848'
sail*CLI>
I should also mention that Site B has open & close times active. The above call was attemped during the closed period direct to a call group with direct extentions (for testing due to the time of night) back at site A.
The same call was placed with a mobile number in the same group with the same result except the mobile rang.
Is there something I am missing or is it that I just don't understand how the system works properly yet?