Hi,
When one trunk was trying to dial a number, it includes 011 prefix, then that trunk fails, it will go to the next failover trunk. However, the next dial strips out the 011... How can I change it so it doesn't do that?
This is the log:
Quote:
[May 15 16:52:42] NOTICE[25420]: channel.c:4176 __ast_read: Dropping incompatible voice frame on SIP/204.11.xxx.xxx-00031689 of format ulaw since our native format has changed to 0x8 (alaw)
-- SIP/voipvoip_new_out-00031688 is making progress passing it to SIP/204.11.xxx.xxx-00031682
-- Executing [852580xxxxx@custom-a2billing:3] AGI("SIP/204.11.xxx.xxx-00031689", "a2billing.php,1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/204.11.192.133-00031689> Playing 'music.gsm' (language 'en')
-- SIP/voipvoip_new_out-00031688 is making progress passing it to SIP/204.11.xxx.xxx-00031682
-- AGI Script Executing Application: (DIAL) Options: (SIP/hgc_out/011918049148000,60,RrL(2147483647:61000:30000))
> Limit Data for this call:
> timelimit = 2147483647 ms (2147483.647 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/hgc_out/011918049148000
== Everyone is busy/congested at this time (1:0/0/1)
-- AGI Script Executing Application: (DIAL) Options: (SIP/voipvoip_new_out/918049148000,60,RrL(2147483647:61000:30000))
> Limit Data for this call:
> timelimit = 2147483647 ms (2147483.647 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
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