This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729
However, the Dial application gives the following error:
-- AGI Script Executing Application: (DIAL) Options:
(SIP/t564/1XXXXXX4332,,HR)
== Using SIP RTP CoS mark 5
[Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio
format found to offer. Cancelling call to 1XXXXXX4332
-- Couldn't call t564/1XXXXXX332
== Everyone is busy/congested at this time (0:0/0/0)
I've checked to ensure that both formats are loaded into Asterisk:
voip2*CLI> module show like 729
Module Description
Use Count
format_g729.so Raw G729 data 0
1 modules loaded
voip2*CLI> module show like 723
Module Description
Use Count
format_g723.so G.723.1 Simple Timestamp File Format 0
1 modules loaded
So I'm at a bit of a loss as to why Asterisk is complaining that
there's no audio format found to offer.
----------------------------------------------------------------
This message was sent using Lightspeed.ca's Advanced Webmail.
--
_____________________________________________________________________
-- 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
