[asterisk-users] No audio format found to offer.

[Ernie Dunbar]

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.

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