Perhaps do this instead?
allow=g723
allow=g729
disallow=all
On 06/29/2011 05:57 PM, Ernie Dunbar wrote:
> 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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>
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