[Asterisk-Users] G.729 codec doesn't seem to work *even* after installing the license

Adam Hart adam at teragen.com.au
Thu Jul 15 19:38:50 MST 2004


try sip debug and see what each side is offering in codecs (make sure yo 
u have allow=g729

Walter Klomp wrote:
> Hi,
> 
> I am trying to post this again as I am getting no answers and the
> support at digium.com bounces...
> 
> (I have searched the whole list and can't find the answer either) 
> 
> I have installed a 5 user license for G.729 and want to route calls through
> Asterisk from my G.729 phone to Cisco AS5300 also using G729. 
> 
> Both Cisco and the phone connect using this codec if I do not force the call
> to go through *
> 
> However if I say canreinvite=no in the sip.conf for either of these gadgets,
> the call will fail with No compatible codecs!
> 
> I have bought a 5 user license just to try and fix this, apparently it
> doesn't work. I want to protect the Cisco gateway from unauthorized use, but
> still using a
> cost-effective codec such as g.723 or g.729 ? 
> 
> [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec
> Translator)
> 
>   == G.729 Host-ID:
> 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa
> 
>   == Found license 'G729-700241AB' providing 5 channels
> 
>   == Found total of 5 G.729 licenses
> 
>   == Registered translator 'g729tolin' from format G729A to SLINR, cost 2
> 
>   == Registered translator 'lintog729' from format SLINR to G729A, cost 12
> 
> 
> I was hoping by letting it ring out, I would get a voice-mail message, but
> that doesn't work either...
>  
> 
> *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No
> compatible codecs!
> 
>     -- Executing Dial("SIP/67.23.212.25-0814f830", "SIP/334|20") in new
> stack
> 
>     -- Called 334
> 
>     -- SIP/334-26f8 is ringing
> 
>     -- Nobody picked up in 20000 ms
> 
>     -- Executing VoiceMail("SIP/67.23.212.25-0814f830", "u334") in new stack
> 
>     -- Playing 'vm-theperson' (language 'en')
> 
>   == Spawn extension (default, 4084, 2) exited non-zero on
> 'SIP/67.23.212.25-0814f830'
> 
>  
> I have dropped this question at the asterisk user list some days ago, but
> it's being ignored... (or nobody has the answer)
> 
> Can anybody shed some light on this ?
> 
> Warmest Regards,
> 
> Walter Klomp
> 
> 
> 
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