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

[Walter Klomp]

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