Hi,
I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX
(security) to an IP phone which supports g729, and vice versa. Both Cisco
and the phone talk 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
didn't work. How can I route calls through the Asterisk as I want to protect
the Cisco gateway from being used without me knowing about it, 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
*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'
Appreciate your ideas.
Walter.
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