[Asterisk-Users] codec negotiation

Eduardo Goncalves eduardo at acenet.com.br
Tue Dec 16 11:08:00 MST 2003


Hi list,

	I'm with a little problem on codec negotiation between a cisco827 and
asterisk.

	My sip.conf is like that: 

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm 
allow=alaw
allow=ulaw
;disallow=all

and cisco like that:

dial-peer voice 6 voip
 destination-pattern 0T
 session protocol sipv2
 session target ipv4:<asterisk-ip>
 dtmf-relay rtp-nte
 codec g711alaw
 no vad   
!         

	When I try to make a call, cisco shows codec g711alaw, but asterisk
shows codec g729A (i have the licenses) and there is no audio. When I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.

	The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok). 

	Why this negotiation problem happens?

Thanks
Eduardo



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