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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users