hi, i have this topology pstn+(e1)asterisk1<->asterisk2<->sip client asterisk1,asterisk2 allow (g729,alaw) sip client prefer g729, then alaw can you someone describe codec negotiation when call for sip client arrive from pstn? (can i set g729 for calls from pstn? ) thanks --------------------------------------- Marek Cervenka =======================================
Hi, asterisk will negotiate codecs for both parties independently (use sip show peer <peer> and look for "codec order" entry), so, if you have prefered codec g729 for your sip phone/peer, asterisk will use them (regardles of codec setting for other party - if codecs does not match, asterisk will try to transcode between) imho ;-) PJ