I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles. If Asterisk knows the trunk only supports ulaw why would it offer g722 to the phone.
Ryan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
