I set it under the sip profile for the box sending calls to asterisk. [BREKEKE] type=peer context=wholesale host=x.x.x.x nat=no canreinvite=no progressinband=yes dtmfmode=rfc2833 insecure=port disallow=all allow=g729
Thanks, Dave George -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Philipp von Klitzing Sent: Friday, June 11, 2010 5:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no ring back 180 with SDP Hi! > I tried no, yes and never in the sip profile for that carrier and it did > not make a difference. > > Look at "progressinband=" in sip.conf. Just to make sure: Maybe you forgot the SIP RELOAD? Are you 100% sure inbound calls arrive with the peer that you set progressinband for? Verify this using SIP DEBUG. Sip peer matching can be quite confusing in Asterisk. Philipp -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
