On Friday 27 Jul 2012, Mitul Limbani wrote: > I think its not inbound call its outgoing, and during call progress > the remote end events are not passing back to sip.
Possibly. I've faced problems identical to the OP's when trying to connect a SIP call to the PSTN without first Answer()-ing it. Regards, -- Raj -- Raj Mathur || [email protected] || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _____________________________________________________________________ -- 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
