Hi everybody Hope I picked the right mailing list. If not, please tell me.
We've got a problem with call forwardings. It's exactly the same problem as described in bug 12708, which is resolved by now. Situation: Caller -> asterisk -> call forward to mobile (packet2packet bridge) Quote from original bug reporter: 'One issue that we have noticed repeatedly is that there is a large delay between when a call is answered and when voice traffic actually flows. The delay is also asymmetrical and of the scope of about 2 seconds. This is very noticeable as calling someone generally misses the entire greeting. Call flow essentially goes like this: start call -> ringing -> answered (other party start talking “welcome to company this is Cameron”) -> their voice flows 2 seconds later and we hear “ameron” If we talk they can't here anything either at the beginning.' My * version: 1.6.2 sip.conf: .... nat=yes canreinvite=no .... extensions.conf: exten => my_nr,1,Dial(SIP/my_mobile...@my_provider,,r) Thanks in advance for any help. Kind regards, tylmann -- _____________________________________________________________________ -- 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
