Admittedly I didn't read your SIP debug (on the mobile), but do you have reinvite=no set for the extensions and SIP trunks (providers)?
This sounds on the surface like a classic case of the Mondays. Erm reinvites I mean. <snip> 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. <SIP Debug Output> 7. MoH stops, 8. Office user gets no audio 9. Incoming call is silent, and then call is dropped 10. Office user gets fed up of saying ‘hello??!?’ and hangs up. </snip> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
