Hi to all. I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and I've some problem with outgoing calls: there is a big delay for bidirectional audio flow. By traces, I've observed that several 200 OK SIP messages are sent by my SIP Provider until ACK is riceved. Maybe the 200 OK messages sequence freezes Asterisk introducing delay for biderctional audio flow. Can anyone tell me if there is some option to set in order to manage sip messages time or similar? Moreover..when I attempt to make an outgoing call with option canreinvite=yes, Asterisk notifies the follow message? Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819b240', 10 retries! Can anyone tell me what it does mean and how to fix it? Thanks, -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************