Lately, I have been experiencing unexpected hangups just when the a call has been established. This effects a small percentage of all calls coming from sip phone which are terminated on a zap pri channel. I turned on sip and pri debugging and it almost looks like the ACK message coming back from the sip agent in response to the "200 ok" message from the asterisk box which signaled the successful call setup would trigger a DISCONNECT message on the zap pri side. Asterisk spits the following line on the console just before issuing the DISCONNECT:
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peeerstate Connect Request I suspect this might have to do with the sip agents (all Grandstream ATAs/phones) as not all my users are affected. Has anybody of you seen this before? Thilo _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
