[Asterisk-Users] Dropped calls
Thilo Salmon
salmon at netzquadrat.de
Wed Apr 14 07:45:13 MST 2004
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
More information about the asterisk-users
mailing list