Shot in the dark here ... Do you have:
canreinvite=no Set in sip.conf for the SIP phones in question ? Ciao, -b Quoting Steve Foy <[EMAIL PROTECTED]>: > Hi, > > I've got a fairly working Asterisk setup, with a few minor glitches, one of > which is very very irritating. > > Sometimes, during a call, the remote end just drops off. We're using > software > SIP phones (SJPhone) connecting to * then out through analogue lines with > X100P cards. > > There is nothing in the logs and nothing on the console, the call just seems > to 'go away'! > > Can anyone shed any light on this? ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. _______________________________________________ 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
