Are the 2 SIP UA's configured for the same codec?
Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 ----- Original Message ----- From: "Kevin" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 02, 2003 11:05 AM Subject: [Asterisk-Users] Sip call dropping > I'm having an issue with a connection between two sip phones, specfically sjphone, The two phones connect just fine, but after about 5 sec the call is dropped. On a side note a call does'nt got dropped between sip/sjphone and the outside line with a wx100p card. The communcation is on a full 100mbit network. I have a text file of the debug output of the call. > > Kevin, > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
