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,
Are the 2 SIP UA's configured for the same codec? Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 to-talk.com 508-295-2826 ----- Original Message ----- From: "Kevin" <kevin@honeycomb.net> To: <asterisk-users@lists.digium.com> 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, specficallysjphone, 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 > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users >
Sjphone is set for Remote preferences for "Codec Preference Selection" Do you I want it Local preferences? Kevin, -----Original Message----- From: Michael Kane [mailto:mkane@to-talk.com] Sent: Wednesday, July 02, 2003 12:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sip call dropping Are the 2 SIP UA's configured for the same codec? Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 to-talk.com 508-295-2826 ----- Original Message ----- From: "Kevin" <kevin@honeycomb.net> To: <asterisk-users@lists.digium.com> 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, > specficallysjphone, 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 > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users