[Asterisk-Users] Sip call dropping

[Michael Kane]

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" <kevin at honeycomb.net>
To: <asterisk-users at 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, 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,
>
>
>
>
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