[Asterisk-Users] Sip call dropping

[Kevin]

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 at to-talk.com] 
Sent: Wednesday, July 02, 2003 12:56 PM
To: asterisk-users at 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
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,
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users