[Asterisk-Users] SIP problem with asterisk

[Tjardick van der Kraan]

canreinvite=no

in the sip entry of the phone on the inside of the NAT.

See other posts on the list.

Just search with google and add site:lists.digium.com

Greetings,

Tjardick

----- Original Message ----- 
From: "George Lin" <glin at cosini.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, September 25, 2003 9:33 AM
Subject: [Asterisk-Users] SIP problem with asterisk


> 
> 
> HI List,
> 
> I have two SIP phones. one is 6002, w:
> 
> which is behind a NAT, and another is 5009 which has public IP .
> 
> When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and
> 5009 did hear from 6002.
> 
> And in the sip debug, I see following message
> 
> Sip read:
> SIP/2.0 481 CallLeg/Transaction Does Not Exist
> 
> 
> and I specified in sip.conf
> [6002]
> 
> type=friend
> host=dynamic
> nat=1
> qualify=yes
> 
> [5009]
> 
> type=friend
> host=dynamic
> nat=1
> qualify=yes
> 
> Can anyone help me what can be wrong ???
> 
> Thanks,
> 
> George Lin
> 
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>