[SR-Users] domain not found in domain table

[Kamrul Khan]

Hi,
We have a Kamalio a configuration that forwards calls to a freeswitch server (adding record-route).  The SIP signaling becomes successful till Kamailio receives 200 OK with session description from our carrier. After kamailio forwards the 200 OK with session description to Freeswitch, Kamailio drops all the ACK and BYE coming from freeswitch. From Kamailio log we found it says,
Aug 26 18:20:15 tyran /usr/sbin/kamailio[17980]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:17 tyran /usr/sbin/kamailio[17979]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:21 tyran /usr/sbin/kamailio[17979]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:25 tyran /usr/sbin/kamailio[17980]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:29 tyran /usr/sbin/kamailio[17980]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:33 tyran /usr/sbin/kamailio[17979]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:37 tyran /usr/sbin/kamailio[17980]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:41 tyran /usr/sbin/kamailio[17979]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain tableAug 26 18:20:45 tyran /usr/sbin/kamailio[17980]: ERROR: <script>: 1.2.3.4 BYE sip:1.2.3.4:5063;transport=udp;lr=on -- domain "4.3.2.1" is not found in domain table
We are not sure why it can send all the other messages and cant find the domain only for ACK and BYE. Please help. Thanks in advanced.
> Date: Tue, 26 Aug 2014 12:02:50 -0500
> From: a_villacis at palosanto.com
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?
> 
> El 25/08/14 18:28, Alex Balashov escribió:
> > On 08/25/2014 07:25 PM, Alex Villací­s Lasso wrote:
> >
> >> However, I do not find an equivalent to bridge mode in the rtpengine
> >> command-line parameters.
> >
> > Bridging mode of this type is not supported by rtpengine.
> >
> If this is true, then mediaproxy-ng/rtpengine should not be announced in the Kamailio documentation (http://www.kamailio.org/docs/modules/4.1.x/modules/rtpproxy-ng.html) as a "drop-in" replacement. At the very least, this requires a documentation fix.
> 
> How would somebody implement the following scenario using rtpproxy or mediaproxy-ng/rtpengine ?
> 
> - Server with 2 or more interfaces, at least one of which is public, and at least one of which is private (LAN)
> - Public interface runs webserver that publishes web phone (SIP.js or similar) for websocket
> - Webserver runs kamailio with access to both public and private interfaces
> - Websocket managed by kamailio, for SIP.js signaling
> - Private interface gives access to LAN where at least one traditional SIP client (UDP port 5060) registers with kamailio
> - Phone call initiated through websocket should contact SIP client in private LAN after proper authentication.
> 
> Can this be done at all with current technologies? How?
> 
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