26 Aug
2014
26 Aug
'14
8:58 p.m.
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(a)palosanto.com
To: sr-users(a)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:
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?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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.