Brekeke SIP Server 2.0.7.2/217
2. Java version:
1.5.0_11
3. OS type and the version:
Windows 2003
4. UA (phone), gateway or other hardware/software involved:
LinksysSPA942+SNOM360, Vegastream GW, Telepo SIPPbx
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 5
6. Your problem:
Linksys SPA 942 and SNOM 360 are on one side of Brekeke and Telepo SIPPBX+VegaStream router on the other side.
UAs successfully REGISTER with PBX through Brekeke.
An incoming call from PSTN/Vegastream passes PBX, which is adding a Record-Route set on the INVITE:
Code: Select all
INVITE sip:xmadu@42.80.210.197:5062 SIP/2.0
..
Record-Route: <sip:77.231.72.72;lr;transport=UDP;x-handler=RouterServlet;x-route-id=e2acf70050052f4b>
Record-Route: <sip:127.0.0.1;lr;transport=TCP;x-handler=RouterServlet;x-route-id=e2acf70050052f4b>
..
Code: Select all
SIP/2.0 180 Ringing
..
Record-Route: <sip:42.80.210.197:5062;lr>
Record-Route: <sip:77.231.72.72;lr;transport=UDP;x-handler=RouterServlet;x-route-id=e2acf70050052f4b>
Record-Route: <sip:42.80.210.197:5062;lr;transport=TCP;x-handler=RouterServlet;x-route-id=e2acf70050052f4b>
..
Can this behavior be prevented with some dialplan?
We do have an existing PBX that has trafic that cannot be affected by the dialplan or other global changes.