1. Brekeke Product Name and version:
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:
The the UA (Linksys) phone is ringing and responds with a 180 Ringing. When this message is passing through the BSS the record-route set is destroyed!
The 127.0.0.1 record route set is replaced with the ipaddress of the BSS. It's not very nice to touch some one else's record-route set!! 127.0.0.1 is used on the SIP PBX to loop through an initial step (on 127.0.01) to take care of billing and other stuff.
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.
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:
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:
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.