[OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

[Peter Kust]

I am currently testing an OpenSIPS/Asterisk combination with a GenBand eSBC (Quantix QFlex).

My basic architecture looks like this

Phone (Cisco SPA525G2) --> OpenSIPS proxy --> Asterisk Media Server
Asterisk Media Server --> OpenSIPS proxy --> GenBand QFlex eSBC (-->PSTN)

The GenBand is handling both the SIP and RTP protocols, which means the Asterisk Media Server is sending the RTP stream direct to the GenBand.

A problem arises on inbound calls (from PSTN through GenBand to OpenSIPS/Asterisk).  During the call setup the GenBand sends a SIP ACK message directly to my Asterisk server, which seems to be causing the Asterisk server to send the BYE message at the end of the call directly to the GenBand instead of via the OpenSIPS proxy.  The result is that the external call end point (i.e., my cell phone), never gets a BYE message and that call leg stays open.

In the OK message from the proxy to the GenBand, the Contact header contains the IP address of my Asterisk server, and not the proxy.  I am being told this is what prompts the GenBand to send to the Asterisk server and not the proxy.