Routing Asterisk -> Proxy -> SIP phone for extensions

We have an asterisk PBX and are looking to insert a proxy in the signalling path between a SIP phone and the Asterisk.

The intention is to monitor the registration and signalling, we are not concerned with the RTP.

We have successfully redirected the registration request and responses using a proxy, but cannot seem to force the asterisk to route call signalling for extensions via our proxy.

Our understanding of RFC3261 was that adding a “maddr=x.x.x.x” line to the contact in the registration request would force the asterisk to route signalling via our proxy for that request but it does not seem to work.

Should this work? Or, is there another way to force inbound signalling for our extensions to be routed via our proxy? (Outbound routing from extension -> proxy -> asterisk) is handled by setting the proxy in the sip phone.


In proxy you mean SER or openser SIP proxy ?

I’m using to route sip phones to asterisk and rtpproxy for the rtp packages.

The SIP Proxy will be our own product built on the ReSIProcate stack. We have the ability to manipulate headers as required, but we seem unable to force the Asterisk to route INVITEs destined for an extension through our proxy to the extension.

Our aim is to monitor the signalling between any device and the Asterisk PBX, and also to provide third party call control via our proxy.