I am working with this configuration:
PC <==> WebRTC/Asterisk <==> SIP Phone
I have a WebRTC interface with an Asterisk server (SipML5).
I send INVITE from my PC to the Asterisk (encapsulated in Websocket) and then, the Asterisk server transmits this INVITE with SIP to the destination.
In this configuration, I need to make calls to “urn:service:sos@IP”
I would like to modify the INVITE message, at the level of Asterisk, to:
- remove the prefixe “sip:” in the field “To” and in the Request-URI,
- be able to send an invite to “urn:service:sos@IP”.
For the moment, when I send the INVITE to “urn:service:sos”,
Asterisk adds “sip:” to my INVITE message and don’t copy the “:service:sos”.
So the INVITE look like this: " INVITE sip:urn@IP", and the Field “To” is the same.
I know I can not modify the INVITE with the function SipAddHeader.
But is there a solution that allow me to modify SIP headers at my convenience?