We are trying to work with Asterisk to deploy a solution and ran in to an issue. To allow SIP OPTIONS (and calls) to pass correctly, we need to be able to update the contact host URI.
Is it possible to do this using CHAN_SIP instead of PJSIP and if so, can you please advise how we can do it?
Example of OPTIONS request as follows:
---
Reliably Transmitting (NAT) to X.X.X.X:5061:
OPTIONS sip:domain.tld SIP/2.0
Via: SIP/2.0/TLS X.X.X.X:5061;branch=z9hG4bK1d3f350b;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as55b84229
To: <sip:sip.domain.tld>
Contact: <sip:[email protected]:5061;transport=tls>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Wishful thinking
Date: Mon, 16 Mar 2020 12:43:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
No, not possible.
Possible on kamailio or opensips.