[Asterisk-Users] Params on SIP URI REGISTER/INVITE

[Lenny Tropiano / asterisk.org Mailing list]

We're doing some SIP development and have a question on "additional parameters"
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).

What we're experiencing is the INVITE doesn't included these parameters
and they get dropped when the INVITE is sent to the 10.1.1.97 address.

Ideas?  Supported?  SIP Bug?

REGISTER sip:test1.mydomain.com SIP/2.0
Via: SIP/2.0/UDP10.1.1.97:5060;branch=z9hG4bKd1f1eb5acc28043b83a28ca2ee1e5f15,SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bk-8c166b93
From: "JohnDoe" <sip:2016 at test1.mydomain.com>;tag=6f0ecbcb3a5e62c4
To: "JohnDoe" <sip:2016 at test1.mydomain.com>
Call-ID: 6c4a901f-fb034a80 at 192.168.0.2
CSeq: 1 REGISTER
Max-Forwards: 69
Contact: "JohnDoe"<sip:8994f3ae412133e2056757d2fcccfbed5f9642da at 10.1.1.97:5060;maddr=192.168.0.2;clport=5061>;expires=3600
User-Agent: Sipura/SPA2000-1.0.15
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER
Content-Length: 0