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

[Olle E. Johansson]

Lenny Tropiano / asterisk.org Mailing list wrote:

> 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?
It's a bug. This is the same problem as the Snom Line= problem and some
middleman box traversal errors. The SIP proxy should really send the same
Contact: header as the client provided.

By this time, this is a well-known hidden feature of chan_sip2. I've added
some code in that test version of the SIP channel to save the contact:
supplied by the client on register and using all of that on subsequent
invites. However, I haven't got feedback enough if that patch worked as
expected or not to port it back into chan_sip in CVS.

Sadly, I've been on holiday and during that period chan_sip and Asterisk
changed a lot (not any surprice) so I have to update chan_sip2 quite a lot
to compile with cvs_head. This will propably happen next week, after we
got the Astricon registration up and running :-)

I would very much like feedback on that part of chan_sip2 in order to
decide whether I should backport it to chan_sip in CVS.

/Olle