[Asterisk-Dev] SIP Peer handling for outbound calls

[Kristopher Lalletti]

Hey everyone,

 

I've been poking at the chan_sip.c for a little while now, in an attempt to
implement a first prototype of a SIP nat proxy hack.

 

Up to date, I've got my asterisk to register on my SIP provider (that was
the easy part) through their NAT proxy, since the registration procedure was
self-contained and pretty-much simple.

 

However, I'm attempting to implement the same hack (using an alternate
hostname/port to send the IP packets without altering SIP messages) for
outbound calls, however, I have yet to identify the source in the code where
the SIP message is actually transmitted, or, where the destination host/peer
are defined to then be delivered. 

 

For a while, I was lead to the impression that is was in the create_addr()
function, however, that function only seems assembles the sip message, while
copying data from the possibly found peer to the pvt structure.

 

Anyone have some pointers, or a description of the process when a call is
initiated from asterisk towards an outbound SIP peer defined in the sip.conf
? All I want is to send the SIP messages to an alternate host/port rather
than the ones defined in the host= and port= parameters of the sip.conf

 

Thanks

Kris

 

Ps: My C is functional, but not great; I've succumbed too much to
framework-based languages.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20040619/e4255ece/attachment.htm