SIP calls are not connected to each other. In the Asterisk architecture, the SIP "call" is connected to an owner channel. Each call has a pair of a "technology" driver structure (tech_pvt) - like IAX, SIP, H323, ZAP and they connect to a generic Asterisk channel that is the "owner".
So when Bob calls Alice, both on SIP devices, the incoming SIP call connects with an Asterisk channel that connects (bridges) to another Asterisk channel that connects to another outbound SIP call. SIP signalling always ends in Asterisk and is not forwarded in any way "through" Asterisk. So by reading chan_sip, you will not discover much, you need to dig deeper into the channel and pbx interface. In chan_sip, you can check the code for transfers that handle all these channels and redirecting them in various ways. Good luck exploring! /Olle _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
