[asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?

[Maxim Sobolev]

Johansson Olle E wrote:
> Let's say we have an outbound SIP call from Asterisk,
> adressed to a SIP proxy. The SIP proxy forks back to
> Asterisk for Voicemail after 30 secs so we have an
> incoming call with the same call ID as an outbound
> and it's not a loop. The other branch of the call goes
> to a cell phone, so that branches back to Asterisk to
> be forwarded over ZAP somewhere. Now we have
> three SIP calls in Asterisk with the same call ID and
> something that is very likely that it will happen.

This is very valid scenario. What I end up doing in Sippy B2BUA is 
treating any new incoming branch as a separate session and create 
separate UASes for all of them. Since Asterisk is essentially SIP UA it 
probably makes sense for it too.

As it was correctly pointed out it's job of the SIP UAC to BYE unwanted 
sessions when multiple 200 OKs arrive.

-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com