[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts

[Irakli Natsvlishvili]

Hmmm... Folks, I beg you pardon, if I'm telling something which was said 
before, but actually I have not found this anywhere, neither on 
Voip-info.org or in several Asterisk's docs.

So, here is the statement:

If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them 
will ALWAYS go via Asterisk.

I.e. Asterisk WILL NOT issue Re-INVITE even if:

 1. Both UAs have canreinvite=yes in their SIP.CONF
 2. Both UAs have same codecs
 3. There are no t, T settings in Dial command.

I'd like to have a confirmation from * developers about this statement.

I.N.