Hello!
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.
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