[Asterisk-Users] An interesting call path observation..

[Philipp von Klitzing]

Hi!

> If SIP/U2 transferred the call to an extension that made use of the
> "switch statement"... What would the call path be?
> 
> Would the call traffic go from A1 in A2 back out of A2 to A3?
> ...or would it be "switched" and go directly from A1 to A3?

The theory - as far as I was able to find out - involves:

transfer=yes/no in iax.conf
canreinvite=yes/no in sip.conf

Next to that you might have codec issues involved, i.e. if the server A1 
has a different set of allowed codecs than A2 and A3. I am not sure if 
incompatible codecs result in a) an aborted call or b) a routed call even 
if transfer=yes.

Cheers, Philipp