[Asterisk-Users] Transfer

[Philipp von Klitzing]

Hi!

> Can * transfer call if I use canreinvite=yes in sip.conf?
> Can * start "automon" (recording) if I use canreinvite?
> 
> If answers are no, then which one did you chouse for your configuration? 
> Do you use "canreinvite=yes" so you can't do those stuff or you don't 
> use this so you have high processor load?

It is not only re-invite that determines what happens to your media path, 
there are also Dial() arguments like t,T,w,W (and possibly some more) 
that can force it go through Asterisk. The same applies to codec 
settings, i.e. if you need Asterisk in between to transcode e.g. from 
g729 to alaw then obviously the rtp stream has to go thru Asterisk.

Next to that: Try to switch both your phone and your Asterisk config to 
dtmfmode=info (SIP INFO) and see if automon recording will work that way 
even if you have canreinvite=yes - it could work since in this case DTMF 
is transmitted as SIP message; I have to admit that I am not 100% sure if 
with canreinvite=yes Asterisk will also be completely cut off from the 
SIP signalling stream, but I think it'll still be in the loop - haven't 
tried it myself.

For your transfer question: You'll have to use t or T in Dial in order to 
permit transfer, which in turn means your rtp traffic will be forced thru 
Asterisk no matter what your canreinvite= settings looks like.
You might want to look at if and how your SIP phone supports native 
transfer by itself.

Cheers, Philipp