[Asterisk-Users] Re: Transfer

[Tomislav Parcina]

In article <43AC21E4.28566.C77D21 at klitzing.pool.informatik.rwth-
aachen.de>, klitzing at pool.informatik.rwth-aachen.de says...
> 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.

Thank you for explaining this to me.

> 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.

I have found out that the problem is with Cisco phones (7905 and 7940). 
With ZAP (analog) phones and with Grandstream it works fine.

I'm opening new thread for this one => Cisco dtmf

> 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.

My phones support nativ transfer but I would like to implement this 
feature.


-- 

Tomislav Parcina
ime.prezime at email.t-com.hr