[Asterisk-Users] Dial command to connect two channels and bypass asterisk server

[turby]

this is not usefull for public enviroment. clients behind nat does not
work...
turby

 
  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nitin Gupta
Sent: Tuesday, February 14, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial command to connect two channels and
bypass asterisk server


thanks for the information Peter, its really helpful. Also I have one more
question - do you have any idea how many such simultaneous calls can an
asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)? 
 
Thanks,
Nitin

 
On 2/13/06, Peter Fern <pete at keypoint.com.au> wrote: 

You can enable this on a per-peer basis with:

sip peers:
canreinvite=yes

iax peers:
notransfer=no 

Check the iax.conf.sample and sip.conf.sample files for usage.

Nitin Gupta wrote:

> Hi I was wondering if its possible to make Dial command bridge two
> channels and after bridging bypass asterisk, so that the voice doesn't 
> need to pass through my asterisk server.
>
> For e.g., I have a user dialed in and he verifies himself and then
> dials an international extension, after the call connects I don't want
> the call to pass through asterisk server anymore. Is there any command 
> already there for any particular channel type?
>
> Thanks,
> Nitin
>
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