If Asterisk is in the public network, it will work. The problem is when Asterisk is behind NAT and one of the client is also behind the same NAT.
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of turby
Sent: Tuesday, February 14, 2006 6:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

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

 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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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