[asterisk-users] How to link 2 existing calls

[Moises Silva]

Not sure if it can help you, but check this patch:

http://bugs.digium.com/view.php?id=5841

Is for a new application called "Bridge" meant to bridge 2 channels.

On 8/7/06, Leon Sun <leon at timestelecom.ca> wrote:
> Hi,
>
> I searched web for few hours and couldn't find any solution about linking 2
> calls from Asterisk. This is scenario.
>
> 1. A call has been connected from A pstn gateway to my Asterisk waiting with
> music.
> 2. Meanwhile, B call has been connected from B pstn gateway to my asterisk
> waiting with music.
> 3. My asterisk has an application that issues a request to link A call and B
> call.
> 4. Asterisk should issue a re-invite to both A and B gateway and let them
> exchange RTP directly. Asterisk should still be working as SIP proxy to
> collect signaling(like bye).
>
> Would please anyone suggest how to do step 3 and 4? I wouldn't prefer
> conference room type since I like RTP packets go through gateway directly.
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Hadley Rich
> Sent: Sunday, August 06, 2006 1:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Ring Groups
>
> On Monday 07 August 2006 06:36, Chris Hembrow wrote:
> > I am new to asterisk, and learning as I plod along. Currently, I am
> > trying to work out how to create a ring group without using AMP.
>
> You should check out the book - 'Asterisk: The Future of Telephony' -
> published by O'Reilly it's available to buy or download. It will give you a
> good starting point.
>
> > I set my inbound line to ring multiple lines by using
> > Dial(SIP/101,SIP/102) but when I answered the call, the lines which
> > didn't answer became locked with no dialtone, as though on a call.
>
> That dial line should be Dial(SIP/101&SIP/102) - the comma (or a pipe, |)
> separates what to dial from the options to the dial command. typing 'show
> application dial' from the Asterisk CLI will get you all the gory details.
>
> > I thought that setting up a ring group might help, but could only find
> > references to creating them through AMP.
>
> 'Ring Group' is just an AMP term, you are going about it the right way
> above.
>
> HTH
>
> hads
>
> --
> http://nicegear.co.nz
> New Zealand's VoIP supplier
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