On 02/23/2012 01:48 PM, Jonas Kellens wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 & B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider >>> Asterisk A1 >>> Asterisk B1 & Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in
the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my Asterisk A1
(which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-0000000e answered SIP/VOXBONEin-0000000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-0000000d and
SIP/Mast-0000000e
This indicates that it *is* working. Asterisk has setup a 'native' RTP
bridge between these two call legs. If they accept the re-INVITES that
are sent, then the media will flow directly between them.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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