<br><br><div class="gmail_quote">On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens <span dir="ltr"><<a href="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:<br>
</div><div class="im"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On 01/20/2012 08:07 AM, Jonas Kellens wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
<br>
I want to place an Asterisk-server A in front of 2 other<br>
Asterisk-servers (B1 & B2).<br>
<br>
This first Asterisk-server A needs to send incoming calls to one of the<br>
2 available Asterisk-servers (B1 or B2) behind it.<br>
<br>
So I want the first Asterisk-server A to accept the call, and based upon<br>
some checks in the dialplan send the call through to one of the other<br>
Asterisk-servers (B1 or B2) which further handle the call.<br>
<br>
The first Asterisk-server A then needs to pull itself from the<br>
media-path. There's no further need for this Asterisk to stay within the<br>
audio-path.<br>
<br>
1. Is this possible ?<br>
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer<br>
definition of Asterisk B1 and Asterisk B2 ?<br>
<br>
So I have :<br>
<br>
Provider >>> Asterisk A1 >>> Asterisk B1 & Asterisk B2<br>
<br>
I want the audio to go directly from Provider to server B1 when the call<br>
has been set up.<br>
</blockquote>
<br>
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.<br>
<br>
</blockquote>
<br></div>
Hello again,<br>
<br>
this is currently not really working.<br>
<br>
I see on the Asterisk CLI that the call streams through my Asterisk A1 (which should stay out of the media path) :<br>
<br>
[Feb 23 22:24:47] -- Called Mast/980419<br>
[Feb 23 22:24:47] -- SIP/Mast-0000000e answered SIP/VOXBONEin-0000000d<br>
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-0000000d and SIP/Mast-0000000e<br>
*CLI><br>
*CLI> core show channels<br>
Channel Location State Application(Data)<br>
SIP/Mast-000000 (None) Up AppDial((Outgoing Line))<br>
SIP/VOXBONEin-000000 980419@VOXBONEin Up Dial(SIP/Mast/980419)<br>
2 active channels<br>
1 active call<br>
<br>
Peer VoxBone and peer Mast should re-invite and leave this Asterisk out of the media path on call answer.<br>
<br>
These are my SIP peer definitions :<br>
<br>
[VOXBONEin]<br>
type=peer<br>
host=XX.XX.XX.XX<br>
context=VOXBONEin<br>
disallow=all<br>
allow=alaw<br>
allow=gsm<br>
canreinvite=yes<br>
qualify=yes<br>
dtmfmode=rfc2833<br>
<br>
[Mast]<br>
type=peer<br>
host=XX.XX.XX.XX<br>
defaultuser=Mast<br>
secret=guessme<br>
disallow=all<br>
allow=alaw<br>
allow=gsm<br>
canreinvite=yes<br>
qualify=yes<br>
dtmfmode=rfc2833<br>
<br>
<br>
Am I missing a setting ? Using Asterisk 1.6.2.22<br>
<br></blockquote><div><br>The Asterisk server still stays in the SIP Signaling path of the call, just media does not flow through the server. You can verify this by running a SIP debug and looking at the media endpoints.<br>
<br>If you don't want the call to maintain in the server at all, you need to look into doing 302 redirects.<br><br>Regards,<br>Jared<br></div></div>