<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#ffffff">
On 02/24/2012 10:51 PM, Jared Geiger wrote:
<blockquote
cite="mid:CAHuchRA+vF+W0c0sqkWmtKpNDMn1eVNySrunr0uSJpqZ3_506g@mail.gmail.com"
type="cite"><br>
<br>
<div class="gmail_quote">On Thu, Feb 23, 2012 at 2:48 PM, Jonas
Kellens <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
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: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
On 01/20/2012 08:07 AM, Jonas Kellens wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
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.</div>
</div>
</blockquote>
<br>
What is it that I should be looking for in the SIP debug information
? Is it in the SDP-body ?<br>
<br>
<br>
Kind regards,<br>
Jonas.<br>
</body>
</html>