[asterisk-users] Asterisk NOT in the media path
Jonas Kellens
jonas.kellens at telenet.be
Wed Feb 29 06:25:42 CST 2012
On 02/24/2012 10:51 PM, Jared Geiger wrote:
>
>
> On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> 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
> *CLI>
> *CLI> core show channels
> Channel Location State Application(Data)
> SIP/Mast-000000 (None) Up AppDial((Outgoing Line))
> SIP/VOXBONEin-000000 980419 at VOXBONEin Up Dial(SIP/Mast/980419)
> 2 active channels
> 1 active call
>
> Peer VoxBone and peer Mast should re-invite and leave this
> Asterisk out of the media path on call answer.
>
> These are my SIP peer definitions :
>
> [VOXBONEin]
> type=peer
> host=XX.XX.XX.XX
> context=VOXBONEin
> disallow=all
> allow=alaw
> allow=gsm
> canreinvite=yes
> qualify=yes
> dtmfmode=rfc2833
>
> [Mast]
> type=peer
> host=XX.XX.XX.XX
> defaultuser=Mast
> secret=guessme
> disallow=all
> allow=alaw
> allow=gsm
> canreinvite=yes
> qualify=yes
> dtmfmode=rfc2833
>
>
> Am I missing a setting ? Using Asterisk 1.6.2.22
>
>
> 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.
What is it that I should be looking for in the SIP debug information ?
Is it in the SDP-body ?
Kind regards,
Jonas.
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