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@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
Regards,
Jonas.
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