On 02/24/2012 10:51 PM, Jared Geiger wrote:
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens <[email protected] <mailto:[email protected]>> 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@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 1.6.2.22The 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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