Hmmm. Let me rephrase your question:
"Dear List: How do I make server b and c do what I want when I have no control
over b or c?"
Enough said.
-K
----- Original Message -----
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Thursday, November 12, 2009 6:45 PM
Subject: Re: [asterisk-users] Termination Question
That could work, but I have no control over server B, not server C !
From: [email protected]
[mailto:[email protected]] On Behalf Of Karl Fife
Sent: Friday, November 13, 2009 3:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Termination Question
I have no first-hand experience with the fussy idiosyncrasies, but the BIG
PICTURE is to have server A set up the call, and then "reinvite" the media
directly from B to C. The call control messages flow to server A, the media
goes directly. If you don't have "NAT traversal Kung-Fu", I suggest using
IAX2 over SIP.
-K
----- Original Message -----
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Thursday, November 12, 2009 6:10 PM
Subject: Re: [asterisk-users] Termination Question
So how can I let A makes a PEER connection between B & C, and ONLY log the
call information?
Thanks.
From: [email protected]
[mailto:[email protected]] On Behalf Of Karl Fife
Sent: Thursday, November 12, 2009 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Termination Question
...and with a packet switched transport layer, the 'hairpin' route through
A may create problematic levels of latency--latency that would perhaps NOT have
been problematic on a classic circuit switched route, so it's definitely
advisable to nail up a connection between b and c.
-K
----- Original Message -----
From: Tarek Sawah
To: Asterisk Users
Sent: Thursday, November 12, 2009 8:28 AM
Subject: Re: [asterisk-users] Termination Question
for the sake of bandwidth you are supposed to connect each two servers
together.. otherwise calls between B && C will have to go through A .
-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286 USA: +1 347 562 2308
--------------------------------------------------------------------------
From: [email protected]
To: [email protected]
Date: Thu, 12 Nov 2009 16:13:10 +0300
Subject: [asterisk-users] Termination Question
Hello,
I would like to know how the following scenario works:
I have 3 Asterisk servers, A,B & C, each one is located in a different
country.
Asterisk A is the main one, and both B & C are connected to it.
My question is, when a call is originated from B to C, it will have to go
through A, but does A makes a peer connection between B & C to eliminate
bandwidth and latency, or the call has to go through A ???
Thanks.
--------------------------------------------------------------------------
Windows 7: Unclutter your desktop. Learn more.
--------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
----------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
