[asterisk-users] Termination Question
B.Masoud @ SH
info at saudihome.com
Thu Nov 12 18:10:00 CST 2009
So how can I let A makes a PEER connection between B & C, and ONLY log the
call information?
Thanks.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <mailto:tareksawah at hotmail.com>
To: Asterisk Users <mailto:asterisk-users at lists.digium.com>
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: info at saudihome.com
To: asterisk-users at lists.digium.com
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.
<http://go.microsoft.com/?linkid=9690331&ocid=PID24727::T:WLMTAGL:ON:WL:en-U
S:WWL_WIN_evergreen:112009>
_____
_______________________________________________
-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091113/b0476543/attachment-0001.htm
More information about the asterisk-users
mailing list