[asterisk-users] Termination Question

[B.Masoud @ SH]

That could work, but I have no control over server B, not server C !

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <mailto:info at saudihome.com>  

To: 'Asterisk Users Mailing List - <mailto:asterisk-users at lists.digium.com>
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: 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 




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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.

 

 


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