[asterisk-users] Termination Question

[Karl Fife]

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: 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 

    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: 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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