2 sip calls bridged causes delay in voice

Discussion in 'General' started by dinaras, Mar 1, 2011.

  1. dinaras

    Jul 21, 2010
    Likes Received:
    Hello, i am trying to set up a click to call solution for a client.
    He wants to use sip providers in order to terminate calls. I use AMI in order to generate the first leg of the call and after this is answered call is sent to a dialplan which calls the second leg and the conversation can start.

    I have tried to implement this both with elastix 1.6 and elastix 2 (asterisk 1.4 and asterisk 1.6)

    I have also tried using reinvite and not using reinvite.

    The difference was that asterisk 1.4 would mess a bit the beginning of the call when reinviting and asterisk 1.6 dont. But apart from that i did not see any better performance regarding latency in the conversation itself.

    The dial plan i use is

    exten => s,1,Wait(1)
    exten => s,2,Background(pls-wait-connect-call)
    exten => s,3,Set(CALLERID(num)=XXXXXXXXXXXX)
    exten => s,4,Dial(SIP/voipprovider/XXXXXXXXXXXXX)
    exten => s,5,Hangup() ; No available circuits

    The same setup if using Dahdi trunks works with no delay at all.

    Do you have any idea how i can make this work, am i doing something wrong.

    I can send you more details and sip debug traces if needed

    Thank you very much in advance for your help

    Bets regards

  2. fmvillares

    Sep 8, 2007
    Likes Received:
    your voip provider is the delayed part...try getting a sip trace and you will see the delays...

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