Asterisk not disconnecting after hanup

Discussion in 'General' started by mobzone, Apr 4, 2010.

  1. mobzone

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    Hi

    We am looking to fix a small issue on my elastix setup.. We currently are experiencing a problem where asterisk is not disconnecting calls which have been hung up by the user. All calls come in via inbound sip trunks and and are routed via outbound sip providers. We are using a2billing 1.6 with Elastix and the issue has just started occuring in the past few weeks. Our provider has changed the way the caller ID comes into the system and we had also changed the date/time of the server via the Elastix interface. The issue started at around the same time as the changes described had been made.

    I am not sure if anyone else has experienced this. It had not occured previously and the sudden occurance of this issue has us somewhat perplexed. I know that we can set RTp timeout however we need to find the solution to the problem so as not to affect the billing in a2billing

    Any help would be appreciated
     
  2. fmvillares

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    its a problem of your provider, not asterisk related as asterisk hungs up when it releases a channel sip in this case...in the case of the provider if they dont release the far end asterisk will not hung up...it happens a lot with voip garage carriers and cheap itsp´s....
     
  3. mobzone

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    Hi

    Thank you for the response. I have contact my inbound provider who has provided the following response

    "We currently have in place a timer which will drop any call that has no RTP in either direction for 30 seconds.

    This timer should then always send a disconnect/BYE from us even if the call is not disconnected properly for whatever reason."

    I am still experiencing this issue and dont know how to resolve it.

    Could it be my termination provider? I am using a Betamax wholesale termination i.e Myvoiptraffic.com

    Could this be the problem...

    Anyone has the correct sip setting to use?Could that be the problem?

    The other thing is that the issue started when I changed the time of my server and rebooted..Could that be a problem and do I require a hard reset?
     
  4. fmvillares

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    there is no settings for rtp disconnect or to disconnect sip sessions in sip conf...only timers...nat and other things...
    its your provider...happens a lot...you could also put a Dial max time restriction in a manual dial settings of dialplans...but if you dont know how to program extensions by hands....you re fried..for the moment
     

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