[Asterisk-Users] Re: Call does not terminate correctly

[ProvoCityPower]

This a re-submittal hoping for some input:
  We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our client gateway Vendor sees it:

  1.  The first call is initiated.  (CRCX)  The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately.  All other CAs that we've worked with do not instruct us to go to sendrecv mode until the number has been completely dialed.

  2.  The call is terminated when hung up.  The call agent responds to this, but it never tells us to delete the connection and we continue to stream audio.

  3.  The next call is attempted.  We are now, not in the state that the call agent thinks we should be in and we are streaming audio to a UDP port that is now closed since the CA tore down the first call.

  4.  The unit is rebooted. (The T2 is hard reset)  The RSIP that is sent to the call agent basically resets the state machine and now the T2 and CA are in sync.  

  I'm not sure why this is happening, but maybe Asterisk can help.  It's clearly something in their code, but I can't really tell any more than that.

  Our sequence of events:

  1) Made first phone call to cell phone. Call was successful left it on for a few minutes. Tried punching all kinds of digits while on the call. Hung up.

  2) Made second call. Picked up handset, was receiving dial tone. Tried first digit and received the error (buzzing sound from the handset) . The digit tone goes haywire and repeats itself over and over again (I think this is what creates the buzzing tone).  Tried to make call while this was taking place. Hung up. 

  3) Reset T2.

  4) Made three-four more additional calls all worked after resetting T2. 

  Any input would be greatly appreciated.

  Thanks,
  Jeff






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