[Asterisk-Users] Call does not terminate correctly

Darren McIntosh dmcintosh at optushome.com.au
Tue Dec 9 05:55:37 MST 2003


> Message: 4
> From: "ProvoCityPower" <jwilson at provocitypower.com>
> To: <asterisk-users at lists.digium.com>
> Date: Mon, 8 Dec 2003 20:18:12 -0700
> Subject: [Asterisk-Users] Re: Call does not terminate correctly
> Reply-To: asterisk-users at lists.digium.com
>
> 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.
>
I agree * shouldn't really go to sendrecv until the B party has answered the
call but I've assumed this is so treatment tones can be played (eg busy tone
seems to be sent via RTP)

>   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.
>

I don't see this behaviour in my setup. Does the call work on-net to another
mgcp endpoint? This is how chan_mgcp ver 1.31 clears down a call to the
asterisk milliwatt tone:
endpoint               asterisk
=================
ntfy hd         ->
                   <-    200ok
                   <-    mdcx recvonly
200ok        ->
                  <-    dlcx
250ok        ->

You don't mention how you are accessing the PSTN? Are you interworking a
couple of protocols here?

>   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. =20
>
>   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.=20
>
>   3) Reset T2.
>
>   4) Made three-four more additional calls all worked after resetting =
> T2.=20
>
>   Any input would be greatly appreciated.
>
maybe a trace might help.

darren





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