asterisk users - Jul 2004 - Asterisk does not disconnect SIP call

Hello everybody,
my situation is the following: I have an ISDN telephone connected to a 
HFC ISDN card on an asterisk server. The asterisk server is behind a 
NAT, but all the ports (i.e. 5060 and the range specified in rtp.conf) 
are forwarded to the asterisk machine. I am using the German SIP 
provider Sipgate.de. The sip commands show that I am registered properly 
with Sipgate.

My problem is that when I want to call via the Sip provider a real phone 
number (ISDN phone >> SIP), I get a ring tone. When I now decide to hang 
up (f.e. when nobody answers), the called telephone continues to ring 
forever. This error shows up:
"app_dial.c : 362 wait_for_answer: Unable to forward frame"

If the other party answers and I am the first one to hang up, the call 
sometimes does not get cancelled as well. The called party has to hang 
up first to really disconnect the call. This error is not yet 
reproducable, as I said, sometimes it works and asterisk hangs up correctly.

I am using the lastest cvs version of asterisk that automatically 
installs with the install script of the brifstuff von www.junghanns.net.

Any suggestions how to solve it?

Christopher

Try to install a newer cvs version of Asterisk CVS Head.
There were some bug fixes on 07/28/2004 for more than one file.
I personally use bristuff 0.1.0RC2k, I had made update with cvs update -A.
It was going very well (08/03/2004) besides one file
asterisk/res/res_features.c.
You have to correct the merge conflicts in it.

I can only recommend this, if you are not really going to use the parking
functionality.
It can happen, that merging + fixing res_features.c will actually break the
logic.

One the other way, I don't have any SIP hangup problem :)

Thomas
----- Original Message -----
From: "C.B." <equinox@bach-online.de>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, July 31, 2004 5:30 PM
Subject: [Asterisk-Users] Asterisk does not disconnect SIP call

> Hello everybody,
> my situation is the following: I have an ISDN telephone connected to a
> HFC ISDN card on an asterisk server. The asterisk server is behind a
> NAT, but all the ports (i.e. 5060 and the range specified in rtp.conf)
> are forwarded to the asterisk machine. I am using the German SIP
> provider Sipgate.de. The sip commands show that I am registered properly
> with Sipgate.
>
> My problem is that when I want to call via the Sip provider a real phone
> number (ISDN phone >> SIP), I get a ring tone. When I now decide to
hang
> up (f.e. when nobody answers), the called telephone continues to ring
> forever. This error shows up:
> "app_dial.c : 362 wait_for_answer: Unable to forward frame"
>
> If the other party answers and I am the first one to hang up, the call
> sometimes does not get cancelled as well. The called party has to hang
> up first to really disconnect the call. This error is not yet
> reproducable, as I said, sometimes it works and asterisk hangs up
correctly.
>
> I am using the lastest cvs version of asterisk that automatically
> installs with the install script of the brifstuff von www.junghanns.net.
>
> Any suggestions how to solve it?
>
> Christopher
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>