[Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off

[James Dennis]

Make sure you are using the most up-to-date CVS. Also make sure that the sip
entry for your ATA has "canreinvite=no". I have a similar setup, except I'm
using a 7960 instead of a ATA and it works fine.

-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of Mikael Andersson
Sent: 10 April 2003 5:24 PM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off


At 14:02 2003-04-09 -0500, Brian Capouch wrote:
>Just an FYI.  I had had a previous problem using an ATA186 to make
>outbound calls over PSTN link.  Calls, virtually all of them, would 
>randomly cut off sometime in the first 6-8 minutes.
>
>Since I constantly upgrade I don't know if it was fixed by an upgrade 
>or
>my removing "callprogress" detection in the conf.
>
>But with latest CVS, and callprogress turned off, the problem is back.
>Seems to affect all calls after some random period of time.
>
>Can send debug info if necesssary; nothing of note shows on CLI; just
>shows the other side hanging up.
>
>Thanks.
>
>B.

I have a similar bug;

my config:

PSTN -> SIP Provider -> Asterisk -> ATA186


Incoming calls from PSTN to ATA doesn't generate a ringing tone, but the 
ATA rings.. When I answer on the ATA the call is cut off ?

Any suggestions ?

/Mike


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