Posted: Wed Apr 09, 2003 7:02 pm Post subject: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off
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.
Posted: Thu Apr 10, 2003 4:23 pm Post subject: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off
At 14:02 2003-04-09 -0500, Brian Capouch wrote:
Quote:
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 ?
Posted: Thu Apr 10, 2003 5:02 pm Post subject: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off
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@lists.digium.com
[mailto:asterisk-dev-admin@lists.digium.com] On Behalf Of Mikael Andersson
Sent: 10 April 2003 5:24 PM
To: asterisk-dev@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:
Quote:
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 ?
Posted: Thu Apr 10, 2003 6:45 pm Post subject: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off
At 18:02 2003-04-10 +0100, James Dennis wrote:
Quote:
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.
I have the latest CVS.. and the ringingtone bug, appeard a few days ago
when I updated.
Also, I have a little config question. How do I set canreinvite=no on my
Sip-provider ?
I have it on the client (ATA)
I do not have any ISDN BRI card or any PSTN hardware in my *. I depend on
SIP provider.
Also I was wondering how to use about 200 incoming SIP lines ? from the
same provider.
Posted: Fri Apr 11, 2003 7:10 am Post subject: [Asterisk-Dev] CVS-04/09/03-13:12:31 SIP Calls cut off
On Thu, 10 Apr 2003, Mikael Andersson wrote:
Quote:
I have the latest CVS.. and the ringingtone bug, appeard a few days ago
when I updated.
I've lost the context - you don't get ringback when you dial out through
your SIP provider? Vonage by any chance?
I looked at someone else's trace and Vonage doesn't send a "180
Ringing". Apparently they also don't play ringback in-band.
Quote:
Also, I have a little config question. How do I set canreinvite=no on my
Sip-provider ?
[vonage]
type=peer
canreinvite=no
Presumably?
Quote:
Also I was wondering how to use about 200 incoming SIP lines ? from the
same provider.
I reckon that depends on the provider. Perhaps they just keep sending
calls to you anyway - I think Packet8 works this way. Alternatively,
you'll need to open lots of accounts and register= all of them.
In practice Packet8 uses G723.1 - which is a problem for *.
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