similar to: RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel1.2.3

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kevin P. Fleming > Sent: Tuesday, January 31, 2006 3:41 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] RE: [Asterisk-Announce] Asterisk 1.2.4 and > Zaptel1.2.3 > > Damon Estep

Does anyone know what date this memory leak was introduced and/or how to check source code for it? I am running a pre-1.2 CVS head version and would like to know if the potential problem exists. > -----Original Message----- > From: asterisk-announce-bounces@lists.digium.com [mailto:asterisk- > announce-bounces@lists.digium.com] On Behalf Of The Asterisk Development > Team > Sent:

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

Hi all, I just updated zaptel and asterisk, but after doing CVS update of asterisk (from CVS-HEAD 2005-08-10 17:10:53) I got a file conflict message conflicts: ast_expr2.h ast_expr2f.c I ignored them, and then tried to compile asterisk: make clean, make , make install. But then the problem shows again, leaving this message: ast_expr2f.c:1784: warning: no previous prototype for

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c -> ast_expr2f.o [CC] ast_expr2.c -> ast_expr2.o [CC] strcompat.c -> strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o -> aelparse aelparse.o(.text+0x3029): In function `ael_yylex':

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

In the real world (or at least in my world) we use undersubscribed internet connections that come with a service level agreement (SLA) that guarantees that the jitter, delay, and packet loss with be within defined parameters in the service agreement. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a

Don't feal bad about not reading. I yell at my 10 y.o. about it all the time. READ, NO more TV, READ!!! > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 10:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 1:48 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the > sametimewithdifferentcaller ID number? > >

The issue I have had with all other FC3 kernels apart from the 2.6.9 one was that the zaptel build would throw lots of warnings up. This would have the knock on of hanging the system, spinlock I think the problem was, on a modprobe -r. Lee ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Tuesday, January 31, 2006 8:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels > atthesametimewithdifferentcaller ID number? > > > >

Hi All, I am trying to upgrade an old Asterisk installation to 1.4.2 (it's currently running CVS-08/02/04-15:15:26) but have hit a couple of problems. The first was easily fixed. I got "storage size of sin isn't known" errors whilst compiling streamplayer.c, but after seeing http://bugs.digium.com/view.php?id=4908#32012 I manually added "#include <netinet/in.h>"

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

Below is my try of installation to the latest CVS. I have not updated 3 months. What do I miss since last time? bye Ronald /usr/local/src/asterisk # make clean; make update; make install build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp