similar to: asterisk 1.4 segfaults when trying to use mixmonitor

Hi. I have been using asterisk-1.6.2 and if I update the version -- using svn -- to around May 19 or after, when I dial a digit on my fxs port which is on an X400p card, asterisk seg faults. If I go back before about this date, this problem does not occur. The dahdi version is svn 7445. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 15:44, covici at ccs.covici.com wrote: > > Rowland penny <rpenny at samba.org> wrote: > > > >> On 29/12/15 13:59, covici at ccs.covici.com wrote: > >>> Hi. I am having problems accessing subdirectories on a samba share. I > >>> am using windows 10 build 10586 and linux kernel

Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 13:59, covici at ccs.covici.com wrote: > > Hi. I am having problems accessing subdirectories on a samba share. I > > am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba > > 4.2.7. I have two shares, one called audio and the other called > > myshare. I cannot access the subdirectories

Hi. I am having problems accessing subdirectories on a samba share. I am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba 4.2.7. I have two shares, one called audio and the other called myshare. I cannot access the subdirectories in the myshare share. Here are the definitions. [myshare] comment = root directory path = / #fake oplocks = yes writable = yes printable =

OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an asterisk from the other end of a call, I here it forever till the call hangs up. Looks like it starts the vldtmf, but never ends it from the logs. I am using Digium 400P rev I with one fxs and one fxo module. Any way around this one? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How

Hi. I am having a problem with asterisk 1.4 and flashing a zap fxo line on a tdm 400p. in 1.2 I was able to type *0 after flashing the hook on my extension if it were talking to a zap fxo line, the fxo would flash and I could pick up call waiting or whatever -- now in 1.4 this does not work. Do I have to do anything in features.conf to fix this or what? Thanks much. -- Your life is like a

Hi. Ever since I bought my Digium 400P with 1 FXS and 1FXO module, once in a while I hear what sounds like a touchtone in my ear on a phone hooked up to the FXS module. This was not heard by the other side, and although it was annoying, it was not too much of a problem till I was using the asterisk 1.4 (rev 46317) and the beta of zaptel 1.4 (rev 1536). Doing this, the toutchtone noises once

Hi. I was using asterisk 1.2 on a box with sip phones attached and a long distance T1 line as the phone provider. We did a successful test of *1 allowing one-touch recording as set in the features.conf. Because of deadlock issues I decided to try 1.4 (latest svn as of yesterday) and the deadlock went away, but when we tried to use the *1 it was sent over the bridged channel rather than being

The minute asterisk tries to execute an agi, it gets utils.c write error broken pipe and so hangs up the call. Anyone know what is going on? I am using kernel 2.6.27 with dahdi trunk if that makes a difference. thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at

Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com

What procedure did you follow to revert back to the old version? It sounds like your binary has been revereted, but the modules it needs to load are still the 13.24.0-rc1 modules... --- Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian 8 to DEbian 9 to get a more recent operating system and

On 29/12/15 15:44, covici at ccs.covici.com wrote: > Rowland penny <rpenny at samba.org> wrote: > >> On 29/12/15 13:59, covici at ccs.covici.com wrote: >>> Hi. I am having problems accessing subdirectories on a samba share. I >>> am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba >>> 4.2.7. I have two shares, one called audio and

Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am

I am using a server with asterisk and I am trying to prioritize voip traffick -- I am a newbie at this traffic shaping, so please bear with me. I used the script below and what happens is thatall traffic in the bulk class stops after a couple of minutes. Also, should I include the ports for rtp in the filter statements with the ports 5061 and 4569? Note I have a fairly big pipe -- supposed to

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici