similar to: what is the possible cause of maximum pbx stack exceeded

Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission

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

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

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

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

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 =

Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by

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

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

Hi. For several months now asterisk will mysteriously stop inserting records into cdr database. I am using mysql and the asterisk addons 1.6.2 to accomplish this. Sometimes there is a strange error about column names, but often there is no error, it just stops. I just have to restart asterisk to get things going again, so I am stumped as to what is happening, or even how to troubleshoot. I

I am using asterisk 1.6.0 and -- not all the time -- when a caller comes in and my ivrdials an extension, the ring he gets sounds like a modem handshake instead of the normal ring tone and it only sounds once even if the phone is not picked up. Anyone seeing this -- the logs look fine as far as I can tell. -- Your life is like a penny. You're going to lose it. The question is: How do you

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

Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated

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

Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

Hi. I hava a variable and in 1.6 I set the string variable to "" and it got the null string. In 1.8, it gets the quotes, I have to set it to nothing at all to make it get the null value. -- 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

Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question

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

Is the list going to be the same after sangoma take over digium? On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp <jcolp at digium.com> wrote: > On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > > that forum supposed to supersede this mailing list ? > > Both remain available but the community