similar to: Problem with Digium 400P and asterisk 1.4

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

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 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

Thank you, very interesting! As I understand the Digium card is used as a interrupt source for Asterisk? Is there a diagnostic tool available ? Anybody else experienced a simmialr problem? Thank you! HB > From: > covici at ccs.covici.com > Date: > Sat, 12 Dec 2009 19:04:23 -0500 > To: > Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at

I am using asterisk CVS 10842 and a TDM 400p withanfxs and an fxo module and when I dial the fxs channel it rings for a second and then says no answer after 20 seconds. I also have the latest Zaptel drivers. Here is a log snippet. Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Executing Dial("SIP/15712523171-09c1", "ZAP/1|20|trwW") in new stack Feb 25 00:53:11 VERBOSE[2015]

Hi. I am using 1.2.5 and I have an extension using a zap fxs channel on a 400P Digium card. Now when thatextension is dialed with a timeout of 20 seconds it rings for about half a second and then the log says noone picked on after 20000 seconds and so it goes to voicemail. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do

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

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

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

Has anyone figured out how to power a Digium TDM 400P card in a Dell 1750 server? I opened the server and noticed that there is no access to 4 pin power to power the card. Is there some sort of adapter that I need to power the Digium card in a Dell Server? I see that the 1750 is listed on the Wiki. How have others powered the TDM400P in a Dell 1750?

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

Hi. Using asterisk 1.4 svn 21112, when I try to use the mixmonitor feature I get the following in the log file. [Aug 11 09:22:54] WARNING[32057] file.c: Tried to write non-voice frame [Aug 11 09:22:54] WARNING[32057] channel.c: Failed to write data to channel monitor write stream After several sequences like that one it segfaults, I cannot find the core file, so I am doing something wrong, but I

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

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

The legacy PBX I'm working with does in-band disconnect notification -- it sends a * touchtone when the line is hung up. I've been trying to get this to work with Asterisk. I added a * extension to my menu context that plays "Goodbye" and hangs up. This works fine if I manually press *, but it never triggers when I hang up and the PBX sends it. I've plugged in an analog

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

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

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 =