similar to: DAHDI hangup detection

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a

Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk

I've got a Digium Wildcard TDM410P with one POTS line and three extensions. One of the extensions is connected to a fax modem. This kind of works, but there's a gotcha. If I set faxdetect=incoming in chan_dahdi.conf, then incoming faxes do get routed to the modem and this all works, but outbound faxes fail, with a message like this: [Oct 29 21:57:09] WARNING[16732][C-00000000] app_dial.c:

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

Hello Unless I overlooked it, the "Asterisk Reference Information Version 1.6.1.6" at www.asterisk.org/docs doesn't include instruction on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P with a single FXO module www.openvox.cn/products/show.php?itemid=20&lang=2). I'd like to know... 1. How to start Dahdi manually. Is this the right way? modprobe

Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -rvvvv I see the port pick

Greetings, This may be a dumb question, but here goes. When I was on 1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to line supervision on my POTS lines using a TDM400P/TDM410P. Since upgrading to the DAHDI branches of 1.4 (SVN and 1.4.26.1), I've only been able to duplicate the success of the 1.4.21 functionality once. To test what I'm talking

I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX "hangs up" asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [s at macro-dialout-trunk:19] Dial("DAHDI/1-1", "DAHDI/g0/9239220,300,") in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the

Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

Hello everybody, I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0. Problem I face is the following one: CLI> module load chan_dahdi.so == Registered application 'DAHDISendKeypadFacility' == Registered application 'ZapSendKeypadFacility' == Parsing

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

Dear Supports, I was attempting to install BRI Card(OpenVox B800P) with wcb4xxp in NT mode .But I can not make it worked! Could you please give me some hints? Thanks in advance! Here are my environments: ------------------------------ CentOS-5.3 Kernel-2.6.18-164.el5 asterisk-1.6.1.6 dahdi-linux-complete-2.3.0+2.3.0 libpri-1.4.10.2 OpenVox-B800P(the first port is spliting into

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

OK, I know it's only just out today but this is what I get when compiling dahdi-linux. make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware' make -C /lib/modules/2.6.31.5-0.1-default/build

I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has

Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***********************Enviroment********************************** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ******************************************************************** In my case , I set prot 1