similar to: FXO Hangs up automatically

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Hi All; To overcome the echo problem, what mainly I have to do in the configuration other than the following line in the system.conf under dahdi directory? echocanceller=mg2,1-16 1) How can I know if the digium card supporting echo cancellator? 2) If I am getting a message in the consol that unable to enable the echo cancelator, then what does it means? The hardware is not supporting echo

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading "The future of telephony" and this [1] document looking for information about

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]

Hello I setup Asterisk@Home with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can

Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP & IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI> module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550

Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. The power cable is plugged into the card, the port is configured to use fxo-signalling. Also, immediate=no. Here's the files: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand

after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI> module load chan_dahdi.so receive the following: signalling must be specified before any channels are. CLI> Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling Error[4663]: chan_dahdi.c:10946 build_channels: Unable to reconfigure channel '1' Error[4663]:

I have had a TDM400 with 2 FXO and 2 FXS working for ages (>12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg -vvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03:

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected