similar to: Hangups transferring call from Intertel system

Hello I have installed my new E100P card in the same machin with two TDM04B cards. I have followthe instructions on http://www.digium.com/index.php?menu=configuration#T_E100P_PRI This is my zapata.conf span=1,0,0,cas,ami fxsls=1-15,17-31 dchan=16 fxsks=32-39 loadzone = us defaultzone=us I do *modprobe zaptel wct1xxp wcfxs ans seems to work ok. * But when i do ztcfg -vvv i got Zaptel

Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support

Hello everyone - first time poster, long time lurker. (sounds like a radio morning program, I know). I'm attempting to get my InterTel Axxess (w/v9.0 software) to play nice with my Asterisk implementation. Asterisk 1.2.6 is running on a Fedora Core 4 x86 box. I've tried getting the Axxess to talk SIP to Asterisk, but InterTel's SIP implementation is, well-let's say, incomplete.

Hello, I am having problems configuring a TDM400P 4 Port FXO card with groundstart signalling. The box has 2 X100P's, 1 TDM04B and 1 TDM40B. I can configure it for loopstart and it works, just not groundstart, which I need for this installation. What am I missing? Thanks! Mark /etc/zaptel.conf #two X100P FXO Cards # fxsls=1-2 # TDM400P 4 Port FXO (TDM04B) fxsgs=3-6 # TDM400P 4 Port

Hi all, My office recently purchased an InterTel Axxess system with the IPRC card for VoIP. To our suprise, this card allows the InterTel endpoints and MGCP endpoints to work, but not SIP clients. I was really expecting to get a SIP softphone working with this setup, but that appears to require our vendor to sell us a SIP gateway and licenses at a not yet determined price. With this

Hi.. I've got an asterisk setup with an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5 seconds before asterisk realizes the line has been hung up and drops the call.. this causes my SIP phone to continue ringing, and occassional phantom voice mail messages to be left.. I'm located in good old standard North America, with a regular Verizon residential

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

*** Now you just need a "phone" or "station" (FXS) device and a method of connecting it. *** So... essentially I need to spend another $125 US on a Wildcard TDM400P so I can plug in an analog phone? On the X100P description it says: "The X100P supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). It can detect ringing and

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I setup the FXO with kewlstart signalling, and the outgoing call is never registered as 'Answered' and the dial

Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with <ZT_SPANCONFIG failed on span 2: No such device or address (6)>

Hi All, I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled "Q. How do I configure my TDM40B and X100P?", I'm receiving the following error: WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module): Ignoring rxwink ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module):

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says "Unable to load module chan_zap.so" no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0:

Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Details: Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: India This is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the zaptel.conf, trying zaptel v1.2(with

Hello, I am having an issue with hangups being handled within Asterisk. Right now, when an inbound call hits the Asterisk box, Asterisk picks up the call just fine. When the caller enters an extension to call, the Asterisk dials out on Zap/3 and rings the extension with no problem. If the extension is answered, there is no problem. If the caller hang's up before the phone is answered,