similar to: providing battery reversal from Asterisk to legacy pbx

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

Hello, I thought that my Digium TDM400P would be the right hardware to support the zaptel timer, and put the following IAX.CONF entry to test, (trunk=yes) in the example below [VHAX] type=peer auth=md5 username=whoknows jitterbuffer=yes ;trunk=yes secret=terriblesecret host=4.5.6.7 qualify=1200 disallow=all allow=ulaw allow=gsm ;allow=g711u ;allow=g711a But, it didn't work. So I had to

hello, using a legacy PBX to access a Asterisk Zap channel (Legacy PBX FXS --> FXO application Asterisk/TDM400P) I want to be able to "flash" the asterisk pbx. However by pressing the FLASH button on the extension connected to the Legacy PBX gets me the flash features on the Legacy PBX, not on the Asterisk PBX side. I thought of using the following codes (listed below) from

hi, we have just received our first shipment of digium cards, FXO + FXS combinations, and collected all the hardware for our custom clone server which will house our test-bed for asterisk. I'm based in Dhaka, Bangladesh so you will understand we may not always be able to get all the hardware we need, when we need them and in the manner that other people are used to.. from a catalog or from a

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Hello list. I'm going te explain my trouble. I have my asterisk with a TDM400P with 4 FXS channels. Two ports are connected to a Panasonic PBX (it's working fine), and others two ports are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected to a CO port (where i had a pstn line). When I call to the Alcatel PBX, the asterisk show me in it console that es ringing but

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

I have a system that I am trying to get a port on a Sangoma A104d card connected to an Adtran 850 with 5 FXS modules and 1 FXO module. A problem I am having is figuring out what cable should be used from the port on the Sangoma to the JP2 port on the Adtran. Tried was a cross-over T1 (1->4, 2->5, 4->1, 5->2) as well as a straight T1 (1->1, 2->2, 4->4, 5->5). Neither one

Em (20:59:21), Asterisk Users Mailing List - Non-Commercial Discussion escreveu: >On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote: >> >> Does Somebody know why no load modules to FXS? I used zaptel-1.0.7 >> version. > >What color are the modules? >Post your /etc/zaptel.conf > >-- >Mike Mike, The color is green, red ans bluw. Link

[resend in plain text format] > On Tue, 2004-12-07 at 09:34 -0600, asterisk-users- > > request@lists.digium.com wrote: > > > I've been struggling with a test * install for a couple months now in a > > > small office and am just about ready to give up on it. It's not that the > > > system itself is a problem. I've got everything (attendant,

Hello, I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of 123-4567 and strip out the first four digits, and then dial whatever number digits remain. If I only have three digits (000-999) and have a mix of channels (ZAP, SIP, IAX2) could someone please point out how I can use a single DIAL command to just dial the extension regardless of the type of channel. .. For each

> On Tue, 2004-12-07 at 09:34 -0600, asterisk-users- > > request@lists.digium.com wrote: > > > I've been struggling with a test * install for a couple months now in a > > > small office and am just about ready to give up on it. It's not that the > > > system itself is a problem. I've got everything (attendant, voicemail, > > > FXS

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [

zaptel compiles fine (make linux26) but when I modprobe ztdummy I get the follow error. [root@localhost zaptel]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy System info: Linux localhost.localdomain 2.6.9-5.0.5.ELsmp #1 SMP Thu May 19 19:24:44

Hi all, Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [root@server1 zaptel-1.0.7]# modprobe wctdm ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but