similar to: Zaptel, analog phone, and call waiting

Hi I have been searching the wiki and mailing lists and I cant see where my config is incorrect. I have a Digium tdm11b (1 fxo + 1 fxs) this is the output of cat /etc/zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

Hi, I have a analog phone connect to a WCTDM card. It used to work fine. Now recently, after several conf change and power restart, it stops working. Whenever I pickup the phone, instead of hearing the dial tone, I hear a busing beeping tone, like a machine gun is firing. :) However, from asterisk console, I do see a a OffHook/OnHook message, but whatever I dial in the phone keypad seems not

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1:

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

I have the following entry in zapata.conf, but I don't get stutter dialtone when there is a message waiting. Suggestions? Please? callgroup=1 pickupgroup=1 callerid="Paul mahler" <100> context=inside mailbox=100 channel => 1 Thanks, Paul

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

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

Help! I am having a problem getting wall waiting to work. I have a X100p and a TDM400 card with and analog phone attached (as well as a number of SIP phones) running on RedHat 9.0 box. The PSTN line has the call waiting feature. If I am on the phone, I get a beep indicating a second call is waiting. However, there is no way, no how, that I can get it to switch to the second line! Flash button,

Hi all, Using Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q with a Junghanns quadBRI (2 spans connected) and a Digium TDM400 for extensions Shouuld I be worried about these lines that keep showing up in my /var/log/messages? qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 qozap: dropped audio card 1 cardid 0 bytes 6 z1 50 z2 28 qozap: dropped audio card 1 cardid 0 bytes 6 z1 59 z2 37 .. ....

Hello list. I have a question about how to read the incoming calls' callerid on an FXO interface of a TDM 400 analog card; (it's one of those RED modules). Now -may this is the complexity adding step..- I have a GSM gateway attached to this FXO thing; incoming calls are processed as they should. But both when peeking on the CLI, as well as in the phone display I do not see the caller id.

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

Hi, We've got IAX softphones, GrandStream VOIP phones and zaptel connected analog phones. Caller id, internally, works just fine (as long as I use numeric only callerids) for IAX and grandstream. Is there a way to have the analog phones' LCD display show the caller id? These are plain old regular analog phone, that if I had callerid from my telco would show on the screen. thanks

> -----Original Message----- > From: C F [mailto:shmaltz@gmail.com] > Sent: Tuesday, January 11, 2005 4:38 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] internal caller id on analog phones > connected tozap > > How are the analog phones connected to * ? this is where the setting > should be. They're connected to

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

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