similar to: Data over Voice through Asterisk

Dear all, I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time" *CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack -- Called 1/150 -- Zap/1-1 answered

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

What are the meaning of these Zap show channel output? Caller ID string: Owner: <None> Real: <None> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0,

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1

Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When

Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)

I have a TDM400P with fxo and fxs modules, and local extension calls work fine but I can not dial-out. I am getting both problems on FreeBSD and Linux, and plugging a regular telefon set works fine. Any ideas of what I am missing? thanks micko Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default)

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> I recently upgraded my Asterisk system to Dahdi and now I have an echo problem.<br> <br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a HARDWARE echo

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hi guys, This has been bugging me for days and I can't figure out what's happening. Using the same channel on DAHDI, I can't make a call after previous one being hung up by remote side. However, it works like it should if my end hangs up the call. To make things interesting, DAHDI, or astribank rather, seems to "reset" itself after receiving an incoming call on that DAHDI

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools