similar to: Echo on Zaptel hardware (Wildcard 100XP)

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

Hello, I have some problem to start asterisk. First I have followed a lot of tutorials to complete correctly the install process. Now it works when I type zttool I can see when I am or not connected to the PSTN. But, I run asterisk with vvvv verbose and I can't see the call detection. There is no detection of the call. I have a X100P card FXO with only one line. So only one channel I

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

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*

I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs). With Trixbox out of the mix and a regular phone connected I get the CID fine yet Trixbox shows 'unknown': dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'ringall' Here is my Zapata.conf if it helps: ############################# ; ; Zapata telephony

Can't get a Digium TDM04B working. Asterisk is running. I seem to have setup the trunks OK. But whenever I make an outgoing call get the 'all circuits are busy now' message. If I call in nothing happens at all! Here is my zapata.conf file: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks fxsks=1-4

Hey all, posted this the other day, but re-read it & realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using tT but transfers are only available when I'm the recipient of the call, not

Hi, I am using asterisk 1.2.5 and have some problems with asterisk connected with an E1 card to our PRI. Dialling in and out generally works. When someone dials in from a mobile phone, all numbers are sent as a block, and the called extension rings as intended. when someone picks up his phone handset, waits for a dial tone, and then dials in manually, the call will be redirected to the

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

Hello, I have another little problem with my ZAPs channels, in fact, when I received a call, I heard no sound while in the CLI, sound is played: -- Starting simple switch on 'Zap/4-1' -- Executing [s at from-zaptel:1] Answer("Zap/4-1", "") in new stack -- Executing [s at from-zaptel:2] BackGround("Zap/4-1", "hello-world") in new stack --

Hi all I wonder if anyone can give a little insight into this; asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless, ISDN30. 10 zactive channels Incoming calls work fine no problems tested 43 DIDs all working. zaptel.conf # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 zapata.conf [channels] language=en context=from-pstn

Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing

Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages: *CLI> Warning, flexibel rate not heavily tested! Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2 Channel 4 unblocked Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2

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

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the

Hi All i'm using sangoma card. connected to E1, my wanpipe file as #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Fri May 27 00:25:04 GMT+7 2005 # # Note: This file was generated automatically # by /usr/sbin/wancfg program. # # If you want to edit this file, it is # recommended

Did you compile and install libpri *before* Asterisk? I had same problem (among others) b/c I didn't install in the correct order. Try the awesome asterisk_update.sh shell script. Are you trying to emulate CPE or NET? Try signalling=pri_cpe Check for whitespace behind the statement, zapata.conf seems bitchy about whitespace. hth -----Original Message----- From: Steve Totaro

I've spent all day today trying to fix an echo problem and I've made no ground whatsoever. I have Digium TDM400 with 3 FXO & 1 FXS. I've tried this computer at two completely different sites with different phone providers. I've tried compiling & installing different versions of Zaptel (currently running 1.4.0, started at 1.2.9 and worked my way up). I've tried