similar to: some problems with asterisk and E1

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config

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

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

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

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

Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] =>

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

Hi all, I have a strange problem, regarding zap channels and cdr. I am using asterisk bristuffed version Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium, Inc. and others. with two billion ISDN cards. I also installed asterisk addons, last stable version via cvs internal calls, or calls starting from internal sip or iax phone are recorded in the cdr all without any

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 --

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

Hi there, I have a new box setup using the latest version of FreePBX and the latest SVN of Asterisk 1.2 as of yesterday. Incoming calls from our PRI work fine. However, outgoing calls gives me the operator saying "The call cannot be completed as dialed" after two rings. Here's an outgoing call from extension 271: -- Executing Set("SIP/271-09f61dc0",

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

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

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

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

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or