similar to: Answered Call marked as "NO ANSWER"

supervision. Verify if for those "numbers" the CO revert the line polarity when callee answer. callprogress=no is a good test too. Jorge Ra??l G??mez C. wrote: > Hi list, > > I'm having problems transferring certain calls made by the attendant > between the PSTN and to an internal extension. Although, transfers > between the majority of the calls ends successfully.

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? Actually, the problem is that my telco provider doesn't offer an uniform method for answer/disconnection supervision, and by that I mean, some of it's line (I think) offer a polarity reversal, but other lines (of the same service provider) do not offer anything at all, so the answer of a call

Hi list, I've just compiled and installed Asterisk 1.4.16 and when I try to run "zap show" I get the message "*No such command 'zap show'*". I have a Sangoma Remora A400D with 2 FXS / 10 FXO ports, I've installed the latest wanpipe too. zaptel-1.4.7.1 was compiled from the wanpipe installation, so I don't know what's happening here!!! Any ideas???

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

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

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009 ; If you edit this file and execute

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hi All This may be a silly question but... I have this dahdi_genconf generated file: ; Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 4 22:05:29 2014 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to

I have Asterisk working with one FXO card (clone x100P card PCI card). I am trying to add 2 more cards so my question is - do I just increase the channel count on the zaptel.conf and zapata.conf files? [original] /etc/zaptel.conf fxsks=1 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] language=en context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 im having an error message when in running asterisk with the tdm card in. here's the error from the console of asterisk: [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get span status: Inappropriate ioctl for device [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:10466 build_channels: Unable to register channel

> I have 2 FXO channels from which I want to route incoming calls to > different contexts in extensions.conf. I edited the context entries in > dahdi-channels.conf and created matching entries in extensions.conf. > One channel is routed to the new context as I want, but the other > channel is stuck going to the default "from-pstn" context no matter what > I do. >

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading "The future of telephony" and this [1] document looking for information about