similar to: Zaptal and Fedora Core 2 and losing GSM playback

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message "the person you are calling, is currently unavailable" This is an ubuntu machine, with a sangoma card, with

So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1

Hi All I am working on asterisk + digium developer card , it has on FXO and one FXS I want to work asterisk in the following way 1>FXO connected to PSTN line 2>the calls coming to PSTN line should be received 3>SPI clients should be able to call outside through PSTN 4>There is no phone connected to the FXS In this case , do i need to bother about configuring FXS , in

Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hi, I would like to use the asterisk box with zapta card to enable some conferencing. I would like to use only TDM connections without VoIP. I'd like also use the Meetme app. I have some questions: 1. Does any one use it for a few conference rooms at ones ? 2. Is it possible to restrict the number of users connected to one conference room ? Regards. Pawel.

It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To:

Sorry about last posting, typo... I just added 2 Digium X100P cards. When my * box boots, it found them and configured them. When I enter genzaptelconf, it comes back with the following error: line 13: Unable to open master device '/dev/zap/ctl' Unloading zaptel hardware drivers: Removing zaptel module: rmmod: module zaptel is not loaded

Interestingly enough I had this same problem today.... 1. I created the directory and permissions for the directory " /var/lib/asterisk/festivalcache/ " (per the comment in the festival.conf file) 2. I had to comment out some things in the festival.conf file: the "host" line, the "port" line, and the "festivalcommand" line. I have also noticed the

Hello. I've just installed asterisk-1.4.21.2 zaptel-1.4.12.1 chan_ss7-1.0.10 libpri-1.4.7 I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers. My OS: Ubuntu 8.04 Server Kernel: 2.6.24-16-server I am getting a choppy GSM playback and too many defunct AGI processes when channel closes. i am using Perl or PHP, also 'agi-test.agi' going to defunct too... I was able to playback GSM

I'm new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side. GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

Do the Zaptel cards actually have a timer that it supplies to Asterisk or is the phone company supplying the timer to the card that is then passed to Asterisk?

I've spent the past week experimenting with Asterisk@Home 2.6, and then Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have an entirely IP (hard & soft)phone setup (IAX and SIP) so I have no requirements to support any Digium PCI cards, etc. All in Asterisk works extremely well except for one thing: Playback of sounds (GSM format) such as an ivr greetings, sound terrible.

Hi - I think I might have seen this problem on the list before, so I'm sorry if this is a duplicate, but I couldn't find it when searching through the archive.... I'm just setting up a new machine with asterisk. It's a RH9 box, and I've tried the RC2 tarball, the 1.0 CVS and the 1.0 RPM's from nacs.net (thanks). My config is basically the sample barebones sip setup