similar to: Asterisk 1.2.16 - No Caller ID

Has anyone been able to successfully solve the following issue: WARNING[21725]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Mar 11 01:26:53] WARNING[21725]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Since we updated asterisk from 1.2.13 to asterisk 1.2.16 the module went away so we updated

how can I fix this just started ...... Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid

Hello All, I have tried numerous versions of asterisk from asterisk at home to compiling it myself through the cvs server. I don't understand it works fine with the intel p2 box but not the faster via cyrix box. Is it the processor or something? Regards, Otis Surratt Jr. / otis@ocosa.com

Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the "s" extension and background application but not sure how? Any help would be appreciated!! -- Otis

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

HI Here is my info: Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my

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

Hi, I've successfully installed Asterisk 1.0 on Fedora Core 2 with the 2.6.8 kernel. I have two other computers running X-lite connecting to it. I've been able to set them up so I can dial extensions "123" and "124" to talk between them. I'm able to access the default "1000", "500", and "600" extensions and they all seem to work.

I've isolated this problem the furthest that I can, and I'm now convinced this is a bug in asterisk. I have a context in extensions.conf like so: [my_context] exten => _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL}) exten => _X.,2,GOTO(my_other_context|${EXTEN}|1) exten => h,1,DeadAGI(my_agi_cleanup) For the purposes of this scenario, my_agi simply will try to HANGUP the channel to

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:

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

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:

I've run into a problem with dir.create on R2.2.0 Windows XP 2002 SP 2. setwd("d:/") print(dir.create("d:\\otis-sim\\rdata", recursive=T)) print(dir.create("d:\\otis-sim\\", recursive=T)) Both return false and fail to create the directories. setwd("c:/") print(dir.create("d:\\otis-sim\\rdata", recursive=T)) Returns true and succesfully

I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk -> ATA 186 MGCP --> analog phone. What changes to I need to make to my mgcp.conf and extensions.conf file to allow answering of Call waiting calls? And how do you answer call waiting calls with the system. I have usecallerid=yes, hidecallerid=no,

Greetings, I have a couple of questions and figured I would put them all in one message to not spam the list as much as possible. I have searched voip-info, google and the list archives for all of these questions. If I have missed the correct response, please accept my apologies. I have been stuck on these for a long time and I am really hoping that the other users out there will be able to