similar to: zapata.conf channel variable question

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

Hi, I'm unable to pick up parked calls after they are transfered. I get the "transfer" message when I press # and then I'm told "701" The extension I'm dialing goes to the on hold music. I'm disconnected, I hang up, dial "701" and I see this message on the console "Everyone is busy/congested at this time" I just have the default

Hi, I have 2 asterisk boxes as Gateway, in this arrangement. (PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE) everything works great, in both directions (receiving and making calls), but when i get a call on the (ANALOGPHONE), I haven't been able to transfer it to another extension of the PANASONIC PBX using the flash key. I've tried the using the t T options on

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

Hi I've problem, i cant get tone on a FXS ZAP channel my configuration are: -- zaptel.conf -- fxoks=1 --zapata.conf -- [channels] immediate=yes context=bell signalling=fxo_ks channel=1 --extensions.conf -- [home] exten => 500,1,Dial(IAX2/guest@misery.digium.com/s@default) [bell] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,Dial(${PHONE},16,tr) ANY IDEA?

I have a Digium TDM400b11, 1FXO [port2] & 1FXS [port 1] When I reload the chan_zap I get: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be specified before any channels are. Apr 9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: load_module

I'm having a problem defining my channels Here is my zapata.conf ----------------------------------------------------------------------------------------------------- ; Zapata telephony interface ; ; Configuration file [channels] ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

On one installation of asterisk, I have a display on the console when I have a incoming call on my zaptel card. every digit was displayed, this was great. Does anyone know how I can get this back? Thanks

From: Malcom Kemp Sent: Wednesday, April 25, 2007 11:19 AM To: 'asterisk-users@lists.digium.com' Subject: AsteriskNOW generation of zapata.conf file I am a new user of Asterisk, and an trying to use AsteriskNOW. The test system is dual processor, so I am using the beta 4 version. I am currently trying to manually configure, as the GUI does not seem to let me accomplish what I need.

Hello, Let's say you would like to define, for every SIP peer, a value which would set, for instance, the maximum daytime calls number. Extension 101 would get a 2 value, extension 102 would also get 2, extension 103 would get 1, and so ... How is it best to proceed as those values : - shall be usable from dial plan, - shall be set when system starts up. Now I would simply use database set

Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

Hi, I'm interested in a callback feature where I can dial my Asterisk, then hangup and Asterisk will call me back and I can then place phone calls or whatever I want to do. And also, if I've got voicemail I want Asterisk to call me back as well. Are there any scripts for this available? Any help would be apreciated! Best regards, Christian

Can anyone explain to me the difference in zaptel.conf and zapata.conf? I'm trying to get a real clear understanding of them but its getting a little murky in places. I will be setting up a PBX running asterisk with 2 T100P cards. I will be bringing a 23 channel PRI into one card and connecting the other card to a Nortell 24 channel FXS channel bank. As I understand it zapata.conf is

Feel free to hit me with a cluestick; maybe it will jar something lose... I'm running a stock *@home 0.6 install and after four days of excrutiatingly annoying debugging, I finally came across the obvious problem. The zaptel driver is reading /etc/zaptel.conf; which by default configures one FXS Kewlstart channel. Asterisk, according to "full" log, will see the first channel, then

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have