similar to: Adding a DELAY to an ADSI script

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

Hello list, I have reason to believe that ADSI can be spoken to phones even when they're onhook. Is this true? Does anyone know? Right now, I'm having trouble figuring out how to do anything to an onhook channel other than ring it. Does this require a magic application and some serious voodoo? Any pointers? I got onto this idea because I noticed that whenever I got voice mail on a

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook switching. It dials perfectly, but does not seem to be changing the onhook/offhook state appropriately. It changes sometimes, but it's not really reliable. For example: When I booted the machine, it started as onhook. It remained "onhook" through the entire first call (which was silent on both ends --

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the

Does anyone know where I can get a list of ADSI functions.. Example *70 (No Call Waiting), Flash = Flash, Hold = ??? Thank you, -gcc

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

Does anybody know if there are any external telephone-keypads for sale anywhere? (containing the keys 0-9, *, # and onhook/offhook would do) I am looking for a keypad to control a softphone and would prefer the controls to be in the physical world instead of as a window. Sincerely, Markus Hakansson

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hi, I have a analog phone connect to a WCTDM card. It used to work fine. Now recently, after several conf change and power restart, it stops working. Whenever I pickup the phone, instead of hearing the dial tone, I hear a busing beeping tone, like a machine gun is firing. :) However, from asterisk console, I do see a a OffHook/OnHook message, but whatever I dial in the phone keypad seems not

Hi, I want to do a scripted 'restart when convenient' on a daily basis. This used to work, but since i've upgraded to Asterisk 11.7 it seems it's never convenient to restart the server. My question: how can i tell *why* it's not convenient to restart the server? It used to be some colleague left the receiver OffHook or something like that, but even when i'm fairly

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

Hi all, I would like to make a script/program that would be able to show lots of status information from my analog FXO lines (and FXS lines in the near future). Example of interesting status information: - Hook status: is there a call being made with that zap? - Voltage status: cable connected, voltage values (if possible), line ringing? - RX/TX Volume status I'm using a TDM400 card with

Hi all, I've been told that part of best practices is to noload all modules that you don't use. Adsi being one of them (are there actually people using adsi & asterisk?) I noloaded it in /etc/asterisk/modules.conf: noload => res_adsi.so noload => app_adsiprog.so When I start * (this is cvs head of today by the way) I get: [app_voicemail.so]Aug 31 17:25:49 WARNING[4961]:

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

I sent this yesterday from my wrong email account. yeah, Thank you. I screwed up by only looking at adsi.conf, and not asterisk.adsi. Once I saw that, things started falling into place. Are you able to pass arguments to the phone so that for every call there is unique data on the screen. I see that works for caller id with $Call1p and $Call1s, but what If I obtain an OrderID from my