similar to: outbound dialing / wait for keypress?

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Asterisk will try calling both at once. As soon as one is answered it cancels the call to the other. What you can do is for extension 101 to put it in it's own context and then call the agi from the h extension. So something like this: [from-internal] exten = 514316XXXX,1,Answer() same => n,Playback(hello) same => n,Dial(LOCAL/100 at extensions&LOCAL/101 at extensions) [extensions]

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

hi there. i'm developing a kind of collaborative phone system in which groups of users need to navigate phone menus together, and run into problems with the '#' character. here's a scenario: 1) person_a dials into my asterisk server. 2) asterisk parks the call, and initiates an outgoing call to person_b 3) after verifying person_b (by requiring a keypress), asterisk bridges

I get congestion (same error) with exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) not dialing 1 exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 1 exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 9 All the same == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [9544790554 at To_Airspring:1]

Hi! is it possible to make a drag dependent on a KeyPress event? I couldn''t find any information on that on the script.aculo.us website. My scenario is the following: I have a tree with several nodes (like windows explorer). Right now i use Drag & Drop to move a node below a new parent. Now i want to implement a copy mechanism which allows the user to copy a node through Drag

Hi all, we are writing a cross browser Javascript control that enhances the HTML text input element in order to accept only digits (more or less). Supposing that ''domText'' is the DOM element corresponding to the HTML text input, we wrote the following: Event.observe(domText, "keypress", this.onKeyPress.bindAsEventListener(this)); where this.onKeyPress =

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

I have a little call recording script that I am running and it works fine, but I have one problem. I get CDR when a user calls into the extension, but I do not get CDR for the call that it makes outbound on # 17. Any idea why? Here is the extensions info: [default] exten => 2211,1,Answer exten => 2211,2,Wait(1) exten => 2211,3,Playback(/etc/asterisk/recording/getshop) exten =>

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

Hi, I spent some hours working my way through the WIKI and a number of other documentations, but after all, three questions are still left: 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my Grandstream BT 100 rings when I diall a regular phone number. Is there any need for me to configure the zapata.conf for the ISDN card to get a special asterisk feature or is this

That sound's like the right thing to do, you'd probably have contexts related to what phones could do "unrestricted","localdialling","extensionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension

> -----Original Message----- > From: Adam Goryachev [mailto:mailinglists@websitemanagers.com.au] > Sent: September 7, 2004 8:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap > interface? {clip} > Have you considered adding the r option to the Dial command, so they > might hear ringing

Hi, I am trying to write a dialplan that will use Dial() to call two local extensions. One extension will run an AGI script (a continuous background process, running until hangup), the other will connect the active channel to Jack() (also running as continuous process, until hangup). This is my current dialplan attempt: --------------------- [from-internal] exten = 514316XXXX,1,Answer() same

Hi everybody We are having periodically troubles with the outbounds calls, seem like the PBX cannot end to dial the entire string of numbers This is output from the PBX., few minutes after all work fine again ... :(, and few minutes after the same problem appears. Thanks in advanced ... Regards This is the PBX aout and my zaptel, zapata confs. Runing asterisk 1.0.9 libmfcr2-0.0.2

I''m desperately trying to update a div when a key is pressed in a text field. Eg. : <input type="text" id="field"> <div id="area"></div> <%= observe_field :field, :url => {:action => :myAction}, :update => :area, :with => "text_param", :on => ''keypress'' %> This does not seem to work, and I

Is anyone else finding it frustrating to do in Ruby client-side things that would be simple in Javascript? It seems like all of the "helpers" involve server (controller) communication. For example, having a text field that just echos below what the user types. Is this the only way to do it? <div id=''word_panel''> <%= text_field_tag

I have two server Server_A(outbound call) for agent login and agent make a outbound call from here and pass into server Server_B call extension.conf exten => _91XX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR) exten => _91XX.,n,hangup() Server_B[192.168.53.197] for call forwarding extension.conf exten =>