Displaying 20 results from an estimated 10000 matches similar to: "Click to dial apps always show from "asterisk""
2007 Aug 11
1
LumenVox Speech Recognition
Hello All,
While looking for solution to solve my Callback DTMF problem, I came
across LumenVox Speech Recognition software.
Has anyone tried out? Need some feedback before I purchase it... Please
help...
Cheers,
Nitesh
2007 Jul 29
2
Dial from Phonebook of Evolution or Thunderbird
Hi,
does anyone know about a plugin that allows dialling a contact from the
phonebook of evolution or T-bird?
--
Alexander Topolanek
http://www.topolanek.at
2007 May 16
1
getting call status using Manager API
I am originating a call using the "Originate" action in the Manager API. It
calls one party, then when they answer does the "Dial" application and calls
another party and connects the two.
Is there a way using the Manager API to check back later on the status of
this call (is it still up, etc.)?
I have found the "Status" API action, but I don't know how to get
2006 Nov 01
3
Manager API - Originate Call - Need Help
Hi all,
How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number?
I can originate a call from my SIP-network using this parameters in Originate call command :
Channel = SIP/0041435215301
Context = default
Exten = 00982166501553
Priority = 1
CallerID = 0041435215301
this works with out any problems I initiate a call from one of my
2006 Oct 20
1
Escape from Voicemail
I used to have fonality and I could press * when I got to someones voice
mail to go back to the menu. I assume I add that to the dialplan but
how? Thanks
BTW I went back to 1.2.12 and transfer works and DTMF works and it seems
to be much better for now.
Thanks for you help
Jason
2006 Oct 30
2
Fxo box for asterisk ?
Hi
do you know if they have "external Box" (not internal card) for
connect Analog Line and Pri/Isdn to asterisk for incomming and
outgoing calls ...
Thanks
2007 Mar 15
2
voip-info.org is back!
Looks like the site is back up. Don't all hit it at once, it might go down
again ;-)
Sean
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2007 Mar 19
10
Microsoft launches first PABX
http://www.crn.com.au/story.aspx?CIID=76033&eid=4&edate=20070320
The company developed Response Point to work alongside traditional phone
systems or voice-over-IP systems.
Continuing its recent foray into the market for digital communications
products, Microsoft on Monday introduced its first packaged digital
phone system for small business.
Anyone know anything about it?
2007 Jan 17
4
windows mobile 5 softphone for square screen devices
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
2006 Oct 13
5
Cisco 7970 SIP won't update?
Does anyone know what triggers the 7970 to update its config? I
was able to get it to update to SIP, but the config I used initially
won't go away. I am making small changes to the SEPxxx.cnf.xml file and
rebooting the phone, the phone is downloading the (TFTP) new config
file, but I don't see any change on the phone itself.
I've looked at the VersionStamp and incremented that, but
2006 Nov 13
3
FW: Desktop integration
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000066">
Hi Dean,<br>
<br>
I will check that site - thanks for the hint.<br>
The biggest problem I see with
2005 Feb 10
4
Why echo occurs
Hi all,
Can someone give me a simple rational explanation why a $5 analog
handset gives me no echo whatsoever on an analog PSTN line, but
PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require
software-based echo cancellation. Surely a $5 analog handset does not
have an "echo canceller".
The echo I mean is when I hear myself while talking to another party.
I have heard
2009 Jun 15
2
Click-to-dial CTI for Windows
Hello guys,
Is there a decent click-to-dial CTI which works well with Asterisk?
We have vanilla asterisk implementation and I have tried a few (ADA,
Outcall etc) but they have poor documentation and don't work very well.
We are looking for an application which can allow us to dial a number
from Outlook and IE/Firefox for outbound calls and get a pop-up for
inbound calls with call history
2012 Jun 14
2
Polycom, Dial Specific Number on Handset Pickup
Hi All,
I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom.
I would be greatly appreciate is some is able to tell me how this is accomplished.
Regards
David.
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2005 May 25
8
What does Asterisk need in the way of a GUI?
We are two programmers who are passionate for Asterisk and we will be
dedicating the next three months towards programming for Asterisk and
would like to get some input from everyone on what they feel Asterisk
is lacking or needs based on what is not currently a part of it or
available through third parties. Hopefully, by asking up front we
won't be wasting our time on something nobody wants
2007 Jun 27
4
Using MSAccess to dial on a Zap line
Hello,
We use MS Access 2000 (I know, we're migrating away from it) as an application
to automatically dial phone numbers. The old phone system we have allowed the
call representative using the application to take their phone off hook, push
a button in the app, and the app would send the phone number to the phone
system and dial the number. We are moving to Asterisk for our main phone
2014 Dec 29
5
chan_sip and 2 devices under same extension - transferring call endpoint(s)
Hi,
(please excuse me for lack of proper jargon usage and the vagueness of description...)
i use Asterisk 11.12.1, (well... as included in FreePBX),
I have several extensions that can register 2 separate devices (chan_sip)
( FreePBX calls this Devices & Users mode : Users are extension/internal number,
devices are the 'SIP Accounts' for the internal 'endpoints' )
(this
2006 Feb 20
1
Dial timeouts and SIP 302 redirects
I have SIP handsets which allow the user to forward a call to another number
after a specified interval of ringing time. On the SwissVoice this is
refered to as CFNR (Call Forward on No Response). What actually happens is
that after a specified period of time (default 15 seconds), the handset
sends back a "302 Moved Temporarily" response to Asterisk.
The problem is that when Asterisk
2007 Oct 04
1
Infuriating problems: no dial tone, dropped calls, no voice: 1.2.13 and 1.4.11
Hi
I've had an asterisk setup for the past 15 months, based on the debian
asterisk packaging. Until late August of this year, I had no problems
once initial setup was complete- the system worked essentially
flawlessly.
Since August I have been having exceedingly infuriating intermittent
problems that are causing me occasional periods of nasty trouble:
1. No Dial Tone. Every Sunday night at
2005 May 16
2
Help with extensions - can't dial 700
I have been working on integrating some FXS ports into my dial plan
delivered via a channel bank and testing with an analog handset.
The receptionist is on Extension 700. All other SIP phones are 7XX.
>From a SIP phone I can dial 700 and all other extensions.
>From the analog handset I can dial any other extension but not the 700
number. Weird? Yep.
The CLI does not show any dialing when I