similar to: Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI

Displaying 20 results from an estimated 20000 matches similar to: "Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI"

2010 Oct 25
2
Pop-up for MS Outlook 2007 recommended
Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Apr 11
1
outCALL- the open source Asterisk integration applicaiton for Microsoft Outlook
Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed
2010 Dec 28
1
OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a
2007 Aug 04
0
Outcall 1.40 released
Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for
2003 Oct 13
1
Call Parking and Paid Digium software modifications
Hello, I'm considering paying Digium to do a modification to Asterisk so that calls can be parked on specific user-defined numbers(transfer to 701 and it's parked on 701, transfer to 702 and it's parked on 702) instead of the way Asterisk currently does call parking(transfer to 700 and then it tells you where it put the call 701-720). What would be the price range for this feature to
2003 Oct 13
0
Call Parking and Paid Digium software modifi cations
That is how many old PBX phone systems work and it is that way our users are used to working with the phone system. Another issue with the way Asterisk callparking currently works is that there is only one call-park orbit, you cannot use a different set of numbers for a different call park instance(i.e. 700 goes to 701-720 AND 740 goes to 741-750). We also have several Grandstream phones which
2008 May 14
0
Register And Get Paid $50 Dollar Free
* Follow This Step : 1) Log In - http://www.AWSurveys.com/HomeMain.cfm?RefID=Apocalto 2) Create A Free Account ( Register ) 3) Start Your Survey Money * Redeem To Paypal Account * --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Spinoffs" group. To post to this group, send email to
2007 Dec 05
7
Asterisk SIP Microsoft Outlook Integration
Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael -------------- next part -------------- An HTML attachment was
2023 Jul 02
1
Get channel variables via ARI/AMI
>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session
2009 Apr 23
1
Dial-out via AMI
Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think
2023 Jul 02
1
Get channel variables via ARI/AMI
On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI
2007 Oct 21
0
Hanging up all call on a device via CLI/AMI/AGI
Hello, never posted to a mailing list before. I've been trying to work out this problem for quite awhile now. I have a PHP script which is run whenever an emergency situation happens. The script connects to the AMI and originates calls to previously defined "emergency" extensions. I'm looking for a way to disconnect all calls on a device if it is in use, in order to deliver the
2023 Jun 26
1
Get channel variables via ARI/AMI
On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > On 6/26/23 9:00 AM, Joshua C. Colp wrote: >> On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: >> >> I am connecting to the ARI with subscribe all, so I can see >> channels being created.  I now want to extract a variety of >> header variables (at the moment the from and to tag).  I
2023 Jun 26
1
Get channel variables via ARI/AMI
I am connecting to the ARI with subscribe all, so I can see channels being created. I now want to extract a variety of header variables (at the moment the from and to tag). I tried to read them from the ARI but Asterisk refuses since the channel is not in a stasis app. Is there a way to read these from either the ARI or AMI ? I'm trying not to modify the dialplan. Thanks Brian
2012 Dec 03
1
Query list of defined channel variables via AMI
Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get.
2014 Jul 21
1
Hold ,UnHold Via AMI
Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -------------- next part -------------- An HTML attachment was scrubbed... URL:
2023 Jul 02
1
Get channel variables via ARI/AMI
>> You use the AMI action Getvar[1] which allows channel variables and dialplan functions. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar I actually tried that, and although I get “success” I never get useful data. For example: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-2-In-00000025 Variable: channel(pjsip,call-id)
2023 Jun 26
2
Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > I am connecting to the ARI with subscribe all, so I can see channels being > created. I now want to extract a variety of header variables (at the > moment the from and to tag). I tried to read them from the ARI but > Asterisk refuses since the channel is not in a stasis app. > > > > Is there a way
2023 Jul 02
1
Get channel variables via ARI/AMI
On Sun, Jul 2, 2023 at 4:39 PM TTT <lists at telium.io> wrote: > >> You use the AMI action Getvar[1] which allows channel variables and > dialplan functions. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar > > > > > I actually tried that, and although I get “success” I never get useful > data. For
2023 Jun 27
1
Get channel variables via ARI/AMI
I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy