Welcome to call.center™
call.center™ is specifically designed to be the only phone app that you will ever need. This means that we incorporate all the key components for robust business, professional and personal communications into one seamless app, with plentiful features integrating standard and advanced telephony services.
The call.center™ app presents a unique and innovative drag-to-call user interface (UI), where all the necessary functions are displayed front and center, smoothly integrating and optimizing voice operations with the workflow processes.
call.center™ is available for multiple operating system platforms including iOS, Android and Windows, allowing you to standardize on this app wherever you go.
App_konference is a channel-independent conference application. It features efficient audio mixing algorithms and comprises a set of enhancements needed to migrate a MeetMe application to AppConference. The goals of this
project are scalability and stability focused on voice.
Conference is a good alternative to the standard MeetMe because it does not require a timing source to be available. Lack of timing can be a pain when Dahdi / Zaptel hardware is NOT present, including solving problems when using virtual hardware that cant support timing cards.
Starting with app_conference the project was forked as app_conference did not include the alterations that were submitted to their project, app_konference now contains bug fixes and performance enhancements over app_conference however as app_Konference is focused on voice the video component is deprecated. The video support have been fork in a new asterisk application : Vonference, it is available from this link Github Voximal/Vonference
app_konference is not in the Asterisk standard distribution and can be found at
There is no configuration file. Conferences are created on-the-fly.
NAME: whatever you want to identify the conference
FLAGS: dialplan flags, see Flags.txt for a comprehensive list
MAXUSERS: limit conference participants to max_users
TYPE: conference type identifier
SPY: channel name to spy
VADSTART: “probability” to use to detect start of speech.
VADCONTINUE: “probability” to use to detect continuation of speech.
VIDEOSTART: length of speech before assuming that a member is speaking
VIDEOSTOP: length of silence before assuming that a member has stopped speaking
Mute/no receive options:
‘C’ : member starts with video muted
‘c’ : member starts unable to receive video
‘L’ : member starts with audio muted
‘l’ : member starts unable to receive audio
Speex preprocessing options (right now only for Zaptel members):
‘V’ : enable speex preprocessing Voice Activity Detection
‘D’ : enable speex preprocessing De-noise
‘A’ : enable speex preprocessing Automatic Gain Control
‘T’ : member connects through Zaptel, so speex preprocessing should be enabled
‘X’ : enable DTMF switch: video can be switched by users using DTMF. Do not use with ‘S’.
‘R’ : enable DTMF relay: DTMF tones generate a manager event
If neither ‘X’ nor ‘R’ are present, DTMF tones will be forwarded to all members in the conference
Moderator/video switch options:
‘M’ : member is a “moderator”. When a moderator quits, all members are kicked and the conference is disabled.
‘S’ : member accepts VAD controlled video switching. Do not use with ‘X’.
‘z’ : member can “linger”. When the member is currently transmitting video and becomes silent and nobody else is speaking, we stay on it.
‘o’ : enable special behavior when in 1 and 2 member situation (one on one video). …
FreePBX provides a web-based, user-friendly administrative interface to Asterisk and is a standardized implementation of Asterisk (i.e. dialplan) that is maintainable, flexible and extensible.
For more information or to purchase a FreePBX Appliance please visit FreePBXdistro.org
Please Note: The companies listed below are not certified FreePBX Resellers and add themselves at there own discretion.
- World Wide Asterisk Support
- Specializing in Asterisk based solutions.
- Asterisk Support & Staffing
- Live Website Support Available
Asterisk Support- Live and Free
.e4 | Technologies
11138 South West Bayshore Drive
Traverse City, MI 49684
Remote AMP installations, ISDN integration
Fon: 0049 180 5925 4444 01
Fax: 0049 180 5925 4444 17
Remote AMP installations, pc2phone, prepaid cards, virtual pbx, callback, web lcr manager – all asterisk based.
“Communicatii Libere” S.R.L.
Contacts on site
Instalacion de FreePBX, AMP, Trixbox,Asterisk en espaÃ±ol o ingles. …
Since its creation in 2009, the company Ulex Innovative Systems develops solutions to interact with the users through their phones.
At first supplier of technologies and services, Ulex releases now his solutions and turnkey systems, integrating the knowledges bound to the technologies of phone voice servers.
- Voximal : VoiceXML interpreter for Asterisk : We are the creators of Voximal , an integrated solution of phone voice server. It is an engine for the creation of vocal services. It is based on Asterisk and our VoiceXML interpreter. It supports numerous features among which the voice synthesis and recognition.
- Chan_rtmp : RTMP channel for Asterisk : RTMP Channel is an open source module for Asterisk. It can handle phone calls via the RTMP protocol. It is also a video conference component based on Web FlashPlayer© client or Android / iOS applications.
- Our expertise is at your service to help you to realize your projects. We assist you in their implementation .
- Our commitment is to provide technical support on all our integrated softwares. According to your contract service, we provide a full diagnosis.
- We work with you to develop new applications or improve existing services .
- Our technical environment is based on GNU/Linux, Asterisk, FreePBX, Android, FlashBuilder, PHP, combined with other free projects less popular.
- The integration is complete and continuous. The operation is guaranteed by creating packages for each environment.
- If you choose to deploy in a cloud infrastructure, we are with you to control your outsourced resources.
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We have been alerted to business continuity issues affecting the customers of
VOIPITS and its subsidiaries (MyPBX.com and DollarDID.com). As the original
supplier of services to your Virtual PBX provider, it is DIDWW’s priority to ensure
continuity, reliability and customer satisfaction. We are therefore reaching out
to you to ensure that you are able to keep your services up and running.
To accomplish this, we will need to port your numbers and services to either:
Please contact us or a certified reseller as soon as possible to avoid service interruption.
If you have any questions, contact DIDWW at firstname.lastname@example.org
or any certified reseller of your choice directly.
Execute a VoiceXML document over Asterisk (Based on the Voximal VoiceXML browser).
The application use Asterisk internal API (Prompt / DTMF / Record) and installed applications.
It replaces the old Vxml application.