-some minor UI bug fixes (still not perfect)
-button for droids without a hardware call button
If you don't have access to an Asterisk machine and the manager interface, this application will be of no use to you.
The MiniPBX App allows you to modify Asterisk / Trixbox / Freepbx / Elastix phone extension features such as Do Not Disturb, Call Waiting, Call Forwarding (Unconditional, Unavailable & Busy) at the tap of a finger .
The MiniPBX App will also show you the most recent 15 Voice mails left to your extension and allow you to initiate a Voice Mail callback at the tap of a button to your desired call forwarding number.
MiniPBX App compliments the mobility of an Asterisk VOIP based PBX. If you are not at your desk or on the road you can have full control of these phone features via your mobile phone or tablet and set your preferences accordingly at the touch of a button.
MiniPBX App allows you to retrieve voice mail at the touch of a button - no menus, simply see the 15 most recent Voice Mails that have been left for you and tap them to initiate a callback to your desired call forwarded number
Never miss a call. If your mobile battery is running low or you simply prefer to use a different phone, you can use the MiniPBX App to forward calls to a different location and never miss that important call.
The application requires that users have login credentials to the Asterisk Recording Interface (ARI) portal, which is installed by default with Asterisk distributions such as FreePBX Distro, Trixbox and Elastix.
This version is fully tested and compatible with Trixbox 2.8 and Freepbx 2.8, 2.9, 2.10 & 2.11 .
Enable / Disable Do Not Disturb
Enable / Disable Call Waiting
Enable / Disable and update Call Fowarding Unconditional number
Enable / Disable Call and update Fowarding Unavailable number
Enable / Disable Call and update Forwarding Busy number
View last 15 Voice Mails left to your extension
One tap initiation for Voice Mail callback
Secure SSL support where servers are https enabled.
ATTENTION: it can make your international, long-distance and mobile calls CHEAPER!!!
All what you need is only once entry the data of used by you "calling cards" (access number, PIN etc.) and to define rules and exceptions to choice them.
You can proceed with using your favorite apps of contacts or dials-up. LS Phone Cards catches "expensive" phone numbers due to preset rules and redirect them to access numbers from "calling cards".
With the ability to use multiple "calling cards" you can make your calls EVEN CHEAPER! For example, when calling to France you can use a card "ABC" (because it is the cheapest tariff for calls to this country), and for calls to China card "XYZ" (also because of the price or better communication).
Features of this program:
The concept of LS Phone Cards is "don't force to do the user that, what you can do themselves". This program, for example, can:
* to identify codes of countries by name;
* to identify the type of number (mobile, landline etc.) in those countries, where it's possible, using only the phone number;
* to identify the regional prefix , that stays before the national code of the country (for example, 011 in the USA, 00 in the EU).
* automatic switching between rules at BORDER CROSSINGS (not for CDMA!)
* can adapt this rule to the type of number from the phone book. For example, when you have free phone calls within your network "XYZ" you can change the type of number in your phone book from "mobile" to "XYZ" and then register appropriate exception in LS Phone Cards. After that the app will dial this number directly.
* regex (expert mode)
You can use ANY COMBINATION from the following phone cards: easybell, WORLD CALLS, WORLD ACCESS, GLOBAL CONNECT, Bingo, GO BANANAS, BALABOLKA, FOX FLAT, Goupil, Linea, MobileCall, Mox, Merkur, Rebtel, vonage and more other!
Keywords: Calling card, Telephone card, Phone card, Callthrough, Callthrougher, Prefixer, DTMF, #, Hash, Betamax, VOIP, Cheap call, Call, Call by Call, Callback, indirect access (IA), dial around service, GSM, CDMA, SIP
Why Smart Assistant?
Smart Assistant was designed to allow the customization of the routing of calls to an extension. Specially designed having in mind people who spend most of their time out of the office, but need to have at hand all of the integrated communication technologies available in its corporate IPPBX.
Moreover, it makes it possible for you to take that "critical phone call" if you have an unexpected departure.
How is Smart Assistant different?
It gives the user the possibility to create multiple scenarios, allowing to create flexible conditions and actions to let you do more than just forward, transfer calls or set a presence status as "not available".
Users can pre-configure settings per location, caller, or event. Settings can be activated automatically, manually, or with the use of proximity devices and NFC tags.
”Smart Assistant is innovative, combining several technologies, which include VoIP, Asterisk, Presence, NFC and others to maximize the functionality of current IPPBXs considering the location of the user at any given moment.
By itself it is responsible for establishing the availability and re-direction of calls without the user having the need to constantly reprogram the calls routing in its IP phone and/or IPPBX”.
How it works?
Smart Assistant works in synchronization with an Elastix unified communications server. You must install the Administrative addon of Smart Assistant on the Elastix server in operation and perform basic configuration. Then the user is ready to redirect calls efficiently.
For more details visit the official web page: http://elx.ec/sassistant
Just try lite version before buy full one. Lite version is intended to test product in order to fit your needs and to check own device compatibility.
Lite version ship the following limitations.
- Only 5 Peers shown in the main screen.
- Just 1 queue will be handled.
- Supervised Agents are limited to 3.
- 50 debugging lines will be shown in dialplan debugger.
- Only 1 call is shown in real time call quality debugger.
Lesula Asterisk Contact Center Supervisor Tool let you monitor your Asterisk installation. No middle-ware software needed, LesulaCC will directly connect to your Asterisk Manager Interface (AMI), giving you a real-time management solution which brings information together and provides coherent and useful answers.
No additional configuration needed (apart manager.conf privileged account), LesulaCC is able to connect every Asterisk Box.
LesulaCC allows Contact Center Supervisors to monitor or be alerted to Queues performance in real-time, wherever they are located.
LesulaCC main features are:
- Agents and Peer status supervising (Idle, Alerting, Busy, Caller/Callee Number, etc.).
- Inbound and outbound call volume.
- Call Detail.
- Call Actions (Just Hangup at this moment).
- Talking time.
- Handle time.
- Queues Call Details.
- Agents Call Details.
- Abandoned calls.
- Asterisk Dialplan Realtime Debug.
- Call Quality Realtime Debug.
LesulaCC can also store multiple Contact Center profiles a let Supervisor automatically connect them.
Tech Fusion IT-c team can offer you support to engineer and deploy a complete IP-PBX solution.
For info feel free to mail to firstname.lastname@example.org
Media5-fone is the best and most comprehensive Mobile VoIP SIP Softphone application that enables you to make and receive free or inexpensive phone calls over the Internet using a Wi-Fi or 3G connection.
You will need to have a VoIP SIP account with your company, or sign up for an account with your favorite SIP provider to use the Media5-fone.
The Media5-fone is compatible with Enterprise IP-PBX and SIP servers from:
• Aastra, Asterisk, Avaya, Broadsoft, Ericsson, FreePBX, FreeSwitch, Kamailio, Mediatrix 4100iPBX Series, Nokia-Siemens, Nortel CS1K, CS2k, CS5200, OpenSIPS, Panasonic, Samsung, Siemens OpenScape and HiPath, sipXecs, Sylantro, And many moreÉ
• Easy to configure:
- Flexible Codec selection and prioritization.
- Authentication Username.
- Dial plan.
- Pre-configured list of SIP Service Providers.
- Selectable ringtone.
- Customizable tab bar.
- Acoustic Echo Cancellation / Automatic Microphone Gain Control.
- Highly interoperable.
• Telephony/Communications Features:
- Make and Receive VoIP Calls over Wi-Fi and 3G.
- SIP URI Dialing support (e.g. sip:email@example.com).
- Audio routing (Loudspeaker / Handset / Wired Headset / Bluetooth Earpiece) / Mute.
- Complete access to the native contact list.
- Favorites / Speed Dial.
- Call History / Contact management.
- Voicemail integration / notification (SIP MWI).
- Hold / Resume / Redial.
- In call contact picture.
- Configurable DTMF (SIP INFO / RFCs 2833/4733 / RTP Inband).
- Included: G.711 (uLaw & ALaw), iLBC.
- Purchasable: Enhanced G.711, G.722 (Wideband), iSAC (HD), G.729 Annex-A & B.
- Support for UDP and TCP.
- Call Waiting / 2nd Call / Call Toggle / Call Transfer / Conference.
- Switch between Multiple SIP accounts (one registration at a time).
- Secured SIP Transport (TLS).
- Secured Encrypted Media (SRTP) with SDES.
- Languages: English, French, German, Spanish, Portuguese, Japanese.
• See the "Media5-fone support" link below
• Or email firstname.lastname@example.org
• Media5 Corporation is not a telephony provider; you must previously obtain a SIP account from your SIP IP-PBX IT administrator or from your SIP Service Provider. Media5-fone IS OPEN (UNLOCKED) AND THEREFORE NOT ATTACHED TO ANY SIP TELEPHONY PROVIDER.
• Allowing VoIP over 3G - Important note: Some mobile network operators may prohibit or restrict the VoIP (Voice over Internet Protocol) over their data network or impose additional fees and/or charges when using VoIP over their network. Please ask your mobile network operator before enabling the VoIP over 3G feature. Please contact the Media5-fone Technical Assistance Center for any concern related to VoIP over 3G.
Vimphone is an open source Voice Over IP (VOIP) / SIP client for Android. The source can be downloaded at: https://bitbucket.org/vimtura/vimphone-android/ and contribution is welcomed. The Linphone for Android project was forked in March 2013, and so it is based on version 2.0. Which, as of this time is all of the current Linphone for Android features. Credit for the logo goes to arrioch for doing his "Blawb" icon theme from which it came (http://arrioch.deviantart.com/art/Blawb-208106783).
The reason for the fork was out of all the SIP clients for Android, Linphone for Android seemed the best. However, it still included some bugs and features that we didn't like. The core features of Linphone for Android are:
- Sleek interface with support for multiple accounts.
- Support for the following audio codecs: speex, G711, ILBC, GSM, SILK, G722.
- Support for the following video codecs: VP8, H264 and mpeg4.
The improvements / enhancements made in Vimphone are (not a complete list):
- Improvements to the call answering UI, with support for specialised SIP headers (currently to display Asterisk queue details - yet to be documented).
- Fixed the in-call transfer / multi-call features.
- Added feature to pause/hold before transfer or adding another party.
- Added feature to mute DTMF when dialling number to transfer / add.
- Fixed a bug to enable switching between WIFI / 3G seamlessly (using the "re-invite" SIP feature).
- Re-skinning as "Vimphone"
- Enhanced the setup assistant.
- Added feature to disable screen auto-rotation (useful if you are laying down).
- Fixed "notifications" (integration into Android for status icons and so on). This includes utilization of the phones "lights" (LEDS) for status display. In the previous version the icon always appeared the same regardless of status (at least on my version of Android).
- Added feature to allow "exiting".
- Fixed bugs with call history display (it was almost impossible to click on the "details" icon as the target area was tiny).
- Added call interception so the standard android dialler can be used and calls can be routed via Vimphone automatically (based on various conditions).
- Probably lots more? I'll document more as I remember :)
- Feel free to bug me for feature requests, or join in if you want to help!
- Current Features
o - Call Monitor - view your call history
o - VoiceMail - list and listen to your voice mail messages
o - Feature Codes - View the list of available feature codes
o - Follow Me - view and change your Follow Me number list and settings
o - Phone Features - view and change all phone features like call waiting, DND and call forwarding
o - Phone Settings - view and change all phone settings like password and notifications
- Features in development:
- Call Monitor
o - Add call to phone contacts
- VmX Locator
Register your site, then distribute the app to your customers and staff.
-=[ Free, Site and Enterprise plans available ]=-
Visit: http://www.PBXLinQ.com for more information.
** PBX LinQ Mobile App is currently only available in English.
We are evaluating the potential of producing a multilingual version.
We will keep you informed of any decisions or news related to this.
If you have any problems or issues...
Please use our support portal to submit a ticket.
We will work to resolve any issues as quickly as possible.
- Version details
v3 (3.1) - November 27, 2013
- Fixed issue with destination value displayed in Call Monitor.
- Internal library update (from v1 to v1.2, including jQuery Mobile update from v1.3.0 to v1.3.2)
v2 - Internal release
v1 (1.1) - June 9, 2013
- Initial release
PBXLinQ.com (a division of e-City Solutions Inc)
2 Bloor Street West, Suite 700, Toronto, ON M4W 3R1
CA/US Toll-Free: +1 (855) 909-LINQ
UK National: +44 (20) 3519-LINQ
Worldwide: +1 (567) 263-LINQ
Some important keywords: PBX LinQ, FreePBX, Asterisk
Develop a custom SIP softphone app using ABTO Software’s VoIP SIP SDK and sell it on Android Market as your own product. Please check our demo app to try the basic softphone features. Using VoIP SIP SDK you can add any softphone features that you need. Contact us at www.voipsipsdk.com to receive more details.
Support both md5 and sha1 hashes.
Has the ability to store the last passphrase, which will be encrypted using AES with the phone's IMEI and IMSI to generate the key (hence the requirement of android.permission.READ_PHONE_STATE)
Source available at http://android.f00d.nl/opiekey/