-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.
NOTE: CONTACT US ABOUT BRANDING / CUSTOMIZATION OF THIS APP FOR YOUR COMPANY!!
“Asterisk” is a registered trademark of Digium Inc.
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.
CONTACT US ABOUT BRANDING THIS APP FOR YOUR COMPANY!!
WARNING: Please use only versions from the appstore, we have reasons to believe some of the (russian) .apk files floating on the web contain a trojan that might steal username and password and make fraudulent calls. The official zoiper from google play is safe to use. If you want all features without paying, don't pirate it but ask your friends for the unlocking cheatcode instead. First rule about the gold club: don't publicly publish the cheatcode, but share it privately at will.
Contact us for whitelabel versions with your logo and company or for our VoiP SDK if you want to build your own solution or visit http://www.zoiper.com/en/voip-sdk
Available for both SIP and IAX systems, Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications.
Want to distribute it to your users ? Use the free automatic provisioning system on http://oem.zoiper.com and avoid tedious manual configurations!.
IMPORTANT: Zoiper softphone is a standalone client-side software VOIP phone application and is not bundled together with a voip service. To make and receive voip calls using Zoiper, you must subscribe to any SIP or IAX based service provider across the globe.
Zoiper's key features include:
- Bluetooth support (beta)
- Lowest battery usage with highest reliability / stability on google play
- Lowest latency of all android softphones
- Excellent audio quality, even on older devices
- Supports calling over 3G and WIFI
- Multiprotocol with SIP and IAX support, compatible with all RFC
- Background / multitasking support
- Native dialer integration
- Integration with the native android contact list
- Speakerphone mute and hold
- UDP and TCP transports (use TCP for better battery life!)
- Supports g711 (ulaw, alaw), speex, iLBC and gsm codecs
- Supports sending of DTMF
- DNS SRV
- Built-in echo cancellation
- STUN support
- Change ringtone per account
- call waiting
- Video support (gold users only)
- ZRTP / TLS support (gold users only)
- Call Transfer (gold users only)
- Wideband audio (gold users only)
Zoiper is also available as customized branding solutions or VoIP SDK, please contact us for more information.
- Phones that are unstable because they have broken audio implementations:
Alcatel one touch series (poor audio drivers cause system freezes and crashes)
HTC one m8 (causes freezes.)
We tried to contact both alcatel and HTC to help them fix the issue, they didn't bother to reply. If you can help, let us know.
Warning: using Zoiper as a default dialer may interfere with dialing 911 emergency services.
If zoiper does not work with your phone, please contact us at firstname.lastname@example.org and we will try to support your phone in future versions.
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
IMPORTANT NOTE: Bria is a standalone softphone and not a VoIP service. A SIP server or subscription with a SIP-based VoIP provider is required to make calls.
• Highly secure, SIP-based softphone with exceptional voice quality
• Purchase in-app Premium Features like Video Calls, Presence and Messaging or audio codecs to enhance your mobile softphone experience
• Pre-defined VoIP available when adding new accounts
• Multitasking support for background operation, such as fielding incoming calls while using other applications
• HD audio codecs including G.722, OPUS and SILK
• Supported accessories include headsets and headphones. Bluetooth™ support is dependent on Android device and operating system
• Available in the following languages: English, Chinese (Simplified and Traditional), French, Japanese, Korean, Portuguese, Russian and Spanish.
• Winner of the 2011 Product of the Year Award from Communications Solutions. For the full list, see http://www.counterpath.com/awards.html
STANDARD PHONE FEATURES
• Multiple account support for up to 12 accounts on any SIP-compliant server
• Contact List leveraging the device’s native contact directory
• Call display and voicemail indicator
• Speakerphone, mute and hold functions
• Call history with a list of received, missed and dialed calls
• Call recording
• Ringtones and contact avatars
• Dial plan support
• Multiple call support – swap between two active calls, merge and split calls, transfer calls (attended and unattended)
• Audio codecs include G.711a/u, G.722(HD), iLBC, GSM, OPUS and SILK
• Automatic codec selection to ensure optimal call quality
• Support for DTMF: the ability to enter numbers to use an auto attendant via RFC 2833, SIP INFO and in-band
• VPN support
• Advanced settings that enable users to control network traversal and media options
• Security and encryption via TLS and SRTP
• DNS SRV record lookups and call quality statistics
PREMIUM FEATURES – OPTIONAL IN-APP PURCHASE
• Messaging and Presence – with XMPP and SIP SIMPLE support
• Video Calls – using H.264 and VP8 video codecs
• G.729 Audio Codec
• AMR-Wideband Audio Codec
For more information on Bria Android Edition’s features, please visit: http://www.counterpath.com/bria-android-edition.html
IMPORTANT VOIP OVER MOBILE/CELLULAR DATA NOTICE
Some mobile network operators may prohibit or restrict the use of VoIP functionality over their network and may also impose additional fees, or other charges in connection with VoIP. You agree to learn and abide by your cellular carrier's network restrictions. CounterPath Corporation will not be held liable for any charges, fees or liability imposed by your carrier for use of VoIP over Mobile/Cellular Data.
CounterPath's Bria mobile products provide handling designed to redirect emergency calls to the Native Cellular Dialer when possible on a best reasonable commercial efforts basis, however this functionality is also dependent on the operating system of the mobile phone which is outside of our control and subject to change at any time. As a result, the official position of CounterPath is that CounterPath’s Bria product is not intended, designed, or fit for placing, carrying or supporting Emergency Calls. CounterPath will not be liable for any costs or damages arising either directly or indirectly from the use of the software for Emergency Calls. Using Bria as a default dialer may interfere with dialing emergency services.
DISA (Direct Inward System Access) allows you to provide an internal tone to external callers. This means that you can call your Asterisk system and dial a number as if you were using an internal position.
The DISA Auto Dialer application will dial your central, the password and the number of your recipient. It also manages the automatic choice in IVR menu.
3CXPhone for 3CX Phone System is an Android SIP client for 3CX Phone System V12 that allows you to make and receive office calls on your Android Smartphone from anywhere. Save on telecommunication costs by making internal office calls free of charge. Keep your mobile number confidential by using the office caller ID. Never miss an important call by answering office calls from anywhere.
3CXPhone for 3CX Phone System requires 3CX Phone System v12. Make sure you install the latest 3CX Phone System from http://www.3cx.com/downloads/3CXPhoneSystem12.exe or update to the latest 3CX Version 12 update. More information is available at http://www.3cx.com/blog/releases/unparalleled-mobility-with-3cxphone-for-android
3CXPhone for 3CX Phone System 12 gives you the following features:
-Make and receive calls via 3CX Phone System Version 12
-Hide your Mobile Phone Number
-Use a Single Number - Calls made to your desk phone will ring on your smartphone, either directly, after a delay or based on your status.
-Telephone bill savings – use company infrastructure to save on call costs. Call colleagues via an internal call rather than via the mobile network
-Seamless out of office or in office detection
-Integrated 3CX Tunnel bypasses remote firewall or provider issues
-Transfer & Hold calls
-Easily change your status
-Provision extension details via email
-Ability to take multiple calls simultaneously
-3CX Push notifications from within 3CX Phone System
-Ability to see presence of colleagues
-View call history
-Company phone book
-Easily setup conference calls
-HTTPS and HTTP connections to 3CX Phone System IP PBX
-Auto provisioning via Plug and Play or Email Attachment
-Configure your call forwarding
-View recorded calls
The new version V2.0 for Android, iPhone and iPad, available immediately, includes significant enhancements in the new ergonomic shape with a curved face interface, the integration of the address book and an account creation assistant.
Three major new capabilities have been added : a text messaging feature (chat) with delivery status notification, multiple calls and audio conferencing. In addition, Linphone supports now audio with speex , G711 ,ILBC, GSM, SILK, G722, OPUS, and video with VP8, H264 and mpeg4 codecs.
Exit button is located in the Settings->About page.
* High performances
* Rewriting/filtering rules for integration with Android
* SIP SIMPLE for Messaging
* Record calls
* Simple configuration
* Fancy UI inspired from 4.x Holo theme
* Many codecs (HD codecs, optimized codecs)
* Supported crypto : TLS for SIP and SRTP/ZRTP for media
More codecs (as the experimental Opus) and themes are available in plugins !
About permission required by CSipSimple :
Please report your bugs to the bug tracker of the project : it helps us to make a better software !!!
Users make the apps : It's opensource ! This application is distributed under GPLv3 license terms (http://www.gnu.org/licenses/gpl-3.0.txt). You can also try the development version (nightly builds) available on the project website if you experiment problems with this version.
keywords : SIP, codecs, voip, GPL, dialer, voice over IP, free softphone, sip softphone, sip phone, telephony, softphone software free, voip phone, softphone sip, pbx
-Multiple SIP accounts (from advanced settings -> general settings)
-Auto adapt to your environment (device/CPU/network/server and peers capabilities)
-Minimal CPU and battery usage while idle or running in the background. Size is only 3 MB.
-Call divert: mute/hold/forward/transfer/conference
-HD Audio, wideband and all common audio codec: G.711 (PCMU/PCMA), GSM, speex, iLBC, G.729
-Improved audio quality: AGC, AEC, PLC, noise reduction, silence suppression
-NAT/firewall traverse capabilities (with STUN, ICE, tunnel and related technologies)
-IM (chat), DTMF (RFC2833 and SIP INFO), Voicemail, Voice recording, Balance display, Caller ID
-Transport protocols: UDP/TCP/TLS. Support for DNS SRV
-VoIP tunneling and encryption (optional/automatic) with auto transport selection: UDP/TCP/TLS/HTTP/VPN (VoIP over HTTP)
-Peer to peer encrypted VoIP media
-VoIP to native dialer integration and integration with the native contact list
...and many more
Customized softphone builds are available for VoIP service providers. Contact email@example.com. Details: http://www.mizu-voip.com/Support/Wiki/tabid/99/topic/Customized%20Android%20softphone/Default.aspx
Please use the forum for bug reports:
Visit the homepage for more details about this SIP client:
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/