Support Pages

The new AGEphone Desktop 2 is SIP based internet telephony softphone that enables users to choose from a host of Voice over IP (VoIP) service providers. Based on our own, compact size microSIP stack it offers uncompromised performance on desktop, mobile and embedded systems alike.

As many first time users are not familiar with VoIP, these pages offer a thorough overview of the softphone. The "Installation" menu point to the left shows how to get AGEphone Desktop 2 up and running, "Howto" shows common usage scenarios and "In Detail" describes every option available.

Apart from that, you can find some more answers to common questions in the FAQ. If what you are looking for isn't there either and you have other questions or suggestions, please feel free to turn to our contact page, the forums, give us a call or chat with us directly in the box to the left.

In general, we don't expect major difficulties running AGEphone Desktop on any Windows 2000 / XP / Vista / Seven system that fulfills the minimum system requirements given below. That said, we strongly recommend that you download and test the Free Trial Version before purchasing:

System Requirements

OS	Windows XP / Vista /  7
CPU	x86 800 MHz (1 GHz recommended)
Memory	512 MB (1 GB recommended)
HDD	10 MB for program files
Connection	Up- and Downstream of each 80kbps or above
Other	Speakers and microphone, headset or handset

Version History

Ver.2.04 [2009.10.09]
* Fixed hangup issue when recording calls

Ver.2.03 [2009.05.11]
* Audio buffer problems fixed
* SPEEX audio quality improvements
* State modus selection fixed (works instantly now)
* Startup settings: "Hide on start" checkbox added
* Twitter: "Load history at startup" checkbox added

Ver.2.02 [2009.04.15]
* Audio quality improvements
* STUN supported NAT traversal issues fixed

Ver.2.01 [2009.02.17]
* Address book import bug fixed
* Volume control bug fixed
* SDP less INVITE packet compliance

Ver.1.85 [2009.10.02]
* SIP-protocol bugfixes
* Backup server option added

Ver.1.83 [2008.08.04]
* No ringtone on call fix

Ver.1.80 [2008.07.02]
* microSIP stack update
* AIR-GUI: Supports transparency effects
* Paging function extension: Auto-answer over call-info header
 
Ver.1.75 [2008.04.04]
* Management of more than 10 audio devices possible.
* Basic QoS options added
* extended handset support

Ver.1.66 [2007.06.25]
* VGA and WVGA support (AGEphone WM5)
* Hardware keyboard support (AGEphone WM5)
* Improved audio buffer.

Ver.1.65 [2007.05.25]
* Improved Address Book Filter
* Session Timer Handling improved
* Fixed several smaller problems

Ver.1.64 [2007.04.13]
* Popup on incoming call improved
* USB handset support improved
* fixed a problem with session timer (UAC)
* Implementation of standby mode (Mobile)

Ver.1.63 [2007.03.28]
* Fixed security vulnerabilities when receiving malformed SIP packets

Ver.1.62 [2007.03.16]
* Fixed security vulnerabilities when receiving malformed SIP packets
* Fixed a problem where calls could not be answered with the enter key
* Fixed a problem when adding contacts from the call history to the address book

Ver.1.61 [2007.10.03]
* Improved address book
* Different USB handsets can now be selected in the setup menu
* Fixed Direct Sound input issue on Vista
* IPv6 global addresses can now be selected in the interface menu

Ver.1.60 [2007.02.28]
* DirectSound support
* IPv6 support
* Improved dial plan settings for international calls
* Local music on hold
* New, improved address book

Ver.1.56 [2006.12.15]
* Registration to servers improved
* Fixed a problem with INVITE cancel

Ver.1.55 [2006.11.24]
* Fixed a problem with multiple users on one computer
* Audio device control improved

Ver.1.54 [2006.11.10]
* Windows Vista support
* Audio device control improved

Ver.1.53 [2006.11.05]
* Improved voice quality
* Fixed a problem with call forwarding

Ver.1.52 [2006.10.17]
* Improved voice quality during recording
* Fixed a problem with the settings screen

Ver.1.51 [2006.10.04]
* Fixed a problem with the address book
* Fixed a problem with call forwarding

Ver.1.50 [2006.09.29]
* Added call forwarding and "Provider Crossover" function
* Improved jitter buffer and voice quality
* Improved UPnP connectivity

Ver.1.42 [2006.08.18]
* Added function to automatically execute programs on call connection / disconnection
* Improved SIP stack
* Fixed an issue that occurred when entering the serial number in the PDA version

Ver.1.41 [2006.08.11]
* Improved Outlook support (a popup is shown when an Outlook contact calls)
* Enabled input of IP-adresses for P2P-connections with standard phone keys
* Fixed an issue that occurred when entering the serial number
* Fixed an issue concerning P2P-connections

Ver.1.40 [2006.07.25]
* Corrected a security issue with possible buffer overflow
* Improved voice quality (ptime = 10)
* Faster startup
* Multilanguage support improved

Ver.1.38 [2006.07.05]
* Automatic selection in case of multiple network interfaces
* Multilanguage support improved
* Volume control improved

Ver.1.37 [2006.06.14]

* Added German language interface
* UPnP connectivity improved
* Corrected some smaller internal problems

Ver.1.36 [2006.05.02]
* Improved NAT traversal algorithm
* Improved SIP stack
* Corrected some smaller internal problems

Ver1.35 [2006.03.20]
* Improved UPnP stack
* Corrected some other problems

Ver1.34 [2006.03.03]
* Improved NAT traversal algorithm
* Recording of complete communication is now possible

Ver1.33 [2006.02.06]
* Increased support of IP telephone servers for use within extension numbers
* Supports calls from an increased number of network environments

Ver1.32 [2006.01.17]
* Increased number of supported IP telephone services
* Improved NAT traversal engine

Ver1.31 [2005.12.28]
* Increased number of UPnP supporting routers
* When shutting down, program now waits until communication is completely closed
* Corrected problem that in a specific environment dialing could not be canceled

Ver1.30 [2005.12.22]
* Improved the SIP stack
* Native implementation of UPnP
* Added calling rule function (call melodies)

Ver1.24 [2005.10.20]
* Added support for serverless calls
* Added "Click to Reply", "Click to Disconnect" Function
* Corrected some other problems

Ver1.23 [2005.09.28]

* Altered answering machine mode so that it remembers ready status on shutdown
* Corrected problem that with some services an error occurred when calling
* Corrected some other problems

Ver1.22 [2005.09.01]
* Added RFC3581 function
* Added NAT Mode

Ver1.21 [2005.08.25]
* Improved UPnP connectivity

Ver1.20 [2005.08.03]

* Added skin switching function
* Improved the SIP stack

Ver1.13 [2005.07.26]
* Improved GUI drawing management
* Added audio visualizer
* Added function for backing up settings

Ver1.12 [2005.07.07]
* Improved the SIP stack

Ver1.11 [2005.06.15]

* English support
* Official support of USB Internet Phone Model IP-001
* Improved address book function
* Improved quality of RFC2833

Ver1.10 [2005.05.24]
* Added vibration mode
* Improved connectivity to IP telephone service providers
* Improved NAT traversal algorithm
* Improved DTMF send quality

Ver1.04 [2005.04.26]
* Made P2P communication with AGEphone PDA version possible
* Improved NAT traversal algorithm

Ver1.03 [2005.03.24]

* Improved DTMF send method
* Now displays status of NAT traversal and register status of each service
* Corrected problem of one way only communication with some audio devices

Ver1.02 [2005.03.14]
* Voice of communication partner can now be recorded
* Enabled communication up from a 29.2kbps band
* Corrected problem of one way only communication with some audio devices

Ver1.01 [2005.03.03]
* Support of call progress tone
* Corrected problem that audio device could not close normally
* Added function to edit phone numbers entered in phone book