🚀 go-pugleaf

In uk.telecom.voip john hamilton <bluestar95@mail.invalid> wrote:
> The vcf load up on the desktop and you have to open up each one in
> alphabetical order. Not reasonable if you have several hundred. ( Is this
> their idea of a joke?). The vcc I cannot open. I don't use messenger, but
> am willing to install it; if this is the only way I can access the list
> when I want to.

Looks like Messenger is the only way to get them out, though you then have a
problem of exporting them from Messenger.  It appears the CTT is an XML
format, so you might have some mileage extracting them directly (eg perhaps
you can persuade Excel to parse the XML as a table).

On the subject, use a generic SIP softphone configured for Voipstunt - that
way, it's easy to change provider when they put their prices up (which they
have - you may well find one of their sister brands is cheaper now).  The
only thing the Windows Voipstunt (etc) client is useful for is registering
numbers for CLID - do that once and everything else can be done via a
softphone and the website.  I find X-lite version 2 (not version 3) good (if
a bit quirky), but YMMV.

Theo

http://ipphonesdk.com

VoIP SIP SDK:

• g729 and g723 Codec´s support
• Multiple and single Codec selection support
• Failure codes support (get SIP Message Response Code, SIP Message
Response Text)
• RTP/RTCP Port setting (for inbound RTP traffic)
• Reduce audio latency and audio latency settings (properties:
MinPrefetchCount,    MaxPrefetchCount, MaxRTPPackets)
• Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped,
OnRemoteMediaStarted,    OnRemoteMediaStoped)
• Get used codec per line
• Custom Ringtone (play wav) support (property: RingtoneFile)
• Play wav to a selected phone line (methods: StartPlayingAtLine,
StopPlayingAtLine)
• Redirect Call to other phone line
• Load and Save Configurations (methods: LoadConfiguration,
StoreConfiguration)
• Complete new, re-written and updated samples with source code
• and much more!

 Here is a list of the main features of the VoIP SIP SDK::

• Easily make and receive SIP (Session Initiation Protocol) based
phone calls through any    SIP gateway or SIP compliant IP-Telephony
service provider
• VoIP conferencing with crystal clear sound even for both low and
high-bandwidth users
   G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16 and
g729 & g723 Codec
• Open standards-based and interoperable with all of the major
equipment vendors
• UDP and TCP support
• Multi-party voice conference support/ Conference split and join,
locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling
• Integrated STUN, TURN and ICE support
<• Comes with new sample SIP Proxy Server    to provide in bundle with
the SIP    Client ActiveX a ready up own SIP VoIP    and Instant
Messaging network solution.
• P2P support for directly connections    between 2 SIP clients
without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted    SIP account settings in
your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
   on-the-fly - also during a conversation/    conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DND (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or    suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCM    WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the    remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB
included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NAT/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully sample applications for various programming languages such as
sample source code    for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and
Delphi
• For .NET framework as well and all development environments with
ActiveX support

Easy, familiar, event-driven call control ActiveX of VoIP SIP SDK:
• Easy to use; quick development
• Support for .NET framework and all development environments with
ActiveX support
• Very easy to incorporate

Rich call control feature set of VoIP SIP SDK:
• Multi-party voice conference support (Conference split/ join,
locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• SIP Instant messaging
• Locally mixed conferences
• Hold/Mute
• Call transfer
• Call forwarding and rejection

Industry leading SIP support of VoIP SIP SDK:
• RFC3261 compliant SIP stack
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support

Comprehensive configuration support of VoIP SIP SDK:
• Select media input/output devices (on-the-fly as well during a
conversation/conference)
• Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE)
• SIP proxy

Advanced digital voice processing features of VoIP SIP SDK:
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… VoIP SIP SDK supports even much more!

http://ipphonesdk.com