Newsgroup: it.tlc.telefonia.voip
Titolo: softphone

qualcuno sa se esiste un provider che offre numeri locali gratuiti in
ingresso, con numerazione USA, con la possibilita di inserire i
parametri in un softphone?


www.ipkall.com
 ti dà un numero gratuito americamo sotto forma di DID
dopo di che ti crei un account gratuito su 
www.voxalot.com
 e ricevi le
chiamate
numero americano -> IPKALL -> VOXALOT -> tuo softfone
se hai già un account betamax (voipstunt, voipbuster, poivy, ecc) puoi
usare
quello per ricevere usando il loro softfone, inserisci il ipKall il
tuo username
e il provider (
sip.voipstunt.com
, ecc.)
se hai bisogno di altri chiarimenti fammi sapere

The award-winning VoIP SDK is a powerful and highly versatile set of
tools designed to dramatically accelerate development of SIP
applications. VoIP SDK enable flexible, rapid application development
options for creating desktop softphone clients, Rich Internet
Applications (RIA) and telephony extensions into existing enterprise
and web applications, It's complies with IETF and 3GPP standards,
delivers high performance, and provides advanced API layer for full
user control and flexibility.
VoIP SDK built on open standards, leveraging SIP (Session Initiation
Protocol) to create, run and terminate multi- media sessions. They
are
built on popular codecs for voice and video (include G.711, G.722.1,
G.
723.1, G.729(b), GSM, iLBC, SPEEX(WB), AMR-WB, H.263, H.263-1998 and
H.
264), and the SIP SIMPLE standard for Instant Messaging and Presence.
Typically, customers that build their own graphical user interface
and
applications. The SDK has components for managing audio and video
media, as well as signaling and call control, IM - XMPP and SIMPLE.
A commercial license for the VoIP SDK which consists of complete DLL,
LIB, documents and one years full support (including e-mail technical
support, patches and upgrades). With the commercial license you have
the right to distribute your applications (not this SDK) to any
amount
of end-users as you like
Download here the free VoIP Enterprice SDK trial:
http://www.sipvoipsdk.com/sip_voip_sdk.zip
More informations 
http://www.sipvoipsdk.com

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

My project required to integrate dialer
with voice playing ability, fax detection and digit recognition
I bought
<a href="
http://voipcore.com
">
http://voipcore.com
</a>
and that guys provide great support