SIP
SIP, the so-called Session Initiation Protocol, is on everyone's lips. SIP allows an IP-based communication - without the need for traditional phone lines.
Forwarding from your Asterisk PBX
So if you are already running a SIP-capable PBX (eg Asterisk), but you do not have the time or know-how is lacking to develop complex voice applications, then you can easily let your calls be handled by our services and servers.
Voice applications via SIP
We create a voice dialog and provide a SIP address to which you forward your callers. When the dialog is finished we either disconnect the call or send it back to your PBX. Results from the dialog will be posted in real time via HTTP or SIP signaling.
Additional features
We offer you some extensions:
- Optional web interface for analysis and parametrization of your module
- Upload audio, Audio download
- VoiceXML usage - just in case you'd prefer to program on your own
- Forwarding to fixed and mobile network numbers worldwide
- Free forwarding to external SIP addresses
- Any other ideas? Talk to us.
IP infrastructure
We support all major audio codecs and are connected with a carrier-neutral Internet backbone - for optimal performance and minimal loss of quality.
The cost of a SIP-based voice application
You pay a one-time fee for the creation of your individual voice application, and a low minute-fee for the use of our platform - simple and transparent.
