Results 1 - 15 of 532 for voip call center. (0.017 seconds)
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Call Shop Billing solution will help you start generating new revenue stream by providing low cost telecommunication services. In a typical call shop operation customers come to the location of the shop, make a telephone call from one of the phone booths to any place in the world and get charged upon completion. -
Terrasoft Call Center Server is a separate module of popular Terrasoft CRM that provides tools for registration and analyzing of incoming and outgoing calls, redirecting incoming calls to selected Terrasoft CRM users, process calls from different contacts according to the established rules. -
COMIREL released full functional TSP TAPI Service Provider for 2 SNOM VoIP phones. These features are supported: Make call - Answer call - Call termination - Swap hold calls - Hold - Unhold - Third party conference - Consultations - Interconnection - Forwarding by condition (always, if busy, after time). AsnomTAPIduo supports 2 SNOM phones independently and parallel. All active phone VoIP profiles are made available as TAPI lines. -
You can get rates from different vendors ( reseller / wholeseller/ Partners) and script will calculate those rates based on your settings and gives you one compiled rates, You can get different rates for different customers and types of customers like wholeseller, resellers e.t.c -
PrettyMay Call Center for Skype (PMCCS) is a 100% software-based Skype PBX that replaces traditional proprietary hardware PBX / PABX. It allows SMBs to quickly and affordably implement a Skype PBX (aka PABX) system with auto-attendant, interactive voice response (IVR), Automatic Call Distribution (ACD), call recording and personalized voicemail capabilities - and a lot more as well. -
Answering Machine for scripting your own professional call center business scripts using a voice modem. Features Caller-ID, Wave Playback, Wave Recording, Digit Monitoring, POP3 Email Manipulation, Speech Recognition and Synthesis. -
SkypeTransfer extends the basic capabilities of SkypeTM and enables you to transfer Skype calls to anyone in your Skype contact list. A great tool designed to enable Skype manual call routing. Using SkypeTransfer, the operator can place the caller on 'Hold', 'Transfer' to another Skype contact or 'Return' to the original caller. -
Designed for networking and telecom engineers, admins and communication technology educators and students, this VOIP technology quick guide covers all related technologies: SIP, H.323 and xGCP signaling architecture and technologies, VOIP transport and QoS technologies, and video and audio codecs. -
conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast. -
conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast. -
conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast. -
conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast. -
conaito VoIP ActiveX library for developers of VoIP audio applications, such as voice chat, conference, VoIP, providing real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast -
SkypeAttendant is a virtual attendant to automatically answer your SkypeIn or Skype to Skype calls. Callers simply say the name of the person they are calling and SkypeAttendant automatically transfers the Skype call to that person. SkypeAttendant uses text to speech technology which means you can quickly customize all voice prompts using a simple text editor. Visit www.skypetransfer.com for a live demo. Click the Skype 'I'm Online' button. -
An easy to use Internet phone for Mac OS X using VoIP to provide crystal clear voice quality in an attractively sleek interface. The quick configuration will allow you to get talking to other iSoftPhone users in seconds. iSoftPhone is integrated with the Mac OS X address book to make calls and manage contacts. Simply dial the number of the person you wish to talk with or select them from contacts to call.