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Hi there,<br />
I'm using call manager 7.1.3. I've got 10 sites where I want them to be set up in such a way that when one of the users at a site dials '9', it transfers the call to the receptionist. Note that if Office A rings 9, it should not transfer the call to receptionist at Office B. Everybody office shud be able to call their receptionist only when they dial 9. Now, I know that can be achieved by defining 10 separate partitions and 10 separate CSS (with each partition only dialable from that CSS) but is there a way to accomplish the task with lesser config in Call manager. Can we use local route group ? 
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Hi there,
I'm using call manager 7.1.3. I've got 10 sites where I want them to be set up in such a way that when one of the users at a site dials '9', it transfers the call to the receptionist. Note that if Office A rings 9, it should not transfer the call to receptionist at Office B. Everybody office shud be able to call their receptionist only when they dial 9. Now, I know that can be achieved by defining 10 separate partitions and 10 separate CSS (with each partition only dialable from that CSS) but is there a way to accomplish the task with lesser config in Call manager. Can we use local route group ? 

Call Manager solution required for routing to reception

Hi there,
I&#39;m using call manager 7.1.3. I&#39;ve got 10 sites where I want them to be set up in such a way that when one of the users at a site dials &#39;9&#39;, it transfers the call to the receptionist. Note that if Office A rings 9, it should not transfer the call to receptionist at Office B. Everybody office shud be able to call their receptionist only when they dial 9. Now, I know that can be achieved by defining 10 separate partitions and 10 separate CSS (with each partition only dialable from that CSS) but is there a way to accomplish the task with lesser config in Call manager. Can we use local route group ? 

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

Voice Over IP

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

IP Telephony

A router is a networking device that forwards data packets between computer networks. Routers perform the "traffic directing" functions on the Internet. The most familiar type of routers are home and small office cable or DSL routers that simply pass data, such as web pages, email, IM, and videos between computers and the Internet. More sophisticated routers, such as enterprise routers, connect large business or ISP networks up to the powerful core routers that forward data at high speed along the optical fiber lines of the Internet backbone. Though routers are typically dedicated hardware devices, use of software-based routers has grown increasingly common.