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General SIP Configuration

Making Outbound Calls (Termination)

*****NOTE*****This document is deprecated. Please see OnSIP Trunking. 07/30/14

Server
Our SIP server for PSTN Gateway service is sip.jnctn.net

Authentication
The standard approach is used to challenge all requests (the Proxy-to-User Authentication scheme as outlined in Section 22.3 of RFC 3261). Customers are identified by their username. If identification by username fails, the authorization username is used. The authentication domain is "jnctn.net". Registration is not required to terminate calls.

*** Please note that your VOIP password is not the same as the password used to login to the Junction Networks web site. Your VOIP password can be found by logging into www.jnctn.com and going to the VOIP page. ***

Caller-Id
If the username is a valid phone number (ten or eleven digits), it is used as the caller-id number associated with a call. Otherwise, if the display name is a valid phone number, it is used as the caller-id number associated with a call.

Caller-id name information cannot be propagated to the PSTN. On the PSTN, the caller-id number is matched to the number database of the local telephone company to determine the name provided with caller-id. As such, caller-id name information cannot be reliably delivered. However, the display name is propageted where it can be - on-network calls for example.

Numbering
The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'. International numbers need to be prefixed with "011".

DTMF
DTMF is supported as defined in RFC 2833.

Example SIP Call Flow

   User   Junction Networks
     |                                       |
     |  INVITE F1                            |
     |-------------------------------------->|
     |  407 Proxy Authorization Required F2  |
     |<--------------------------------------|
     |  ACK F3                               |
     |-------------------------------------->|
     |  INVITE F4                            |
     |-------------------------------------->|
     |  100 Trying F5                        |
     |<--------------------------------------|
     |  180 Ringing F6                       |
     |<--------------------------------------|
     |  200 OK F7                            |
     |<--------------------------------------|
     |  ACK F8                               |
     |-------------------------------------->|
     |         Both Way RTP Media            |
     |<=====================================>|
     |  BYE F9                               |
     |<--------------------------------------|
     |  200 OK F10                           |
     |-------------------------------------->|
     |                                       |

Receiving Inbound Calls (Registration & Origination)

Server
Our SIP server for PSTN Gateway service is sip.jnctn.net

Registration
Registration is a common SIP procedure. It allows us to learn the current location of a customer and thus where to send their calls. The standard approach is used to authenticate registrations (the Proxy-to-User Authentication scheme as outlined in Section 22.3 of RFC 3261). Customers are expected to register using their Junction Networks username and password. The authentication domain is "jnctn.net".

*** Please note that your VOIP password is not the same as the password used to login to the Junction Networks web site. Your VOIP password can be found by logging into www.jnctn.com and going to the VOIP page. ***

Trunking Configuration - sip.jnctn.net
Inbound calls from sip.jnctn.net are addressed to the phone number dialed (the DID). Calls are delivered to the most recently registered contact. The DID may be any that the customer subscribes to. As this configuration allows the routing of incoming calls based on the DID that was called, registering with sip.jnctn.net can be helpful if a customer subscribes to more than one DID. This server is recommended for use with SIP servers and IP PBXes.

Unauthenticated Inbound Calls
Customers are required to receive unauthenticated inbound calls. Inbound calls originate from the server with which a customer is registered.

DTMF
DTMF is supported as defined in RFC 2833.

Miscellaneous

Consolidating Accounts Under a Single Bill
Multiple accounts can be consolidated under one customer account for billing purposes (to share a single balance, for example). Please contact your Junction Networks sales representative for assistance.

Other Alternatives
We do provide a range of SIP configuration alternatives - one of which may be more suitable for your situation than our standard setup. In particular, we can provide specialized configurations with respect to authentication (for example, authentiation based on IP address only), codecs, dtmf, as well as custom failover scenarios. If you wish to use another scheme or have special needs, please contact us, and a Junction Networks sales representative will get back to you shortly.

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