Syntellect IVR SIP Trunking Configuration Guide Using AccessLine SIP Trunking

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Juliet Goodman
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1 And Syntellect IVR SIP Trunking Configuration Guide Using AccessLine SIP Trunking Interfacing AccessLine SIP Registrar and Syntellect SCP v8 IVR Platform By Jeff Thorness Introduction These Notes describe the procedures for configuring Session Initiation Protocol (SIP) Trunking between service provider AccessLine and a Syntellect IVR system. The AccessLine SIP Trunking service referenced within these Notes is designed for business customers. The service enables local and long distance PSTN calling via standards-based SIP trunks directly as an alternative to legacy analog or digital trunks, without the need for additional TDM enterprise gateways and the associated maintenance costs.

2 Prerequisite for Configuring Syntellect IVR with AccessLine SIP Trunks AccessLine Configuration Information You should have the configuration information from Accessline before you start the install. This information will provide settings for the SIP connection(s). The information you will receive from Accessline will be composed of the following information. Your order will be different from this information: Primary codec: G.711 Secondary codec: N/A SIP TRUNK ID dgwsid10941 PASSWORD ******** DNS NAME usbc.accessline.com IP ADDRESS PORT 6060 Call processing has been setup to allow up to 2 simultaneous calls. The following 911 callback number has been configured in our network as the default for this location; please test and make sure you are able to present this as the 911 callback number and program it to route appropriately on inbound This can also be used for testing inbound calls and has been setup as the default for outbound CLI. The following additional test number has also been provisioned

3 Set Up Instructions: 1) Within the Syntellect Console, drill down to the Dialogic Boards Driver and under HMP Startup Options set the following: a. IP Address: {value of the IP address for your IVR (ECS) } b. SIP Port: 6060 c. Max # of IP Calls: {qty of licensed HMP ports} d. Optionally-> reserve some of those IP resources for outbound e. Detect DTMF over SIP: No

4 2) Click on the last HMP board definition (the one labeled HMP (IP) and set the following values: a. Global Call IP protocol: SIP b. Fast Start and Fast Answer can both be either Yes or No Yes by default c. DTMF Transmission mode: RFC_2833 d. Codec 1 type: G.711 u-law

5 3) Click on the SIP Digest Authentication and set the following values: a. Realm: accessline b. Identity: sip: + {SIP TRUNK ID} <- IP address of Registrar c. Username: {SIP TRUNK ID} d. Password: {Password}

4) Drill down on Registration to SIP and then add a domain by right clicking on the right hand side of the screen and then choose Add from the menu. Options: a.

6 4) Drill down on Registration to SIP and then add a domain by right clicking on the right hand side of the screen and then choose Add from the menu. Options: a. Realm/Domain: {IP Address of Registrar} + :6060 in order to use port 6060 b. Registration interval: 60 <- This keeps the link alive by re-registering once per minute c. Accept other defaults for hops and register users

5) Drill down on SIP and then the real/domain that you just created and then click on Users. From there, right click on the right hand side of the screen and Add a user using the following parms: a.

7 5) Drill down on SIP and then the real/domain that you just created and then click on Users. From there, right click on the right hand side of the screen and Add a user using the following parms: a. Address: {SIP Trunk < - Registrar s IP Address b. Contact: {SIP Trunk +{your ECS IP Address} c. Register: Yes Once you have all these values set, you should be able to restart your CP Engine and accept phone calls. If you want to make outbound SIP calls, continue on.

1) Within your script, you must set your @CallingParty system variable in order to authenticate yourself with your registrar before making a call.

8 1) Within your script, you must set system variable in order to authenticate yourself with your registrar before making a call. Failure to set this value correctly will result in a 407 proxy authentication failure and the call will not initiate. sip: +{SIP TRUNK <-Registrar s IP Address

9 2) In the Make a Call dialogue, use SIP notation for the dial string: a. sip: +{phone# to <- IP address of Registrar and Port 6060

10 Fax Line(s) Setup FAXing is not supported by the Syntellect IVR system. Emergency (911) Hunt Group AccessLine configures a unique 911 callback number per customer location. The 911 callback number is registered with our E911 provider. When AccessLine receive a 911 call from the PBX, we forward the call to our E911 provider with the registered 911 callback number as the CLI (regardless of what was passed from the PBX). The 911 callback number is used by the Emergency responders to call back to the location.

NEC: SIP Trunking Configuration Guide V.1

NEC: SIP Trunking Configuration Guide V.1 FOR MORE INFO VISIT: CALL US EMAIL US intermedia.net +1.800.379.7729 sales@intermedia.net 2 NEC: SIP Trunking Configuration Guide V.1 TABLE OF CONTENTS Introduction...