Configuring Avaya IP Office 500 for Spitfire SIP Trunks This document is a guideline for configuring Spitfire SIP trunks onto Avaya IP Office 500 and includes the settings required for Inbound DDI routing and Outbound CLI presentation. The settings contained within have been tested and are known to work at the time of testing. SIP trunk details such as account number and password will be provided separately.
Spitfire SIP Configuration for Avaya IP Office 500v1 and v2 at Software level 6.0.xx Preliminary checks SIP trunk Licences generated and installed o IPO LIC SIP TRNK RFA 1 [202967] o IPO LIC SIP TRNK RFA 5 [202968] o IPO LIC SIP TRNK RFA 10 [202969] o IPO LIC SIP TRNK RFA 20 [202970] VCM base modules installed on IP 500v1 or v2 Main Unit (VCM32, VCM64 or built-invcm10 on Combination card) Caveats with Spitfire SIP Trunking on Avaya IP Office running on software level 6.0.xx Spitfire SIP Trunking relies on an RPID (Remote Party Identification) header for outbound CLI presentation. At level 6.0.xx the Avaya IP Office does not include a RPID header in an outbound SIP INVITE thus Spitfire will always present the registration account as the default CLI for all outbound calls. Spitfire SIP Trunking does not support anonymous@xxxx.xxx as a mechanism for withholding outbound CLI. In order to withhold CLI over a Spitfire SIP Trunk, please prefix the outbound call with 141. Only G711 ALAW or ULAW codecs are accepted For purposes of this document LAN1 on the IP Office has been used and assigned an IP address of 10.0.0.1/24 with an internal default gateway on 10.0.0.254. The IP Office is behind a NAT ed router. Spitfire SIP Trunk 442031234567 with registration has been used as an example. The default? dial short-code has been left to route to the default Main ARS table (ARS Table ID 50).
Setting up Spitfire SIP via Avaya Manager tool. Enabling LAN 1 for SIP Trunks 1. On the System\LAN1\VoIP tab, tick the SIP Trunks Enable box and click OK
2. Click on Line option in left-hand tree. Right Click and select new and click on SIP Line
a. Tick the Registration Required box b. In the ITSP Domain Name field enter spitfiretsp.net c. In the ITSP IP Address field set the IP Address to 83.218.143.16 d. In the Call Routing Method drop-down box select To Header e. In the Use Network Topology info drop-down box select None and click OK
3. Click on SIP Credentials tab and click on the Add button a. Enter supplied registration account in User name and Authentication Name fields e.g. 442031234567 b. Enter supplied password c. Set Expiry field to 5 and click OK
4. Click on VoIP tab a. Set Compression Mode to G.711 ALAW 64K b. Tick Re-invite Supported tickbox and click OK
5. Click on the SIP URI tab and click on the Add button a. On the Registration drop-down box select the newly created registration account e.g. 442031234567 b. If the default Incoming Group and Outgoing Group numbers do not conflict with any other existing trunk types accept the default values, otherwise enter new group numbers. c. Set Max Calls per Channel to correspond with the number of channels ordered with the Spitfire Trunk and click OK
6. Select IP Route from Left-hand tree, right click and select new a. Enter default route 0.0.0.0 with mask 0.0.0.0 via gateway 10.0.0.254 and destination to LAN1. If a default route already exists via a different Gateway then enter IP route as 83.218.143.16 with 255.255.255.255 mask via Gateway 10.0.0.254 and destination to LAN1. b. Click OK
7. Click On Incoming Call Route from left-hand tree, right click and select new a. On Line Group Id set it to corresponding group, created under the SIP URI form (See point 5) b. Set Incoming Number to supplied DDI Range numbers. The IP Office will, in default, resolve the incoming number from right to left but for simplicity the full number is used e.g. 442031234567 c. Select Destination tab and select from the drop-down box the required internal DN e.g. 2000 Main Group. Click OK
8. Click On ARS from left-hand tree a. Select Main ARS form b. Set Dial Delay Time to 3 seconds. This field will depend on the type of handsets used on the IP Office. The IP Office, in default, does not support enbloc dialling although on the 16xx and 96xx series IP Phones enbloc dialling can be enabled. The delay time is an inter-digit delay allowing the user enough time to complete the number in full. After last digit has been entered and after 3 seconds of no further digits, the IP Office will assume that the number has been dialled in full. c. Remove the default? route. Add new route in format as table below. Keep in mind that local dial plans must include the STD code e.g. for the London Area 3, 7 and 8 are local numbers and will need to be prefixed with 020. The screenshot below will show London local number routes. 0N 0N @spitfiretsp.net Dial 0
9. Save Configuration and set for immediate reboot