Avaya IP office

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'''Avaya's IP Office IP500 V2''' Control Unit is a communication solution that can scale up to 272 telephones and 8 T1/E1 trunks. This system supports 8 expansion modules providing a combination of up to 272 analog, digital, and IP telephones, with capacity for 16 analog trunks or 8 digital trunks (192 T1/PRI channels or 240 E1 channels). Optionally, SIP trunks are also supported. This guide was kindly provided by our customers, it was tested and currently working for them, however, you may need to make modifications according to your own needs. ==Configuring an Avaya IP Office IP500v2 (Version 9.0)== This guide applies specifically to the Avaya IP Office platform running on an IP500v2 running the software release 9.0. Settings on different versions and editions of the Avaya IP Office platform will be similar but may difer slightly. ===Prerequisites=== *Ensure that you have already installed the required VCM modules to support sufficient voice channels. *Ensure that you have already added the required licenses for SIP trunks. *This guide assumes that you have already configured the proper network settings, that you already have network access to your system, that your system already has access to the Internet, and that DNS resolution already functions correctly from within the system. ===Configuring the network=== Under the LAN that is connected to the Internet, on the VoIP page, "SIP Trunks Enable" must be checked. Further down the page, take note of the incoming RTP port range. In your firewall, open port 5060 UDP and the port range from the RTP settings, 10000-20000 UDP by default, and forward them to your system's Internet facing LAN IP. [[File:1_LANVoIP.png|800px|thumb|left|Configuring LAN- click to enlarge]] <br clear="all" /> If you experience an issue with the RTP or NAT ports, you could set the Avaya's default ports (Minimum: '''46750''' and Maximum: '''50750''') '''NOTE: While STUN is not recommended, you can follow these instructions to set it up, if you require it:''' On the Network Topology page, set a public STUN server, "'''stun.stunprotocol.org'''" for example, the click "'''Run STUN'''". The detection process will take several minutes. Be patient. Once detection has complete, the "'''Firewall/NAT Type'''", the "'''Public IP Address'''", and the "'''Public Port'''" will automatically be filed in. Set a "'''Binding Refresh Time'''" to something appropriate, such as 60 seconds. Check "Run STUN on startup" to ensure that the system detects any changes to the public IP address if using a the system with a dynamic public IP. [[File:2_LANNetworkTopology.png|800px|thumb|left|LAN network topology - click to enlarge]] <br clear="all" /> Once the network configuration settings are applied and the firewall is properly configured, the system is now ready to accept the voip.ms SIP trunk. ===Adding the voip.ms SIP trunk=== In the "Line" section of the system, add a new "SIP Line". Choose a "Line Number", in this case we chose this to be Line 1. Enter the voip.ms server associated with your DID under the "ITSP Domain Name". In this case, montreal4.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers]). Choose a "Prefix", usually 9. Set the "Association Method" to use "VIA" in order to match against the resolved IP address from the ITSP domain name. [[File:3_SIPLine.png|800px|thumb|left|SIP line - click to enlarge]] <br clear="all" /> Add your voip.ms account credentials under the "SIP Credentials" page using the "User name" and "Password" sections. '''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 100000 or 100000_sub) '''Password:''' Your VoIP.ms SIP Password Set an "Expiry" to something low such as 5 minutes, and check "Registration required". [[File:4_SIPCredentials.png|800px|thumb|left|SIP credentials - click to enlarge]] <br clear="all" /> After adding your account credentials, select them on the "SIP URI" page in the "Registration" dropdown. To allow incoming calls, set the "Local URI", "Contact", and "Display Name" to a star ("*"). Set the Incoming Group and Outgoing Group as appropriate for your configuration. [[File:5_SIPURI.png|800px|thumb|left|SIP URI - click to enlarge]] <br clear="all" /> For a more efficient interchange with voip.ms, set the "Codec Selection" to custom and choose only the codecs supported by voip.ms and check "Re-invite Supported". [[File:6_SIPVoIP.png|800px|thumb|left|SIP - click to enlarge]] <br clear="all" /> If your networking and firewall were correctly setup, and all the previous settings applied, you should now have access to both outgoing as well as incoming voip.ms calls. Enjoy! ==Configuring an Avaya IP Office 500 v2 (Version 11.0.4)== This guide applies specifically to the Avaya IP Office platform running on software release 11.0.4. Settings on different versions and editions of the Avaya IP Office platform will be similar but may differ slightly. The following are notes and screenshots are amendments to the information that exists for version 9.0, along with relevant notes around settings that were required in order to get IP office features working correctly with the Voip.MS trunks. My screenshots will differ slightly as the system they are taken from requires no trunk access digits to be dialed, and uses e164 mode for outbound calls. The system also sits on the public internet so some sip settings may look slightly different. Setup notes: Some users on this system have their own DIDs and present those numbers on outgoing calls. Most users do not, and present the main incoming DID. This is all configured on the users' SIP tab. Mobile twinning is fully functional and shows the original caller data on the twinned device. ===Adding the voip.ms SIP trunk=== Under sip line configuration, remove all prefixes. Populate ITSP domain name [[File:Sipline1.PNG]] Populate ITSP Proxy Address [[File:Avayasipline2.png]] Inbound calls configuration. You set up an inbound call route for each incoming DID, and assign them to incoming group number (101 in this example) [[File:Avayasipline3in.png]] Outgoing calls configuration. "Use Internal Data" references the data on the SIP tab for each user when sending outgoing caller ID. In order for twinned calls (user mobility tab) from external callers to work, you have to configure the diversion header as shown. [[File:Avayasipline3out.png]] [[File:Avayasipline4.png]] [[File:Avayasipline6.png]] [[File:Avayasipline7.png]]

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