manual alu oxo user guide

AT&T IP Flexible Reach Alcatel-Lucent OXO Configuration Guide

Alcatel-Lucent OmniPCX Office R7.1 Configuration Guide

For Use with AT&T’s IP Flexible Reach Service

Version 1 / Issue 1 Date July 28, 2009

www.alcatel-lucent.com


2 of 11 Alcatel-Lucent OmniPCX Office R7.1 Configuration Guide

TABLE OF CONTENTS

1 Introduction............................................................................................................................. 3 2 Special Notes .......................................................................................................................... 3 3 Overview................................................................................................................................. 3 4 Configuration Guide ............................................................................................................... 4 5 Troubleshooting .................................................................................................................... 11


Alcatel-Lucent OmniPCX Office R7.1 Configuration Guide of 11

1. Introduction This configuration guide describes how to configure Alcatel-Lucent OmniPCX Office (OXO) to connect to AT&T’s Flexible Reach Service. OXO must be equipped with VoIP board to configure SIP trunk. OXO was tested with release R7.1 build 28.001. This is a general description of OXO configuration for ATT SIP trunk purpose for more details please refers to PublicSIPtrunking_ConfigurationGuide_OXO_ed1 document and Expert documentation for R710.

2. Special Notes T.38 Fax is supported with the following exception. Fax is not supported with Cisco TDM gateways. Emergency 911/E911 Services Limitations While AT&T IP Flexible Reach services support E911/911 calling capabilities in certain circumstances, there are significant limitations on how these capabilities are delivered. Please review the AT&T IP Flexible Reach Service Guide in detail to understand these limitations and restrictions.

3. Overview OXO was tested in the following configuration:

OXO was configured to support 8 SIP trunk channels, has 3 IP phones, 1 UA phone, 1 SG3 fax machine and 1 analog modem: Multitech MT9234ZBA-USB used for SG3 fax and softphone Pimphony ver. 6.3. We have connected 1 POTS line to simulate rollover when SIP trunk is down.



4. Configuration Guide

4.1. Checking the software version OXO needs to be run on release 710.028.001 or higher to compliant with ATT SIP service. To check OXO software please go to: Data Saving & Swapping > SW-Downloading

OXO

Private Side Public Side

FAX

Customer Site

Customer firewall

IP BorderElement

PSTN

Customer Premises

Phones and server in private address space.

Managed Router does NAT.

Customer sites connect to ATT IP Border Element

Application Servers, Network

Gateways, etc.

BVOIP Network



4.2. Installation number and DIDs Installation number must be declared under: Numbering > Installation Number

DID numbers should be assign to stations under: Numbering > Dialing Plan > Public Dialing Plan



4.3. IP Configuration By default system has the following IP configuration: Main CPU: 192.168.92.246 VoIP Master: 192.168.92.248 Default GW: 192.168.92.246 To change IP setting you need to go to: Hardware and Limits > LAN/IP configuration > LAN Configuration to change default GW:

And Hardware and Limits > LAN/IP configuration > Boards to change CPU and VoIP master IPs:

After the change system needs to be reset.



4.4. Enabling SIP for VoIP and setting trunk channels By default system has enabled H.323 as VoIP protocol and no channels are assigned to VoIP trunk. VoIP protocol must be switched to SIP and the number of channels must be increased under: Voice Over IP > VoIP: Parameters > General

After this changed system prompt to reset VoIP board. Additionally the noteworthy label SimullpAlt must be changed to 01, in order to do this please go to: System Miscellaneous > Memory Read/Write > Other Labels

Click on Details change value from 00 to 01 click on Modify and Write:



4.5. Configuring VoIP trunk and trunk group The VoIP trunk must be configured as public under: External Lines > List of Accesses > VoIP: Details and check the Public trunk box Also please choose desired number of channels for SIP trunk:

To create trunk group select: External Lines > List of Trunk Groups

- create a trunk group containing the VoIP trunks - verify Traffic sharing settings and barring matrix



Next the List Index needs to be created for new VoIP Trunk Group under: Numbering > Automatic Routing Selection > Trunk Group List

- create a list index for trunk group

Now the ARS route must be created to send all number to VoIP trunk group list: Numbering > Automatic Routing Selection > Automatic Routing: Prefixes

- add a line that routes prefix 1 for example to the VoIP trunk with following settings:

Activation – YES Network – Pub Prefix – 1 Ranges – 0-9 Substitute – leave blank TrGpList – enter the trunk group list index Called(ISVPN/H450) – het Destination – SIP Gateway IP Type – Static IP Address – in test scenario: 207.242.225.200 Gateway Alive Protocol – SIP Option Gateway Alive Timeout – enter desired time period in seconds Gateway Bandwidth – choose right value depends on number of VoIP channels and

carrier bandwidth



Codec/Framing – Default - In a default configuration (Codec/Framing set to default), there is no preferred codec/framing on the Alcatel-Lucent OmniPCX Office Communication Server side. The selected codec/framing depends on the remote party Index of Gateway – choose gateway index

In above example you can see that for prefix 1 main line and subline were added to

rollover calls to another ATT SIP gateway IP: 207.242.225.201 and if both IP gateways down calls are sent to POTS lines. SIP Option is used for a keep alive mechanism. Note that AT&T customer care will provide the customer with the 2 AT&T SIP endpoints during provisioning. To create Gateway Parameters in OMC go to: Numbering > Automatic Routing Selection > Gateway Parameters - add new index with following settings:

RFC 3325 – Yes Remote SIP port – 5060 Index of SIP Numbers Format – 1 DNS – Disabled

To create SIP Public Numbering in OMC go to: Numbering > Automatic Routing Selection > SIP Public Numbering - configure index 1 with following settings:

Calling Format (Outgoing) – National Calling Prefix (Outgoing) – leave blank Called Format (Outgoing) – National/International Called Prefix (Outgoing) – leave blank Called Short Prefix (Outgoing) – leave blank Calling Format (Incoming) – National Calling Prefix (Incoming) – leave blank Called Format (Incoming) – National Called Prefix (Incoming) – leave blank



4.6. Creating ARS Prefix To create trunk access prefix for ARS in OMC main trunk group prefix needs to be modified under: Numbering > Dialing Plans > Internal Dialing > Main Trunk Group - modify with following setting:

Start – 9 End – 9 Base – ARS NMT – Drop Priv – No Accept new setting by with Modify key.

5. Troubleshooting

- If calls can’t be established please check gateway alive status under: Numbering > Automatic Routing Selection > Automatic Routing: Prefixes

- If there is problem with resources please check VoIP Counters under: Voice Over IP > VoIP: Traffic Counters

- For additional troubleshooting external ethereal traces are required. If you need a technical assistance please call ALU Technical Support line to open a case, the phone number is +1 (877) 729-7299.

manual alu oxo user guide

Documents

configuration guide

oxo private

following configuration

oxo software

overview oxo

ip phones

software version oxo

atts flexible reach