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
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:
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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 Border Element PSTN
Customer Premises
Phones and server in private address space.
Managed Router does NAT.4.2. Installation number and DIDs
Installation number must be declared under: Numbering > Installation Number
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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:
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
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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
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
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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
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.
