Alcatel-Lucent OmniPCX Office R7.1 Configuration Guide

Transcription

For Use with AT&T’s
IP Flexible Reach Service
Version 1 / Issue 1
Date July 28, 2009
www.alcatel-lucent.com
AT&T IP Flexible Reach
Alcatel-Lucent OXO Configuration Guide
TABLE OF CONTENTS
1
2
3
4
5
Introduction............................................................................................................................. 3
Special Notes .......................................................................................................................... 3
Overview................................................................................................................................. 3
Configuration Guide ............................................................................................................... 4
Troubleshooting .................................................................................................................... 11
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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.
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Customer Premises

Phones and server in private
address space.
PSTN
Managed Router does NAT.
Customer sites connect to ATT IP
Border Element
Application
Servers,
Network
Gateways,
etc.
Customer Site
Private Side
BVOIP
Network
Public Side
Customer firewall
FAX
IP Border
Element
OXO
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
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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
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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:
After the change system needs to be reset.
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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:
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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
- create a trunk group containing the VoIP trunks
- verify Traffic sharing settings and barring matrix
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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
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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
Calling Prefix (Incoming) – leave blank
Called Format (Incoming) – National
Called Prefix (Incoming) – leave blank
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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.
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