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Alcatel-Lucent Omnipcx Enterprise Rel 9.0 Sip Trunk Inter-Connectivity With Paetec Configuration Guide
Alcatel-Lucent Omnipcx Enterprise Rel 9.0 Sip Trunk Inter-Connectivity With Paetec Configuration Guide
Rel 9.0
SIP Trunk Inter-connectivity with PAETEC
Configuration Guide
1 6/30/2009
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not permitted without written authorization from Alcatel-Lucent and PAETEC
Table of Contents
1 Introduction ........................................................................................................................3
2 System Architecture ..............................................................................................................4
3 Software Releases .................................................................................................................5
3.1 OmniPCX4400 Software Release ...........................................................................................5
3.2 PAETEC Software Releases .................................................................................................5
3.3 IP Addresses................................................................................................................... 5
4 Configuration.......................................................................................................................9
4.1 Extensions ..................................................................................................................... 9
4.2 SIP Trunk ...................................................................................................................... 9
4.2.1 SIP Trunk Group.........................................................................................................9
4.2.2 SIP Gateway ........................................................................................................... 15
4.2.3 SIP Proxy ............................................................................................................... 16
4.2.4 SIP External Gateway ................................................................................................ 16
4.3 Quality of Service .......................................................................................................... 19
4.4 Compression ................................................................................................................ 20
4.5 ARS and Discriminator..................................................................................................... 22
4.5.1 NPD ..................................................................................................................... 22
4.5.2 ARS Tables ............................................................................................................. 23
4.5.3 Numbering Command Table ........................................................................................ 28
4.5.4 Call back translator: ................................................................................................. 29
5 Summary of Test Results ....................................................................................................... 30
5.1 Summary of Test Plan Results............................................................................................ 30
5.2 OmniPCX Enterprise limitations ......................................................................................... 30
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1 Introduction
The purpose of this document is to provide a configuration guide for an Alcatel-Lucent OmniPCX Enterprise
interconnected to the public Network through a PAETEC SIP Trunk (LPG Platform). The architecture and
configuration described in this document is based on validation test that took place at PATEC Labs in Houston, TX
between the Alcatel-Lucent OmniPCX Enterprise Rel 9.0 and PAETEC SIP Trunk.
The document also includes the limitations identified during the validation test.
PAETEC will configure SIP trunks on its network and provide customers with IP addresses of SIP Proxy, and phone
numbers assigned to customers before scheduled service activation date. SIP trunks are offered through MPLS
connections with proper QoS to guarantee the security and quality of voice traffic. MPLS configuration will be done by
PAETEC throughout its network and all the way to the CPEs provided by PAETEC.
If you are in need of technical support for PAETECs Dynamic IP SIP Trunking, please contact our Customer Care
department at 877.340.2600 or customercare@paetec.com. When contacting Customer Care, please include the
customer account number.
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2 System Architecture
A-L Call
Server B
Subnet 1: 10.1.x.x
PAETEC SIP
Gateway
10.3.1.10
Subnet 2: 10.2.x.x
A-L Call
Ext. Ext. Server
9090 9091
The Alcatel-Lucent OmniPCX Enterprise (OXE) used for carrying the SIP trunk test consisted basically of:
- One Common hardware shelf MR3 equipped with one Call Server (CPUA), one GD (with MADA-3 daughter card),
one MIX board that included 4 Z ports (for analog phones and faxes) and 8 digital interfaces (for proprietary Alcatel-
Lucent digital sets).
A Gateway Applicative interface (GA) equipped with a MADA-3 daughter board was also included in the MR3 shelf
(for additional compressors if needed). A PRA T1 (for a North America ISDN interface) and APA8 (for analog trunk
interface) were also installed in this MR3 shelf).
- A second Common hardware shelf MR1 equipped with a second Call Server (CPUB) was installed to ensure
redundancy.
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- Both CPUs (A and B) were configured in Spatial Redundancy (over 2 separate IP subnets : 10.1.x.x and 10.2.x.x)
with CPU domain address defined as: oxe.PAETEC.net ; DNS SRV was used for address resolution (as it is mandatory
to use DNS to resolve the OmniPCX Enterprise CPU address in case of Spatial redundancy and not using the IP
address directly).
There are 2 IPTouch phones (4038) and 2 IPTouch phones (4028) connected to the OmniPCX Enterprise on the LAN.
3 Software Releases
3.3 IP Addresses
The table below is an example (for configuration guide purpose) of the CPUs and Gateway Drivers IP addresses:
Host Name
IP Address Network Mask Default Gateway Host Description
OmniPCX4400
10.1.1.10 cpua 255.255.255.0 10.1.1.1 OmniPCX CPU a
10.1.1.11 cpuam 255.255.255.0 10.1.1.1 CPU A role main
10.2.1.10 cpub 255.255.255.0 10.2.1.1 OmniPCX CPU b
10.2.1.11 cpubm 255.255.255.0 10.2.1.1 CPU B role main
10.1.1.12 GD 255.255.255.0 10.1.1.1 Gateway Driver
10.1.1.13 GA 255.255.255.0 10.1.1.1 Gateway Applicative
Driver
The CPU IP parameters can be changed using netadmin-m command on the OmniPCX Enterprise command prompt.
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(1)OXe1> netadmin m
1. 'Installation'
2. 'Show current configuration'
3. 'Local Ethernet interface'
4. 'CPU redundancy'
5. 'Role addressing'
6. 'Serial links (PPP)'
7. 'Tunnel'
8. 'Routing'
9. 'Host names and addresses'
10. 'Copy setup'
11. 'Security'
12. 'DHCP configuration'
13. 'SNMP configuration'
14. 'VLan configuration'
15. 'Node configuration'Use
16. 'Ethernet redundancy'
17. 'History of last actions'
18. 'Apply modifications'
0. 'Quit'
Do you want to activate internal name resolver (y/n default is 'n') ? y Put y only in case of spatial redundancy
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Name used when the CPU role is MAIN (default is xma000001) ? cpuam
Address used when the CPU role is MAIN (default is is 192.168.0.3) ? 10.1.1.11
Name used when the CPU role is MAIN (default is xmb000001) ? cpubm
Address used when the CPU role is MAIN (default is 192.168.0.3) ? 10.2.1.11
The CPU Role main address (which is used by the Verizon SIP Gateway) can also be changed using the netadmin m
command as below:
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In order to change the GD IP parameters, you have to connect on the configuration port of the GD card, login as
admin (password : admin) , type su followed by the command mgconfig.
Note: Please make sure to change the GD IP address before changing the CPU one.
Using this menu you can update the GD IP parameters as well as the role main address on the CPU.
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4 Configuration
4.1 Extensions
The table below shows a list of extensions created for the test with the SIP trunk. All the extensions IP addresses
are configured to be static.
2 SIP Parameters
Define the SIP specific parameters, such as SIP Internal Gateway, SIP Proxy, SIP External Gateway and trusted IP
addresses.
Define the parameters that allow the users to make and receive calls on the SIP trunk.
Comprises the configuration of Access prefix, ARS tables, Dialing discriminator, DID translation, NPD.
The following pages contain snapshots (mgr tool) of the PCX configuration related to the SIP trunk implementation.
The most relevant parameters are highlighted.
The SIP Trunk Group Global parameters are configured under /Trunk Groups ; choose Review/Modify ; select Trunk
Group Id = 10 then <F1> to validate your selection:
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Trunk Group Name : SIP
UTF-8 Trunk Group Name : ---------------------------------------
Number Compatible With : 0
Remote Network : 10
Shared Trunk Group + False
Special Services + Nothing
Node number : 1
Transcom Trunk Group + False
Auto.reserv.by Attendant + False
Overflow trunk group No. : -1
Tone on seizure + False
Private Trunk Group + False
Q931 Signal variant + ISDN all countries
SS7 Signal variant + No variant
Number Of Digits To Send : 0
Channel selection type + Quantified
Auto.DTMF dialing on outgoing call + YES
T2 Specification + SIP
Public Network COS : 31
DID transcoding + True
Can support UUS in SETUP + True
Implicit Priority
Activation mode : 0
Priority Level : 0
Preempter + NO
Incoming calls Restriction COS : 10
Outgoing calls Restriction COS : 10
Callee number mpt1343 + NO
Overlap dialing + NO
Call diversion in ISDN + YES
+------------------------------------------------------------------------------
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The SIP Trunk Group Local Parameters are configured under /Trunk Groups ; Descend Hierarchy ; Trunk Group ;
Review/Modify ; Enter Trunk Group Id = 10 then <F1> to validate your selection:
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B Channel Choice + YES
Channels: Attendant Control (Rsvd) : 0
Redirection For ACD (Dissuasion) + NO
DTO joining + NO
Consultation Call On B Channel + NO
Automated Attendant + NO
Calling party Rights COS : 0
TS Overflow + YES
Number To Be Added : --------
Charge Calling And ADN Creation + YES
Logical Channel + 1__15 & 17__31
Use Split Access + NO
Heterogeneous Remote Network + NO
COS Restrictions - Barring mode + Not Restricted / Not barred
ARS Class of service : 31
External Access Server + NO
CSTA Tracking MCDU Trk : --------
+-------------------------------------------------------------------------+
PAETEC send the complete 10 digits phone number; those digits need to be translated to
the 4 digits extension number; in order to do that; you need to manage the DID
translator by going to:
/Translator/External Numbering Plan/Default DID Tranlator / Create
An ABC-F Trunk Group (ex. Trunk Group Id 11)needs to be created. This trunk group is
usually used for SIP devices (SIP Phones , External SIP Voice Mail etc). This trunk
group has to be defined in the SIP Gateway Parameter (Never Use the ISDN Trunk Group in
the SIP Gateway)
The SIP Trunk Group Local Parameters are configured under /Trunk Groups ; Descend
Hierarchy ; Trunk Group ; Review/Modify ; Enter Trunk Group Id = 10 then <F1> to
validate your selection
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+-Review/Modify: Trunk Groups--------------------------------------------------+
Node Number (reserved) : 1
Trunk Group ID : 11
Trunk Group Type + T2
Trunk Group Name : ABCF
UTF-8 Trunk Group Name : ---------------------------------------
Number Compatible With : -1
Remote Network : 11
Shared Trunk Group + False
Special Services + Nothing
Node number : 1
Transcom Trunk Group + False
Auto.reserv.by Attendant + False
Overflow trunk group No. : -1
Tone on seizure + False
Private Trunk Group + False
Q931 Signal variant + ABC-F
SS7 Signal variant + No variant
Number Of Digits To Send : 0
Channel selection type + Quantified
Auto.DTMF dialing on outgoing call + YES
T2 Specification + SIP
Public Network COS : 31
DID transcoding + False
Can support UUS in SETUP + True
Implicit Priority
Activation mode : 0
Priority Level : 0
Preempter + NO
Incoming calls Restriction COS : 10
Outgoing calls Restriction COS : 10
Callee number mpt1343 + NO
Overlap dialing + YES
Call diversion in ISDN + NO
+------------------------------------------------------------------------------
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+-Review/Modify: Trunk Group----------------------------------------------+
Node Number (reserved) : 1
Trunk Group ID : 11
Instance (reserved) : 1
Trunk Group Type + T2
Public Network Ref. : ------
VG for non-existent No. + YES
Entity Number : 0
Supervised by Routing + NO
VPN Cost Limit for Incom.Calls : 0
Immediate Trk Listening if VPNCall + YES
VPN TS % : 50
CSTA-Monitored + NO
Max.% of trunks out CCD : 0
Ratio analog.to ISDN cost : ------
TS Distribution on Accesses + YES
Quality profile for voice over IP + Profile #1
IP Compression Type + G 711
Use of volume in system + YES
Announcement for dial tone + NO
Private to Public Overflow + YES
End-to-end dialing + NO
DTMF end-to-end signal. + NO
Trunk group used in DISA + NO
DISA Secret Code : ----
Routing To Manager + NO
Trunk COS : 31
Sending of Progress message + YES
No. of digits unused (ISDN) : 0
B Channel Choice + YES
Channels: Attendant Control (Rsvd) : 0
Redirection For ACD (Dissuasion) + NO
DTO joining + NO
Consultation Call On B Channel + NO
Automated Attendant + NO
Calling party Rights COS : 0
TS Overflow + YES
Number To Be Added : --------
Charge Calling And ADN Creation + YES
Trunk COS : 31
Sending of Progress message + YES
No. of digits unused (ISDN) : 0
B Channel Choice + YES
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Channels: Attendant Control (Rsvd) : 0
Redirection For ACD (Dissuasion) + NO
DTO joining + NO
Consultation Call On B Channel + NO
Automated Attendant + NO
Calling party Rights COS : 0
TS Overflow + YES
Number To Be Added : --------
Charge Calling And ADN Creation + YES
Logical Channel + 1__15 & 17__31
Use Split Access + NO
Heterogeneous Remote Network + NO
COS Restrictions - Barring mode + Not Restricted / Not barred
ARS Class of service : 31
External Access Server + NO
CSTA Tracking MCDU Trk : --------
+-------------------------------------------------------------------------+
The SIP Gateway is configured under /SIP/SIP Gateway; Review/Modify ; 2nd Instance = 1 then <F1> to validate.
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DNS type + DNS SRV
SIP DNS1 IP Address : 10.3.1.100
SIP DNS2 IP Address : 192.168.2.20
SDP in 18x + True
Cac SIP-SIP + False
INFO method for remote extension + False
Dynamic Payload type for DTMF : 101
The SIP Proxy is configured under /SIP/SIP Proxy ; Review/Modify ; 2nd Instance =1 the <F1> to validate
The SIP External Gateway is used Inbound and Outbound SIP calls.
For Inbound calls, It is a must (as per the OXE requirement) to define an External Gateway for each IP address of
fqdn on the received SIP messages (either in the P_Asserted_Identity or the From objects). Otherwise, Inbound calls
will not be routed in by the OXE.
The parameter SIP Remote Domain should include this IP address or the fqdn of the SIP SBC.
During the validation test with PAETEC, all Inbound calls and Ounbound calls were going through the SBC with the
IP address 10.1.3.10 from which SIP messages were received (either in the P_Asserted_Identity or in the From
header) was defined as External Gateway for Inbound calls and Outbound Calls (Gateway Id 1)
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Note: The OmniPCX Enterprise continuously monitors the connectivity to each Gateway by managing the parameter
Supervision timer. The default value of this parameter is 0 (which means that the OmniPCX Enterprise does not
check the status of the Gateway), once this parameters is managed to a value (in seconds), the OmniPCX
continuously checks the connectivity by the end of the interval and puts the link out of service once the
connectivity is lost. The link is put back in service once the connectivity is found back alive.
The SIP External Gateways are managed under /SIP/ SIP Ext Gateway ; Choose Review/Modify OR Create ; then
enter the SIP Ext Gateway Id in case you are modifying (In this configuration only the Id 1 is managed as there was
not a fallback Gateway)
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Minimal authentication method + SIP None
INFO method for remote extension + False
Send only trunk group algo + False
To EMS + False
Dynamic Payload type for DTMF : 101
+------------------------------------------------------------------------------+
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4.3 Quality of Service
PAETEC has requested to implement QoS on the SIP Trunk with TOS = 46 and 802.1p = 5. To implement that an IP
Quality of Service COS is configured ( Category 1), this by going to /IP ; Choose Descend Hierarchy; Choose IP
Quality Of Service COS ; Choose Review/Modify ; Enter IP QoS COS =1 ; press <F1> to validate
The GD is then configured with the QoS COS by going to /Shelf ; Choose Descend Hierarchy; Choose Descend
Hierarchy again then Choose Ethernet Parameters; Choose Review Modify; Enter Shelf address : 1; Board address :
0; press <F1> to validate:
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Review/Modify: Ethernet Parameters
Node Number (reserved) : 1
Shelf Address : 1
Board Address : 0
Shelf - Board Address : 1-0
Interface Type + GD
Board IP Address : 10.1.1.12
IP NetMask : 255.255.255.0
Default Gateway IP Address : 10.1.1.1
IP Quality of service : 1
Cryptographic box address : ---------------------------------------
Board Ethernet Address : 00:80:9F:38:83:34
Interworking with Gatekeeper + NO
Gatekeeper ID : -1
Numb. of sig. channels IP Phones : 0
Numb. of sig. channels inter-ACT : 0
IP Domain Number : 0
E164 Number List Index : -1
Gateway H323 name : --------------------------------
4.4 Compression
PAETEC has requested to use G729 as compression on the SIP. To configure that you need to configure the default
system compression and the IP domain extra-domain compression as below:
Choose Compression Type in the parameter System Options , press <F1> to validate
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Review/Modify: Compression Parameters
[ All instances ]
[ Set filters ]
[ Select attributes ]
Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
System Option + Compression Type
Change the parameter Compression Type to G729 then press <F1> to validate
The next step is to change the compression type for the IP domain for which the IP Phones belong to (by default IP
Domain 0). The SIP trunk compression type is changed by managing the parameter Extra Domain Compression
Type.
To manage that go to /IP/Descend Hierarchy/IP Domain/Review-Modify ; choose the IP Domain 0 then press <F1> to
validate
Review/Modify: IP domain
Node Number (reserved) : 1
Instance (reserved) : 1
IP Domain Number : 0
IP Domain Name : --------------------
Country + Default
Intra-domain Coding Algorithm + Without Compression
Extra-domain Coding Algorithm + With Compression
FAX/MODEM Intra domain call transp + NO
FAX/MODEM Extra domain call transp + NO
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Domain Max Voice Connection : -1
IP Quality of service : 0
Contact Number : ------------------------------
Backup IP address : ---------------------------------------
Trunk Group ID : -1
IP recording quality of service : 0
Time Zone Name + System Default
Calling Identifier : ------------------------------
Supplement. Calling Identifier : ------------------------------
SIP Survivability Mode + NO
4.5.1 NPD
The ARS tables use NPD 33 which is configured as below. The NPD configuration can be reached under
/Translator/External Numbering Plan/Numbering Plan Discriminator (NPD); Choose Review/Modify or Create (if you
are creating it); Enter Description Modifier Id = 33 (in case of Review/Modify)
To configure an ARS Table, go to /Translator/Automatic Route Selection/ARS Route List/ ; Choose Descend
Hierarchy ; Choose ARS Route; Choose Review/Modify or Create ; in case of Review/Modify enter the ARS Route list
and Route number followed by <F1> to validate:
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Review/Modify: ARS Route
Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
ARS Route list : 0
Route : 1
Name : SIP
Trunk Group Source + Route
Trunk Group : 10
No.Digits To Be Removed : 0
Digits To Add : ------------------------------
Numbering Command Tabl. ID : 1
VPN Cost Limit : 0
Protocol Type + Dependant on Trunk Group Type
NPD identifier : 33
Route Type + Public
ATM Address ID : -1
Preempter + False
Quality
Enter Voice and Fax
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Review/Modify: ARS Route
Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
ARS Route list : 1
Route : 1
Name : SIP
Trunk Group Source + Route
Trunk Group : 10
No.Digits To Be Removed : 0
Digits To Add : ------------------------------
Numbering Command Tabl. ID : 1
VPN Cost Limit : 0
Protocol Type + Dependant on Trunk Group Type
NPD identifier : 33
Route Type + Public
ATM Address ID : -1
Preempter + False
Quality
Enter Voice and Fax
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Review/Modify: ARS Route
Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
ARS Route list : 2
Route : 1
Name : SIP
Trunk Group Source + Route
Trunk Group : 10
No.Digits To Be Removed : 0
Digits To Add : ------------------------------
Numbering Command Tabl. ID : 1
VPN Cost Limit : 0
Protocol Type + Dependant on Trunk Group Type
NPD identifier : 33
Route Type + Public
ATM Address ID : -1
Preempter + False
Quality
Enter Voice and Fax
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Review/Modify: ARS Route
Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
ARS Route list : 4
Route : 1
Name : SIP
Trunk Group Source + Route
Trunk Group : 10
No.Digits To Be Removed : 0
Digits To Add : ------------------------------
Numbering Command Tabl. ID : 2
VPN Cost Limit : 0
Protocol Type + Dependant on Trunk Group Type
NPD identifier : 33
Route Type + Public
ATM Address ID : -1
Preempter + False
Quality
Enter Voice and Fax
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4.5.3 Numbering Command Table
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4.5.4 Call back translator:
External Call back translator is managed to display the correct calling number format and to be able to recall an
unanswered number. This is configured
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5 Summary of Test Results
The test plan has revealed the limitations and anomalies in the next sections
- The OmniPCX Enterprise does not support Fax over G-711. It only supports T-38 protocol.
- When the OmniPCX Enterprise local User transfers an incoming or outgoing call to another external party.
The display on this external party will always show the Number of the local user and not the external party
he/she is connected to.
- The OmniPCX Enterprise does not support feature codes on SIP Trunk (Not needed as features are available
locally on the PBX)
END OF DOCUMENT
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