Alcatel Lucent Configuration Guide

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Alcatel-Lucent OmniPCX Enterprise Rel 9.0 SIP Trunk Inter-connectivity with PAETEC Configuration Guide...

Description

Alcatel-Lucent OmniPCX Enterprise Rel 9.0 SIP Trunk Inter-connectivity with PAETEC Configuration Guide

Author Alcatel-Lucent / PAETEC

Date 08/11/2009

Nature Creation of this document

Comments Ed.1

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Table of Contents Introduction ........................................................................................................................3 System Architecture ..............................................................................................................4 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 1 2 3

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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 Configuration & Support 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 PAETEC’s Dynamic IP SIP Trunking, please contact our Customer Care department at 877.340.2600 or [email protected] When contacting Customer Care, please include the customer account number.

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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

Ext. 9090

Ext. 9091

A-L Call Server

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 AlcatelLucent 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. 4

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- Both CPU’s (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.

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3.1

Software Releases

OmniPCX4400 Software Release

Release: Software version:

3.2

9.0 H1.301.28b

PAETEC Software Releases PAETEC’s Dynamic IP SIP Trunking Solution Components 6.4.0.8 LGP MGC/MG Acme Packet Session Border Controller

3.3

6.1.0 Patch 1

IP Addresses

The table below is an example (for configuration guide purpose) of the CPUs and Gateway Drivers IP addresses:

IP Address 10.1.1.10 10.1.1.11 10.2.1.10 10.2.1.11 10.1.1.12 10.1.1.13

Host Name OmniPCX4400 cpua cpuam cpub cpubm GD GA

Network Mask

Default Gateway

Host Description

255.255.255.0 255.255.255.0 255.255.255.0 255.255.255.0 255.255.255.0 255.255.255.0

10.1.1.1 10.1.1.1 10.2.1.1 10.2.1.1 10.1.1.1 10.1.1.1

OmniPCX CPU a CPU A role main OmniPCX CPU b CPU B role main Gateway Driver Gateway Applicative Driver

The CPU IP parameters can be changed using netadmin-m command on the OmniPCX Enterprise command prompt. Note: This configuration needs to be performed on both CPU’s 5

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(1)OXe1> netadmin –m Alcatel e-Mediate IP Network Administration =========================================== 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' What is your choice ? 1 Do you want to erase the existing setup (y/n default is 'n') ? y Do you intend to use IP/X25 on your system (y/n default is 'n') ? n Warning: the node name must be unique. Enter node name (default is alcateloxeus) ? oxe The node name has to match the name defined in the DNS server Do you want to activate internal name resolver (y/n default is 'n') ? y Put y only in case of spatial redundancy Enter local CPU position (a or b) for duplicated configuration or n for single CPU (default is 'n') ? a Use a if you are configuring cpu a and b if you are configuring cpu b Ethernet interface setup ======================== CPU name (default is xa000001) ? cpua CPU address (default is is 192.168.0.2) ? 10.1.1.10 Network mask (default is 255.255.255.0) ? 255.255.255.0 6

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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 CPU redundancy setup ==================== Twin CPU name (default is xb000001) ? cpub Twin CPU's address (default is is 192.168.0.2? 10.2.1.9 Network mask (default is 255.255.255.0) ? 255.255.255.0 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 Default router setup ==================== Do you use an external gateway as default router (y/n default is n) ? y Default router name (default is router) ? routera Default router address (default is is 192.168.0.1) ? 10.1.1.1 Twin router setup ================= Default router name (default is routerb) ? routerb Default router address (default is 192.168.0.32) ? 10.2.1.1 Please wait, working ... The setup has been correctly achieved. 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. eMGD login: admin Password: [[email protected] admin]$ su [[email protected] admin]# mgconfig Welcome to the e-MGD IP configuration tool -----------------------------------------MAC address 00809F388334 1. Startup mode Static 2. IP address 192.168.0.28 3. Netmask 255.255.255.0 4. Gateway address 192.168.0.1 5. CS role address 192.168.0.26 6. CS redundancy role address 0.0.0.0 7. Crystal number 1 (Manual allocation) 8. IP QoS menu 9. BACKUP menu 10. Duplex and speed mode menu 11. Download binaries timeout 1200 12. Passive CS address 0.0.0.0 0. Exit Using this menu you can update the GD IP parameters as well as the role main address on the CPU.

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4.1

Configuration

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. DN 8090 8091 8092

4.2

Type Extension Extension Extension

Set type 4038 4038 4028

DID 585-754-9090 585-754-9091 585-754-9092

IP address 10.1.1.100 10.1.1.101 10.1.1.102

SIP Trunk

The PCX configuration for SIP trunking consists of the following parameters: 1 – SIP Trunk Group By Defining a trunk group with virtual accesses used for the SIP calls. Basically, it determines the number of channels available for SIP communications (i.e. number of simultaneous calls). 2– SIP Parameters Define the SIP specific parameters, such as SIP Internal Gateway, SIP Proxy, SIP External Gateway and trusted IP addresses. 3– ARS and Discriminator 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. 4.2.1

SIP Trunk Group

The SIP Trunk Group Global parameters are configured under /Trunk Groups ; choose Review/Modify ; select Trunk Group Id = 10 then to validate your selection: +-Review/Modify: Trunk Groups--------------------------------------------------+ ¦ ¦ ¦ Node Number (reserved) : 1 ¦ ¦ Trunk Group ID : 10 ¦ ¦ ¦ ¦ Trunk Group Type + T2 ¦ 9

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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 to validate your selection: +-Review/Modify: Trunk Group----------------------------------------------+ ¦ ¦ ¦ Node Number (reserved) : 1 ¦ ¦ Trunk Group ID : 10 ¦ ¦ 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 + Always VoIP ¦ ¦ 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 11

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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 ┌─Create: Default DID num. translator ───────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ First External Number : 5857549090 │ │ │ │ First Internal Number : 8090 │ │ Range Size : 3 │ │ Unique Internal Number + NO │ │ │ └──────────────────────────────────────────────────────────────────────┘ 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 to validate your selection 12

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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 ¦ 14

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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 : -------¦ ¦ ¦ +-------------------------------------------------------------------------+

4.2.2

SIP Gateway

The SIP Gateway is configured under /SIP/SIP Gateway; Review/Modify ; 2nd Instance = 1 then to validate. ┌─Review/Modify: SIP Gateway───────────────────────────────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ │ │ SIP Subnetwork : 11 Enter the Trunk Group Id of the ABC-F Trunk Group and the remote network entered in the ABC-F Trunk Group │ SIP Trunk Group : 11 │ │ IP Address : 10.1.1.1 │ │ Machine name - Host : oxe This name is automatically managed same as the Node name in netadmin. Make sure that this is the same as the name managed in the DNS server │ SIP Proxy Port Number : 5060 │ │ SIP Subscribe Min Duration : 1800 │ │ SIP Subscribe Max Duration : 86400 │ │ Session Timer : 1800 │ │ Min Session Timer : 900 │ │ Session Timer Method + RE_INVITE │ │ DNS local domain name : COMPANY.COM │ 15

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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 │ │ └──────────────────────────────────────────────────────────────────────────────┘ 4.2.3

SIP Proxy

The SIP Proxy is configured under /SIP/SIP Proxy ; Review/Modify ; 2nd Instance =1 the to validate ┌─Review/Modify: SIP Proxy─────────────────────────────────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ │ │ SIP initial time-out : 500 │ │ SIP timer T2 : 4000 │ │ SIP connection duration : 180000 │ │ Recursive search + False │ │ Minimal authentication method + SIP None │ │ Authentication realm : --------------------------------------- │ │ Only authenticated incoming calls + False │ │ Framework Period : 3 │ │ Framework Nb Message By Period : 25 │ │ Framework Quarantine Period : 1800 │ │ │ └──────────────────────────────────────────────────────────────────────────────┘

4.2.4

SIP External Gateway

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) 16

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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) +-Review/Modify: SIP Ext Gateway-----------------------------------------------+ ¦ ¦ ¦ Node Number (reserved) : 1 ¦ ¦ Instance (reserved) : 1 ¦ ¦ SIP External Gateway ID : 1 ¦ ¦ ¦ ¦ Gateway Name : --------------------------------------- ¦ ¦ SIP Remote domain : 10.3.1.10 ¦ ¦ PCS IP address : --------------------------------------- ¦ ¦ SIP Port Number : 5060 ¦ ¦ SIP Transport Type + UDP ¦ ¦ RFC3262 Forced use + False ¦ ¦ Belonging Domain : oxe.COMPANY.COM ¦ ¦ Registration ID : --------------------------------------- ¦ ¦ Registration ID in P_Asserted + True ¦ ¦ Registration timer : 0 ¦ ¦ SIP Outbound Proxy : --------------------------------------- ¦ ¦ Supervision timer : 30 ¦ ¦ Trunk group number : 10 ¦ ¦ Pool Number : -1 ¦ ¦ Outgoing realm : --------------------------------------- ¦ ¦ Outgoing username : --------------------------------------- ¦ ¦ ¦ ¦ Outgoing Password : ---------¦ ¦ Confirm : ---------¦ ¦ ¦ ¦ Incoming username : --------------------------------------- ¦ ¦ ¦ ¦ Incoming Password : ---------¦ ¦ Confirm : ---------¦ ¦ ¦ ¦ RFC 3325 supported by the distant + True ¦ ¦ DNS type + DNS SRV ¦ ¦ SIP DNS1 IP Address : 10.3.1.100 ¦ ¦ SIP DNS2 IP Address : 192.168.2.20 ¦ ¦ SDP in 18x + True ¦ 17

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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 to validate ┌─Review/Modify: IP Quality Of Service COS───────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ IP QoS COS : 1 │ │ │ │ 8021Q Used + False │ │ 8021p Priority : 5 │ │ VLAN ID : 0 │ │ TOS/diffServ : 46 │ │ UDP Lost : 7 │ │ UDP Lost Reinit : 7 │ │ UDP Keep-alive : 15 │ │ │ └────────────────────────────────────────────────────┘ 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 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: To configure the default system compression, go to /system/Descend Hierarchy/Other System Parameters/Descend Hierarchy/Compression Parameters/Review Modify Choose “Compression Type” in the parameter “System Options” , press 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 to validate

┌─Review/Modify: Compression Parameters──────────────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ System Option + Compression Type │ │ │ │ Compression Type + G 729 │ │ │ └────────────────────────────────────────────────────────────────────────┘ 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 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 │ 21

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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

ARS and Discriminator

PAETEC required 10 digits dialing for local and long distance call and open dialing for International calls. By default, the discriminator used is 0 for external calls using the SIP Trunk Group. When a user place an external call using the ARS Professional Trunk Grp Seizure prefix “9”, depending on the number on the dialed number and ARS list is chosen as per the list below. Each ARS list has only one route (only route ID 1) which uses the Trunk Group ID 10 and the SIP External Gateway 1 (using the dialing command table. The ARS tables used are: ARS 0 Æ for international calls ARS 1 Æ for long distance calls ARS 4 Æ for calls to 911, 311, 411 & 711 The Dialing Command Tables used are: Table 1 Æ for ARS 0,1 . Dialing sequence = “I”, external SIP gateway = 1 Table 2 Æ for ARS 4. Dialing sequence = “9I”, external SIP gateway = 1 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)

┌─Review/Modify: Numbering Plan Description (NPD)──────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Description identifier : 33 │ │ │ │ Name : SIP │ │ Calling Numbering plan ident. + NPI/TON: ISDN Unknown │ │ Called numbering plan ident. + NPI/TON: ISDN Unknown │ 22

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│ Authorize personal calling num use + False │ │ Install. number source + NPD source │ │ Default number source + NPD source │ │ Called DID identifier : -1 │ │ Calling/Connected DID identifier : -1 │ │ Installation number : --------- Change as per the DID plan │ Default number (num. inst. sup.) : 9090 │ │ │ └──────────────────────────────────────────────────────────────────────┘ 4.5.2

ARS Tables

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 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

The Numbering Command Tables are configure by going to /Translator/Automatic Route Selection/Number Command Table; Choose Review/Modify or Create; Enter the Table Id if you review/modify:

┌─Review/Modify: Numbering Command Table───────────────────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Table ID : 1 │ │ │ │ Carrier Reference : 0 │ │ Command : I │ │ Associated Ext SIP gateway : 1 │ │ │ └──────────────────────────────────────────────────────────────────────────────┘ ┌─Review/Modify: Numbering Command Table───────────────────────────────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Table ID : 2 │ │ │ │ Carrier Reference : 0 │ │ Command : 9I │ │ Associated Ext SIP gateway : 1 │ │ │ └──────────────────────────────────────────────────────────────────────────────┘

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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 ┌─[ 4 ] Instances: Ext.Callback Translation─┐ │ │ │ -> A │ │ DEF │ │ │ └───────────────────────────────────────────┘ ┌─Review/Modify: Ext.Callback Translation────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Basic Number : A │ │ │ │ No.Digits To Be Removed : 1 │ │ Digits To Add : 9011 │ │ │ └────────────────────────────────────────────────────┘ ┌─Review/Modify: Ext.Callback Translation────────────┐ │ │ │ Node Number (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Instance (reserved) : 1 │ │ Basic Number : DEF │ │ │ │ No.Digits To Be Removed : 0 │ │ Digits To Add : 9 │ │ │ └────────────────────────────────────────────────────┘

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5

Summary of Test Results

5.1

Summary of Test Plan Results

A comprehensive validation test plan has been performed including: -

Testing General call Processing Testing Basic Call Failures (extensions busy) Testing the Outgoing Dialing Plan Testing DTMF Detection Testing Call Transfers scenarios Testing Call Forwarding scenarios Testing Calls on hold Testing Call Waiting Testing 3 way-calls Testing PBX Voice Mail Testing faxes Testing redundancy and call hold

The test plan has revealed the limitations and anomalies in the next sections

5.2

OmniPCX Enterprise limitations

The following are the limitations in the OmniPCX Enterprise: -

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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