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Document Overview

This document provides a configuration guide for Sonus SBC 1000/2000 (Session Border Controller) when connecting to Alcatel IP Phones 40x8EE Series.

The configuration guide supports features outlined in the Alcatel-Lucent web page.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000 and Alcatel IP Phones.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.

 

Requirements

The following equipment and software were used for the sample configuration provided:

 

Equipment

Software Version

Sonus Networks

SBC 1000



V5.0.2 build 405

Third-party Equipment

Alcatel 4008, 4028 & 4068

NOE : 4.33.71 – SIP : 2.10.60

Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 1000/2000.


Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-Party Product Features

None

 

Prerequisites

None

 

Verify License

SIP Registrar Licenses are needed to have IP Phones registered with SBC 1000/2000.


 

Alcatel 40x8EE SIP-Touch Configuration

Follow the different steps below to configure the Alcatel IP Phones for the interconnection scenario. If you have DHCP and TFTP you can proceed with full automatic configuration. If you need to manually configure the phone, press the keys “i” then # during boot to access the phone configuration.

Make sure the TFTP server has the Alcatel configuration files, for each of the phones with the format as described below. Two files can be used: One for general/common configuration for all phones named “sipconfig.txt”. And a file per device where the name must be “sipconfig-<MACaddress>.txt” where all phone specific parameters will be specified.

 

40x8EE SIP-Touch sets sample configuration
###########################################################################


#                                                                                                                                                                 #


#       40x8EE SIP-Touch sets sample configuration file                                       
#


#                                                                                                                                                                 #

###########################################################################


 


###########################################################################

## Syntax information :
##

## All parameters are grouped by sections (the section name is specified

## with the syntax[section1]). The content is UTF-8 encoded

## Each configuration item uses a single line with a "key=value" pattern

## "Key" content starts at the first none-space character and ends with "="
## or " ="
## "Value" content starts at the first none-space character and ends at the
## "\n\r"
## If the value is empty, the terminal should use the default value instead
## Lines starting with "#" will be ignored as a comment 


## The items of a list are separated by a ","
###########################################################################

[dns]
###########################################################################
## The primary DNS IP
address HAS TO BE FILLED
## If no DNS, use the
SIP proxy address instead
###########################################################################

   dns_addr=192.168.123.101
   dns2_addr=
   hostname=Alcatel

[sip]
###########################################################################
## Domain name : IP
address, FQDN or domain name (see the SIP proxy config)
###########################################################################

   domain_name=

###########################################################################

## Primary SIP proxyand SIP registrar settings
##
## Proxy address : IPaddress, FQDN or domain name
## Registrar address: IP address, FQDN or domain name (usually, the proxy)
## SIP proxy UDP port: usually 5060
## SIP registrar UDPport : by default 5060

###########################################################################

   proxy_addr=192.168.123.63
   proxy_port=5060
   registrar_addr=192.168.123.63
   registrar_port=5060
   outbound_proxy_addr=
   outbound_proxy_port=

###########################################################################

## Redundancy settings 
##
## Proxy address : IPaddress, FQDN or domain name
## Registrar address: IP address, FQDN or domain name (usually, the proxy)
## SIP proxy UDP port: usually 5060
## SIP registrar UDPport : by default 5060
##
sip_transport_mode_survi : Transport mode in PCS mode 

##          0 = UDP or TCP
##                  1 = UDP
##                  2 = TCP

###########################################################################

   proxy2_addr=
   proxy2_port=
   registrar2_addr=
   registrar2_port=
   outbound_proxy2_addr=
   outbound_proxy2_port=
   pcs_addr=
   pcs_port=
   sip_transport_mode_survi=0
   option_timer=120

###########################################################################

## Global SIPparameters
## Transport mode : 0= UDP or TCP
##                  1 = UDP
##                  2 = TCP

## local_rtp_port :RFC3605 is not supported in this release, so 
## only default value can beused
## PRACK type : 0 =PRACK supported
##              1 = PRACK required
##              2 = PRACK disabled
## Codec settings : 0= G711 (PCMU)
##                  4 = G723.1
##                  8 = G711 (PCMA)
##                 18 = G729A


###########################################################################


   register_expire=3600
   register_retry=300
   local_sip_port=
   sip_transport_mode=0
   local_rtp_port=42000
   local_rtcp_port=42001
   prack_type=0
   preferred_vocoder=8,0,4,18

###########################################################################
## SIP authentication
##
## Realm : If noauthentication, leave empty
## Authentication name : HAS TO BE FILLED 
## If no authentication, PUT A VALUE LIKE none
## Authenticationpassword : If no authentication, leave empty

###########################################################################


   authentication_realm=192.168.123.63
   authentication_name=1001
   authentication_password=1001

###########################################################################

## SIP number name info 
##

###########################################################################

   user_name=1001
   display_name=Alcatel4028

## Voicemail settings
##

## Voice mail URI :directory number of the voice mail

## user name :directory number of the set

## MWI URI : completeSIP URI of the voice mail

###########################################################################

   voice_mail_uri=
   message_waiting_indication_uri=

[qos]
###########################################################################

## SIP & RTPDIFFSERV : [0,63]


## SIP QOS Tickets :0 = disable
##           1 = enable

## SIP QOS Tickets
Target : domain name
###########################################################################

   sip_diffserv=0
   rtp_diffserv=0
   sip_qos_tickets_enable=0
   sip_qos_tickets_target=


[sntp]
###########################################################################

## SNTP server settings (can be OXE or an external server)
##

## Timezone construction : UT::60:032902:102503

## GMT delta : 60 = + sixty minutes from GMT time

##          Daylight saving start (mmddhh) :032902 = 29 March 2am

##          Daylight saving end (mmddhh) : 102503= 25 October 3am 

## The daylight saving settings HAVE to be changed each year.

###########################################################################

   sntp_addr=192.168.123.101
   timezone=UT::60:032902:102503

[telnet]
###########################################################################
## Password to access the telnet session


###########################################################################
   telnet_password=000000


[arp]
###########################################################################


## Mode : 0 = ARP spoofing disabled
##        1 = ARP spoofing enabled


###########################################################################

   arp_spoofing_enable=1
   arp_spoofing_timer=30


[init]
###########################################################################

## For IP Touch with
SIP binary in 1.xx, 2.00.10 and 2.00.20, equal or greater than 2.00.81

##     mode 0 = SIP
##     mode 1 = NOE
##

## For IP Touch with SIP binary 2.00.30 to 2.00.80
##     mode 0 = NOE
##     mode 1 = SIP

###########################################################################

   application_mode=0



[audio]
###########################################################################

## Tone country : 0 =
English


##                1 = French
##                2 = German
##                3 = Italian
##                4 = Spanish
##                5 = Dutch
##                6 = Portuguese
## DTMF type : 0 =RFC2833
##             1 = In-band
##             2 = SIP INFO

## DTMF level / RLR
handset / SLR handset / Sidetone handset : 
## 0 = 0db, 1 = +3db, 2 = +6db, 3 = -3db, 4 = -6db

## VAD / DTMF
feedback / Hearing Aid : 

##       0 = VAD not used
##       1 = VAD used


###########################################################################


   tone_country=1
   dtmf_type=0
   dtmf_level=0
   dtmf_avt_payload_type=101
   vad=0
   dtmf_feedback_enable=0
   rlr_handset=0
   slr_handset=0
   sidetone_handset=2
   hearing_aid_enable=0


[appl]
###########################################################################

## Password to access
the administrator menu on the phone (digits only) 

## Power priority : 1= critical
##                  2 = high
##                  3 = low

## Time format : 0 =24 hours format
##               1 = AM / PM

## Speed dial numbers
(first and last name, URI)
###########################################################################

   admin_password=000000
   bluetooth_parameters=blue
   supported_language=0
   remote_forward_code=
   remote_forward_deactive_code=
   power_priority=
   asset_id=
   time_format=
   speed_dial_1_first_name=
   speed_dial_1_last_name=
   speed_dial_1_uri=
   speed_dial_2_first_name=
   speed_dial_2_last_name=
   speed_dial_2_uri=
   speed_dial_3_first_name=
   speed_dial_3_last_name=
   speed_dial_3_uri=
   speed_dial_4_first_name=
   speed_dial_4_last_name=
   speed_dial_4_uri=

[admin]
###########################################################################

## Global admin parameters
## Binary polling timer : 10 min - 65535 min (default 480 min)

## Config polling
timer : 5 min - 65535 min (default 60 min)
###########################################################################

   binary_polling_timer=480
   config_polling_timer=60
   disable_pc_port=0
   activate_vlan_filter=0


 

 

SBC 1000 Configuration

The following configuration steps provide an example of how to configure the Sonus SBC 1000/2000 to interoperate with Alcatel IP Phones 40x8EE Series:

  1. Local Registrar
  2. Signaling Group
  3. Transformation Table
  4. Call Routing Table
  5. Tone Table

1. Local Registrar

SIP provides a registration function that allows users to upload their current locations for use by proxy servers. Registration creates bindings in a location service for a particular domain that associates an address-of-record URI with one or more contact addresses.

Registration entails sending a REGISTER request to a special type of UAS (User-Agent Server) known as a registrar. A registrar acts as the front end to the location service for a domain, reading and writing mappings based on the contents of REGISTER requests. This location service is then typically consulted by a proxy server that is responsible for routing requests for that domain.

Select Settings > SIP > Local Registrars to access the Local Registrars screen.

Figure : Local Registrar

 

2. Signaling Group

Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. These groups are the entity to which calls are routed, as well as the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, this will specify protocol settings and link to server, media and mapping tables.

Select Settings > Signaling Groups to access the Signaling Groups configuration screens.

Figure : Signaling Group

3. Transformation Table

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or into a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially selected from there. In addition, Transformation tables will be configurable as a reusable pool that Action Sets can reference.

Select Settings > Transformation to access the Transformation configuration screen.

Figure : Transformation

 

4. Call Routing Table

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

Select Settings > Call Routing Table to access the Call Routing Table configuration screen.

Figure : Call Routing Table

 

5. Tone Table

Tone tables allow the Sonus SBC 1000/2000 Administrator to customize the tones a user hears when placing a call. You can modify the tone to match your local PSTN or PBX. The default tone table is configured for the values used in the United States for the following categories: Ringback, Dial, Busy, Congestion, Call Waiting, Disconnect, and Confirmation.

Select Tone Tables > Default Tone Table to access the Tone Table configuration screen.

Figure : Tone Table

 

 

Test Results

 

Figure : Test Results

S.NoProcedureObservationResultComment
1

Simple call

Call from Alcatel phone 1 to phone 2.

Calling number and/or name presented on phone depending of phone capacity.

Audio both way.

Pass 
2

Hold/Retrieve call

With Music file uploaded & configured on SBC

Pass

 
3

Forward call

Forward call set on Alcatel phone 2 toward Alcatel phone 3. Call from Alcatel phone 1 rings directly phone 3. Displayed caller on phone 3 is phone 1 with no reference to forward condition.

Pass

 
4

Transfer call

Call already established between Alcatel phone 1 and phone 2 can be transferred to an Alcatel phone 3 by pressing Transfer key on one of the Alcatel phone.

Pass

 
5

Conference call

Establish a first call between Alcatel phone 1 and Alcatel phone 2. Then make a second call from phone 1 toward Alcatel phone 3. When established pressing Conference key on phone 2 brings to a 3 party conference, hosted by the Alcatel phone 2.

Pass

 
6

DTMF

DTMF typed on Alcatel phones are passed to the other connected phones.

Pass 


Conclusion

These Application Notes describe the configuration steps required for Sonus SBC 1000/2000 to successfully interoperates with Alcatel IP Phones 40x8EE Series. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results

 .