• No results found

OBS-BIV (FR): SIP TRUNK

N/A
N/A
Protected

Academic year: 2021

Share "OBS-BIV (FR): SIP TRUNK"

Copied!
28
0
0

Loading.... (view fulltext now)

Full text

(1)

Technical Bulletin

|

Alcatel-Lucent OmniPCX Office

TC1788 ed.01

OBS-BIV (FR): SIP TRUNK

CONFIGURATION GUIDE OXO R820

This document details the configuration necessary for the ALCATEL-LUCENT OXO R820 to inter-operate with the BIV SIP trunk offer of the Operator OBS in France.

Revision History

Edition 1: March 15, 2013 creation of the document

Legal notice:

Alcatel, Lucent, Alcatel-Lucent and the Alcatel-Lucent logo are trademarks of Alcatel-Lucent. All other trademarks are the property of their respective owners.

The information presented is subject to change without notice.

(2)

Table of contents

1 GENERAL ... 3

1.1REFERENCES ... 3

1.2SCOPE OF THE DOCUMENT ... 3

1.3SCOPE OF ALCATEL-LUCENT’S SUPPORT ... 4

1.4SOFTWARE/HARDWARE VERSIONS ... 4

1.5FEATURE &SET COMPATIBILITY LIST ... 4

1.5.1 Supported Sets ... 4

1.5.2 Supported Features ... 5

1.5.3 Restrictions ... 5

2 SYSTEM GENERAL INFO AND BASIC SET UP ... 6

2.1PRE-REQUIRED INFORMATION ... 6

2.2SYSTEM CONNECTION PROCEDURE ... 7

2.3NETWORK CONFIGURATION ... 7

2.4(PUBLIC)NUMBERING CONFIGURATION ... 7

2.4.1 Installation numbers ... 7

2.4.2 DDI numbers ... 8

3 DETAILED SIP TRUNK CONFIGURATION ... 9

3.1SIP ABILITY CHECK ... 9

3.2SIPTRUNK CREATION ... 9

3.2.1 Management of VoIP channels ... 9

3.2.2 “Physical” Access associated to the SIP Trunk ... 11

3.2.3 Hosting System Trunk Group ... 11

3.3INTERNAL NUMBERING PLAN ... 13

3.4TRAFFIC SHARING AND BARRING (REMINDER) ... 13

3.5ARS CONTEXT ASSOCIATED TO THE SIPTRUNK ... 13

3.5.1 ARS Trunk Groups Lists ... 13

3.5.2 Gateway/ Registrar Proxy parameters ... 14

3.5.3 ARS prefixes for remote Gateway with Static-IP ... 17

3.5.4 ARS prefixes for remote Gateway with Dynamic-IP ... 17

3.5.5 SIP format of public phone numbers ... 19

3.6VOIPMISC PARAMETERS... 20

3.6.1 SIP Timers ... 20 3.6.2 Fax (T38) ... 20 3.7NOTEWORTHY ADDRESSES ... 21 3.7.1 Debug Labels... 21 3.7.2 Other Labels ... 23 3.8MISCELLANEOUS CONFIGURATION ... 24

3.8.1 Calling Identity when External Call Forward ... 24

(3)

1 General

The purpose of this document is to provide the main configuration steps of the Omni PCX Office IPBX (OXO) to ensure inter-operation with the SIP BIV commercial offer delivered by the Operator OBS.

1.1 References

Alcatel-Lucent documentation (available from the Business Partner Web Site): [1] Alcatel-Lucent OmniPCX Office Communication Server- Expert Documentation [2] Alcatel-Lucent OmniPCX Office Communication Server- Public SIP Trunking

1.2 Scope of the document

This guide focuses the configuration of OXO “SIP trunk” feature. It is intended for normal-skilled engineers who are familiar with the OMC config-tool and can manage the basic set up of the IPBX which is not covered herein (e.g. TDM trunk and management of local phones).

Apart from the OMC screen pictures, the configuration parameters /values are put in green with the sign



placed in front. Some examples:



International_Prefix = "00" (quoted value for a typed-field in OMC)



RTP_Direct = true (non-quoted value when taken from a user pick-list in OMC)



ARS_IP_Address = N/A (value is N/A when the parameter is disabled in OMC)

The unambiguous name given to the parameters is proper to the present document and is derived from the OMC naming. When terminated with “_example”, a parameter is site-dependent and the value provided needs to be customized. Examples:



Subnet_Mask_example = "255.255.255.0"



SIP_Trunk_Channels_example = 8

Because the examples given can be from a real customer system, for privacy/ security reasons, some parameters (i.e. IP addresses, logins, public phone numbers) may be masked or partially masked with asterisks. Examples:



GW_Password = "*******"

(4)

1.3 Scope of Alcatel-Lucent’s support

Depending on the status of its approval process, the present SIP trunk solution may receive either Limited Availability (LA) or General Availability (GA) level of support from Alcatel-Lucent.

The support delivered to Alcatel-Lucent Business Partners is strictly delimited by the approved interop context and system configuration that are detailed in this document. This support covers the protocol and functional aspects of the SIP trunk but not the audio quality of calls for the part incumbent on the Operator or linked to the client's IP

infrastructure.

1.4 Software/ Hardware versions

INFRASTRUCTURE COMPONENT

VERSION

(minimal compatible)

OXO system ALZQO820/035.001

OMC Management Software OMC820/19.1a

1.5 Feature & Set Compatibility List

The two tables hereafter list the main inter-operation features and the models of sets that are supported in the solution described. The "Supported" value corresponds to:

- "YES" => supported by Alcatel-Lucent (i.e. tests done or not necessary for the related context) - "NO" => not supported (i.e. tests are KO or have not been done or are Not Applicable) - "WR" => (With Restrictions") restricted support as detailed in section 1.5.3

For an item stated as not supported and mentioning the remark "Tests not done": In case you would like or need to have this item passed to the supported category and, if you are ready to collaborate with our approval process: simply send an email to "[email protected]" and ask for the details related to the certification of the required item.

1.5.1 Supported Sets

The following table lists the models of sets supported.

SUPPORTED SETS (OXO/ OBSR820GA)

SUPPORTED * (YES / NO / WR)

REMARKS

Analogue Z (POTS) YES

Desktop IP 40x8 series YES

Desktop UA 40x9 series YES

Desktop MyICphone 8082 YES

Appli MyICmobile IPhone YES

Appli MyICmobile Android YES

(5)

1.5.2 Supported Features

The following table lists the main inter-operation features supported.

INTER-OP FEATURE (OXO/ OBSR820GA)

SUPPORTED * (YES / NO / WR)

REMARKS

Type of Topology YES Remote Hosted NAT

SIP Registration YES

SIP Authentication YES

GW Dynamic Mgmt. Mode YES

Direct RTP YES

Outbound Basic Call YES

Inbound Basic Call YES

Dynamic Codec Mgmt. YES

Call Release YES

Internal Call Transfer YES

Internal Call Forward YES

Call Hold YES

Inbound Call to DDI YES

Reception of DTMF YES

Emission of DTMF YES

Emergency Calls YES

Caller’s Repertory YES

CLIP (Inbound and Outbound) YES

CLIR (Inbound and Outbound) YES

External Call Transfer YES

External Call Forward YES

Conference with 2 External YES

Outbound Fax T38 NO T38 not supported by Provider

Inbound Fax T38 NO T38 not supported by Provider

Outbound Fax G711 YES

Inbound Fax G711 YES

Dynamic Call Routing YES

Busy State YES

Preannouncement YES

1.5.3 Restrictions

(6)

2 System General Info and Basic Set Up

2.1 Pre-required information

The tables below gather all necessary information that should be collected before starting the OXO configuration.

When the “Example Value” is empty, the parameter is either not relevant or not mandatory for the configuration with this SIP Operator.

IP PARAMETERS

Data Type Parameter Example Value

IP Data subnetwork OXO CPU - Data IP 192.168.10.2

Default Gateway 192.168.10.1

Netmask 255.255.255.0

Data VLAN

IP Voice Subnetwork OXO CPU - Voice IP 192.168.10.2

Default Gateway 192.168.10.1

Netmask 255.255.255.0

Voice VLAN

Table of IP infrastructure parameters (to be customized for the customer site) SIP TRUNK PARAMETERS

Data Type Parameter Example Value

Numbering Plan Installation Number 29***4330

Public DDI range 4331 - 4339

SIP-Trunk Data SIP Operator GW IP

Registered Username 33123454340

Login +3329***[email protected]

Password *******

SIP Operator Domain sip.osp.com SIP Customer Domain sip.osp.com

SIP Realm

Outbound Proxy imspcf211gm.sip.osp.com Registrar IP address

Registrar name sip.osp.com

Resolving DNS server 1 172.22.246.212 Resolving DNS server 2 172.22.246.213

(7)

2.2 System Connection procedure

In order to configure the IPBX, it's necessary to use the OMC software at the Expert level session

2.3 Network configuration

The IP configuration is to be set before going further. This is done using the OMC menu Hardware and Limits -> LAN IP Configuration.

192.168.10.2

255.255.255.0 192.168.10.1

The “Boards” Tab shows the IP address for the main CPU and the default gateway. The main CPU IP address must be edited there



Main_CPU_IP_Address_example = "192.168.10.2"

The “LAN Configuration” Tab lets you edit the values of default gateway IP address and the subnet mask



Def_Router_Address_example = "192.168.10.1"



Subnet_Mask_example = "255.255.255.0"

Any modification of the LAN-IP parameters requires a further system reboot (“Warm Reset”) to validate the change !!

2.4 (Public) Numbering configuration

As for any type of trunk, the public numbering used for SIP trunk is first ruled by the general numbering configuration of the PBX.

2.4.1 Installation numbers

(8)

- Configuration for “Installation Numbers”:

Installation Number :



Instal_Number_example = "29***4330”

International Prefix :



International_Prefix = "00"

International Code :



International_Code = "33"

Intercity Prefix :



Intercity_Prefix = "0"

Intercity Code :



Intercity_Code_example = ""

Recall Prefix :



Recall_Prefix = "0"

Alternative System CLIP :



Alternative_System_CLIP_example = ""

2.4.2 DDI numbers

Go to OMC:.. Numbering -> Numbering Plans – Public Numbering Plan Tab...

Subscriber 4340 4349

4331 - 4339

4331 - 4339

- Configuration for “Public Numbering Plan”:

(9)

3 Detailed SIP Trunk Configuration

Illustrated with OMC screenshots, this chapter describes the OXO configuration parts corresponding to the public SIP trunk and the hosting IP infrastructure.

3.1 SIP ability check

The trunk is the link between the IPBX and the network. A specific SW licence is mandatory to enable IP trunks on the system.

The OMC menu Hardware and Limits -> Software Key Features gives an overview of the allowed features.

In the Multi-site Tab, check the number of “IP Trunks” really activated (number of channels available for the VOIP trunk).

3.2 SIP Trunk creation

The VoIP trunk uses a specific signaling protocol (i.e. SIP) and physical resources embedded in the IPBX (i.e. DSP’s). To create the SIP trunk in the system, it's first necessary to allocate some of the DSP resources to it.

3.2.1 Management of VoIP channels

Go to OMC Voice Over IP -> VOIP:Parameters - General Tab to check/ edit the requested parameters.

(10)

-

the VOIP protocol in use :



VOIP_Protocol = SIP

-

from the amount available in the system, the number of channels dedicated to VoIP trunks:



VoIP_Trunk_Channels_example = 8

- the RTP routing mode and codec management for IP extensions:

“RTP Direct” :



RTP_Direct = true

“Codec pass-through for SIP trunk”:



Trunk_Codec_Passthru = false

“Codec pass-through for SIP phones”:



Phone_Codec_Passthru = true

- the IP Quality of Service :

“IP Quality of Service” :



IP_QoS_example = 00000000

With OMC menu Voice Over IP -> VOIP:Parameters - DSP Tab , also define the global control of Echo cancellation and the VAD parameters of VoIP calls.

1

2

  

Set DSP parameters as follows:

“Echo Cancellation” :



DSP_Echo_Cancel = true

(11)

3.2.2 “Physical” Access associated to the SIP Trunk

From OMC External Lines -> List of access, select the VoIP access associated to the VoIP trunk.

1

2

3

  8

Then, configure the access parameters:

“Public trunk” option :



Public_Trunk = true

number of channels allocated to the SIP trunk “VoIP-Trunk Ch.”:



SIP_Trunk_Channels_example = 8

“Alternative CLIP/COLP Number” :



Trunk_Alternative_CLIP_example = “”

3.2.3 Hosting System Trunk Group

(12)

Use the OMC menu External Lines ->List of Trunk Groups is used for that.

Carry out the selections and push-button steps 1 to 6 above. As a variant, at step 2, you can decide to include the SIP trunk access into the OXO’s main Trunk Group (i.e. step 2a for index #1) or include it into one

secondary Trunk Groups (e.g. step 2b for index #2).

The SIP trunk can be placed freely into one or several Trunk Groups of the system (having it within several Trunk Groups can be useful to manage traffic and particular sharing between users).

(13)

3.3 Internal Numbering Plan

Accessible from OMC Numbering -> Numbering Plans menu, the internal numbering plan is the place where dialing of internal phones is first analyzed by the system.

The example configuration here defines access to the internal ARS table for a number dialed that starts with digit 0. The Drop value associated indicates that the number analysis must be continued in the ARS Prefix table without keeping the initial digit 0.

3.4 Traffic Sharing and Barring (reminder)

Though not described here, correct configuration of traffic sharing, barring and feature rights mechanisms is necessary to allow call features and outbound traffic over the SIP trunk.

3.5 ARS context associated to the SIP Trunk

In-depth configuration of the SIP trunk is carried out via the ARS context associated to the trunk.

3.5.1 ARS Trunk Groups Lists

To enable voice calls via the ARS system, it’s necessary to have ARS Trunk Groups created via the OMC menu Numbering -> Automatic Routing Selection -> Trunk Groups Lists.

(14)

Carry out the selections and push-button steps 1 to 5 above. At step 4, you need to select the line index corresponding to a System Trunk Group previously defined at section 3.2.3 (i.e. selecting 4a for the Main Trunk Group or 4b for the secondary Trunk Group of index #2).

3.5.2 Gateway/ Registrar Proxy parameters

(15)

In the picture above, the gateway at index 1 is configured to be associated to the SIP trunk - Set GW parameters (part 1) as follows:

“Login” :



GW_Login_example = "+3329***[email protected]"

“Password” :



GW_Password = "*******"

“Domain Name” :



GW_Target_Domain = "sip.osp.com"

“Realm” :



GW_Realm = "sip.osp.com"

“RFC 3325” :



GW_RFC3325 = Yes

“Remote SIP Port” :



GW_Remote_SIP_Port = Dynamic (see Note below)

“SIP Numbers ..” : index number 1 shown as example (detailed at Ch. 3.5.5)

Default value of Remote SIP port is 5060. After completing the further parameters, this value will automatically change to “Dynamic” after the parameter DNS is set to “DNS SRV”.

- Set GW parameters (part 2) as follows:

“DNS” :



GW_DNS_Mode = DNSSRV

“Primary DNS...” :



GW_Prim_DNS = "172.22.246.212"

“Secondary DNS” :



GW_Sec_DNS = "172.22.246.213"

“Outbound Proxy” :



GW_Outb_Proxy = "imspcf211gm.sip.osp.com"

“Fax” :



GW_Fax_Mode = G711

“Registration Req.” :



GW_Reg_Requested = Yes

(16)

5

4

3

2

1

Gateway Parameters (right part) 5060 sip.osp.com 3600 sip.osp.com

- Set GW parameters (part 3) as follows:

“Registrar IP ..” :



GW_Reg_IP_Address = “” (see Note after)

“Port :



GW_Reg_Port = "5060"

“Registrar Name” :



GW_Reg_Name = "sip.osp.com"

“Expire Time” :



GW_Reg_Expire_Time = "3600"

“Local Domain ..” :



GW_Local_Domain_Name = "sip.osp.com"

(17)

3.5.3 ARS prefixes for remote Gateway with Static-IP

Not Applicable. Configuration of dynamic DNS mode described at section 3.5.4 is required instead.

3.5.4 ARS prefixes for remote Gateway with Dynamic-IP

The ARS Prefix table is the entry place of the ARS system to start the analysis of dialed numbers. Go to the OMC menu Numbering -> Automatic Routing Selection -> Automatic Routing Prefix.

As illustrated in the picture below, you need first to configure the route line for standard calls (national and international): use the Add function to create a new line and then, configure the line parameters as indicated. Note that the parameters relative to the right part of the line are depicted a further in this doc section.

(18)

1 3 2 4

(left part)

het SIP-GW Dynamic

Line 1: standard calls starting with digit 0 (national and international calls)

Line 2: for external emergency numbers (e.g.112). This line is linked internally to the country emergency numbers specified in the system flag table “EmergNum“ (refer to Technical Bulletin TC80) Line 3 and 4: for short-format external numbers (country dependent). The complete list of short numbers will require one or several ARS lines depending on the country considered

o Line 3: example for short numbers that begin with digit 3 (e.g. 3611, 3900, …) o Line 4: example for short numbers that begin with digit 1 (e.g. 11, 118712, …)

In area 2, the “Calling” and “Called/PP” fields must be set as shown in the example, in order not to interfere with other SIP parameters for numbering.

In area 3 and 4, values must also be respected :

“Called (ISVPN/..)”:



ARS_Called_Mode = het

“Destination” :



ARS_Destination = SIP-GW

3 4

1 2 5 6 7

Dynamic >=1024

kBit/s Default

“IP Type”:



ARS_IP_Type = Dynamic (see note below)

“IP Address”:



ARS_IP_Address = N/A (see note below)

“GW Alive Protocol”:



ARS_GWalive_Prot = N/A (see note below)

“GW Alive Timer”:



ARS_GWalive_Timer = N/A (see note below)

“Gateway Bandwidth”: to be configured according to the customer IP infra (real bandwidth available for VoIP calls)



ARS_GW_Bandwidth_example = >=1024 kBit/s

“Codec/Framing”:



ARS_Codec_Framing = Default

The Gateway Alive Status is modified when OXO performs a contact with the concerned gateway The gateway Parameters Index field is accessible after the Gateway parameters menu is completed (step 3.5.2)

(19)

3.5.5 SIP format of public phone numbers

The menu Numbering -> Automatic Routing Selection -> SIP Public Numbering is the place to define which number format is supposed to be received or sent over the SIP trunk.

1

2

3

(left part) 

Canonical + National/International

- Configuration for outgoing calls - calling number (From):

“Format”:



SIPnum_Out_Calling_Format = Canonical

“Prefix”:



SIPnum_Out_Calling_Prefix = "+"

- Configuration for outgoing calls - called number (To):

“Format”:



SIPnum_Out_Called_Format = National/International

“Prefix”:



SIPnum_Out_Called_Prefix = ""

“Short Prefix”:



SIPnum_Out_Called_Short_Prefix = ""

- Configuration for Incoming calls - calling number (From):

“Format”:



SIPnum_Inc_Calling_Format = Canonical/International

“Prefix”:



SIPnum_Inc_Calling_Prefix = ""

- Configuration for Incoming calls - called number (To):

“Format”:



SIPnum_Inc_Called_Format = Canonical/International

“Prefix”:



SIPnum_Inc_Called_Prefix = "+"

(20)

3.6 VOIP Misc Parameters

3.6.1 SIP Timers

Go to OMC menu Voice Over IP -> VOIP:Parameters/SIP presented below

1000 4000

6

Check/ Set values as follows:

“Timer T1”:



SIP_Timer_T1 = 1000

“Timer T2”:



SIP_Timer_T2 = 4000

“Number of Retries”:



SIP_N_Retries = 6

3.6.2 Fax (T38)

(21)

3.7 Noteworthy Addresses

Some PBX specific tuning may not be accessible directly via dedicated OMC screens. In such case, the tuning is done via internal control parameters (flags) also called “Noteworthy addresses” or “Labels”. Access to the system Labels is made with OMC menu System Miscellaneous -> Memory Read/Write.

Noteworthy addresses have a customized factory default value which is dependent of the SW installation target selected for OXO (country dependent).

The different flags detailed hereafter are key for the configuration of the SIP trunk and therefore they need to be checked/ edited with special care. When a flag needs to be modified, click on the 'modify' then on the 'write' buttons to send the new value to the IPBX. Also, if specified in the document, a further system warm reset may be requested for some of the flags to have the new value taken into account.

3.7.1 Debug Labels

Go to System Miscellaneous ->Memory Read/Write > Debug Labels > and check/ edit values carefully as detailed below.

MultAnsRei controls emission of Re-INVITE messages during SDP negotiation



Flag_MultAnsRei = 00

VOIPnwaddr is a 100-byte-size table of flags controlling most of the VoIP/SIP config parameters not found

(22)

000: 00 00 01 01 00 0A 00 00 008: 00 00 13 C4 00 00 00 00 010: 04 00 01 00 01 00 01 01 018: 00 00 00 00 01 01 00 00 020: 00 00 00 01 00 00 01 93 028: 00 00 00 00 00 00 00 00 030: 00 00 00 00 00 00 00 00 038: 00 00 00 00 00 00 00 00 040: 00 00 00 00 00 00 00 00 048: 00 00 00 00 00 00 00 00 050: 00 00 00 00 00 00 00 00 058: 00 00 00 00 00 00 00 00 060: 00 00 00 00  Line 1:



Flag_VOIPnwaddr_Line1 = 00 00 01 01 00 0A 00 00 Line 2:



Flag_VOIPnwaddr_Line2 = 00 00 13 C4 00 00 00 00 Line 3:



Flag_VOIPnwaddr_Line3 = 04 00 01 00 01 00 01 01 Line 4:



Flag_VOIPnwaddr_Line4 = 00 00 00 00 01 01 00 00 Line 5:



Flag_VOIPnwaddr_Line5 = 00 00 00 01 00 00 01 93 Line 6:



Flag_VOIPnwaddr_Line6 = 00 00 00 00 00 00 00 00 Line 7:



Flag_VOIPnwaddr_Line7 = 00 00 00 00 00 00 00 00 Line 8:



Flag_VOIPnwaddr_Line8 = 00 00 00 00 00 00 00 00

(23)

3.7.2 Other Labels

Go to System Misc ->Memory Read/Write > Other Labels section and configure labels as shown below.

DtmfDynPL: Sets the payload value for DTMF



Flag_DtmfDynPL = 65

PrefCodec indicates if a particular codec has preference



Flag_PrefCodec = 00 00

PrefFramin indicates the preferred length of RTP packets



Flag_PrefFramin = 00

AlCodLst permits to select one specific default codec list



Flag_AlCodLst = 16

SIPInDspNm controls display of received “CNIP” name when receiving a call



Flag_SIPInDspNm = 03

SIPOgDspNm controls the “CNIP” name sent over the network for outgoing calls



Flag_SIPOgDspNm = 03

FaxPasCd configures codec pass-through preference for fax G711 (between G711a and G711u)



Flag_FaxPasCd = 0F FF

VipPuNuA controls the global public/ private numbering model. Value must be set to 00.



Flag_VipPuNuA = 00

ExtNuFoVoi controls usage of the Installation Numbers table. Value must be set to 22.



Flag_ExtNuFoVoi = 22

SimulIpAlt controls usage of the local dial tone.



Flag_SimulIpAlt = 00

(24)

3.8 Miscellaneous Configuration

3.8.1 Calling Identity when External Call Forward

This configuration permits to define the calling CLIP that is sent by OXO to the “forwarded-to” destination (case of an incoming external call coming to an internal user who has an external diversion engaged). The control can be made globally for the PBX or extension by extension.

From the menu System Misc -> Feature Design, tab Part 1 & tab Part 2; check the parameters: "CLI for external diversion":



CLI_Ext_Diversion = true

"CLI is diverted party if ext..":



CLI_is_Diverted_Party = false

 

CLI for external diversion

CLI is diverted party if external caller

For one extension taken from the menu Subscribers/Basestation List, check the similar parameter: "CLI is diverted party":



CLI_is_Diverted_Party = false

 

CLI is diverted party

  

CLI is diverted party

(25)

4 SIP trunk Configuration Abstract

This chapter gathers the rough value of OXO parameters (three following tables).

Table 1

CONFIG PARAMETER VALUE REMARK Item_IP_Infra

Main_CPU_IP_Address_example "192.168.10.2" Value given as example Def_Router_Address_example "192.168.10.1" Value given as example Subnet_Mask_example "255.255.255.0" Value given as example

Item_System_Numbering_Plans

Instal_Number_example "29***4330” Value given as example International_Prefix "00" International_Code "33" Recall_Prefix "0" Intercity_Prefix "0" Intercity_Code_example "" Alternative_System_CLIP_example ""

DDI_Range_example "4331 - 4339" Value given as example

Item_VoIP_General

VOIP_Protocol SIP

VoIP_Trunk_Channels_example 8 Value given as example

RTP_Direct true

Trunk_Codec_Passthru false

Phone_Codec_Passthru true

G711_MOH true

IP_QoS_example 00000000 Value given as example

DSP_Echo_Cancel true DSP_VAD false T38_UDP_Redundancy 1 T38_Fax_Framing 0 SIP_Timer_T1 1000 SIP_Timer_T2 4000 SIP_N_Retries 6 Public_Trunk true

SIP_Trunk_Channels_example 8 Value given as example Trunk_Alternative_CLIP_example 8 Value given as example

Item_System_Misc

CLI_Ext_Diversion true

(26)

Table 2

CONFIG PARAMETER VALUE REMARK Item_ARS_Prefixes ARS_Called_Mode het ARS_Destination SIP-GW ARS_IP_Type Dynamic ARS_IP_Address N/A ARS_GWalive_Timer N/A ARS_GWalive_Prot N/A

ARS_GW_Bandwidth_example >=1024 kBit/s Value given as example

ARS_Codec_Framing Default

Item_ARS_GW_Parms

GW_Login_example "+3329***[email protected]" Example (partially masked) GW_Password "*******" (masked)

GW_Target_Domain "sip.osp.com"

GW_Realm ""

GW_RFC3325 Yes

GW_Remote_SIP_Port Dynamic DNS mode => Dynamic

GW_DNS_Mode DNSSRV GW_Prim_DNS "172.22.246.212" GW_Sec_DNS "172.22.246.213" GW_Outb_Proxy "imspcf211gm.sip.osp.com" GW_Fax_Mode G711 GW_Reg_Requested Yes

GW_Reg_Username_example "+3329***4330" Example (partially masked)

GW_Reg_IP_Address "" N/A GW_Reg_Port "5060" GW_Reg_Name "sip.osp.com" GW_Reg_Expire_Time "3600" GW_Local_Domain_Name "sip.osp.com" Item_ARS_SIP_Public_Numbering_Plan SIPnum_Out_Calling_Format Canonical SIPnum_Out_Calling_Prefix "+" SIPnum_Out_Called_Format National/International SIPnum_Out_Called_Prefix "" SIPnum_Out_Called_Short_Prefix "" SIPnum_Inc_Calling_Format Canonical/International SIPnum_Inc_Calling_Prefix "" SIPnum_Inc_Called_Format Canonical/International SIPnum_Inc_Called_Prefix "+"

(27)

Table 3

(28)

Follow us on Facebook

and

Twitter

Stay tuned on our Facebook and Twitter channels where we inform you about : New software releases

New technical communications AAPP InterWorking Reports Newsletter

Etc.

twitter.com/ALUEnterpriseCare facebook.com/ALECustomerCare

Submitting a

Service

Request

Please connect to our eService application at :

https://businessportal.alcatel-lucent.com/alugesdp/faces/gesdp/customerSupport/CustomerSupport.jspx Before submitting a Service Request, make sure that :

In case a Third-Party application is involved, that application has been certified via the AAPP You have read through the Release Notes which lists new features available, system requirements, restrictions etc. available in the Technical Knowledge Base

You have read through the Troubleshooting Guides and Technical Bulletins relative to this subject available in the Technical Knowledge Base

References

Related documents

While using the digital trunk and data trunk, users can monitor the connection status via UCM6510 web UI.  Monitor interface status under web

Primary SIP Trunk Trunk Group Centurylink VoIP Network PSTN TDM Voice Network HQ Sonus NBS Sonus NBS PSTN GW Backup Location. Backup SIP Trunk

Navigate to Voice > Trunks > Shared Line Accounts to create an SLA for each connected analog trunk account (FXO interface).. Enter the Name and select the Associated Trunk

Register the MyPBX A as an extension in MyPBX B via VOIP (SIP/IAX2) Trunk, so the extensions in MyPBX A can make calls to MyPBX B’s extensions via this ‘Special’ trunk. Use the

Whenever a match line is found, the call is conveyed thru the specific trunk gateway (GW index) associated to this line. On OMC, go to menu Numbering -> Automatic Routing

Numbering -> Automatic Routing Selection -> Trunk Groups Lists. In this menu, new lines are created after clicking the mouse right button and selecting function “Add”.. selecting

The menu Numbering -> Automatic Routing Selection -> SIP Public Numbering is the place to define the format of phone numbers transmitted over the SIP

Numbering -> Automatic Routing Selection -> Trunk Groups Lists. In this menu, new lines are created after clicking the mouse right button and selecting function “Add”.. selecting