Application Notes for Configuring Open Text Fax Server, RightFax

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Open Text Fax Server,

RightFax Edition with Avaya Aura

®

Communication Manager and Avaya Aura

®

Session Manager via SIP Trunk Interface

- Issue 1.0

Abstract

These Application Notes describe the procedures for configuring the Open Text Fax Server,

RightFax Edition with Avaya Aura

®

Communication Manager and Avaya Aura

®

Session

Manager using a SIP trunk interface.

Open Text Fax Server, RightFax Edition is a software based fax server that sends and receives fax calls over an IP network. In the tested configuration, Open Text Fax Server, RightFax

Edition interoperated with Avaya Aura

®

Session Manager to send/receive faxes using SIP trunk facilities.

Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect

Program at the Avaya Solution and Interoperability Test Lab.

RAB; Reviewed:

SPOC 2/2/2011

Solution & Interoperability Test Lab Application Notes

©2011 Avaya Inc. All Rights Reserved.

1 of 43

RFax_SM6_SIP

1. Introduction

These Application Notes describe the procedures for configuring Open Text Fax Server, RightFax

Edition with Avaya Aura

®

Communication Manager and Avaya Aura

®

Session Manager using SIP trunks.

Open Text Fax Server, RightFax Edition is a software based fax server that sends and receives fax calls over an IP network. Open Text Fax Server, RightFax Edition utilizes the Brooktrout SR140

T.38 Fax over Internet Protocol (FoIP) virtual fax board software from Dialogic. In the tested configuration, Open Text Fax Server, RightFax Edition interoperated with Avaya Aura

®

Session

Manager to send/receive faxes using a SIP trunk interface.

1.1. Interoperability Compliance Testing

The compliance test tested interoperability between RightFax and Session Manager by making intrasite and inter-site fax calls to and from a RightFax server that was connected (at each of the two sites in the test configuration) to Session Manager via SIP interface. Specifically, the following fax operations were tested in the setup for the compliance test:

 Fax from/to RightFax to/from fax machine at a local site

 Fax from/to RightFax to/from fax machine at a remote site

 Fax from/to RightFax to/from RightFax server at a remote site

In the compliance test, both ISDN-PRI trunks and SIP trunks directly connecting two

Communication Manager systems connected the Main Site, and Remote Site.

Faxes were sent with various page lengths and resolutions. For capacity, a large number of 2-page faxes were continuously sent between the two RightFax servers. Serviceability testing included verifying proper operation/recovery from failed cables, unavailable resources, restarts of Session

Manager as well as RightFax reboots. Fax calls were also tested with different Avaya Media

Gateway media resources to process the fax data between locations. This included the TN2302AP IP

Media Processor (MedPro) circuit pack and the TN2602AP IP Media Processor circuit pack in the

Avaya G650 Media Gateway; the integrated Voice over Internet Protocol (VoIP) engine of the

Avaya G450 Media Gateway, and the Avaya MM760 Media Module installed in the Avaya G450

Media Gateway.

1.2. Support

Technical support for RightFax can be obtained by contacting Open Text at

 Phone: (800) 540-7292

 Email: [email protected]

 https://cslogin.opentext.com/login/

RAB; Reviewed:

SPOC 2/2/2011



2 of 43

RFax_SM6_SIP

2. Configuration

The test configuration was designed to emulate two separate sites with multiple Port Networks at one site, and modular Gateway resources at the other site. Figure 1 illustrates the configuration used in these Application Notes.

2.1. Configuration Details

In the sample configuration, Communication Manager Servers and Gateways at the two sites were connected via SIP and ISDN-PRI trunks. Faxes were alternately sent between the two sites using these two facilities. Connections to Session Manager were via SIP trunk facilities, and the RightFax servers communicated directly with Session Manager via SIP.

Two separate Session Manager Servers were used to connect the RightFax Servers to each site.

ISDN-PRI T1 Trunk

SIP Trunk

Analog

SIP Trunk

Analog

Avaya G650

Media Gateway

(Port Network 1)

Analog

Fax Machine

(20000)

Avaya 4621SW

IP Telephone

(20003)

Analog

Fax Machine

(6006)

Avaya Aura

®

Communication Manager on

S8300 Server with

G450 Media Gateway

10.64.10.67

10.

64.

40.

42

Avaya Aura

®

Session Manager on

Avaya S8800 Server

10.64.22.12

Avaya Aura

®

Communication Manager on

Avaya S8800 Server

Layer 3 Router

0.3

4.2

.6

10

10

.64

Open Text Fax Server,

.10

.22

RightFax Edition

.17

0

(Ext. 40000)

Layer 3 Router

10.6

4.10

.170

Avaya G650

Media Gateway

(Port Network 2)

Avaya Aura

®

Session Manager on

Avaya S8800 Server

Avaya 9630

IP Telephone

(6001 - 6003)

Avaya 9630

SIP Telephone

(6010 - 6012)

(Registered to SM

Open Text Fax Server,

RightFax Edition

(Ext. 45000)

SIP Trunk

Main Site Remote Site/Simulated PSTN

Figure 1: RightFax interoperating with Session Manager via SIP Trunk

The Main Site had an Avaya S8800 Server running Communication Manager with two Avaya G650

Media Gateways. Each media gateway was configured as a separate port network in separate IP network regions. The RightFax server at this site communicated with Session Manager via SIP. In turn, Communication Manager used a SIP Trunk which terminated on a CLAN circuit pack in port network 2 to communicate with Session Manager. IP media resources were provided by Media

Processor (MedPro) circuit packs. Two versions of the MedPro circuit pack were tested in this configuration: TN2302AP and TN2602AP. Endpoints at this site included an Avaya 4600 Series IP

Telephone (with H.323 firmware), and an analog fax machine.

The Remote Site had an Avaya S8300 Server running Communication Manager in an Avaya G450

Media Gateway. The RightFax server at this site communicated with Session Manager via SIP. On

RAB; Reviewed:

SPOC 2/2/2011



3 of 43

RFax_SM6_SIP

the Avaya G450 Media Gateway, the signaling and media resources supporting a SIP trunk connected to Session Manager were integrated directly on the media gateway processor. Endpoints at this site included Avaya 9600 Series IP Telephones (with H.323 and SIP firmware), and an analog fax machine.

The IP telephones were not involved in the faxing operations, they were present in the configuration to verify the effect VoIP telephone calls had on the FoIP faxing operations.

Outbound fax calls originating from RightFax were sent to Session Manager first, then to

Communication Manager, via the configured SIP trunks. Based on the dialed digits, Communication

Manager directed the calls to the local fax machine, or the inter-site trunks (ISDN-PRI or SIP) to reach the Remote Site. Inbound fax calls to RightFax were first received by Communication

Manager from the local fax machine or from across either ISDN-PRI or SIP trunks connected to the

Remote Site. Communication Manager then directed the calls to RightFax via the configured

Session Manager SIP trunks.

RAB; Reviewed:

SPOC 2/2/2011



4 of 43

RFax_SM6_SIP

3. Equipment and Software Validated

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

Equipment

Avaya S8800 Servers (at both sites)

Avaya S8800 Server (at Main Site)

Avaya G650 Media Gateway (at Main Site)

- 2 CLANs

- 2 MedPros – TN2302

- 2 MedPros – TN2602

Avaya S8300D Server (at Remote Site)

Software/Firmware

Avaya Aura

Avaya Aura

Avaya Aura

®

®

Session Manager 6.0

System Manager 6.0

®

Communication Manager 6.0 SP1

R016x.00.0.345.0 with patch 18567

TN799DP - HW01 FW38 & HW13 FW 38

TN2302AP - HW20 FW120

TN2602AP - HW02 FW057

Avaya Aura

®

Communication Manager 6.0 SP1

R016x.00.0.345.0 with patch 18567

Avaya G450 Media Gateway (at Remote

Site)

Avaya 9620 IP Telephone (SIP)

Avaya 9630 IP Telephone (H.323)

Avaya 4621SW IP Telephone (H.323)

Analog Fax Machines

Open Text Fax Server, RightFax Edition on

Windows Server 2008R2

Dialogic Brooktrout SR140 Fax Software

 Boston Bfv API

 Boston Driver

 Boston SDK

 Boot ROM

Avaya one-X

30.14.0/1

®

Deskphone Edition SIP 2.5

H.323 3.11

H.323 2.9

-

9.4 Feature Pack 1 Service Release 3 v6.2.04 (Build 12) v6.2.00 (Build 4) v6.2.04 (Build 12)

6.2.1B9

RAB; Reviewed:

SPOC 2/2/2011



5 of 43

RFax_SM6_SIP

4. Configure Avaya Aura

®

Communication Manager

This section describes the Communication Manager configuration necessary to interoperate with

Session Manager and Open Text Fax Server, RightFax Edition. It focuses on the configuration of the SIP trunks connecting Communication Manager to the Avaya SIP infrastructure with the following assumptions:

 The examples shown in this section refer to the Main Site. Unless specified otherwise, these same steps also apply to the Remote Site using values appropriate for that location.

 The configuration of Communication Manager was performed using the System Access

Terminal (SAT). After the completion of the configuration, the save translation command was used to make the changes permanent.

4.1. Steps to Configure Communication Manager

The procedures for configuring Communication Manager include the following areas:

 Verify Communication Manager License (Step 1)

 Identify IP Interfaces (Step 2)

 Administer IP Network Regions (Steps 3 – 6)

 Administer IP Node Name (Step 7)

 Administer IP Network Map (Step 8)

 Administer IP Codec Set (Steps 9 – 10)

 Administer SIP Signaling Group (Step 11)

 Administer SIP Trunk Group (Steps 12 – 13)

 Administer Public Unknown Numbering (Step 14)

 Administer Route Pattern (Step 15)

 Administer AAR Analysis (Steps 16 – 17)

RAB; Reviewed:

SPOC 2/2/2011



6 of 43

RFax_SM6_SIP

Step Description

1. Verify Communication Manager License

Use the display system-parameters customer-options command to verify that the

Communication Manager license has proper permissions for features illustrated in these

Application Notes. Navigate to Page 2, and verify that there is sufficient remaining capacity for SIP trunks by comparing the Maximum Administered SIP Trunks field value with the corresponding value in the USED column.

The license file installed on the system controls the maximum permitted. If there is insufficient capacity, contact an authorized Avaya sales representative to make the appropriate changes display system-parameters customer-options Page 2 of 11

OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 12000 96

Maximum Concurrently Registered IP Stations: 18000 1

Maximum Administered Remote Office Trunks: 12000 0

Maximum Concurrently Registered Remote Office Stations: 18000 0

Maximum Concurrently Registered IP eCons: 414 0

Max Concur Registered Unauthenticated H.323 Stations: 100 0

Maximum Video Capable Stations: 18000 0

Maximum Video Capable IP Softphones: 18000 0

Maximum Administered SIP Trunks: 24000 298

Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0

Maximum Number of DS1 Boards with Echo Cancellation: 522 0

Maximum TN2501 VAL Boards: 128 2

Maximum Media Gateway VAL Sources: 250 0

Maximum TN2602 Boards with 80 VoIP Channels: 128 0

Maximum TN2602 Boards with 320 VoIP Channels: 128 2

Maximum Number of Expanded Meet-me Conference Ports: 300 0

(NOTE: You must logoff & login to effect the permission changes.)

RAB; Reviewed:

SPOC 2/2/2011



7 of 43

RFax_SM6_SIP

Step

2. Identify IP Interfaces

Description

Use the list ip-interface clan and list ip-interface medpro commands to identify IP interfaces in each network region. Interfaces in cabinet 01 (port network 1) as indicated in the Slot field are in IP network region 1 as indicated in the Net Rgn field.

Testing with the TN2302 and TN2602 circuit packs were done separately. When testing with the TN2302, the TN2602 was disabled (turned off) and vice versa as indicated in the

ON field. list ip-interface clan

IP INTERFACES

Skts Net Eth

ON Slot Code/Sfx Node Name/ Mask Gateway Node Warn Rgn VLAN Link

IP-Address

-- ---- -------- --------------- ---- --------------- ---- --- ---- ----

y 01A03 TN799 D CLAN1A /24 Gateway001 400 1 n 1

10.64.22.16

y 02A03 TN799 D CLAN2A /24 Gateway001 400 2 n 2

10.64.22.19 list ip-interface medpro

IP INTERFACES

Net

ON Slot Code/Sfx Node Name/ Mask Gateway Node Rgn VLAN Virtual Node

IP-Address

-- ----- -------- --------------- ---- --------------- --- ---- ---------------

n 01A02 TN2302 MEDPRO1A /24 Gateway001 1 n

10.64.22.15

n 02A02 TN2302 MEDPRO2A /24 Gateway001 2 n

10.64.22.18

y 01A04 TN2602 MEDPRO1A-2 /24 Gateway001 1 n

10.64.22.17

y 02A04 TN2602 MEDPRO2A-2 /24 Gateway001 2 n

10.64.22.20

RAB; Reviewed:

SPOC 2/2/2011



8 of 43

RFax_SM6_SIP

Step

3. Administer IP Network Region 1

Description

The configuration of the IP network regions (Steps 3 – 6) was already in place and is included here for clarity. At the Main Site, the Avaya G650 Media Gateway comprising port network 1 and all IP endpoints were located in IP network region 1.

Use the display ip-network-region command to view these settings.

A descriptive name was entered for the Name field.

 IP-IP Direct Audio (Media Shuffling) was enabled to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya Media

Gateway. This was done for both intra-region and inter-region IP-IP Direct Audio.

This is the default setting. Media Shuffling can be further restricted at the trunk level on the Signaling Group form.

 The Codec Set field was set to the IP codec set to be used for calls within this IP network region. In this case, IP codec set 1 was selected.

 The default values were used for all other fields.

At the Remote Site, all IP components were located in IP network region 1 and the IP network region was configured in the same manner as shown below. display ip-network-region 1 Page 1 of 20

IP NETWORK REGION

Region: 1

Location: Authoritative Domain: avaya.com

Name: PN1

MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes

Codec Set: 1 Inter-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? n

UDP Port Max: 3329

DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y

Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS

Audio PHB Value: 46 Use Default Server Parameters? y

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 6

Audio 802.1p Priority: 6

Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS

H.323 IP ENDPOINTS RSVP Enabled? n

H.323 Link Bounce Recovery? y

Idle Traffic Interval (sec): 20

Keep-Alive Interval (sec): 5

Keep-Alive Count: 5

RAB; Reviewed:

SPOC 2/2/2011



9 of 43

RFax_SM6_SIP


4. Administer IP Network Region 1 – Continued

On Page 4, codec sets are defined for inter-region calls. In the case of the compliance test at the Main Site, calls from IP network Source Region 1 to IP network region 2 (dst rgn

2) used codec set 1. The default values were used for all other fields. At the Remote Site, only one IP network region was used, so no inter-region settings were required. display ip-network-region 1 Page 4 of 20

Source Region: 1 Inter Network Region Connection Management I M

G A t

dst codec direct WAN-BW-limits Video Intervening Dyn A G c

rgn set WAN Units Total Norm Prio Shr Regions CAC R L e

1 1 all

2 1 y NoLimit n t

3

4

5

At the Main Site, IP network region 2 was created for Port Network 2 in a similar manner as IP network region 1 shown in Step 3 but with a different name. This was the network region used for H.323 Trunk connections to the RightFax server.

12

13

Note the default UDP Port Min and UDP Port Max settings, these were used in the

15 configuration of the RightFax server described in Section 5. display ip-network-region 2 Page 1 of 20

IP NETWORK REGION

Region: 2

Location: Authoritative Domain: avaya.com

Name: PN2

MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes

Codec Set: 1 Inter-region IP-IP Direct Audio: yes

UDP Port Min: 2048 IP Audio Hairpinning? n

UDP Port Max: 3329

DIFFSERV/TOS PARAMETERS

Call Control PHB Value: 46

Audio PHB Value: 46

Video PHB Value: 26

802.1P/Q PARAMETERS

Call Control 802.1p Priority: 6

Audio 802.1p Priority: 6

Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS

H.323 IP ENDPOINTS RSVP Enabled? n

H.323 Link Bounce Recovery? y

Idle Traffic Interval (sec): 20

Keep-Alive Interval (sec): 5

Keep-Alive Count: 5

RAB; Reviewed:

SPOC 2/2/2011



10 of 43

RFax_SM6_SIP


6. Administer IP Network Region 2 – Continued

The inter-region codec setting was created similarly to Step 4. display ip-network-region 2 Page 3 of 19

Source Region: 2 Inter Network Region Connection Management I M

G A e

dst codec direct WAN-BW-limits Video Intervening Dyn A G a

rgn set WAN Units Total Norm Prio Shr Regions CAC R L s

1 1 y NoLimit n all

2 1 all

3 3 y NoLimit n all

7. Administer IP Node Name

Use the change node-names ip command to create a node name that maps to the Session

Manager IP address. This node name is used in the configuration of the SIP trunk signaling group in Step 11. change node-names ip Page 1 of 2

IP NODE NAMES

Name IP Address

CLAN1A 10.64.22.16

CLAN2A 10.64.22.19

CM-Remote 10.64.21.111

DemoSM 10.64.20.31

Gateway001 10.64.22.1

MEDPRO1A 10.64.22.15

MEDPRO1A-2 10.64.22.17

MEDPRO2A 10.64.22.18

MEDPRO2A-2 10.64.22.20

TR18300 10.64.10.67

8. Administer IP Network Map

Session Manager and the RightFax server were configured to be located in an IP network region other than the default region 1; the region was assigned using the change ip-

network-map command. In the case of the compliance test, the IP addresses for these resources at the Main Site were assigned to IP network region 2 as shown in the example below. At the Remote Site, Session Manager and the RightFax server were located in the default IP network region 1, so it did not require an IP address map entry. change ip-network-map Page 1 of 63

IP ADDRESS MAPPING

Subnet Network Emergency

IP Address Bits Region VLAN Location Ext

--------------------------------------------- ------ ------ ---- -------------

FROM: 10.64.20.31 / 2 n

TO: 10.64.20.31

FROM: 10.64.22.170 / 2 n

TO: 10.64.22.170

RAB; Reviewed:

SPOC 2/2/2011



11 of 43

RFax_SM6_SIP

Step

9. Administer IP Codec set

Description

Use the change ip-codec-set 1 command to verify that G.711MU or G.711A is contained in the codec list. The example below shows the value used in the compliance test. display ip-codec-set 1 Page 1 of 2

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)

1: G.711MU n 2 20

10. Administer IP Codec set – Fax settings

On Page 2, set the FAX Mode field to t.38-standard. This is necessary to support the

RightFax server assigned to IP network region 2. The Modem Mode field should be set to off.

Leave the FAX Redundancy setting at its default value of 0. A packet redundancy level can be assigned to improve packet delivery and robustness of FAX transport over the network (with increased bandwidth as trade-off). Avaya uses IETF RFC-2198 and ITU-T

T.38 specifications as redundancy standard. With this standard, each Fax over IP packet is sent with additional (redundant) 0 to 3 previous fax packets based on the redundancy setting. A setting of 0 (no redundancy)

is suited for networks where packet loss is not a problem

. This setting should match the redundancy settings in Brooktrout SR140 configuration; otherwise Brooktrout SR140 will negotiate T.38 redundancy to the most common denominator (no redundancy in this case). change ip-codec-set 1 Page 2 of 2

IP Codec Set

Allow Direct-IP Multimedia? n

Mode Redundancy

FAX t.38-standard 0

Modem off 0

TDD/TTY US 3

Clear-channel n 0

RAB; Reviewed:

SPOC 2/2/2011



12 of 43

RFax_SM6_SIP

Step

11. Administer SIP Signaling Group

Description

For the compliance test, a signaling group and the associated SIP trunk group was used for routing fax calls to/from the RightFax server via Session Manager. For the compliance test at the Main Site, signaling group 12 was configured using the parameters highlighted below. All other fields were set as described in [3].

 The Group Type was set to sip.

 The Transport Method was set to the recommended default value of tls (Transport

Layer Security). As a result, the Near-end Listen Port and Far-end Listen Port are automatically set to 5061.

 The Near-end Node Name was set to CLAN2A, the node name that maps to the IP address of the CLAN circuit pack used to connect to Session Manager. Node names are defined using the change node-names ip command (see Step 7 above).

 The Far-end Node Name was set to demoSM. This node name maps to the IP address of the Session Manager server as defined using the change node-names ip command.

 The Far-end Network Region was set to 2. This is the IP network region which contains Session Manager and RightFax.

 The Far-end Domain was set to avaya.com. This domain is sent in the headers of SIP

INVITE messages for calls originating from and terminating to Session Manager using this signaling group.

 Direct IP-IP Audio Connections was set to y. This field must be set to y to enable

Media Shuffling on the trunk level (see Step 3 on IP-IP Direct Audio).

 The DTMF over IP field was set to the default value of in-band.

 The default values were used for all other fields. change signaling-group 12 Page 1 of 1

SIGNALING GROUP

Group Number: 12 Group Type: sip

IMS Enabled? n Transport Method: tls

Q-SIP? n SIP Enabled LSP? n

IP Video? n Enforce SIPS URI for SRTP? y

Peer Detection Enabled? y Peer Server: SM

Near-end Node Name: CLAN2A Far-end Node Name: demoSM

Near-end Listen Port: 5061 Far-end Listen Port: 5061

Far-end Network Region: 2

Far-end Domain: avaya.com

Bypass If IP Threshold Exceeded? n

Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n

DTMF over IP: in-band Direct IP-IP Audio Connections? y

Session Establishment Timer(min): 3 IP Audio Hairpinning? n

Enable Layer 3 Test? y Initial IP-IP Direct Media? n

H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

RAB; Reviewed:

SPOC 2/2/2011



13 of 43

RFax_SM6_SIP

Step

12. Administer SIP Trunk Group

Description

For the compliance test, trunk group 12 was used for the SIP trunk group for routing fax calls to/from Session Manager. Trunk group 12 was configured using the parameters highlighted below. All other fields were set as described in [3].

On Page 1:

 The Group Type field was set to sip.

 A descriptive name was entered for the Group Name.

 An available trunk access code (TAC) that was consistent with the existing dial plan was entered in the TAC field.

 The Service Type field was set to tie.

 The Signaling Group was set to the signaling group shown in the previous step.



The Number of Members field contained the number of trunks in the SIP trunk group. It determines how many simultaneous SIP calls can be supported by the configuration. Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call.

 The default values were used for all other fields. change trunk-group 12 Page 1 of 21

TRUNK GROUP

Group Number: 12 Group Type: sip CDR Reports: y

Group Name: PN2 to demoSM COR: 1 TN: 1 TAC: *012

Direction: two-way Outgoing Display? n

Dial Access? n Night Service:

Queue Length: 0

Service Type: tie Auth Code? n

Member Assignment Method: auto

Signaling Group: 12

Number of Members: 50

RAB; Reviewed:

SPOC 2/2/2011



14 of 43

RFax_SM6_SIP


13. Administer SIP Trunk Group – continued

On Page 3:

 Set the Numbering Format field to public. This field specifies the format of the calling party number sent to the far-end.

 Default values may be used for all other fields. change trunk-group 12 Page 3 of 21

TRUNK FEATURES

ACA Assignment? n Measured: none

Maintenance Tests? y

Numbering Format: public

UUI Treatment: service-provider

Replace Restricted Numbers? n

Replace Unavailable Numbers? n

Modify Tandem Calling Number: no

Show ANSWERED BY on Display? y

14. Administer Public Unknown Numbering

Public unknown numbering defines the calling party number to be sent to the far-end.

Use the change public-unknown-numbering command to create an entry that will be used by the trunk groups defined in Steps 12-13. In the example shown below, all calls originating from a 5-digit extension beginning with 2 or 4 and routed across any trunk group (Trk Grp column is blank) were sent as a 5-digit calling number. change public-unknown-numbering 0 Page 1 of 2

NUMBERING - PUBLIC/UNKNOWN FORMAT

Total

Ext Ext Trk CPN CPN

Len Code Grp(s) Prefix Len

Total Administered: 6

5 1 5 Maximum Entries: 9999

5 2 5

5 4 5 Note: If an entry applies to

5 5 5 a SIP connection to Avaya

4 6 4 Aura(tm) Session Manager,

5 7 5 the resulting number must

be a complete E.164 number.

RAB; Reviewed:

SPOC 2/2/2011



15 of 43

RFax_SM6_SIP

Step

15. Administer Route Pattern

Description

Use the change route-pattern command to create a route pattern that will route fax calls to the SIP trunk that connects to the RightFax server.

The example below shows the route pattern used for the compliance test at the Main Site.

A descriptive name was entered for the Pattern Name field. The Grp No field was set to the trunk group created in Steps 12–13. The Facility Restriction Level (FRL) field was set to a level that allows access to this trunk for all users that require it. The value of 0 is the least restrictive level. The default values were used for all other fields. change route-pattern 12 Page 1 of 3

Pattern Number: 12 Pattern Name: To SM

SCCAN? n Secure SIP? n

Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC

No Mrk Lmt List Del Digits QSIG

Dgts Intw

1: 12 0 n user

16. Administer AAR Analysis

Automatic Alternate Routing (AAR) was used to route calls to RightFax via Session

Manager. Use the change aar analysis command to create an entry in the AAR Digit

Analysis Table for this purpose. The example below shows entries previously created for the Main Site using the display aar analysis 0 command. The 3 rd

highlighted entry specifies that 5 digit dial string 40000 was to use route pattern 12 to route calls to the

RightFax fax server at the Main Site via Session Manager. The dial string 45000 (the

RightFax server at the Remote Site) used Route Pattern 15 to route calls between

Communication Managers. change aar analysis 0 Page 1 of 2

AAR DIGIT ANALYSIS TABLE

Location: all Percent Full: 1

Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd

10 4 4 4 aar n

3 5 5 12 aar n

40000 5 5 12 aar n

45000 5 5 15 aar n

RAB; Reviewed:

SPOC 2/2/2011



16 of 43

RFax_SM6_SIP

5. Configure Avaya Aura

®

Session Manager - Overview

This section covers the configuration of Session Manager at the Main Site. Session Manager is configured via an Internet browser using the administration web interface. It is assumed that the setup screens of the administration web interface have been used for initial configurations. For additional information on these installation tasks, refer to [3].

Each SIP endpoint used in the compliance test that registered with Session Manager required that a user and endpoint profile be created and associated with Session Manager. This configuration is not directly related to the interoperability of the products being tested, so it is not included here. These procedures are covered in [3].

This section summarizes the configuration steps that are necessary for interoperating with RightFax.

The test environment was previously configured to enable Communication Manager and Session

Manager at each site to communicate with each other. Details of this configuration are not described in this document, and additional information can be obtained in [3].

The documented configurations were repeated for the Session Manager at the Remote Site using values appropriate for the Remote Site from Figure 1. This includes but is not limited to the IP addresses, SIP domain and user extensions.

The steps used were:

 Create a SIP Entity for the RightFax Server

 Create a SIP Entity Link for the RightFax Server

 Create a Routing Policy

 Create or Modify Dial Patterns

RAB; Reviewed:

SPOC 2/2/2011



17 of 43

RFax_SM6_SIP

5.1. Configure Session Manager - Details

This section summarizes the applicable user-defined parameters used during the SIP installation procedures.

Step

1.

Login

Description

Access the System Manager administration web interface by entering https://<ip-addr>/SMGR as the URL in an Internet browser, where <ip-addr> is the IP address (or FQDN) of the System Manager server.

Log in with the appropriate credentials.

2.

Create a SIP Entity for the RightFax Server

Navigate to Routing\SIP Entities and click New to create an Entity definition. In the screenshot below, the Entity RB_FaxServer was previously created using the following settings.

RAB; Reviewed:

SPOC 2/2/2011



18 of 43

RFax_SM6_SIP


3.

Create a SIP Entity for the RightFax Server - Continued

Enter a descriptive Name such as RB_FaxServer and enter the FQDN or IP Address for the RightFax server as shown below. Select Other for the Entity Type. All other settings were defaults.

4.

Create an Entity Link for the RightFax Server

An Entity Link establishes the details of how Entities will communicate with each other.

Use the Add button to create a new link. In this case, Session Manager at the Main Site,

demoSM, was configured to communicate with RB_FaxServer using UDP protocol over port 5060 as a Trusted Entity.

RAB; Reviewed:

SPOC 2/2/2011



19 of 43

RFax_SM6_SIP


5.

Create a Routing Policy

Navigate to Routing\Routing Policies and click New to create a routing policy for incoming calls to the RightFax server. The illustration below was captured after the

Policy RB_Fax_Server_2 had been created and the following steps will describe how this policy was created.

A Routing Policy consists of a definition of the SIP Entity as Destination, the Time of

Day the policy applies, and the Dial Patterns that will trigger this particular policy.

Below are the settings used for this test. Use the Select or Add buttons to create or use existing definitions for each parameter.

RAB; Reviewed:

SPOC 2/2/2011



20 of 43

RFax_SM6_SIP


6.

Create or Modify Dial Patterns

Associating a dial pattern with a SIP Entity Link instructs Session Manager how to route calls matching the administered Dial Pattern(s). In the test, existing Routing Policies were modified for routing to endpoints or Entities at the Remote Site, and new Dial

Patterns were created to route to the Main Site and Remote Site Fax Servers using the existing and new routing policies.

In the snapshot below, the Dial Patterns were previously created. The applicable patterns were all 5 Digit extension patterns; dialed numbers beginning with 2 (the local analog fax machine at the Main Site), dialed numbers beginning with 40 (to route incoming Fax calls to the RightFax server at the Main Site), dialed numbers beginning with 45 (to route to Communication Manager at the Main Site in order to route via the public network interface between the sites). In addition, an existing 4 digit patterns beginning with 60 was used to route Fax calls to Communication Manager at the Main Site for routing via the public network interfaces to the analog machine at the Remote Site.

The ‘40’ and ‘45’ dial patterns were created for this test, all others were in place in the test environment.

RAB; Reviewed:

SPOC 2/2/2011



21 of 43

RFax_SM6_SIP


7.

Create or Modify Dial Patterns – Continued

The entries required to create the new Dial Pattern for routing calls to the RightFax server at the Main site are illustrated below. The Pattern, Min and Max number of digits, and SIP Domain entries were used for this Dial Pattern definition. Click Add to associate the dial pattern with an existing Routing Policy, in this case the

RB_Fax_Server_2 policy created in Step 5 above. The Originating Location Name

All was used in this case to apply this pattern regardless of originating locations. In the same way, a new Dial Pattern was created and associated with the existing policy to route calls to Communication Manager at the Main Site using the Dial Pattern 45. This was used to route calls from the Main Site Fax Server to the Remote Site Fax Server.

RAB; Reviewed:

SPOC 2/2/2011



22 of 43

RFax_SM6_SIP

6. Configure Open Text Fax Server, RightFax Edition

This section describes the configuration of Open Text Fax Server, RightFax Edition and the embedded Brooktrout SR140 virtual fax board software from Dialogic. It assumes that the application and all required software components, including Brooktrout SR140 and the database software (MSSQL 2008), have been installed and properly licensed. For instructions on installing

RightFax, consult the RightFax Installation Guide [5].

Note that the configurations documented in this section pertain to interoperability between RightFax and the Avaya SIP infrastructure. The standard configurations pertaining to RightFax itself (e.g., administering fax channels) are not covered. For instructions on administering and operating

RightFax, consult the RightFax Administrator‟s Guide [4].

The configuration procedures covered in this section include the following:

 Launch RightFax Enterprise Fax Manager and SR140 Configuration Tool (Steps 1 – 4)

 Configure SR140 IP stack (Step 5)

 Configure SR140 SIP IP parameters (Step 6)

 Configure SR140 T.38 parameters (Step 7)

 Update SR140 configuration file (Step 8)

 Administer RightFax dialing rules (Steps 9 – 11)

 Administer RightFax users (Steps 12 – 15)

The examples shown in this section refer to the Main Site. Unless specified otherwise, these same steps also apply to Remote Site using values appropriate from Figure 1.

RAB; Reviewed:

SPOC 2/2/2011



23 of 43

RFax_SM6_SIP


1. Launch RightFax Enterprise Fax Manager

The RightFax configuration is performed using the RightFax Enterprise Fax Manager.

Launch the RightFax Enterprise Fax Manager from the Windows Start menu. At the main window, highlight the host name of the fax server (created during the installation process) from the navigation menu in the left pane:

RAB; Reviewed:

SPOC 2/2/2011



24 of 43

RFax_SM6_SIP


2. RightFax DocTransport Module

The Brooktrout SR140 was configured during installation. To view or modify the settings, the RightFax DocTransport Module must be stopped. Right-click this module in the lower right pane and select Stop All Services. After all the service modules indicate the stopped status, right-click the RightFax DocTransport Module name again to select Configure Service.

RAB; Reviewed:

SPOC 2/2/2011



25 of 43

RFax_SM6_SIP


3. RightFax DocTransport Module - Continued

In the DocTransport Configuration window that appears, click the Configure

Brooktrout button.

RAB; Reviewed:

SPOC 2/2/2011



26 of 43

RFax_SM6_SIP


4. Brooktrout Configuration Tool

The Brooktrout Configuration Tool – Wizard Mode window gets displayed. Click the

Advanced Mode button in this window.

RAB; Reviewed:

SPOC 2/2/2011



27 of 43

RFax_SM6_SIP


5. Configure IP Stack

A Configure IP Stack window is displayed on first invocation of the Brooktrout SR140 configuration tool (assuming the Brooktrout SR-140 licenses were installed):

Choose SIP and click OK. The following SR140 configuration tool window is displayed.

Note that IP Stack can be viewed/reconfigured from the Brooktrout Configuration Tool menu Options  Configure IP Stack.

RAB; Reviewed:

SPOC 2/2/2011



28 of 43

RFax_SM6_SIP


6. Configure SIP IP Parameters

Important: This step describes configuring the SIP Primary Gateway address using the

Brooktrout Configuration Tool. This method is sufficient if the RightFax Server will communicate with a single SIP gateway or Session Manager. Alternatively, use

RightFax Dialing Rules, described in steps 10-11 below, to specify the address of the SIP

Gateway. You should only use one of these options. Do not use both.

Either option listed above is appropriate if Media Shuffling is enabled in the Avaya communications equipment.

If the RightFax Server needs to be configured to communicate directly with multiple gateways or Session Managers, use only RightFax Dialing Rules. Use a separate

Dialing Rule for gateway.

Navigate to Brooktrout  IP Call Control Modules  SIP in the left navigation menu. Select the IP Parameters tab in the right pane. Configure the fields as follows:

 Primary Gateway –set to the IP address of the Session Manager server, and port number 5060.

 From Value – set to appropriate UserInfo@DomainName. The DomainName should be set to the authoritative domain as configured in Session Manager.

 Contact Address – set to the IP address assigned to RightFax and the port number 5060.

 Username – Required. Default value is a dash („-„) character.

Use default values for all other fields.

RAB; Reviewed:

SPOC 2/2/2011



29 of 43

RFax_SM6_SIP


7. Configure T.38 Parameters

Select the T.38 Parameters tab. Configure the fields as shown below in the screenshot.

Notes:

- “Media Renegotiate Delay Inbound, msec” has been changed from default.

- “Maximum Bit Rate, bps” is set to maximum, 33600.

- “Maximum T.38 Version” is set to “3”.

RAB; Reviewed:

SPOC 2/2/2011



30 of 43

RFax_SM6_SIP


8. Configure RTP Parameters

Select the RTP Parameters tab. Set the RTP codec list value to use only a single codec, either pcmu or pcma to match the codec used in your region.

RAB; Reviewed:

SPOC 2/2/2011



31 of 43

RFax_SM6_SIP

9. Configure RTP Port Range

In the left-hand navigation area, expand Call Control Parameters and click the Module

0x##: SR140 item for your FoIP channels.

Configure the Lowest IP Port Number and Highest IP Port Number values to match the UDP Port Min and UDP Port Max values in the IP Network Region configuration screen in Communication Manager. Note: Communication Manager default port range

is 2048 to 3329; however, the Brooktrout Configuration Tool range only spans 1000 ports. If you set Lowest IP Port Number to 2048, the Highest Port Number should

automatically set to 3048.

RAB; Reviewed:

SPOC 2/2/2011



32 of 43

RFax_SM6_SIP

10. Complete Brooktrout SR140 Configuration

After verifying all the above parameters are properly set, click Save in the button menu.

Exit the Brooktrout Configuration Tool.

In the DocTransport Configuration screen, click the OK button.

From Windows explorer, navigate to the Brooktrout folder in the RightFax install directory (typically Program Files\RightFax\DecTransport\Brooktrout), Open and edit the callctrl.cfg file as follows, then save the updates:

 Verify that the following configuration segment is present; and that rtp_codec matches the value specified in Step 8 above, either “pcmu” or “pcma”.

[host_module.1/rtp]

rtp_codec=pcmu

 Change rtp_ced_enable setting to true under the

[host_modele.1/t.38parameters] header (

... below indicates other entries under the header)

[host_module.1/t.38parameters]

...

rtp_ced_enable=true

...

After making and saving the above updates in the callctrl.cfg file, restart all RightFax service modules by right clicking the RightFax DocTransport Module name in the lower right pane of the RightFax Enterprise Fax Manager window and select Start All

Services (see Step 2).

RAB; Reviewed:

SPOC 2/2/2011



33 of 43

RFax_SM6_SIP

11. Configure Dialing Rules

Note: See Step 6 above for discussion on using RightFax Dialing Rules versus the

Brooktrout Configuration Tool to configure media gateway addresses.

Dialing Rules are used by RightFax to route calls. In the compliance test, a dialing rule was created to route outbound fax calls to the Avaya Media Gateway. In the left navigation menu under the host name of the fax server, navigate to Dialing Plan, rightclick Dialing Rules and select New to create a new rule.

The example below shows the single rule created for the compliance test at site 1. The * in the Pattern field indicates that this rule applies to all dialed numbers. To view the details, double click on the rule in the right pane.

RAB; Reviewed:

SPOC 2/2/2011



34 of 43

RFax_SM6_SIP

12. Configure Dialing Rules - Continued

The Rule Edit window will appear as shown below. The Number Adjustments tab shows the digit string manipulation that is done to each dialed number. In the example below, each number is appended with @10.64.40.42 as indicated in the Append this field. This IP address is for the Session Manager server at the Main Site.

RAB; Reviewed:

SPOC 2/2/2011



35 of 43

RFax_SM6_SIP

13. Configure Dialing Rules - Continued

In the Destination tab of the Rule Edit window, select the correct host name of the fax server for Send via Fax Server. This is typically the local computer name.

RAB; Reviewed:

SPOC 2/2/2011



36 of 43

RFax_SM6_SIP

14. Configure Users

A user is created on RightFax for each incoming fax number. The user represents the fax recipient. To view the list of users, navigate to Users in the left navigation menu under the host name of the fax server. The example below shows a list of 4 users.To view the details of a user, double-click on the user entry in the right pane .

RAB; Reviewed:

SPOC 2/2/2011



37 of 43

RFax_SM6_SIP

15. Configure Users – Identification

The User Edit window will appear as shown below. Select the Identification tab. The example below shows the settings used for the compliance test at the Main Sites. The

User ID field is set to a descriptive name. Appropriate values should be entered or selected for other fields .

RAB; Reviewed:

SPOC 2/2/2011



38 of 43

RFax_SM6_SIP

16. Configure Users – Inbound Routing

On the Inbound Routing tab, the Routing Code field is set to the fax number of the recipient. In the case of the compliance test, this was extension 40000 for the Main Site, and 45000 for the Remote Site (pictured below). Default values may be used for all other fields .

RAB; Reviewed:

SPOC 2/2/2011



39 of 43

RFax_SM6_SIP

17. Configure Users – Outbound Settings

The Default Outbound Settings tab configures various outbound fax call settings.

Configure these settings as appropriate

.

RAB; Reviewed:

SPOC 2/2/2011



40 of 43

RFax_SM6_SIP

7. General Test Approach and Test Results

This section describes the compliance test approach used to verify interoperability of Open Text Fax

Server, RightFax Edition with Avaya Aura

®

Session Manager.

7.1. General Test Approach

The general test approach was to make intra-site and inter-site fax calls to and from RightFax. The inter-site calls were made using SIP or ISDN-PRI trunks between the sites. Faxes were sent with various page lengths, resolutions, and at various fax data speeds. For capacity, a large number of 2page faxes were continuously sent between the two RightFax servers simultaneously. Serviceability testing included verifying proper operation/recovery from failed cables, unavailable resources, and

Session Manager and RightFax restarts. Fax calls were also tested with different Avaya Media

Gateway media resources used to process the fax data between sites. This included the TN2302

MedPro circuit pack, the TN2602 MedPro circuit pack in the Avaya G650 Media Gateway; the integrated VoIP engine of the Avaya G450 Media Gateway and the Avaya MM760 Media Module installed in the Avaya G450 Media Gateway.

7.2. Test Results

Open Text Fax Server, RightFax Edition successfully passed all compliance testing.

7.3. General Observations

Fax calls consume DSP (Digital Signal Processing) resources for processing fax data on the

TN2302AP IP Media Processor (MedPro) circuit pack and the TN2602AP IP Media Processor circuit pack in the Avaya G650 Media Gateway, and the integrated Voice over Internet Protocol

(VoIP) engine of the Avaya G450 Media Gateway. To increase the capacity to support simultaneous fax calls, additional TN2302AP and/or TN2602AP MedPro circuit packs need to be installed in the

Avaya G650 Gateway, and additional Avaya MM760 Media Module or Modules need to be installed in the Avaya G450 Media Gateway. The information contained in the table below indicates DSP capacities/usage in the Avaya media processors. Customers should work with their Avaya sales representatives to ensure that their fax solutions have adequate licenses and DSP resources to match the intended Fax capacity/usage.

Platform

Device

TN2302, G450, MM760

TN2602

DSP Resources per Platform Device

64

64

DSP Resources per FoIP Call

4

1

Note that the SIP trunk group on Communication Manager for connecting to Session Manager at each site, as well as the SIP or ISDN-PRI trunk group for connecting the 2 sites must be configured with adequate number of trunk group members to support the number of simultaneous fax calls intended. On RightFax, adequate number of fax channels must also be appropriately configured for the intended capacity.

RAB; Reviewed:

SPOC 2/2/2011



41 of 43

RFax_SM6_SIP

8. Verification Steps

The following steps may be used to verify the configuration:

 From Communication Manager SAT, use the status signaling-group command to verify that the SIP signaling groups configured in Step 11 of Section 4 are in-service.

 From Communication Manager SAT, use the status trunk-group command to verify that the SIP trunk group configured in Section 4, Steps 12 - 13 is in-service.

 Verify that fax calls can be placed to/from Open Text Fax Server, RightFax Edition server at each site.

 From Communication Manager SAT, use the list trace tac command to verify that fax calls are routed to the expected trunks.

 From System Manager, confirm that the Entity Link between Session Manager and the Open

Text Fax Server, RightFax Edition server is in service.

9. Conclusion

These Application Notes describe the procedures required to configure Open Text Fax Server,

RightFax Edition server to interoperate with Avaya Aura

Aura

®

®

Communication Manager and Avaya

Session Manager. Open Text Fax Server, RightFax Edition successfully passed compliance testing.

10. Additional References

[1] Avaya Aura™ Communication Manager Feature Description and Implementation, Doc # 555-

245-205, Release 6.0, Issue 8.0, June, 2010.

[2] Administering Avaya Aura™ Communication Manager, Doc # 03-300509, Release 6.0, Issue

6.0, June, 2010.

[3] Administering Avaya Aura™ Session Manager, Doc # 03-603324, Release 6.0, Issue 3, August,

2010.

[4] Open Text Fax Server, RightFax Edition Administrator’s Guide, Version 9.4 Feature Pack 1,

January 8, 2010.

[5] Open Text Fax Server, RightFax Edition Installation Guide Version 9.4 Updated for Feature

Pack 1 Service Release 3, November 11, 2010.

Documentation for:

Avaya products may be found at http://support.avaya.com

.

RightFax may be found at https://knowledge.opentext.com/knowledge/llisapi.dll/open/16512673 .

RAB; Reviewed:

SPOC 2/2/2011



42 of 43

RFax_SM6_SIP

©

2011 Avaya Inc. All Rights Reserved.

Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes.

Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya DevConnect

Program at [email protected].

RAB; Reviewed:

SPOC 2/2/2011



43 of 43

RFax_SM6_SIP

Application Notes for Configuring Open Text Fax Server, RightFax

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Open Text Fax Server,

RightFax Edition with Avaya Aura

Communication Manager and Avaya Aura

Session Manager via SIP Trunk Interface

- Issue 1.0

Abstract

1. Introduction

1.1. Interoperability Compliance Testing

1.2. Support

2. Configuration

2.1. Configuration Details

3. Equipment and Software Validated

4. Configure Avaya Aura

Communication Manager

4.1. Steps to Configure Communication Manager

5. Configure Avaya Aura

Session Manager - Overview

5.1. Configure Session Manager - Details

6. Configure Open Text Fax Server, RightFax Edition

7. General Test Approach and Test Results

7.1. General Test Approach

7.2. Test Results

7.3. General Observations

8. Verification Steps

9. Conclusion

10. Additional References

Related manuals

Avaya

SM61