SIP Trunking Configuration Guide for Cisco Unified Communications Manager (CUCM) Version 9.0.1.11005-1 with Cisco Unified Border Element (CUBE)



Similar documents
and 2, implemented With Cisco Unified Border Control Element (CUBE)

IDT / Net2phone SIP Trunking Configuration Guide for Cisco Business Edition 3000 (BE3000) Release with Cisco Unified Border Element Release 8.8.

Motorola TEAM WSM - Cisco Unified Communications Manager Express (CME) Integration

CS3695/M6-109 Lab 8-NPS02 VOIP Sniffing Ver. 8 Rev. 0

Paetec SIP Configuration Guide The missing manual

EarthLink Business SIP Trunking. Cisco Call Manager and Cisco CUBE Customer Configuration Guide

Cisco Unified Communications Manager with Cisco Unified Border Element (IOS 15.4(2)T) using SIP

Cisco CCA Tool SIP Security methods

ADTRAN SBC and Cisco Call Manager Express SIP Trunk Interoperability

Brest. Backup : copy flash:ppe_brest1 running-config

Integra Telecom SIP Trunking: Connecting Cisco Unified Communications Manager 8.5(1) via the Cisco Unified Border Element using SIP

Cisco Unified Communications Manager with Cisco Unified Border Element [CUBE IOS-XE 3.15] on ISR 4K using SIP

Level 3 SIP Trunking: Connecting Cisco Unified Communications Manager 7.1(3) via the Cisco Unified Border Element using SIP

Lab Configuring Syslog and NTP (Instructor Version)

EarthLink Business SIP Trunking. Cisco CUCM 9.1 with CUBE Customer Configuration Guide

Configuring Fax Pass-Through

Cisco ISDN PRI to SIP Gateway

Sprint SIP Toll Free: Connecting Cisco Unified Customer Voice Portal 8.5 via the Cisco Unified Border Element 8.8 using SIP

CenturyLink SIP Trunking: Connecting Cisco Unified Communications Manager via the Cisco Unified Border Element 8.6 using SIP

Cisco IOS SIP Configuration Guide

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5

Verizon IP Trunking Service: Connecting Cisco Unified Communications Manager 6.1(2) via the Cisco Unified Border Element using SIP

Dial Peer Configuration Examples

Intelepeer SIP Trunking: Connecting Cisco Unified Communications Manager 8.5(1) via the Cisco Unified Border Element 1.3 using SIP

Feb, Note: Testing was conducted in tekvizion Labs.

Microsoft Lync 2013 [v ] to Verizon Business SIP Trunk via the Cisco Unified Border Element 10.5 [IOS 15.4(3)M]

Business Talk IP (France and International) connecting:

Time Warner Cable Business Class (TWCBC):

Validated Integrations: CUCM 10.x with xic version 4.0 SU-6 (support included for all 4.0 SU s) Version 4.08

Cisco 2621 Gateway-PBX Interoperability: Lucent/Avaya Definity G3si V7 PBX with Cisco CallManager Using T1 PRI NI-2 for an H.

Configuring Voice and Data Support on VWIC3s

Let's take a look at another example, which is based on the following diagram:

Case Study 1: Registering IP Phones with a remote Call

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

TotalCloud Phone System

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX)

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

Ejemplo de configuración de punta a punta SBC en un Cisco 7600 Series Router

ADTRAN SBC and Cisco Unified Call Manager SIP Trunk Interoperability

Application Note. December 2014 Table of Contents

Configuration Professional: Site to Site IPsec VPN Between Two IOS Routers Configuration Example

BRI to PRI Connection Using Data Over Voice

Implementing Cisco IOS Unified Communications (IIUC)

How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions

Digium Switchvox AA65 PBX Configuration

Simple MPLS network topology for Dynamips/Olive

Avaya IP Office 8.1 Configuration Guide

Configuration Notes 290

How To Configure A Cisco Router With A Cio Router

SIP Trunking Test Results for CudaTel Communication Server

CCNA Exploration 4.0: ESwitching Basic Switching / Wireless PT Practice SBA. Switch S1 S1#sh ru Building configuration...

Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager Issue 1.0

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Panasonic KX-NCP500 IP PBX V2.0502

Avaya one-x Quick Edition Interoperability with Cisco Integrated Services Router (ISR) SIP Gateway - Issue 1.0

Lab 5.3.9b Managing Router Configuration Files Using TFTP

Cisco Unified Communications 500 Series

Lab 7: Firewalls Stateful Firewalls and Edge Router Filtering

Basic Router Configuration Using Cisco Configuration Professional

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1

XO SIP Service Customer Configuration Guide for Interactive Intelligence Customer Interaction Center (CIC) with XO SIP

Felix Rohrer. PT Activity 7.5.3: Troubleshooting Wireless WRT300N. Topology Diagram

Network Scenarios Pagina 1 di 35

Note: This case study utilizes Packet Tracer. Please see the Chapter 5 Packet Tracer file located in Supplemental Materials.

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.

Session Title: Exploring Packet Tracer v5.3 IP Telephony & CME. Scenario

Direct IP Calls. Quick IP Call Mode

Cisco Voice Gateways. PacNOG6 VoIP Workshop Nadi, Fiji. November Jonny Martin - jonny@jonnynet.net

for SS7 VoIP Gateways

Note: As of Feb 25, 2010 Priority Telecom has not completed FXS verification of fax capabilities. This will be updated as soon as verified.

EarthLink Business SIP Trunking. NEC SV8300 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide

1 SIP Carriers Warnings Vendor Contact Vendor Web Site : Versions Verified SIP Carrier status as of 9/11/2011

nexvortex Setup Guide

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0

ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability

Direct Inward Dial Digit Translation Service

Application Note. Note: Testing was conducted in Verizon lab. May 10, Initial Version Table of Contents

Cisco Voice over IP

Optimum Business SIP Trunk Set-up Guide

Juniper Networks WX Series Large. Integration on Cisco

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V p13 Configuration Guide

3CX PBX v12.5. SIP Trunking using the Optimum Business Sip Trunk Adaptor and the 3CX PBX v12.5

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports

Introducing Cisco Voice and Unified Communications Administration Volume 1

AudioCodes Mediant 1000 Configuration Guide

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Allworx 6x IP PBX

EarthLink Business SIP Trunking. ININ IC3 IP PBX Customer Configuration Guide

Lexmark Fax over IP. Pete Davidson. Document Version 1.0. October 4, 2012

Configuring Modem Transport Support for VoIP

VoIP Gateway/IP-PBX Interworking with Skype

IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>>

Cisco Unified Communications Manager SIP Trunk Configuration Guide for the VIP-821, VIP-822 and VIP-824

NetVanta 7100 Exercise Service Provider SIP Trunk

Application Notes Rev. 1.0 Last Updated: February 3, 2015

Implementing Cisco Voice Communications and QoS

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide

SIP Trunking with Elastix. Configuration Guide for Matrix SETU VTEP

Integrating VoIP Phones and IP PBX s with VidyoGateway

Transcription:

SIP Trunking Configuration Guide for Cisco Unified Communications Manager (CUCM) Version 9.0.1.11005-1 with Cisco Unified Border Element (CUBE)

Table of Contents Introduction... 3 Executive Summary..3 Equipment Hardware and Software Requirements...5 Test Configurations...6 CUCM Configurations.. 7 CUBE Configuration Details.... 42 Equipment Configuration Files and Information 48 1

1. Introduction This document contains the test results of the SIP Trunking evaluation performed on the CUCM version 9.0.1.11005-1 connected to a Cisco 2821 Integrated Services Router (ISR) running CUBE software version c2800nm-advipservicesk9-mz.151-4.m6.bin and a Cisco 2432-24FXS Internet Access Device (IAD) acting as a router. The Cisco ATA187 was used to test FAX pass through and FAX Relay. The ATA187 was connected to the Cisco 3560 PoE switch. The Cisco IAD was connected to the Juniper Networks ERX 1400 via a T1 configured for Point-to-Point Protocol (PPP). Cisco 7961 IP phones running SCCP software downloaded from the CUCM were connected to the CUCM via a Cisco 3560 Power over Ethernet (PoE) switch. The Cisco 2821 ISR was also connected to the Cisco 3560. The Cisco 3560 switch is up linked to the Cisco IAD via Fast Ethernet port 0/0. The customer LAN will connect directly to the Cisco IAD via a separate Ethernet port and not the Cisco 3560 switch which is reserved only for Cisco IP phones. 2. Executive Summary This configuration guide provides the settings used during the lab evaluation for the CUCM with CUBE with XO Communications SIP Trunking service. The following is a summary of the issues addressed and the limitations found during the lab evaluation. When configuring the customer for SP1 or SP2, the MoH Server codec must be set accordingly. CUBE SIP Profile Rules are still required to correct the MoH problem. CUBE ip address trusted authenticate feature command is enabled by default. Cisco does not support SIP REFER method for outbound calls. FAX was tested using the Cisco ATA187. FAX pass-through using SP2 does not work properly. CUCM MoH Server Codec Setting: Make sure that we match the codec package to make MoH work properly. When configuring the customer for SP1 or SP2, the MoH Server codec must be set accordingly to G.711ulaw for SP1 and G.729 for SP2 as discussed in detail in section 5.15 of this report. CUBE SIP Profile Rules Are Required to Correct the MoH Problem These configuration commands are still required in the CUBE sip profile section of the configuration for the remote party to hear MoH when the call is placed on hold from a CUCM phone. 2

CUBE ip address trusted authenticate Feature Command is Enabled by Default This is on by default and XO IP address must be put in the proper command to allow the XO IP address. Cisco Does Not Support SIP Refer For Outbound Calls Transfers with CUCM will use traditional reinvites; SIP REFER not used. SP1 and SP2 G.711 Pass-through and T.38 FAX Issues and Workarounds Pkg Codec DTMF Fax 1 G.711 RFC 2833 (in-band DTMF is NOT supported) 2 G.729a/G.711 RFC 2833 (in-band DTMF is NOT supported) G.711 passthrough and T.38 are supported (G.711 passthrough is NOT supported.) T.38 is supported Registration: Static 3

3. Equipment Hardware and Software Requirements for Testing 1. CUCM Server a. Hardware: Cisco MCS 7800 Series Product No MCS7825H3-K9-CMB2 b. Software Version: CUCM 9.0.1.11005-1 2. Cisco 2821 ISR running CUBE software a. Hardware: Cisco 2821 ISR b. 1 PVDM2-32 and 2 PVDM2-48 DSP modules c. IOS software version c2800nm-advipservicesk9-mz.151-4.m6.bin 3. Cisco Unity Connection (CUC) Voice Mail Server a. Hardware: Cisco MCS 7800 Series Product No MCS7825H3-K9-CMB2 b. Software Version: CUC 7.0.1.11000-13-2 4. CUCM and Cisco Unity Connection (CUC) PC GUI Access a. Microsoft Internet Explorer version 8.0.6001.18702 5. Cisco Phones a. Cisco 7961 b. Software Version: SCCP41.9-3-1-1S 6. Cisco 2432-24FXS IAD a. Software Version: c2430-mz.xo 7. Cisco Catalyst 3560 PoE series P-24 a. Software Version: c3560-advipservicesk9-mz.122-44.se2.bin 8. SIP Trunking PBXs Approved for Use a. Cisco: 2811 CME, 2821 CME, Cisco UC520 b. Digium IP PBX c. Avaya: IPO 406, IPO 500 4

4. Test Configurations 4.1. Lab Configuration: CUCM with CUBE running on a Cisco 2821 ISR and the Cisco 2432-24FXS IAD Acting as a Router The diagram below shows the configuration used during lab testing. The CUCM does not support fax directly. External devices such as a Cisco 2800 or 3800 series ISR running CUBE software, a Cisco 2900 or 3900 Series ISR G2 running CUBE software, a Cisco ATA187, or a CUCM SIP trunk to a Right FAX server can be used. The lab tested FAX pass through and T.38 FAX using a Cisco ATA187. No FAX or modem testing was performed on the Cisco 2821 ISR running CUBE software. XO VoIP Network XO Demarc Router Cisco 2821 ISR (CUBE) CUCM (MCS7825H3) Cisco Catalyst 3560 Series PoE-24 Switch TN#5 FAX FXS Port1 TN#6 Analog Phone FXS Port2 CUC (MCS7825H3) Cisco ATA187 TN#1 Cisco IP Phone 7961 TN#2 Cisco IP Phone 7961 TN#3 Cisco IP Phone 7961 Note: Above lab setup only shows main lab network elements. 5

5. Test Bed Configuration Files: CUCM Configuration Details 5.1 CUCM Configuration Details 5.1.1 The following sections contain configuration information that is useful in configuring the CUCM and CUBE. 5.2 CUCM Configuration Details 5.2.1 CUCM Region and Device Pool Configuration Description 5.3 The following sections contain a brief description of the CUCM region and device pool configurations used during testing. 5.3.1 CUCM Region and Device Pool Configuration for SP1 with CUBE 5.4 In this setup there is only one region called the default region and it is configured to use G.711 codec. The device pool used is the default device pool. All CUCM phones and the SIP trunk are assigned to the default device pool and use the G.711 codec to communicate over the default region and over the SIP trunk. 5.4.1 CUCM Region Screen Capture for SP1 with CUBE CUCM Region Information Screen Capture for SP1 with CUBE 6

5.5 CUCM Device Pool Screen Capture for SP1 with CUBE Default Device Pool Configuration Screen Capture Part 1 7

Default Device Pool Configuration Screen Capture Part 2 8

Default Device Pool Configuration Screen Capture Part 3 9

5.6. CUCM Region and Device Pool Configuration for SP2 with CUBE In this setup there are two regions where one is the default region and the other is named the XO g729 region. The default region is configured to use the G.729 codec. The XO g729 region is configured to use the G.729 codec. All CUCM phones are configured in the default device pool and the XO SIP trunk is configured in the XO SIP Trunk Device Pool. All phones communicate with each other over the default region using G.711 or G.722 the largest codecs available. When the phones establish an outbound call or receive an inbound call over the SIP trunk these calls are processed over the XO g729 region configured for G.729 codec. On the CUCM a single SIP trunk can only support one codec. This means that all calls whether they are inbound or outbound over the SIP trunk will establish using only the G.729 codec. 5.6.1. CUCM Region Screen Capture for SP2 with CUBE XO G.729 Region Information Screen Capture for SP2 with CUBE 10

5.7. CUCM Device Pool Screen Capture for SP2 with CUBE G.729 SIP Trunk Device Pool Information Screen Capture Part 1 11

G.729 SIP Trunk Device Pool Information Screen Capture Part 2 12

G.729 SIP Trunk Device Pool Screen Capture Part 3 13

5.8.1. CUCM SIP Profile and SIP Trunk Configuration Screen Captures 5.8.1.1. CUCM SIP Profile Configuration Screen Capture This section contains the SIP Profile used during service package 1 and 2 testing. SIP Profile Configuration Screen Capture Part 1 14

SIP Profile Configuration Screen Capture Part 2 15

SIP Profile Configuration Screen Capture Part 3 16

SIP Profile Configuration Screen Capture Part 4 17

5.9. CUCM SIP Trunk Security Profile Configuration 5.9.1. This section contains the SIP Trunk Security Profile used during SP1 and SP2 testing. CUCM SIP Trunk Security Profile Screen Capture 18

5.10. CUCM SIP Trunk Screen Captures for SP1 This section contains the SIP trunk settings used during SP1 testing. Please note that the Media Termination Point Required box is not checked. This allows CUBE to perform the Early Offer/Delayed Offer (EO/DO) conversion. The screen captures in this section use a 4 digit phone extension. Within the Call Routing Information, the Inbound Calls section has the Significant Digits* set to 4 and the option for Redirecting Diversion Header Delivery - Inbound is checked. Within the Outbound Calls section the Calling Party Selection* is set to Originator and the Caller ID DN is left blank because CUBE is configured to add the NPA-NXX via a SIP profile rule. The option for Redirecting Diversion Header Delivery - Outbound is checked. In the CUCM with CUBE SIP Trunk Screen Capture Part 3 under the SIP Information parameters the Destination Address must be set to CUBE's IP address. CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. CUCM with CUBE SIP Trunk Configuration Screen Capture Part 1 19

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 2 20

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 3 21

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 4 22

5.11. CUCM SIP Trunk Screen Captures with CLID Blocked for SP1 The screen captures in this section show the SIP trunk configuration settings where the caller ID is blocked by using a separate route pattern. In the CUCM with CUBE CLID Blocked SIP Trunk Screen Capture Part 3 under the Outbound Calls section, the Calling Line ID Presentation* and the Calling Name Presentation* fields are set to Restricted. This SIP trunk will block the caller ID for all outbound calls. In the same screen capture under the SIP Information parameters the Destination Address must be set to CUBE's IP address. CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 1 23

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 2 24

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 3 25

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 4 26

5.12. CUCM SIP Trunk Screen Captures for SP2 This section contains the SIP trunk settings used during SP2 testing. Please note that the Media Termination Point Required box is not checked. This allows CUBE to perform the Early Offer/Delayed Offer (EO/DO) conversion. The screen captures in this section use a 4 digit phone extension. Within the Call Routing Information, the Inbound Calls section has the Significant Digits* set to 4 and the option for Redirecting Diversion Header Delivery - Inbound is checked. Within the Outbound Calls section the Calling Party Selection* is set to Originator and the Caller ID DN is left blank because CUBE is configured to add the NPA-NXX via a SIP profile rule. The option for Redirecting Diversion Header Delivery - Outbound is checked. In the CUCM with CUBE SIP Trunk Screen Capture Part 3 under the SIP Information parameters the Destination Address must be set to CUBE's IP address. CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. CUCM with CUBE SIP Trunk Configuration Screen Capture Part 1 27

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 2 28

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 3 29

CUCM with CUBE SIP Trunk Configuration Screen Capture Part 4 30

5.13. CUCM SIP Trunk Screen Captures with CLID Blocked for SP2 The screen captures in this section show the SIP trunk configuration settings where the caller ID is blocked by using a separate route pattern. This SIP trunk will block the caller ID for all outbound calls. In the CUCM with CUBE CLID Blocked SIP Trunk Screen Capture Part 3 under the Outbound Calls section, the Calling Line ID Presentation* and the Calling Name Presentation* fields are set to Restricted. In the same screen capture under the SIP Information parameters the Destination Address must be set to CUBE's IP address. CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 1 31

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 2 32

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 3 33

CUCM with CUBE CLID Blocked SIP Trunk Configuration Screen Capture Part 4 34

5.14. CUCM Phone Configuration Using a Four Digit Extension CUCM Phone Screen Configuration Using a Four Digit Extension 35

5.15. CUCM MoH Server Codec Selection When configuring the customer for SP1 or SP2, the MoH Server Codec setting under CUCM administration, Service Parameters, Cisco IP Voice Media Streaming Application, Clusterwide Parameters, the codec selection displayed under Supported MoH Codecs must be set to G.711ulaw for SP1 and G.729 for SP2 accordingly. G.711ulaw is the system default. The codec must be selected and saved as shown in the screen captures below. The highlighted codec indicates the codec that is currently in use. MoH Server Codec Screen Capture Part 1 36

MoH Server Codec Screen Capture Part 2 37

5.16. CUCM Enterprise Parameters: DSCP Bit Settings for Signaling and Media The screen capture below shows the DSCP bit settings for the CUCM phones for signaling and media. The DSCP for Cisco CallManager to Device Interface* parameter is the signaling setting which is AF31 and the DSCP for Phone Configuration* parameter is the media setting which is EF. The CUCM server must be rebooted for these changes to take effect. 38

5.17. CUCM Conference Bridge Configuration for SP1 The screen capture below shows the CUCM conference bridge resource parameters listed under CUCM Administration, Media Resources for the default CUCM conference bridge. The CUCM software conference bridge was used to verify SP1 three and four way conference bridging test cases. This is a software resource that runs on the CUCM server itself versus the external conference bridge resource for SP2 which must run on CUBE due to the G.729 codec requirement discussed in the next section. This screen can also be used to verify that the CUCM conference bridge resource is registered by checking the registration state of the device. 39

5.18. CUCM Conference Bridge Configuration for SP2 The screen capture below shows the Conference Bridge resource parameters listed under CUCM Administration, Media Resources. An external conference bridge resource is configured using CUBE which registers with the CUCM. This conference bridge resource was used in testing SP2 conference bridging for three and four way conference calls. When the CUCM is configured for SP1 which uses the G.711 codec, the CUCM uses its own software conference bridge. However when the CUCM is configured for SP2 which uses the G.729 codec, the conference bridge must be supported on an external device such as CUBE. This screen can also be used to verify that the CUBE conference bridge resource is registered with the CUCM by checking the registration state. 40

6. CUBE Configuration Details 6.1. The following sections provide a brief description regarding CUBE s flow-through versus flow-around modes for media and which was used during lab testing. 6.1.1. Correcting the CUCM Diversion Header Problem With CUBE When Using 4 Digit CUCM Phone Extensions 6.1.2. A diversion header problem was discovered where the NPA-NXX is not prefixed to the four digit extension number for the user portion of the diversion header. This problem affected PSTN-to-PSTN call transfers that use Call Forward Always (CFA), Call Forward On Busy (CFOB), and Call Forward No Answer (CFNA) where a four digit extension is used for the CUCM phones because the original caller ID is not delivered to the final PSTN destination. When the CUCM is configured to use CUBE running on a Cisco 2821 ISR, a SIP profile can be used to modify the diversion header to correct this problem. For the lab scenario a SIP profile was written for CUBE to modify the diversion header by adding the NPA-NXX to the 4 digit extension in the user portion of the diversion header. What follows is the sip profile rule used during lab testing: 6.1.3. voice class sip-profiles 1 6.1.4. request INVITE sip-header Diversion modify "<sip:(.*)@201.1.3.254>" "<sip:469387\1@201.1.3.254>" 6.1.5. Please note that a customer CUCM environment may have several NPA- NXXs or other phone extension digit configurations that will require different SIP profile rules and translation rules to be designed using CUBE that are unique to each customer environment. 6.1.6. CUBE SIP Profile Rules to Correct MoH Issue 6.1.7. These configuration commands are still required in the CUBE sip profile section of the configuration for the remote party to hear MoH when the call is placed on hold from a CUCM phone. 6.1.8. CUBE SIP Profile Rules for MoH to Work Properly: 6.1.9. voice class sip-profiles 1 6.1.10. request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv" 6.1.11. request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv" 6.1.12. response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv" 6.1.13. CUBE DSCP Bit Settings for Signaling and Media 6.1.14. The CUBE voip dial peer default DSCP bit settings for signaling is AF31 and for media is EF. These settings do not appear in the configuration when a show configuration is executed at the CLI prompt. However, the settings can be displayed by executing the command show dial-peer voice # at the CLI prompt for each voip dial peer where the # represents the dial peer number. 41

6.1.15. Configuring the CUCM and Cisco ATA187 for FAX Pass Through or T.38 FAX Relay 6.2. This section provides CUCM and ATA187 configuration information for configuring the CUCM and the Cisco ATA187 for FAX pass through and T.38 FAX relay. 6.2.1. Configuring the CUCM to Recognize the Cisco ATA187 6.3. Using the Cisco Unified CM Administration, under Device, Phone then click on the Add New button. Under Phone Type* select ATA187 then click next. Enter the MAC Address*, Description and Device Pool* information. In the Product Specific Configuration Layout section, set the Fax Mode* field accordingly and save the configuration. Cisco ATA187 Device Information Configuration Screen Capture 1 42

Cisco ATA187 Device Information Configuration Screen Capture 2 43

6.4. Configuring the Cisco ATA187 Using the Cisco provided instructions for the ATA187, configure the device such that it shows up in the Cisco Unified CM Administration under Device, Phones as registered with the CUCM as shown in the first line item entry under Device Name ATAF02572784B29 in the screen capture below. 44

6.5. Configuring the Cisco ATA187 for FAX Pass Through In the CUCM screen capture above, click on the ATA187 in question, then scroll down to the Product Specific Configuration Layout section and set the Fax Mode* option to Fax Pass-through as shown in the screen capture below and save. No other settings were modified. ATA187 Fax Pass Through Screen Capture 45

6.6. Configuring the Cisco ATA187 for T.38 FAX Relay In the CUCM screen capture above, click on the ATA187 in question, then scroll down to the Product Specific Configuration Layout section and set the Fax Mode* option to T.38 Fax Relay as shown in the screen capture below and save. No other settings were modified. ATA187 Fax Relay Screen Capture 6.6.1. CUBE FAX Pass Through and T.38 FAX Relay Commands This section contains the FAX specific commands that are required in the CUBE configuration for FAX Pass Through and T.38 FAX Relay to work properly. For FAX pass through, under the voice service voip section of the configuration, add the command fax protocol pass-through g711ulaw. For FAX relay under the voice service voip section of the configuration, add the command fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none. 46

7. Equipment Configuration Files and Information This section contains the Cisco 2821 ISR running CUBE, Cisco 3560 PoE, and Cisco 2432-24FXS IAD and Juniper 1400 ERX configuration files used during testing. It also contains a BroadSoft and Sonus Networks PSX screen capture for settings that need to be checked for CUCM with CUBE to work properly in production. 7.1. Cisco 2821 ISR Running CUBE Software The following sections contain the CUBE test configuration used during SP1 and SP2 testing. 7.1.1. CUBE Configuration File for Service Package 1 CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. C2821CUBE#wr t Building configuration... Current configuration : 2982 bytes Last configuration change at 18:42:51 UTC Mon Apr 8 2013 version 15.1 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption hostname C2821CUBE boot-start-marker boot-end-marker no logging buffered enable secret 5 $1$dgKR$TQ5lvonJvCjRAWx7.cyIX0 no aaa new-model dot11 syslog ip source-route ip cef 47

no ip domain lookup no ipv6 cef multilink bundle-name authenticated voice service voip ip address trusted list ipv4 205.158.163.0 255.255.255.0 allow-connections sip to sip no supplementary-service sip moved-temporarily redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback passthrough g711ulaw no fax-relay sg3-to-g3 sip header-passing error-passthru early-offer forced midcall-signaling passthru sip-profiles 1 voice class media 1 voice class codec 1 codec preference 1 g711ulaw voice class sip-profiles 1 request INVITE sip-header Diversion modify "<sip:(.*)@201.1.3.254>" "<sip:469387\1@201.1.3.254>" request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv" request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv" response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv" voice translation-rule 1 rule 1 /^\(...\)$/ /469387\1/ 48

voice translation-profile Sonus_NBS_Outgoing translate calling 1 voice-card 0 dsp services dspfarm crypto pki token default removal timeout 0 license udi pid CISCO2821 sn FHK1023F1J8 archive log config hidekeys redundancy interface GigabitEthernet0/0 description Connection to CUCM3560 port 23 ip address 201.1.3.252 255.255.255.0 duplex auto speed auto interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto no ip forward-protocol nd no ip http server no ip http secure-server 49

ip route 0.0.0.0 0.0.0.0 201.1.3.1 control-plane mgcp fax t38 ecm mgcp profile default dial-peer voice 10 voip description Outgoing dial-peer to Sonus NBS translation-profile outgoing Sonus_NBS_Outgoing destination-pattern.t session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip profiles 1 dtmf-relay rtp-nte no vad dial-peer voice 11 voip description Outgoing dial-peer to CUCM destination-pattern 46938739.. session protocol sipv2 session target ipv4:201.1.3.254 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced dtmf-relay rtp-nte no vad sip-ua sip-server ipv4:205.158.163.138 50

banner login ^C ** Welcome to the Cisco2821 running CUBE **^C line con 0 session-timeout 120 exec-timeout 60 0 absolute-timeout 180 flowcontrol hardware line aux 0 line vty 0 4 exec-timeout 3600 0 password labbgp1 login transport input all scheduler allocate 20000 1000 end C2821CUBE# 7.1.2. CUBE Configuration File for Service Package 2 CUBE's sip-server address must be set to the Sonus Networks NBS signaling IP address. C2821CUBE#wr t Building configuration... Current configuration : 3378 bytes Last configuration change at 19:01:56 UTC Mon Apr 8 2013 version 15.1 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption hostname C2821CUBE boot-start-marker boot-end-marker no logging buffered enable secret 5 $1$dgKR$TQ5lvonJvCjRAWx7.cyIX0 51

no aaa new-model dot11 syslog ip source-route ip cef no ip domain lookup no ipv6 cef multilink bundle-name authenticated voice service voip ip address trusted list ipv4 205.158.163.0 255.255.255.0 allow-connections sip to sip no supplementary-service sip moved-temporarily redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback passthrough g711ulaw no fax-relay sg3-to-g3 sip header-passing error-passthru early-offer forced midcall-signaling passthru sip-profiles 1 voice class media 1 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class sip-profiles 1 52

request INVITE sip-header Diversion modify "<sip:(.*)@201.1.3.254>" "<sip:469387\1@201.1.3.254>" request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv" request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv" response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv" voice translation-rule 1 rule 1 /^\(...\)$/ /469387\1/ voice translation-profile Sonus_NBS_Outgoing translate calling 1 voice-card 0 dsp services dspfarm crypto pki token default removal timeout 0 license udi pid CISCO2821 sn FHK1023F1J8 archive log config hidekeys redundancy interface GigabitEthernet0/0 description Connection to CUCM3560 port 23 ip address 201.1.3.252 255.255.255.0 duplex auto speed auto 53

interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto no ip forward-protocol nd no ip http server no ip http secure-server ip route 0.0.0.0 0.0.0.0 201.1.3.1 control-plane mgcp fax t38 ecm mgcp profile default sccp local GigabitEthernet0/0 sccp ccm 201.1.3.254 identifier 1 version 7.0 sccp sccp ccm group 1 bind interface GigabitEthernet0/0 associate ccm 1 priority 1 associate profile 3 register CFB0018191A4CB0 dspfarm profile 3 conference codec g729ar8 codec g729r8 codec g711ulaw maximum sessions 8 associate application SCCP dial-peer voice 10 voip description Outgoing dial-peer to Sonus NBS 54

translation-profile outgoing Sonus_NBS_Outgoing destination-pattern.t session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip profiles 1 dtmf-relay rtp-nte no vad dial-peer voice 11 voip description Outgoing dial-peer to CUCM destination-pattern 46938739.. session protocol sipv2 session target ipv4:201.1.3.254 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip g729 annexb-all voice-class sip early-offer forced dtmf-relay rtp-nte no vad sip-ua sip-server ipv4:205.158.163.138 banner login ^C ** Welcome to the Cisco2821 running CUBE **^C line con 0 session-timeout 120 exec-timeout 60 0 absolute-timeout 180 flowcontrol hardware line aux 0 line vty 0 4 exec-timeout 3600 0 password labbgp1 login transport input all scheduler allocate 20000 1000 end 55

C2821CUBE# 7.2. Cisco 3560 PoE Configuration File This file contains the device configuration used during the evaluation. CUCM3560#wr t Building configuration... Current configuration : 3963 bytes version 12.2 no service pad service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption hostname CUCM3560 boot-start-marker boot-end-marker enable secret 5 $1$T/F4$rIWfQ68OVzIO7yXqFxU0z1 no aaa new-model system mtu routing 1998 vtp mode transparent ip subnet-zero crypto pki trustpoint TP-self-signed-3561089152 enrollment selfsigned subject-name cn=ios-self-signed-certificate-3561089152 revocation-check none rsakeypair TP-self-signed-3561089152 crypto pki certificate chain TP-self-signed-3561089152 certificate self-signed 01 30820240 308201A9 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 33353631 30383931 3532301E 170D3933 30333031 30303030 56

35305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 35363130 38393135 3230819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100A233 FA77DC4C 79C0503F E8067FFD 629619E3 12F8CE94 ACBAB5B9 A7A6787F DF892A30 8B0ADB48 869FF69D 14D3E262 FCC507AD 7FE382C1 0D1C3527 A1248C79 A0E58516 A8AA5928 7F66A66B B11C8C9B 50AE7253 CFFE55C3 58879021 9790C696 97C57B28 8C387F05 0EBBFF92 B630154D 516B9AAE 6127E87C 490C57B6 E294A5A2 F6A70203 010001A3 68306630 0F060355 1D130101 FF040530 030101FF 30130603 551D1104 0C300A82 0843434D 33353630 2E301F06 03551D23 04183016 801405B2 76A81F94 38A596E0 20B6DCE5 9975EF1A 714B301D 0603551D 0E041604 1405B276 A81F9438 A596E020 B6DCE599 75EF1A71 4B300D06 092A8648 86F70D01 01040500 03818100 6E703269 5AC65206 E33DB499 5D0A7BC5 25331F18 DB839656 96D586DE 25BF6112 81585C1E 08A2331F AFE173E1 587BEE7C 4A726698 C3DA2709 C0E36E45 9F7881BE 8E707424 CB5EE28E 446402E2 0217A5FD F25D5D0D F0A045C0 AD8D8305 79B1C9E5 7DBF1E6B 4AA13B9A 75DF2E79 8A948C0C F8A65AEA 8E2E105C 961ABD0F FFB8A470 quit spanning-tree mode pvst spanning-tree extend system-id vlan internal allocation policy ascending vlan 10,99-101,200 57

interface FastEthernet0/1 description Connection to Cisco Unified Call Manager switchport access vlan 10 switchport mode access speed 100 duplex full interface FastEthernet0/2 description Connection to Cisco Unity Connection Voice Mail switchport access vlan 10 switchport mode access speed 100 duplex full interface FastEthernet0/3 switchport access vlan 10 switchport mode access interface FastEthernet0/4 switchport access vlan 10 switchport mode access interface FastEthernet0/5 switchport access vlan 10 switchport mode access interface FastEthernet0/6 switchport access vlan 10 switchport mode access interface FastEthernet0/7 switchport access vlan 10 switchport mode access interface FastEthernet0/8 switchport access vlan 10 switchport mode access interface FastEthernet0/9 switchport access vlan 10 switchport mode access interface FastEthernet0/10 switchport access vlan 10 switchport mode access 58

interface FastEthernet0/11 switchport access vlan 10 switchport mode access interface FastEthernet0/12 interface FastEthernet0/13 interface FastEthernet0/14 interface FastEthernet0/15 interface FastEthernet0/16 interface FastEthernet0/17 interface FastEthernet0/18 interface FastEthernet0/19 interface FastEthernet0/20 interface FastEthernet0/21 interface FastEthernet0/22 interface FastEthernet0/23 interface FastEthernet0/24 switchport access vlan 10 interface GigabitEthernet0/1 interface GigabitEthernet0/2 interface Vlan1 no ip address shutdown interface Vlan10 description CUCM3560 Telnet Connection ip address 201.1.3.249 255.255.255.0 ip classless ip http server ip http secure-server 59

control-plane banner login ^C ** Welcome to the CUCM 3560PoE **^C line con 0 line vty 0 4 password labbgp1 login line vty 5 15 password cisco login end CUCM3560# 60