Cisco ISDN PRI to SIP Gateway



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

Cisco IOS SIP Configuration Guide

Dial Peer Configuration Examples

Configuring Network Side ISDN PRI Signaling, Trunking, and Switching

Cisco CCA Tool SIP Security methods

Cisco CME SIP Trunk Configuration

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

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

nexvortex Setup Guide

Voice Dial Plans, Configuring Voice Interfaces and Dial Peers

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

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

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

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

Configuring SIP Registration Proxy on Cisco UBE

Call Setup and Digit Manipulation

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

Q&A. DEMO Version

Configuring Fax Pass-Through

VoIP Gateway/IP-PBX Interworking with Skype

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

Quick Installation Guide

Dial Peer. Example: Dial-Peer Configuration

Mediatrix 3000 with Asterisk June 22, 2011

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

Configuring Modem Transport Support for VoIP

Curso de Telefonía IP para el MTC. Sesión 5-1 Implementación de Gateways SIP. Mg. Antonio Ocampo Zúñiga

Cisco Voice over IP

Curso de Telefonía IP para el MTC. Sesión 4-1 Tipos de llamadas. Mg. Antonio Ocampo Zúñiga

Direct Inward Dial Digit Translation Service

Configuring Voice over IP

Voice Call Flow Overview

Avaya IP Office 8.1 Configuration Guide

This topic describes dial peers and their applications.

How To Set Up A Dialogic.Com On A Cell Phone With A Sim Sim Sims On A Sims 2 (For A Simplon) On A Pts 2 ( For A Pty Phone) On An Ipad Or

Mediatrix Gateway 440x Series Quick Configuration Guide

VoIP Configuration Examples

Quality of Service and Bandwidth Management Configuration

Connecting to IP Gateways/PBXs example of a CISCO Gateway

- Basic Voice over IP -

System Components PBX Model. Configuration Tasks

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

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

Peer-to-Peer SIP Mode with FXS and FXO Gateways

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

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

Technical Configuration Notes

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

VoIP Workshop PacNOG3

NodePhone Business Trunks User Manual

Configuration guide for Switchvox and Cbeyond.

CVOICE Exam Topics Cisco Voice over IP Exam # /14/2005

SIP Trunking with Elastix. Configuration Guide for Matrix SETU VTEP

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

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

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

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions

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

NTP VoIP Platform: A SIP VoIP Platform and Its Services

CVOICE - Cisco Voice Over IP

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Application Notes Rev. 1.0 Last Updated: January 9, 2015

nexvortex Setup Guide

Connecting with Vonage

VoIP Configuration. Prerequisite Tasks CHAPTER

Configuring T.38 Fax Relay

Cisco PBX Interoperability: Lucent/Avaya Definity G3si V7 PBX with CallManager using Analog FXS and FXO Interfaces as an MGCP Gateway

Configuring Voice over IP for Cisco MC3810 Series Concentrators

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

First, we must set the Local Dialing Type to 7 or 10 digits. To do this, begin by clicking on Dial Plan under Voice.

Configuring Avaya BCM 6.0 for Spitfire SIP Trunks

IP Office Technical Tip

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

Building a Scalable Numbering Plan

Cisco Unified Communications 500 Series

Vega 100G and Vega 200G Gamma Config Guide

Optimizing Converged Cisco Networks (ONT)

Visocall IP PBX Connection

Link2VoIP SIP Trunk Setup

Case Study 1: Registering IP Phones with a remote Call

Configuring Positron s V114 as a VoIP gateway for a 3cx system

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

Vega 100G and Vega 200G Gamma Config Guide

AT&T Service Tested. Interaction Center (xic)

Configuration Notes 290

Configuring SIP Trunking and Networking for the NetVanta 7000 Series

Special-Purpose Connections

Configuring Voice over IP

Application Note. December 2014 Table of Contents

nexvortex Setup Template

Allo PRI Gateway and Elastix Server

System Components PBX Model. Configuration Tasks

OfficeServ 7100 IP-PBX. SIP Trunking using the Optimum Business Sip Trunk Adaptor and the Samsung

Cisco IOS Voice Gateway PBX Interoperability: Avaya 8500 Communications Manager 2.1 to T1 QSIG with H.323

Voice Over IP. Priscilla Oppenheimer

Configuring SIP Support for SRTP

L2F Case Study Overview

Cisco 1751 Voice-over-IP Quick Start Guide

2N NetStar. 2N NetStar as a PBX Booster. Quick guide. Version 1.00

Configuring Analog and Digital Voice Interfaces

Transcription:

Cisco ISDN PRI to SIP Gateway Supported features Full ISDN E1 emulation Early media support Inbound calling. Type SIP REGISTERED TRUNK Outbound Calling ISDN PRI equivalent Secure Calling via SIP Encrypt platform available Cisco Platforms ratified: Cisco 2800, 3800 Cisco 2900, 3900 Cisco AS5300 Cisco AS5350XM Cisco AS5400 Cisco AS5400XM Cisco UC560 T1/E1 varient (Single E1 only) ** Cisco router chassis requires PVDM resource, voice enabled E1 cards and voice feature enabled software / licence. UC560-T1/E1 is a Small Business PBX and contains a PVDM2-64 and a single E1 card. This device has all the components needed for an ISDN to SIP Gateway, (or as a voice gateway for the BE6000). E1 presentation of Cisco E1 line cards is wired as CPE side. To emulate the ISDN Network at Layer1 use an E1 cross over cable made by swapping pins 1,4 and 2,5 Example configuration snippits were configured on a 2821 router with x2 VWIC2-2MFT-T1/E1 and x3 PVDM2-64. Recommended IOS version is 12.4(24)T3 and later.

Step one: Configure VoIP.co.uk portal for inbound calling type: SIP REGISTERED TRUNK Create a SIP account for the Cisco router. A SIP Account is a username / password pair which a SIP phone / endpoint uses to authenticate itself. 1. On the my.voip.co.uk portal, click on SIP-AOR and create a SIP account for the Cisco router. Give the SIP account a meaningful name like My Cisco gateway. Also make sure the password is complex. 2. Click on dashboard and then Incoming targets. Create a New Incoming Target; TYPE: SIP_REGISTERED TRUNK and again call the incoming target something meaningful such as Route calls to my Cisco gateway. It is recommended that the alphanumeric characters AAA are pre-pended to an incoming call to facilitate call routing on the ISDN/SIP gateway. The Registered Trunk should contain the user template: AAA${e164} 3. Click on the new incoming target and add in the new SIP account you just created. 4. Click on Dashboard / Phone numbers and configure a telephone number to route calls to your new incoming target. Step Two. Setup E1 cards 1. Most Cisco E1 cards are T1/E1 capable and they need to be set for E1 (Europe) card type e1 0 0 card type e1 0 1 network-clock-participate wic 0 network-clock-participate wic 1 network-clock-select 5 E1 0/0/0 network-clock-select 6 E1 0/0/1

Step Three. Create 30 channels on each E1 controller E1 0/0/0 pri-group timeslots 1-31 controller E1 0/0/1 pri-group timeslots 1-31 controller E1 0/1/0 pri-group timeslots 1-31 controller E1 0/1/1 pri-group timeslots 1-31 interface Serial0/0/0:15 description use dial-peer voice 10 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn timer T309 400000 isdn protocol-emulate network isdn incoming-voice modem isdn send-alerting isdn sending-complete no cdp enable interface Serial0/0/1:15 description use dial-peer voice 11 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn timer T309 400000 isdn protocol-emulate network isdn incoming-voice modem isdn send-alerting isdn sending-complete no cdp enable interface Serial0/1/0:15 description use dial-peer voice 12 no ip address encapsulation hdlc

isdn switch-type primary-net5 isdn timer T309 400000 isdn protocol-emulate network isdn incoming-voice modem isdn send-alerting isdn sending-complete no cdp enable interface Serial0/1/1:15 description use dial-peer voice 13 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn timer T309 400000 isdn protocol-emulate network isdn incoming-voice modem isdn send-alerting isdn sending-complete no cdp enable Step 4 Configure region cadence voice-port 0/0/0:15 cptone GB bearer-cap Speech voice-port 0/1/0:15 cptone GB bearer-cap Speech voice-port 0/0/1:15 cptone GB bearer-cap Speech voice-port 0/1/1:15 cptone GB bearer-cap Speech

Step 5 Configure router with Basic SIP settings sip-ua authentication username <SIP username created in Step 1> password <SIP password in step 1> credentials username <SIP username created in Step 1> password <SIP password in step 1> realm proxy.voip.co.uk retry invite 2 registrar dns:proxy.voip.co.uk expires 120 sip-server dns:proxy.voip.co.uk voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 Step 6 ISDN to SIP call routing Inbound POTS dial-peer to accept calls from ISDN dial-peer voice 9 pots description incoming pots dial-peer translation-profile incoming isdn-to-voip incoming called-number.+ direct-inward-dial Outbound VoIP dial-peer to route calls to VoIP.co.uk dial-peer voice 1 voip description Outbound SIP destination-pattern ^ T voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte fax-relay ecm disable fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 1 fallback pass-through g711alaw

no vad Step 7 SIP to ISDN Inbound VoIP dial-peer to accept calls from VoIP.co.uk dial-peer voice 997 voip description incoming voip rtp payload-type cisco-codec-fax-ind 124 voice-class codec 1 session protocol sipv2 session target sip-server incoming called-number ^AAA.+ dtmf-relay rtp-nte fax-relay ecm disable fax nsf 000000 fax protocol t38 ls-redundancy 3 hs-redundancy 1 fallback pass-through g711alaw no vad Outbound POTS dial-peer to route calls to the ISDN PBX: dial-peer voice 10 pots description POTS talking dial peer for E1 #0 translation-profile outgoing voip-to-isdn preference 1 destination-pattern ^AAA.+ port 0/0/0:15 dial-peer voice 11 pots description POTS talking dial peer for E1 #1 translation-profile outgoing voip-to-isdn preference 1 destination-pattern ^AAA.+ port 0/0/1:15 dial-peer voice 12 pots description POTS talking dial peer for E1 #2 translation-profile outgoing voip-to-isdn preference 1

destination-pattern ^AAA.+ port 0/1/0:15 dial-peer voice 13 pots description POTS talking dial peer for E1 #3 translation-profile outgoing voip-to-isdn preference 1 destination-pattern ^AAA.+ port 0/1/1:15 IOS 15.0 and later (Toll fraud feature) voice service voip ip address trusted list ipv4 193.203.210.0 255.255.254.0 Step 8 Set up number translations voice translation-rule 501 rule 1 /^44\(.+\)$/ /\1/ type any national plan any isdn rule 3 /^\+44\(.+\)$/ /\1/ type any national plan any isdn rule 10 /\(.+\)/ /0\1/ type any national plan any isdn voice translation-rule 601 rule 1 /^AAA44\(.+\)$/ /\1/ type any national plan any isdn voice translation-profile voip-to-isdn translate calling 501 translate called 601 Note: translation-rule 601 presents the 10 digit phone number to the PBX. It is normal in the UK for BT to present either the full number minus the leading zero (cities) or the trailing 6 or 4 digits. Example: Phone number AAA1869222500 arrives on the gateway TDM_Gateway#test voice translation-rule 601 AAA441869222500 Matched with rule 1 Original number: AAA441869222500 Translated number: 1869222500

Original number type: none Original number plan: none TDM_Gateway# Translated number type: national Translated number plan: isdn To present the trailing six digits, rule 601 would be re-written: voice translation-rule 601 rule 1 /^AAA441869\(.+\)$/ /\1/ type any national plan any isdn Step 9 Test Test inbound and outbound telephony. Check: DTMF operation Early Media Rinback tones ring ring Some older phone systems required an ISDN alerting message to be sent by the gateway. Unless specifically enabled the Ringback tone may be not heard by the PBX user when dialling out. voice call send-alert