Hosted PBX use case for Service Providers

Size: px
Start display at page:

Download "Hosted PBX use case for Service Providers"

Transcription

1 Hosted PBX use case for Service Providers This use case will show how to implement Hosted PBX Services for Service providers using an Access SBC in Front of the Hosted PBX platform and an SMB SBC at customer premises to provide Security, Transcoding and Local Survivability to existing endpoints (IP Phones). The case can be used also as a sample for Large enterprises with Centralized IPPBX (Data center) and remote branch offices. We will also explain how to enable remote users roaming on Public Networks in a secure manner (Smart Phones, Home Offices, Employees on the field...) The following diagram shows the scenario: As we can see: All external Voice traffic is secure and encrypted (TLS/SRTP) All Internal Voice traffic is standard (SIP/UDP) We will show transcoding between GSM and G711 for Smartphone users Users in Branch office / Customer office will have local survivability, using the local SMB-SBC as a secondary registrar. In more detail here are the specifications of what we will configure in the Access SBC:

2 Let's Start!!!!! First we will create 2 Sip Profiles in our Access SBC, one internal, in the Private Subnet, facing the IPPBX/Softswitch located at The External Sip ( ) profile will be used to receive upper registrations for end points coming from branch offices, via the local SMB-SBC. Note the following attributes: We are going to use a non standard listening port for UDP and TCP (15060). We will see also that we will use for TLS. The only reason we are enabling TCP and UDP in addition to TLS is for testing purposes. We recommend to change to TLS only once everything has been tested and we are going on production. The Internal Sip ( ) profile, listening on standard port will face the Softswicth/IPPBX Now, lest see what is needed in the External Sip Profile: First we need to associate a domain ( mypbx.ddns.net) for upper registration:

3 As we can see: Forward registration is enabled to be able to forward registrar requests to the IPPBX/Softswitch. IPPBX is in standar listening port at on the private network We have enforced 60 seconds expires to be able to quickly detect lost of service Transport between SBC and IPPBX on the internal network will be UDP Now our External SIP Profile nedds to bind the new domain:

4 At this point is importnt to note that the domaing is a full FQDN alse which resolves the public IP of the SBC in the internet. We will autodiscover that IP by using the prefix "host:" in the Sip Profile. Please note: Note we are using host: for external IP address for signaling and RTP mydemopbx.ddns.net Interop section will have default values, as well as timing section For TLS we will use previously loaded self signed certificates We will enable also SRTP Encryption and enforce it for any inbound invite in this profile.

5 Notice that all authentications are disabled, as it is being delegated to the IPPBX. We are also enabling all Far End NAT Transversal techniques and will aneable Full Idenification in session routing:

6 Notice we have associated a Routing plan called Inbound_to_PBX. Here it is:

7 So any inbound call coming in the External Profile will be routed to the IPPBX ( ) in the Private network via the Internal Profile (Internal_SIP_Interface) Now, we need to configure the Internal Profile, which will deliver calls to the PBX coming from Extensions at customer's premises, and receive calls from the IPPBX going out to those extensions.

8 This profile is alocated at the eth1 interface IP No need to reference any external IP address for translation Transport is UDP Standard port is used Now profiles with non default values:

9 Will accept blind authentication as this interface is never exposed and is in a secure Data center private network Will associated a Routing Plan that we will describe later (Outbound_All_Extensions) Will do a header manipulation previous to routing on calls landing on this profile (Domain_Routing_Header) Now, let look at the Header Manipulation rules: In thie manipulation, as an FQDN is used as the domin, and call will be routed to the external network to the customer on premise SMB-SBC, we need to asure the domain is properly passed. So, we will capture the domain name in a variable to be used on the Routing plan. We will use in this case Basic technique:

10 We will capture in kdomain the value " mydemopbx.ddns.net" if the call is coming from the host at (IPPBX) Now let's see the Routing Plan "Outbound_All_Extensions": Notice: We are assiging kdomain to the channel variable domain_name We are enforcing SRTP on the b leg when the call will be bridged We are enforcing TLS on the bridge application In case you are doing test with UDP or TCP you can comment the current bridge sentence and uncomment the one not enforcing TLS In terms of coec we are only using PCMA and PCMU at this point. So Default media profile is as follows: At this point we have finished configuring the Access SBC at the Data Center or Enterprise central Site. Let's now do the configuration for the local SBC at customer premises. The setup for the Branch office will like like this:

11 As we can see we will be using a local domain in addition to the domain form upper registration to the data center. A total of 3 Sip Profiles will be needed: Internal_Phones: used to upper register extension to the central IPPBX Internal_Phones_Survival: to receive secondary registrar and process Sip requests as a secondary proxy for all local SIP end points. External_to_Carrier: to be used to snd and receive SIP to and from the Central IPPBX (via Access SBC) A Local Domain with local authentication is needed:

12 We will also create the USer Credential Information. In this case, just for two extensions (505 and 506)

13 Now we will show configuration for each sip profile: Internal_Phones Profile:

14 Notice: We will do a header manipulation to grant the domain name is passed correctly (Domain_Name_Pass) This internal profile will be associated with a Routing Plan called Outbound_to_Carrier Listening port will be standard Profile will be associated to internal interfce Header Manipulation:

15 Routing Rules: <extension name="from_local_to_pbx"> <condition field="caller_id_number" expression="^5.."> <condition field="destination_number" <action application="export" data="dialed_extension=$1"/> <action application="export" data="sip_secure_media=true"/> <action application="export" data="domain_name=${kdomain}"/> <action application="bridge" 15061;transport=tls"/> </condition> </condition> </extension> Will look like this: Notice: We are enforcing SRTP via the "sip_secure_media=true" We are checking call is coming from an extension via caller_id_number="^5.." <-- this could be just deleted to make it more generic We are enforcing sip domain name in the bridge as well as TLS in Leg B via Externa_to_Carrier profile Now, let's present what to do with the external profile:

16 Transport protocol will be TLS Associated to external interface External public IP address will be announced for signaling and RTP

17 TLS will use a selfsigned certificate SRTP is enabled and enforced for all inbound trafic on this profile

18 NAT is not eneabled except for RTP Adjust to avoid certain cases where audio in one side will not wait for the other side. A Dial plan as been associated for incoming calls to the profile (Inbound_Dialplan) Here we are showing the Inbound dial plan associated to this external profile: At this point Any Endpoint at the Branch Office should be able to register in the central IPPBX and also be able to make calls between extensions. Any VoIP Trafic in the Internet will be tatlly secure with TLS/SRTP TLS and SRTP functionalities are jept at the SBC's levels and no need for any considertion at the IPPBX level or endpoint is needed. Now, we will add one more level of complexity by implementing dual registration at the endpoints in the Branch Office On this excercise we will use SNOM 870 SIP Phones, but it can be implemented in any phone supporting Dual Registration (Polycom, Yealink, AASTRA and Grandstream has been tested. Call for details if needed) So, we will use the secondary domain ( mydemopbxbkp.ddns.net) for the prupose of secondary registrar for the End Points. The way SNOM implements Dual Registration, is by using what they call "Failover identity". So, for example for extension 505 we will create an identity to upper register on the PBX via the local SBC as follows:

19 Notice: Domain in SNOM is communicated in the SIP header via the registrar FQDN, and corresponds to the domain used with the IPPBX /Softswitch We are using the local SBC internal profile as the outbound proxy to be used by the phone. This is identity 1, but we are assigning Identity 2 as the Failover (or secondary registrar) The Failover Identity will look like this:

20 Notice: Domain used is the local domain we created for survivability purposes (mydemopbxbkp.ddns.net Outbound proxy corresponds to a new internal sip profile called (Internal_Phones_Survival) using the listening port associated to this profile (6060) This new profile will have the following configuration:

21

22 Routing rules associated to this profile is the ona named Local_Calls, which will look like this: At this point, you can test by blocking connectivity to the central site and you will notice on the phone that Identity 1 (Extensions 505 and 506) become red to make clear registration have failed. Even so, you can still test calls between the two extensions as the SNOM will start using automatically Identity 2 for the failed Identities. Now are are going to enable users to directly register on the PBX via the access SBC on the central site. This is a typical situation for employees traveling with Softphones on their Laptops or SmartPhones, or even very small braches or home offices, with a few extensions. So, what we will need, as we don't have in this situation an SMB-SMB on premise, is to configure the endpoint to support TLS/SRTP and we will point them directly to the corporate/central SBC for upper registration and SIP requests.

23 No need to make any changes at any of the SBC's as we are going to take advantage of the same domain " mydemopbx.ddns.net" and the External SIP Profile we created for Branch offices. Conceptually this is because from the perspective of the Access SBC there is no difference between and end point coming from a branch (As the local SBC takes the personality of all endpoint behind him), and an isolated end point in the public internet. In both cases the Access SBC will only accept TLS/SRTP traffic. So, let's show how an extension confiuration on the pubic internet will look like, using a SNOM 870, extension 506:

24 Notice: We are using the mydemopbx.ddns.net domain outbound proxy is " mydemopbx.ddns.net:15061;transport=tls", as this FQDN will resolve to the IP Public address for the Access SBC. There is no failover identity defined.

25 Notice: Enabled RTP Encryption At this point you can test between Branch Office extensions and Extension on the Internet. We have acomplished the goals for the use case. Any question, suggestions or comments feel free to me: Enjoy!!!!

Configuration Guide IP-to-IP Application

Configuration Guide IP-to-IP Application Multi-Service Business Gateways Enterprise Session Border Controllers VoIP Media Gateways Configuration Guide IP-to-IP Application Version 6.8 November 2013 Document # LTRT-40004 Configuration Guide Contents

More information

Installation & Configuration Guide Version 4.0

Installation & Configuration Guide Version 4.0 TekSIP Installation & Configuration Guide Version 4.0 Document Revision 6.8 https://www.kaplansoft.com/ TekSIP is built by Yasin KAPLAN Read Readme.txt for last minute changes and updates, which can be

More information

Grandstream Networks, Inc. IPVideoTalk Service Configuration Guide on UCM

Grandstream Networks, Inc. IPVideoTalk Service Configuration Guide on UCM Grandstream Networks, Inc. Table of Contents OVERVIEW... 4 IPVIDEOTALK SERVICE CONFIGURATION ON UCM... 5 Configure SIP Trunk on IPVT10... 5 Configure Grandstream UCM... 5 Configure VoIP Trunk... 5 Configure

More information

SBC Configuration Examples for Mediant SBC

SBC Configuration Examples for Mediant SBC Configuration Note AudioCodes Mediant Series of Session Border Controllers (SBC) SBC Configuration Examples for Mediant SBC Version 7.2 Configuration Note Contents Table of Contents 1 Introduction...

More information

1 SIP Carriers 1.1 CBeyond 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

Microsoft Lync Server 2013 and Twilio SIP Trunk using AudioCodes Mediant E-SBC

Microsoft Lync Server 2013 and Twilio SIP Trunk using AudioCodes Mediant E-SBC Configuration Note AudioCodes Professional Services - Interoperability Lab Microsoft Lync Server 2013 and Twilio SIP Trunk using AudioCodes Mediant E-SBC Version 7.0 Configuration Note Contents Table

More information

1 SIP Carriers. 1.1 LightBound Warnings Vendor Contact Vendor Web Site:

1 SIP Carriers. 1.1 LightBound Warnings Vendor Contact Vendor Web Site: 1 SIP Carriers 1.1 LightBound 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

Acano solution. Third Party Call Control Guide. 07 June G

Acano solution. Third Party Call Control Guide. 07 June G Acano solution Third Party Call Control Guide 07 June 2016 76-1055-01-G Contents 1 Introduction 3 1.1 How to Use this Guide 3 1.1.1 Commands 5 2 Example of Configuring a SIP Trunk to CUCM 6 2.1 Prerequisites

More information

Acano solution. Third Party Call Control Guide. December F

Acano solution. Third Party Call Control Guide. December F Acano solution Third Party Call Control Guide December 2015 76-1055-01-F Contents Contents 1 Introduction... 3 1.1 How to Use this Guide... 3 1.1.1 Commands... 4 2 Example of Configuring a SIP Trunk to

More information

Application Note 3Com VCX Connect with SIP Trunking - Configuration Guide

Application Note 3Com VCX Connect with SIP Trunking - Configuration Guide Application Note 3Com VCX Connect with SIP Trunking - Configuration Guide 28 May 2009 3Com VCX Connect Solution SIP Trunking Table of Contents 1 3COM VCX CONNECT AND INGATE... 1 1.1 SIP TRUNKING SUPPORT...

More information

NEC: SIP Trunking Configuration Guide V.1

NEC: SIP Trunking Configuration Guide V.1 NEC: SIP Trunking Configuration Guide V.1 FOR MORE INFO VISIT: CALL US EMAIL US intermedia.net +1.800.379.7729 sales@intermedia.net 2 NEC: SIP Trunking Configuration Guide V.1 TABLE OF CONTENTS Introduction...

More information

One-Voice Resiliency with SIP Trunking

One-Voice Resiliency with SIP Trunking Configuration Note AudioCodes One Voice for Skype For Business One-Voice Resiliency with SIP Trunking For Branch Sites in Microsoft Lync Server or Skype for Business Environments Version 7.2 Configuration

More information

Unified Communications in RealPresence Access Director System Environments

Unified Communications in RealPresence Access Director System Environments [Type the document title] 2.1.0 March 2013 3725-78704-001A Deploying Polycom Unified Communications in RealPresence Access Director System Environments Polycom Document Title 1 Trademark Information POLYCOM

More information

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya Aura Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya Aura Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017 DMP 128 Plus C V DMP 128 Plus C V AT Avaya Aura Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017 Revision Log Date Version Notes August 6 th 2017 1.0 First Release. Applies to Firmware 1.01.0004.002

More information

Cisco Expressway Session Classification

Cisco Expressway Session Classification Cisco Expressway Session Classification Deployment Guide First Published: December 2016 Last Updated: December 2017 Expressway X8.10 Cisco Systems, Inc. www.cisco.com 2 Preface Preface Change History Table

More information

Grandstream Networks, Inc. UCM6xxx SIP Trunks Guide

Grandstream Networks, Inc. UCM6xxx SIP Trunks Guide Grandstream Networks, Inc. Table of Content INTRODUCTION... 4 REGISTER SIP TRUNKS... 5 Configuration... 5 DID / DOD Configuration... 9 Direct Inward Dialing (DID)... 9 Direct Outward Dialing (DOD)... 10

More information

SBC Configuration Examples

SBC Configuration Examples Configuration Note SBC Configuration Examples Mediant Session Border Controllers (SBC) Version 7.0 Configuration Note Contents Table of Contents 1 Introduction... 7 1.1 Configuration Terminology... 7

More information

SIP Proxy Deployment Guide. SIP Server 8.1.1

SIP Proxy Deployment Guide. SIP Server 8.1.1 SIP Proxy Deployment Guide SIP Server 8.1.1 5/4/2018 Table of Contents SIP Proxy 8.1 Deployment Guide 3 SIP Proxy Architecture and Deployment 4 Supported Features 7 Prerequisites 9 Deploying SIP Proxy

More information

Configuration Note Microsoft Lync Server 2013 & BluIP SIP Trunk using Mediant E-SBC

Configuration Note Microsoft Lync Server 2013 & BluIP SIP Trunk using Mediant E-SBC Enterprise Session Border Controllers (E-SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note Microsoft Lync Server 2013 & BluIP SIP Trunk using Mediant E-SBC October 2013 Document #

More information

OneXS will provide users with a reference server (IP, FQDN, or other means to connect to the service). This must be obtained before setup can begin.

OneXS will provide users with a reference server (IP, FQDN, or other means to connect to the service). This must be obtained before setup can begin. 1 SIP Carriers 1.1 OneXS 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya IP Office Configuration Guide REVISION: 1.2 DATE: JANUARY 9 TH 2018

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya IP Office Configuration Guide REVISION: 1.2 DATE: JANUARY 9 TH 2018 DMP 128 Plus C V DMP 128 Plus C V AT Avaya IP Office Configuration Guide REVISION: 1.2 DATE: JANUARY 9 TH 2018 Revision Log Date Version Notes August 6 th 2017 1.0 First Release: Applies to Firmware 1.01.0004.002

More information

Cisco Expressway Options with Cisco Meeting Server and/or Microsoft Infrastructure

Cisco Expressway Options with Cisco Meeting Server and/or Microsoft Infrastructure Cisco Expressway Options with Cisco Meeting Server and/or Microsoft Infrastructure Deployment Guide First Published: December 2016 Last Updated: October 2017 Expressway X8.9.2 Cisco Systems, Inc. www.cisco.com

More information

Genesys Application Note. AudioCodes SIP Phones With Genesys SIP Server. Document version 1.7

Genesys Application Note. AudioCodes SIP Phones With Genesys SIP Server. Document version 1.7 Genesys Application Note AudioCodes SIP Phones With Genesys SIP Server Document version 1.7 The information contained herein is proprietary and confidential and cannot be disclosed or duplicated without

More information

Configuration information in this document is based on IC version 3.0, so the menus shown may vary slightly from your product implementation.

Configuration information in this document is based on IC version 3.0, so the menus shown may vary slightly from your product implementation. 1 SIP Carriers 1.1 Telepacific 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be

More information

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya Aura Configuration Guide REVISION: DATE: MARCH 7 TH 2018

DMP 128 Plus C V DMP 128 Plus C V AT. Avaya Aura Configuration Guide REVISION: DATE: MARCH 7 TH 2018 DMP 128 Plus C V DMP 128 Plus C V AT Avaya Aura Configuration Guide REVISION: 1.2.1 DATE: MARCH 7 TH 2018 Revision Log Date Version Notes August 6 th 2017 1.0 First Release. Applies to Firmware 1.01.0004.002

More information

Dolby Conference Phone. Configuration Guide for Unify OpenScape Enterprise Express 8.0.x

Dolby Conference Phone. Configuration Guide for Unify OpenScape Enterprise Express 8.0.x Dolby Conference Phone Configuration Guide for Unify OpenScape Enterprise Express 8.0.x Version 3.3 31 July 2017 Copyright 2017 Dolby Laboratories. All rights reserved. Dolby Laboratories, Inc. 1275 Market

More information

Configuration Note. Connecting XO Communications SIP Trunking Service to Microsoft Lync Server Using

Configuration Note. Connecting XO Communications SIP Trunking Service to Microsoft Lync Server Using Mediant 800 MSBG E-SBC, Mediant 1000 MSBG E-SBC and Mediant 3000 E-SBC Media Gateway Configuration Note Connecting XO Communications SIP Trunking Service to Microsoft Lync Server 2010 Using AudioCodes

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager 5.2.1, Avaya Aura Session Manager 6.1 and Avaya Aura Session Border Controller 6.0.3 with AT&T IP Toll

More information

Configuring Sonus SBC 1000/2000 with Microsoft Office 365. Application Notes Last Updated April 16, 2013

Configuring Sonus SBC 1000/2000 with Microsoft Office 365. Application Notes Last Updated April 16, 2013 Configuring Sonus SBC 1000/2000 with Microsoft Office 365 Application Notes Last Updated April 16, 2013 Contents Sonus SBC 1000/2000 Session Border Controller Configuration Notes... 2 Configuration Checklist...

More information

Frequently Asked Questions (Dialogic BorderNet 500 Gateways)

Frequently Asked Questions (Dialogic BorderNet 500 Gateways) Frequently Asked Questions (Dialogic BorderNet 500 Gateways) Q: What is a Dialogic BorderNet 500 Gateway, and what are its main functions? A: A Dialogic BorderNet 500 Gateway consists of a full featured

More information

SBC Deployment Guide Architecture Options and Configuration Examples

SBC Deployment Guide Architecture Options and Configuration Examples Enterprise Session Border Controllers Mediant E-SBC Series AudioCodes SBC Deployment Guide Architecture Options and Configuration Examples Version 6.4 April 2012 Document # LTRT-31620 Deployment Guide

More information

Dolby Conference Phone. Configuration guide for Unify OpenScape Enterprise Express 8.0.x

Dolby Conference Phone. Configuration guide for Unify OpenScape Enterprise Express 8.0.x Dolby Conference Phone Configuration guide for Unify OpenScape Enterprise Express 8.0.x Version 3.2 28 June 2017 Copyright 2017 Dolby Laboratories. All rights reserved. Dolby Laboratories, Inc. 1275 Market

More information

Cisco TelePresence Conductor with Cisco Unified Communications Manager

Cisco TelePresence Conductor with Cisco Unified Communications Manager Cisco TelePresence Conductor with Cisco Unified Communications Manager Deployment Guide TelePresence Conductor XC4.0 Unified CM 10.5(2) January 2016 Contents Introduction 6 About this document 6 Related

More information

Application Note Asterisk BE with SIP Trunking - Configuration Guide

Application Note Asterisk BE with SIP Trunking - Configuration Guide Application Note Asterisk BE with SIP Trunking - Configuration Guide 23 January 2009 Asterisk BE SIP Trunking Table of Contents 1 ASTERISK BUSINESS EDITION AND INGATE... 1 1.1 SIP TRUNKING SUPPORT... 2

More information

Configuration Note. Microsoft Lync Server 2013 & NextGenTel SIP Trunk using Mediant E-SBC. Enterprise Session Border Controllers (E-SBC)

Configuration Note. Microsoft Lync Server 2013 & NextGenTel SIP Trunk using Mediant E-SBC. Enterprise Session Border Controllers (E-SBC) Enterprise Session Border Controllers (E-SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note Microsoft Lync Server 2013 & NextGenTel SIP Trunk using Mediant E-SBC Version 6.8 December

More information

ThinkTel ITSP with Registration Setup

ThinkTel ITSP with Registration Setup January 13 ThinkTel ITSP with Registration Setup Author: Zultys Technical Support This configuration guide was created to assist knowledgeable vendors with configuring the Zultys MX Phone System with ThinkTel

More information

Leveraging Amazon Chime Voice Connector for SIP Trunking. March 2019

Leveraging Amazon Chime Voice Connector for SIP Trunking. March 2019 Leveraging Amazon Chime Voice Connector for SIP Trunking March 2019 Notices Customers are responsible for making their own independent assessment of the information in this document. This document: (a)

More information

Configuration Note Microsoft Lync Server 2013 & tipicall SIP Trunk using Mediant E-SBC

Configuration Note Microsoft Lync Server 2013 & tipicall SIP Trunk using Mediant E-SBC Enterprise Session Border Controllers (E-SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note Microsoft Lync Server 2013 & tipicall SIP Trunk using Mediant E-SBC Version 6.8 December

More information

Configuring SIP Registration Proxy on Cisco UBE

Configuring SIP Registration Proxy on Cisco UBE The Support for SIP Registration Proxy on Cisco UBE feature provides support for sending outbound registrations from Cisco Unified Border Element (UBE) based on incoming registrations. This feature enables

More information

Allstream NGNSIP Security Recommendations

Allstream NGNSIP Security Recommendations Allstream NGN SIP Trunking Quick Start Guide We are confident that our service will help increase your organization s performance and productivity while keeping a cap on your costs. Summarized below is

More information

Microsoft Skype for Business Server 2015 and DTAG SIP Trunk using AudioCodes Mediant MSBR E-SBC

Microsoft Skype for Business Server 2015 and DTAG SIP Trunk using AudioCodes Mediant MSBR E-SBC Configuration Note AudioCodes Professional Services - Interoperability Lab Microsoft Skype for Business Server 2015 and DTAG SIP Trunk using AudioCodes Mediant MSBR E-SBC Version 6.8 Configuration Note

More information

Dolby Conference Phone 3.0 configuration guide for Unify OpenScape Enterprise Express 8.0.x

Dolby Conference Phone 3.0 configuration guide for Unify OpenScape Enterprise Express 8.0.x Dolby Conference Phone 3.0 configuration guide for Unify OpenScape Enterprise Express 8.0.x 11 July 2016 Copyright 2016 Dolby Laboratories. All rights reserved. For information, contact: Dolby Laboratories,

More information

Avaya PBX SIP TRUNKING Setup & User Guide

Avaya PBX SIP TRUNKING Setup & User Guide Avaya PBX SIP TRUNKING Setup & User Guide Nextiva.com (800) 285-7995 2 P a g e Contents Description... 3 Avaya IP PBX Configuration... 3 Licensing and Physical Hardware... 4 System Tab Configuration...

More information

Microsoft Skype for Business Server 2015 and ShoreTel UC System using AudioCodes Mediant E-SBC

Microsoft Skype for Business Server 2015 and ShoreTel UC System using AudioCodes Mediant E-SBC Configuration Note AudioCodes Professional Services - Interoperability Lab Microsoft Skype for Business Server 2015 and ShoreTel UC System using AudioCodes Mediant E-SBC Version 7.0 Configuration Note

More information

Avaya Solution & Interoperability Test Lab. Abstract

Avaya Solution & Interoperability Test Lab. Abstract Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager 7.1, Avaya Aura Session Manager 7.1, and Avaya Session Border Controller for Enterprise 7.2, with AT&T

More information

Configuration Note Microsoft Lync Server 2013 & Windstream SIP Trunk using Mediant E-SBC

Configuration Note Microsoft Lync Server 2013 & Windstream SIP Trunk using Mediant E-SBC Enterprise Session Border Controllers (E-SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note Microsoft Lync Server 2013 & Windstream SIP Trunk using Mediant E-SBC Version 6.8 August

More information

Microsoft Skype for Business Server 2015 and TELUS SIP Trunk using AudioCodes Mediant E-SBC

Microsoft Skype for Business Server 2015 and TELUS SIP Trunk using AudioCodes Mediant E-SBC Configuration Note AudioCodes Professional Services - Interoperability Lab Microsoft Skype for Business Server 2015 and TELUS SIP Trunk using AudioCodes Mediant E-SBC Version 7.0 Configuration Note Contents

More information

Dolby Conference Phone. Configuration guide for Avaya Aura Platform 6.x

Dolby Conference Phone. Configuration guide for Avaya Aura Platform 6.x Dolby Conference Phone Configuration guide for Avaya Aura Platform 6.x Version 3.1 22 February 2017 Copyright 2017 Dolby Laboratories. All rights reserved. Dolby Laboratories, Inc. 1275 Market Street San

More information

Cisco TelePresence Conductor with Cisco Unified Communications Manager

Cisco TelePresence Conductor with Cisco Unified Communications Manager Cisco TelePresence Conductor with Cisco Unified Communications Manager Deployment Guide XC2.2 Unified CM 8.6.2 and 9.x D14998.09 Revised March 2014 Contents Introduction 4 About this document 4 Further

More information

Application Note. Microsoft OCS 2007 Configuration Guide

Application Note. Microsoft OCS 2007 Configuration Guide Application Note Microsoft OCS 2007 Configuration Guide 15 October 2009 Microsoft OCS 2007 Configuration Guide Table of Contents 1 MICROSOFT OCS 2007 AND INGATE... 1 1.1 SIP TRUNKING SUPPORT... 2 2 INGATE

More information

DMP 128 Plus C V DMP 128 Plus C V AT

DMP 128 Plus C V DMP 128 Plus C V AT DMP 128 Plus C V DMP 128 Plus C V AT Interactive Intelligence Configuration Guide REVISION: 1.0.1 DATE: MARCH 7 TH 2018 Revision Log Date Version Notes Feb 9 th 2018 1.0 First Release: Applies to Firmware

More information

UCM6102/6104/6108/6116 Configuration

UCM6102/6104/6108/6116 Configuration UCM6102/6104/6108/6116 Configuration This document introduces manual configuration steps performed for interoperability testing between AccessLine and Grandstream UCM6102/6104/6108/6116. Configuration

More information

Dolby Conference Phone. Configuration guide for Avaya Aura Platform 6.x

Dolby Conference Phone. Configuration guide for Avaya Aura Platform 6.x Dolby Conference Phone Configuration guide for Avaya Aura Platform 6.x Version 3.2 28 June 2017 Copyright 2017 Dolby Laboratories. All rights reserved. Dolby Laboratories, Inc. 1275 Market Street San Francisco,

More information

Configuration Note. Microsoft Lync Server 2013 & ITSP SIP Trunk using AudioCodes Mediant SBC. Interoperability Laboratory. Version 6.

Configuration Note. Microsoft Lync Server 2013 & ITSP SIP Trunk using AudioCodes Mediant SBC. Interoperability Laboratory. Version 6. AudioCodes Mediant Series Session Border Controller (SBC) Interoperability Laboratory Configuration Note Microsoft Lync Server 2013 & ITSP SIP Trunk using AudioCodes Mediant SBC Version 6.8 May 2015 Document

More information

SIP TRUNKING CARRIER CERTIFICATION OXE-SIP configuration

SIP TRUNKING CARRIER CERTIFICATION OXE-SIP configuration OXE version: R11.0.1 K1.400.33 SIP TRUNKING CARRIER CERTIFICATION OXE-SIP configuration System SIP parameters Path: System / Other System Param. / SIP Parameters OXE default value new value (if modified)

More information

Unofficial IRONTON ITSP Setup Guide

Unofficial IRONTON ITSP Setup Guide September 13 Unofficial IRONTON ITSP Setup Guide Author: Zultys Technical Support This unofficial configuration guide was created to assist knowledgeable vendors with configuring the Zultys MX Phone System

More information

Teams Direct Routing. Configuration Checklists for BTIP and Business Talk SIP services. 28 january Teams Direct Routing AudioCodes Checklist 0.

Teams Direct Routing. Configuration Checklists for BTIP and Business Talk SIP services. 28 january Teams Direct Routing AudioCodes Checklist 0. Teams Direct Routing Configuration Checklists for BTIP and Business Talk 28 january 2019 Teams Direct Routing AudioCodes Checklist 0.2 Contents 1 Main certified architectures... 3 1.1 Standalone mode...

More information

Microsoft Teams Direct Routing Enterprise Model and Swisscom SIP Trunk "Smart Business Connect" using AudioCodes Mediant SBC

Microsoft Teams Direct Routing Enterprise Model and Swisscom SIP Trunk Smart Business Connect using AudioCodes Mediant SBC Configuration Note AudioCodes Professional Services Interoperability Lab Microsoft Teams Direct Routing Enterprise Model and Swisscom SIP Trunk "Smart Business Connect" using AudioCodes Mediant SBC Version

More information

Microsoft Skype for Business Server 2015 and Flowroute SIP Trunk using AudioCodes Mediant E-SBC

Microsoft Skype for Business Server 2015 and Flowroute SIP Trunk using AudioCodes Mediant E-SBC Configuration Note AudioCodes Professional Services Interoperability Lab Microsoft Skype for Business Server 2015 and Flowroute SIP Trunk using AudioCodes Mediant E-SBC Version 7.0 Configuration Note

More information

Application Notes for Configuring Avaya IP Office 8.1 with Etisalat SIP Trunk service Issue 1.0

Application Notes for Configuring Avaya IP Office 8.1 with Etisalat SIP Trunk service Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 8.1 with Etisalat SIP Trunk service Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Application Notes for Configuring SIP Trunking between TelePacific SmartVoice SIP Connect and an Avaya IP Office Telephony Solution 1.

Application Notes for Configuring SIP Trunking between TelePacific SmartVoice SIP Connect and an Avaya IP Office Telephony Solution 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between TelePacific SmartVoice SIP Connect and an Avaya IP Office Telephony Solution 1.0 Abstract These Application

More information

Mitel Technical Configuration Notes HO858

Mitel Technical Configuration Notes HO858 TelNet Worldwide, Inc. telnetww.com 1-833-4TELNET Mitel Technical Configuration Notes HO858 rev. 2018-12-12 Configure MiVoice Business 9.0 for use with TelNet Worldwide SIP Trunking Description: This document

More information

Sipdex M200s IPPBX. Embedded. Support Any IP Phone. Softphone and SIP Client App

Sipdex M200s IPPBX. Embedded. Support Any IP Phone. Softphone and SIP Client App Sipdex M200s IPPBX Based on embedded asterisk system, SIPDEX M200s IPPBX is a high quality, stable PBX without any moving parts and a very small footprint required minimum technology knowledge to deploy.

More information

Configuration Note Microsoft Lync Server 2013 & Netia SIP Trunk using Mediant E-SBC

Configuration Note Microsoft Lync Server 2013 & Netia SIP Trunk using Mediant E-SBC Enterprise Session Border Controllers (E-SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note Microsoft Lync Server 2013 & Netia SIP Trunk using Mediant E-SBC Version 6.8 June 2014 Document

More information

AudioCodes OVR with SIP Trunking

AudioCodes OVR with SIP Trunking Configuration Note AudioCodes One Voice for Skype For Business AudioCodes OVR with SIP Trunking for Microsoft Skype for Business Online Version 7.2 Configuration Note Contents Table of Contents 1 Introduction...

More information

FRAFOS ABC-SBC Generic SIP Trunk Integration Guide for ShoreTel 14.2

FRAFOS ABC-SBC Generic SIP Trunk Integration Guide for ShoreTel 14.2 FRAFOS ABC-SBC Generic SIP Trunk Integration Guide for ShoreTel 14.2 FRAFOS GmbH FRAFOS GmbH Windscheidstr. 18 10627 Berlin Germany Email: info@frafos.com WWW: www.frafos.com 11.05.2015 IN # 15023 Table

More information

Spectrum Enterprise SIP Trunking Service Avaya IPO10 with SBC IP PBX Configuration Guide

Spectrum Enterprise SIP Trunking Service Avaya IPO10 with SBC IP PBX Configuration Guide Spectrum Enterprise SIP Trunking Service Avaya IPO10 with SBC IP PBX Configuration Guide About Spectrum Enterprise: Spectrum Enterprise is a division of Charter Communications following a merger with Time

More information

Cisco TelePresence Conductor with Unified CM

Cisco TelePresence Conductor with Unified CM Cisco TelePresence Conductor with Unified CM Deployment Guide TelePresence Conductor XC3.0 Unified CM 10.x Revised February 2015 Contents Introduction 5 About this document 5 Related documentation 5 About

More information

Grandstream Networks, Inc. UCM6XXX Configuration Guide for Remote Extensions

Grandstream Networks, Inc. UCM6XXX Configuration Guide for Remote Extensions Grandstream Networks, Inc. Table of Content INTRODUCTION... 3 NAT CONFIGURATION ON UCM6XXX... 4 Prerequisites... 4 UCM6XXX NAT Settings... 4 Configuring DDNS Settings (Optional)... 5 Configuring NAT Extension

More information

nexvortex Setup Template

nexvortex Setup Template nexvortex Setup Template KERIO OPERATOR October 2015 5 1 0 S P R I N G S T R E E T H E R N D O N V A 2 0 1 7 0 + 1 8 5 5. 6 3 9. 8 8 8 8 Introduction This document is intended only for nexvortex customers

More information

Application Notes for Configuring SIP Trunking between Bandwidth.com SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.

Application Notes for Configuring SIP Trunking between Bandwidth.com SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Bandwidth.com SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.0 Abstract These

More information

Patton Electronics Co Rickenbacker Drive, Gaithersburg, MD 20879, USA tel: fax:

Patton Electronics Co Rickenbacker Drive, Gaithersburg, MD 20879, USA tel: fax: Patton Electronics Co. www.patton.com 7622 Rickenbacker Drive, Gaithersburg, MD 20879, USA tel: +1 301-975-1000 fax: +1 301-869-9293 2012 Inalp Networks AG, Niederwangen, Switzerland All Rights Reserved.

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIP CoE Technical Configuration Notes Configure Mitel 6863/6865 SIP Phone to use with MiVoice Business 8.0 SP2 FEBRUARY 2018 SIP COE HO2459 TECHNICAL CONFIGURATION NOTES NOTICE The information contained

More information

Broadvox Fusion Platform Version 1.2 ITSP Setup Guide

Broadvox Fusion Platform Version 1.2 ITSP Setup Guide November 13 Broadvox Fusion Platform Version 1.2 ITSP Setup Guide Author: Zultys Technical Support This configuration guide was created to assist knowledgeable vendors with configuring the Zultys MX Phone

More information

Configuration Note. AireSpring SIP Trunk & Genesys Contact Center using AudioCodes Mediant SBC. Session Border Controllers (SBC)

Configuration Note. AireSpring SIP Trunk & Genesys Contact Center using AudioCodes Mediant SBC. Session Border Controllers (SBC) Session Border Controllers (SBC) AudioCodes Mediant Series Interoperability Lab Configuration Note AireSpring SIP Trunk & Genesys Contact Center using AudioCodes Mediant SBC Version 6.8 October 2014 Document

More information

Connecting IP-PBX to BroadSoft's BroadCloud SIP Trunk using AudioCodes Mediant SBC

Connecting IP-PBX to BroadSoft's BroadCloud SIP Trunk using AudioCodes Mediant SBC Quick Guide AudioCodes Mediant Session Border Controllers (SBC) Connecting IP-PBX to BroadSoft's BroadCloud SIP Trunk using AudioCodes Mediant SBC Version 7.2 Introduction See Chapter 1 Obtain Software

More information

DMP 128 Plus C V DMP 128 Plus C V AT. Cisco CUCM Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017

DMP 128 Plus C V DMP 128 Plus C V AT. Cisco CUCM Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017 DMP 128 Plus C V DMP 128 Plus C V AT Cisco CUCM Configuration Guide REVISION: 1.1 DATE: SEPTEMBER 1 ST 2017 Revision Log Date Version Notes August 4 th 2017 1.0 First Release: Applies to Firmware Version

More information

Describe the EdgeMarc s VoIP Survivability facility; how it works and how it is configured.

Describe the EdgeMarc s VoIP Survivability facility; how it works and how it is configured. KnowledgeBase Q & A Question Describe the EdgeMarc s VoIP Survivability facility; how it works and how it is configured. Answer Article last updated: January 31, 2007 Based on VOS: v6.7.6 1.0 Overview

More information

A. On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Discover to off.

A. On the VCS, navigate to Configuration, Protocols, H.323, and set Auto Discover to off. Volume: 383 Questions Question No: 1 Which parameter should be set to prevent H.323 endpoints from registering to Cisco TelePresence Video Communication Server automatically? A. On the VCS, navigate to

More information

SIP Trunking. Overview. 1) Network Setup (here)

SIP Trunking. Overview. 1) Network Setup (here) SIP Trunking Overview The SIP Trunking use case allows your PBX to safely connect over the internet to an ITSP. The SBC in this scenaro is providing enhanced security for the corporate network without

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager 6.2, Avaya Aura Session Manager 6.2 and Avaya Session Border Controller for Enterprise with AT&T IP Flexible

More information

Dolby Conference Phone. Configuration Guide for Microsoft Skype for Business

Dolby Conference Phone. Configuration Guide for Microsoft Skype for Business Dolby Conference Phone Configuration Guide for Microsoft Skype for Business Version 3.3 31 July 2017 Copyright 2017 Dolby Laboratories. All rights reserved. Dolby Laboratories, Inc. 1275 Market Street

More information

Sbc Service User Guide

Sbc Service User Guide For Mediatrix Sentinel and Mediatrix 3000 Revision 04 2016-01-13 Table of Contents Table of Contents Configuration notes 5 Call Agents 6 phone_lines_ca Call Agent 8 trunk_lines_ca Call Agent 9 local_users_ca

More information

FreeSWITCH IP PBX with Secure Twilio Elastic SIP Trunking

FreeSWITCH IP PBX with Secure Twilio Elastic SIP Trunking FreeSWITCH IP PBX with Secure Twilio Elastic SIP Trunking (Updated: 3/14/2017) Implementing security mechanisms in the Twilio Elastic SIP trunk provides secure and reliable data transfer between your SIP

More information

SBC Site Survey Questionnaire Forms

SBC Site Survey Questionnaire Forms SBC Site Survey Questionnaire Forms For Design and Deployment of AudioCodes Mediant SBC Product Line This document is intended for the persons responsible for the design and deployment of AudioCodes SBC

More information

OpenScape Business V2

OpenScape Business V2 OpenScape Business V2 Tutorial Support of SIP Endpoints connected via the internet Version 3.1 Definitions HowTo An OpenScape Business HowTo describes the configuration of an OpenScape Business feature

More information

Configuring MediaPack 1288 Analog Gateway as Third-Party SIP Device (Advanced) in Cisco Unified Communications Manager Ver

Configuring MediaPack 1288 Analog Gateway as Third-Party SIP Device (Advanced) in Cisco Unified Communications Manager Ver Configuration Note AudioCodes Professional Services Interoperability Lab Configuring MediaPack 1288 Analog Gateway as Third-Party SIP Device (Advanced) in Cisco Unified Communications Manager Ver. 10.0.1

More information

Unified Communication Platform

Unified Communication Platform fonouc Unified Communication Platform fonouc Unified Communications Service Platform, is a scalable, managed, turnkey solution for carries and service providers, designed to provide multi-tenant business

More information

Application Notes for Configuring Fonolo In-Call Rescue with Avaya IP Office Server Edition using SIP Trunks Issue 1.0

Application Notes for Configuring Fonolo In-Call Rescue with Avaya IP Office Server Edition using SIP Trunks Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Fonolo In-Call Rescue with Avaya IP Office Server Edition using SIP Trunks Issue 1.0 Abstract These Application Notes describe

More information

Configuring Multi-Tenants on SIP Trunks

Configuring Multi-Tenants on SIP Trunks The feature allows specific global configurations for multiple tenants on SIP trunks that allow differentiated services for tenants. allows each tenant to have their own individual configurations. The

More information

Technical White Paper for NAT Traversal

Technical White Paper for NAT Traversal V300R002 Technical White Paper for NAT Traversal Issue 01 Date 2016-01-15 HUAWEI TECHNOLOGIES CO., LTD. 2016. All rights reserved. No part of this document may be reproduced or transmitted in any form

More information

Avaya Session Border Controller Enterprise Implementation and Maintenance Exam

Avaya Session Border Controller Enterprise Implementation and Maintenance Exam 1 Avaya - 3107 Avaya Session Border Controller Enterprise Implementation and Maintenance Exam QUESTION: 1 If the Remote Worker cluster is using a Real Server IP and Real Server Port, over which protocols

More information

Avaya Solution & Interoperability Test Lab. Abstract

Avaya Solution & Interoperability Test Lab. Abstract Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager/Local Survivable Processor 6.3, Avaya Aura Branch Session Manager 6.3, and Avaya Session Border Controller

More information

USER GUIDE. Alcatel OmniPCX Office OXO-Fusion 360 SIP Trunk Programming Guide 11/07/2017

USER GUIDE. Alcatel OmniPCX Office OXO-Fusion 360 SIP Trunk Programming Guide 11/07/2017 Alcatel OmniPCX Office OXO-Fusion 360 SIP Trunk Programming Guide 11/07/2017 Contents: SIP Trunk Programming Guide Step 1: Gather Information...4 Step 2: OXO Programming...5 Step 3: Network Programming...22

More information

Spectrum Enterprise SIP Trunking Service AudioCodes Mediant Series IP PBX Configuration Guide

Spectrum Enterprise SIP Trunking Service AudioCodes Mediant Series IP PBX Configuration Guide Spectrum Enterprise SIP Trunking Service AudioCodes Mediant Series IP PBX Configuration Guide About Spectrum Enterprise: Spectrum Enterprise is a division of Charter Communications following a merger with

More information

v2.0 September 30, 2013

v2.0 September 30, 2013 v2.0 September 30, 2013 This document was written for Iwatsu Enterprise-CS systems with version 8.x software. In some cases, available feature operations may differ from those listed in this document,

More information

Application Note Asterisk BE with Remote Phones - Configuration Guide

Application Note Asterisk BE with Remote Phones - Configuration Guide Application Note Asterisk BE with Remote Phones - Configuration Guide 15 January 2009 Asterisk BE - Remote SIP Phones Table of Contents 1 ASTERISK BUSINESS EDITION AND INGATE... 1 1.1 REMOTE SIP PHONE

More information

EarthLink Business SIP Trunking. ShoreTel 14.2 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. ShoreTel 14.2 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking ShoreTel 14.2 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Optus Evolve Voice SIP Trunking Service with Avaya Aura Communication Manager 7.0, Avaya Aura Session Manager 7.0 and Avaya Session Border

More information

INTERFACE SPECIFICATION SIP Trunking. 8x8 SIP Trunking. Interface Specification. Version 2.0

INTERFACE SPECIFICATION SIP Trunking. 8x8 SIP Trunking. Interface Specification. Version 2.0 8x8 Interface Specification Version 2.0 Table of Contents Introduction....3 Feature Set....3 SIP Interface....3 Supported Standards....3 Supported SIP methods....4 Additional Supported SIP Headers...4

More information