Hosted PBX use case for Service Providers
|
|
- Dennis Chase
- 5 years ago
- Views:
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
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 informationInstallation & 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 informationGrandstream 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 informationSBC 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 information1 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 informationMicrosoft 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 information1 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 informationAcano 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 informationAcano 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 informationApplication 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 informationNEC: 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 informationOne-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 informationUnified 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 informationDMP 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 informationCisco 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 informationGrandstream 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 informationSBC 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 informationSIP 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 informationConfiguration 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 informationOneXS 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 informationDMP 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 informationCisco 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 informationGenesys 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 informationConfiguration 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 informationDMP 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 informationDolby 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 informationConfiguration 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 informationAbstract. 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 informationConfiguring 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 informationFrequently 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 informationSBC 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 informationDolby 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 informationCisco 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 informationApplication 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 informationConfiguration 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 informationThinkTel 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 informationLeveraging 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 informationConfiguration 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 informationConfiguring 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 informationAllstream 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 informationMicrosoft 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 informationDolby 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 informationAvaya 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 informationMicrosoft 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 informationAvaya 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 informationConfiguration 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 informationMicrosoft 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 informationDolby 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 informationCisco 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 informationApplication 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 informationDMP 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 informationUCM6102/6104/6108/6116 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 informationDolby 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 informationConfiguration 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 informationSIP 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 informationUnofficial 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 informationTeams 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 informationMicrosoft 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 informationMicrosoft 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 informationApplication 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 informationApplication 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 informationMitel 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 informationSipdex 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 informationConfiguration 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 informationAudioCodes 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 informationFRAFOS 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 informationSpectrum 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 informationCisco 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 informationGrandstream 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 informationnexvortex 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 informationApplication 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 informationPatton 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 informationTechnical 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 informationBroadvox 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 informationConfiguration 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 informationConnecting 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 informationDMP 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 informationDescribe 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 informationA. 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 informationSIP 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 informationAbstract. 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 informationDolby 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 informationSbc 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 informationFreeSWITCH 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 informationSBC 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 informationOpenScape 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 informationConfiguring 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 informationUnified 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 informationApplication 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 informationConfiguring 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 informationTechnical 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 informationAvaya 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 informationAvaya 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 informationUSER 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 informationSpectrum 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 informationv2.0 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 informationApplication 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 informationEarthLink 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 informationAbstract. 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 informationINTERFACE 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