SFLphone Documentation
|
|
|
- Jeffery Little
- 10 years ago
- Views:
Transcription
1 SFLphone Documentation Release 1.0 SFLphone Team August 18, 2014
2
3 Contents 1 Contents Getting started Setup a secure environment with Asterisk Setup a secure environment with Freeswitch Indices and tables 25 i
4 ii
5 SFLphone is a robust, standards-compliant enterprise softphone, for desktop and embedded systems. It is designed to handle several hundred calls a day. SFLphone is available under the GNU GPL license, version 3. Please visit the official website for a complete list of features: Contents 1
6 2 Contents
7 CHAPTER 1 Contents 3
8 1.1 Getting started 4 Chapter 1. Contents
9 1.1.1 Install SFLphone First, you need to add the official SFLphone PPA (Personal Package Archive). This allows us to push new versions for older distributions. Note: This step is not mandatory as Ubuntu provides SFLphone packages in its universe repository. Solution 1: Software Center Open Software Center (in Unity, press the Windows key and type software center) Figure 1.1: Ubuntu Software Center. In Edit > Software sources, select the Other software tab Click add and enter ppa:savoirfairelinux Click close Click the little arrow next to All Software and select sflphone Select GNOME client for SFLphone and click Install Solution 2: Command-line Add the repository to your software sources: sudo add-apt-repository ppa:savoirfairelinux Now, update the package list: sudo apt-get update You can now install the latest SFLphone version: 1.1. Getting started 5
10 sudo apt-get install sflphone-client-gnome Solution 3: Building from sources (not recommended) Please refer to the instructions here to build SFLphone from source Configuring an existing account The simplest way to configure SFLphone is to use the First Run wizard. In the Unity lens interface (top left dock icon or Windows key), type sflphone After a few seconds, a wizard window should appear on the screen with a welcome message Press Continue to start the wizard Note: You can always return to the installation wizard in SFLphone by clicking the menu Call > Configuration Assistant Figure 1.2: Account configuration wizard: first step. Account Select Register an existing SIP or IAX2 account VoIP Protocols You can select here the communication protocol to use to make calls (if unsure, use SIP). Account settings This step allows you to set up your account, by specifying the hostname, username, password,... 6 Chapter 1. Contents
11 Figure 1.3: Account configuration wizard: Account window. Figure 1.4: Account configuration wizard: VoIP Protocols window. Figure 1.5: Account configuration wizard: Account settings Getting started 7
12 Here are the details of each settings: Fields Alias Hostname Username Password Voic number (optional) Secure communications with ZRTP Press Continue and Apply when you are ready to register your account. Description A name you will remember (Example: Workplace) Usually the server address, (Example: sip.sflphone.org) Usually your phone extension (Example: 123), but may also b Your account password (Warning: will be stored in plain text) Another username you can call to play your voic . Not ev has one Do not check (see security section) Figure 1.6: Account configuration wizard: Review Account registration The last panel displays an overview of your account settings. You may now click on Close, as the installation wizard is finishow now. If you now select Edit > Accounts, your new account should be registered, and appear in green SIP security basics The first thing to know about SIP security is that there is usually none at all. By default, everything is transmitted unencrypted and readable by any software that can grab traffic between you and your peer. If you are using proxies along the way, each of them may decide to ignore certain security options. The second important detail to retain is that SIP security alone (SIPS, SIP-TLS) does not encrypt your communications. SIPS only encrypts the handshake between both peers. To encrypt the media stream (aka, voice) itself, you also need to enable Secure RTP (aka, SRTP). Likewise, only having SRTP enabled may encrypt your audio but will not obscure information about the call itself (i.e. participants, IP addresses, etc.). Most registrar and SIP servers have some level of support for security, however, all implementations are not created equal. As of Asterisk 1.8 (the default in many server operating systems), some security options are not as well supported as they should be. For a safer system, Freeswitch is the best free software option. Asterisk also needs to be recompiled to enable SRTP (see ). This, in turn, will force you to manually update your SIP server everytime a security patch is available, introducing security vulnerabilities of its own. 8 Chapter 1. Contents
13 Important: TODO show security options here 1.1. Getting started 9
14 1.2 Setup a secure environment with Asterisk 10 Chapter 1. Contents
15 1.2.1 Set up a basic Asterisk server Important: Prerequisites: an Ubuntu server or virtual machine Outcome: a basic SIP server with 2 accounts Install Asterisk Install Asterisk with apt-get: sudo su apt-get install asterisk It will ask you for a country code, you can check to get yours. You can check if Asterisk is operational using: service asterisk status Basic configuration Add SIP accounts Now, using your favourite text editor, make a backup of /etc/asterisk/sip.conf and replace it with: [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr= srvlookup=no disallow=all allow=ulaw allow=g722 allow=alaw allow=gsm alwaysauthreject=yes canreinvite=no session-timers=refuse localnet= / [666] 1.2. Setup a secure environment with Asterisk 11
16 callerid=anonymous type=friend host=dynamic secret=123 context=internal [777] type=friend host=dynamic secret=456 context=internal Be sure to update localnet to match your network settings. Run ifconfig command to check your public IP address: To apply the settings, execute rasterisk and type: sip set debug on sip reload To confirm the new diaplan, run: sip show users Dialplan Now, setup a new dialplan to be able to call other users. Edit /etc/asterisk/extensions.conf and add the following lines: [users] include => default include => trunklocal include => iaxtel700 include => trunktollfree include => iaxprovider exten => 777,n,Dial(SIP/777,777,Tt) 12 Chapter 1. Contents
17 [internal] exten => _XXX,1,Dial(SIP/${EXTEN}) Configuring Asterisk encryption In this part, we will use TLS to encrypt the data streams. # Operate as root to install openssl sudo su - apt-get install openssl # Create the directory and get a certificate mkdir /etc/certs cd /etc/certs/ wget sh ast_tls_cert -C pbx.myhostname -O "NSA proof(?) server" -d /etc/certs/ # Now fill out the form. sh ast_tls_cert -m client -c /etc/certs/ca.crt -k /etc/certs/ca.key -C mynsauser.myhostname -O "NSA p chown asterisk:asterisk./ -R ls Note: This How-To has assumed the same computer as server and client for simplicity s sake. In a real-world context, this will rarely be the case. So you will have to safely upload the certificates to the client computer. To do this, the scp command allows you to upload the key over an encrypted connection. In our case, you can simply copy them using cp as it doesn t change anything. scp mynsauser.pem username@hostname:/tmp/ scp ca.crt username@hostname:/tmp/ Now, Asterisk need to be configured to use this certificate for encrypted calls. Backup your current /etc/asterisk/sip.conf, then open sip.conf and replace its contents with: [general] context=internal allowguest=no allowsubscribe=yes allowoverlap=no bindport=5060 bindaddr= ; tlsbindaddr= ; srvlookup=no disallow=all allow=ulaw allow=g722 allow=alaw allow=gsm alwaysauthreject=yes canreinvite=no ;nat=yes session-timers=refuse localnet= / tlsenable=yes tlscertfile=/etc/certs/asterisk.pem tlscafile=/etc/certs/ca.crt tlscipher=tlsv Setup a secure environment with Asterisk 13
18 ;tlsclientmethod=tlsv1 tlsdontverifyserver=no tlsbindaddr= [777] callerid=nsa type=friend secret=nsa host=dynamic transport=tls port=5061 context=internal dtmfmode=rfc2833 insecure = invite,port nat = yes [888] callerid=nsa type=friend secret=nsa host=dynamic transport=tls port=5061 context=internal dtmfmode=rfc2833 insecure = invite,port nat = yes [999] callerid=play type=friend secret=play host=dynamic transport=tls port=5061 context=local dtmfmode=rfc2833 ;insecure = invite,port nat = yes And in extensions.conf, in the [local] section, add: exten => 999,1,Answer exten => 999,2,Playback(tt-weasels) exten => 999,3,Wait(10) exten => 999,4,Hangup Now, run the rasterisk command and type: sip reload dialplan reload Asterisk should now be using TLS for message passing. Warning: The stream itself is not yet encrypted, only the SIP messages are. 14 Chapter 1. Contents
19 1.2.3 Configuring SFLphone with Asterisk Once this is done, execute sflphone-client-gnome. A configuration wizard will launch if you started it for the first time. Select the following values: Account: Register an Existing SIP or IAX2 account, then Next VoIP Protocols: SIP, then Next As for the SIP account settings, input these values: Settings Values Alias My First Account Hostname <your asterisk server IP> Username 666 Password 123 Note: You may run ifconfig to check your IP address. Then Next and Apply. If you go in Edit > Account, you should now see MyFirstAccount as Registered (in green). If you see a Trying... status and run Asterisk on the same computer as SFLphone, you have to change the default port in your account advanced settings (Edit > Accounts > Select your account > Edit > Advanced) Configuring SFLphone security Important: Prerequisites: You have both ca.crt and the.pem certificates For Asterisk to encrypt your stream, select: In Edit > Account > Your account > Security, select ZRTP in SRTP key exchanges (Asterisk need to be compiled with SRTP support). Also select Use TLS transports. In the Advanced tab, set your ca.crt as authority and.pem as user certificate. Uncheck Verify incoming certificate (as a server) Click on Apply You are now done! Warning: Please note that this setup is still vulnerable to Man-in-the-middle attack, but not to packet sniffers Setup a secure environment with Asterisk 15
20 Figure 1.7: Account list window, showing the accounts registration status. 16 Chapter 1. Contents
21 Figure 1.8: The account s advanced settings allows to configure the network interface, port range and registration expiration time Setup a secure environment with Asterisk 17
22 18 Chapter 1. Contents
23 1.3 Setup a secure environment with Freeswitch 1.3. Setup a secure environment with Freeswitch 19
24 1.3.1 Installing Freeswitch Please see the Ubuntu Quick Start Guide for an official guide to installing Freeswitch. Otherwise, to install the latest FreeSwitch from git, run the following commands: # Install all dependencies sudo su - apt-get install git-core build-essential autoconf automake libtool libncurses5 libncurses5-dev make l # Download FreeSwitch via git mkdir -p /usr/local/src cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git cd freeswitch # Build./bootstrap.sh./configure make # You may have to call "make" a few times before it works. # Install in /usr/local make all install cd-sounds-install cd-moh-install Reference: to install FreeSwitch (install all optional packages). Now, it is time to see if FreeSwitch is properly loaded. If you already have SFLphone running on the same computer, make sure to close it first. /usr/local/freeswitch/bin/freeswitch After a few seconds, a shell will appear. To test if SIP is ready, enter: sofia status profile internal This should display something like: Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts , Dialplan XML Context public Challenge Realm auto_from RTP-IP SIP-IP URL sip:mod_sofia@ :5060 BIND-URL sip:mod_sofia@ :5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 NOMEDIA false 20 Chapter 1. Contents
25 1.3. Setup a secure environment with Freeswitch 21
26 LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS Configuring Freeswitch security To have an overview of SIP security, please read the section SIP security basics. cd /usr/local/freeswitch/bin sudo./gentls_cert setup -cn alt DNS:localhost -org sudo./gentls_cert create_server -cn alt DNS:localhost -org This will generate a self signed certificate in /usr/local/freeswitch/conf/ssl/ca/cakey.pem. Important: Self-signed certificates are not entirely secure and void the chain of trust. If you want care about security, please generate a certificate signed by an authority. Now, in /usr/local/freeswitch/conf/vars.xml, enable TLS: Original: <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/> <!-- Internal SIP Profile --> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <!-- External SIP Profile --> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> New: <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/> <!-- Internal SIP Profile --> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> <!-- External SIP Profile --> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> 22 Chapter 1. Contents
27 Now, in the Freeswitch shell, execute: reloadxml Back in /usr/local/freeswitch/bin, it is now time to create certificate for users: sudo./gentls_cert create_client -cn out 1002.pem Be sure to copy/scp the following certificates to all relevants users. Note: This How-to is always using the same computer as server and client for simplicity purpose. In a real context, this will rarely be the case. So you will have to safely upload the certificates to the client computer. To do this, the scp command allow you to create an encrypted upload of the key, thus, invulnerable to man in the middle attack. In this case, you can copy them using cp as it doesn t change anything. scp /usr/local/freeswitch/conf/ssl/ca/cacert.pem username@hostname:/home/username/ scp /usr/local/freeswitch/conf/ssl/agent.pem username@hostname:/home/username/ If you use an alternate upload method, please double check if the target file owned by the same used as the sflphone process (usually your current username) and have 600 permissions. The scp lines will automatically do that for you. Optional: In this how-to, we run Freeswitch as root. This, of course, as Freeswitch is a network facing application, is a potential attack vector. If you change Freeswitch user, do not forget to use: cd /usr/local/freeswitch/ find -iname conf/ssl/ xargs chown myfreeswitchuser:myfreeswitchuser Reference: Configuring SFLphone with Freeswitch Freeswitch already provides a few usable accounts you can use right away. They are numbered from 1000 to 1015 and the default password is The configuration files for every accounts are stored in /usr/local/freeswitch/conf/directory/default/. You can edit accounts individually Configuring SFLphone security Configuring a secure Freeswitch account is trivial. First, make sure you uploaded the cacert.pem and agent.pem as described in the earlier steps. Once this is done, create your account using Edit > Account > New and in the Security tab, check Use TLS transports and select ZRTP in SRTP key exchange. Now, in the Edit dialog, add both your cacert.pem and agent.pem and press OK Setup a secure environment with Freeswitch 23
28 Figure 1.9: Account settings for a Freeswitch account. 24 Chapter 1. Contents
29 CHAPTER 2 Indices and tables genindex modindex search 25
LABORATORIUM 1 Setup and basic configuration of Asterisk BPX on Linux
LABORATORIUM 1 Setup and basic configuration of Asterisk BPX on Linux 1. VM setup Please download Asterisk Virtual Machine from http://kt.agh.edu.pl/~rzym/lectures/ti- SSiZ/VMAsterisk.zip and extract archive.
Setup the Asterisk server with the Internet Gate
1 (9) Setup the Asterisk server with the Internet Gate This guide presents ways to setup the Asterisk server together with the Intertex Internet Gate. Below two different setups are described. Also, please
Skype connect and Asterisk
Skype connect and Asterisk General Configuration Guide Skype for SIP and Asterisk you are new to SIP, Asterisk is a useful, open-source (GPL) platform with which to test and experiment with the Skype for
Telephony with an Asterisk phone system
Telephony with an phone system TALKATIVE An old computer is all you need to build your own do-it-yourself personal phone server. BY MARTIN LOSCHWITZ Technology that supports the easy exchange of audio
Unifying Information Security. Implementing TLS on the CLEARSWIFT SECURE Email Gateway
Unifying Information Security Implementing TLS on the CLEARSWIFT SECURE Email Gateway Contents 1 Introduction... 3 2 Understanding TLS... 4 3 Clearswift s Application of TLS... 5 3.1 Opportunistic TLS...
Mediatrix 3000 with Asterisk June 22, 2011
Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration
Configuration Notes 290
Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...
Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment
Application Note May 2009 Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment 2009 Cisco Systems, Inc. All rights reserved. Page 1 of 20 Contents Introduction 3 Audience 3 Scope
Quick Provisioning Guide for Third-Party PBX
Quick Provisioning Guide for Third-Party PBX Table of Contents Quick Provisioning Guide Table of Contents Chapter 1: Overview...1 Chapter 2: Asterisk Configuration...2 Creating a Phone Extension on Asterisk...2
Using Polycom KIRK Wireless Server 300 or 6000 with Asterisk
Using Polycom KIRK Wireless Server 300 or 6000 with Asterisk Technical Bulletin Version 10 l August 2010 l 14205500 Introduction This document provides introductory information on how to use a Polycom
Micronet VoIP Solution with Asterisk
Application Note Micronet VoIP Solution with Asterisk 1. Introduction This is the document for the applications between Micronet units and Asterisk IP PBX. It will show you some basic configurations in
Atcom MP01 and Elastix Server
Atcom MP01 and Elastix Server Setup Guide http://www.elastix.org 1.0 Setup Diagram This is a setup diagram for a mesh network of Atcom MP01 configuration. When everything is configured we ll be able to
Basic configuration of the GXW410x with Asterisk
Basic configuration of the GXW410x with Asterisk Please note that due to the customizable nature of both the GXW410x and Asterisk and the vast deployment possibilities, these instructions should be taken
Asterisk with Twilio Elastic SIP Trunking Interconnection Guide using Secure Trunking (SRTP/TLS)
Asterisk with Twilio Elastic SIP Trunking Interconnection Guide using Secure Trunking (SRTP/TLS) With the Introduction of Twilio Elastic SIP trunking this guide provides the configuration steps required
Test on IX130 Performance
Test on IX130 Performance This document aims to report the performance of IX130. We design a few scenarios where uses IX130, including gateway and PBX and how well the performance of handling transcoding.
Ryan Brown October 9, 2004 The Burgh Live, LLC. Voice over IP using Asterisk (*)
Ryan Brown October 9, 2004 The Burgh Live, LLC Voice over IP using Asterisk (*) What is Asterisk? * (http://www.asterisk.org www.asterisk.org) ) is an Open Source Private Branch Exchange (PBX) and Interactive
Figure 38-1. The scenario
38. Asterisk Application We offer the application shows that it is convenient and cost saving to implement the free IP-PBX using Asterisk and Vigor 3300V when users want to use the Soft Phone or IP Phone
Using the GS8 Modular Gateway with Asterisk
Zed-3 501 Valley Way Milpitas CA 95035 Using the GS8 Modular Gateway with Asterisk Application note, 96-90002-02, May 2008 USA Voice: +1-408-587-9333 Fax: +1-408-586-9038 www.zed-3.com This document is
Creating a DUO MFA Service in AWS
Amazon AWS is a cloud based development environment with a goal to provide many options to companies wishing to leverage the power and convenience of cloud computing within their organisation. In 2013
Applications between Asotel VoIP and Asterisk
Applications between Asotel VoIP and Asterisk This document is describing the configuring manner of registering and communicating with Asterisk only. Please visit the official WEB of Asterisk http://www.asterisk,
NodePhone Business Trunks User Manual
NodePhone Business Trunks User Manual Contents NodePhone Business Trunks 2 Features 2 Sip Trunking Explained 3 What do I need 3 Costs 3 Additional costs 4 How much bandwidth do I need? 5 Technical information
Department of Communications and Networking. S-38.2131/3133 Networking Technology, laboratory course A/B
Department of Communications and Networking S-38.2131/3133 Networking Technology, laboratory course A/B Work Number 29: VoIP Student Edition Preliminary Exercises and Laboratory Assignments Original document
Quick Start Guide. Cerberus FTP is distributed in Canada through C&C Software. Visit us today at www.ccsoftware.ca!
Quick Start Guide Cerberus FTP is distributed in Canada through C&C Software. Visit us today at www.ccsoftware.ca! How to Setup a File Server with Cerberus FTP Server FTP and SSH SFTP are application protocols
NOC Workshop VoIP in the NOC labs SANOG10
NOC Workshop VoIP in the NOC labs SANOG10 New Delhi, India August 29 - September 2, 2007 Page 1 of 10 Lab Summary NOC Workshop, SANOG10 - VoIP in the NOC We only have limited time for this portion of the
Motorola TEAM WS M Configuring Asterisk PBX Integration
Motorola TEAM WS M Configuring Asterisk PBX Integration Objective The purpose of this document is to provide a guideline on how to configure the WSM/TEAM software as well as an Asterisk-based PBX in order
TEL 500 WRITE UP WEEK 8 FREE PBX SIP LAB SUBMITTED TO: PROF. RONNY BULL BY: ANUSHA ALIGAPALLY
TEL 500 WRITE UP WEEK 8 FREE PBX SIP LAB SUBMITTED TO: PROF. RONNY BULL BY: ANUSHA ALIGAPALLY DATE: 11/05/2014 ABSTRACT: Private Branch Exchange has multiple phones connected to it which are in the same
VoIP Workshop PacNOG3
VoIP Workshop PacNOG3 Rarotonga, Cook Islands June 2007 Labs 1-4, Asterisk Lab 5, INOC-DBA Lab 6-7, Cisco Voice Gateways Lab 8, CODECS Page 1 of 13 Lab Summary Server logins are as you have set up in previous
Software: Sjphone and X-Lite softphones, Redhat Linux 9, Asterisk (downloaded current version mid June 03)
Asterisk IAX - a Newbie s Struggle Produces a How-To IRC: [email protected]/room:#Asterisk E-mail: [email protected] Overview Intrigued by the proposition of an Open Source PBX, I jumped into
Asterisk: The Open Source PBX Solution Adam Olson Systems and network administrators typically deal with
1 1 1 0 1 0 1 0 1 Asterisk: The Open Source PBX Solution Adam Olson Systems and network administrators typically deal with data and functionality such as email communications, Web and database applications,
Configuring Bria 3 Mac for Virtual Contact Center
Configuring Bria 3 Mac for Virtual Contact Center Counterpath s Bria 3 is a softphone application that enables you to manage your Virtual Contact Center VOIP calls easily from your desktop replacing or
VoIP Laboratory C VoIP Billling in a Village Telco
VoIP Laboratory C VoIP Billling in a Village Telco (cc) Creative Commons Share Alike Non Commercial Attribution 3 The Village Telco is composed by four major components: (1) a mesh potato, that acts as
Ciphermail Gateway Separate Front-end and Back-end Configuration Guide
CIPHERMAIL EMAIL ENCRYPTION Ciphermail Gateway Separate Front-end and Back-end Configuration Guide June 19, 2014, Rev: 8975 Copyright 2010-2014, ciphermail.com. CONTENTS CONTENTS Contents 1 Introduction
MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment)
MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment) N.B. Goto MyIC Preferences in the System Toolbar. Description: this may be any appropriate description of the
Using LifeSize Systems with Microsoft Office Communications Server 2007
Using LifeSize Systems with Microsoft Office Communications Server 2007 This technical note describes the steps to integrate a LifeSize video communications device with Microsoft Office Communication Server
Avaya IP Office SIP Configuration Guide
Valcom Session Initiation Protocol (SIP) VIP devices are compatible with SIP-enabled versions of Avaya IP Office (5.0 and higher). The Valcom device is added to the IP Office system as a SIP Extension.
AlienVault Unified Security Management (USM) 4.x-5.x. Deploying HIDS Agents to Linux Hosts
AlienVault Unified Security Management (USM) 4.x-5.x Deploying HIDS Agents to Linux Hosts USM 4.x-5.x Deploying HIDS Agents to Linux Hosts, rev. 2 Copyright 2015 AlienVault, Inc. All rights reserved. AlienVault,
Configuring a Softphone for Windows for Virtual Contact Center
Configuring Bria 3 for Virtual Contact Center Configuring a Softphone for Windows for Virtual Contact Center 8x8 issues softphone licenses to Virtual Contact Center customers allowing them to manage all
Kerio Operator. Getting Started Guide
Kerio Operator Getting Started Guide 2011 Kerio Technologies. All rights reserved. 1 About Kerio Operator Kerio Operator is a PBX software for small and medium business customers. Kerio Operator is based
Using LifeSize systems with Microsoft Office Communications Server 2007. Server Setup
Using LifeSize systems with Microsoft Office Communications Server 2007 This technical note describes the steps to integrate a LifeSize video communications device with Microsoft Office Communication Server
SyncSwitch Quick Start Guide For Making First Test Call V 2.1
SyncSwitch Quick Start Guide For Making First Test Call V 2.1 This version is a revision of V. 2.0, some gramatical mistakes has been corrected on this version. There are no major changes between V.2.0
IPChitChat VoIP Service User Manual
IPChitChat VoIP Service User Manual Document Owner: Netcloud Ltd Prepared By: Michael Date of Issue: 11 th June 2011 Version: V0.5 Copyright 2009 Netcloud Ltd Page 1 of 31 Netcloud are UK specialists in
A SHORT INTRODUCTION TO DUPLICITY WITH CLOUD OBJECT STORAGE. Version 1.12 2014-07-01
A SHORT INTRODUCTION TO DUPLICITY WITH CLOUD OBJECT STORAGE Version 1.12 2014-07-01 PAGE _ 2 TABLE OF CONTENTS 1. Introduction....Page 03 2. System Configuration....Page 04 3. Create Backup Script....Page
Asterisk. Technical Application Notes
Asterisk Technical Application Notes Table of Contents About Asterisk... 1 Purpose, Scope and Audience... 3 Asterisk Deployment Information... 4 Asterisk External IP Address... 4 Sending Calls to Broadvox...
1. Product Information
ORIXCLOUD BACKUP CLIENT USER MANUAL LINUX 1. Product Information Product: Orixcloud Backup Client for Linux Version: 4.1.7 1.1 System Requirements Linux (RedHat, SuSE, Debian and Debian based systems such
VOIP with Asterisk & Perl
VOIP with Asterisk & Perl By: Mike Frager 11/2011 The Elements of PSTN - Public Switched Telephone Network, the pre-internet phone system: land-lines & cell-phones. DID - Direct
1 You will need the following items to get started:
QUICKSTART GUIDE 1 Getting Started You will need the following items to get started: A desktop or laptop computer Two ethernet cables (one ethernet cable is shipped with the _ Blocker, and you must provide
How To Connect A Gemalto To A Germanto Server To A Joniper Ssl Vpn On A Pb.Net 2.Net 3.5.1 (Net 2) On A Gmaalto.Com Web Server
Application Note: Integrate Juniper SSL VPN with Gemalto SA Server [email protected] October 2007 www.gemalto.com Table of contents Table of contents... 2 Overview... 3 Architecture... 5 Configure
Secured Communications using Linphone & Flexisip
Secured Communications using Linphone & Flexisip Solution description Office: Le Trident Bat D 34, avenue de l Europe 38100 Grenoble France Tel. : +33 (0)9 52 63 65 05 Headquarters: 12, allée des Genêts
Free Dynamic DNS account you can use one of your choosing I like DynDNS but there's also No-IP and probably others.
1 of 7 3/26/2009 2:01 PM The 'Point and Click' Home VPN HowTo Guide contact: beakmyn frontiernet net The 'Point and Click' Home VPN HowTo Guide by beakmyn is licensed under a Creative Commons
Guideline for SIP Trunk Setup
Guideline for SIP Trunk Setup with ZONETEL Table of contents Sample sip.conf (it applies to asterisk 1.4.x)...3 Sample elastix setup... 3 Ports required... 4 Caller ID...4 FAQ... 5 After i dial out, the
Software Based VoIP Lab A step by step guide to setting up and configuring an IP-PBX. Donal O Connor DNET 4 [email protected]
Software Based VoIP Lab A step by step guide to setting up and configuring an IP-PBX Donal O Connor DNET 4 [email protected] Introduction Traditionally, a company or individual would have to buy really
Interactive Intelligence CIC 2015 R4 Patch1 Configuration Guide
Interactive Intelligence CIC 2015 R4 Patch1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Copyright 2015 by tekvizion PVS, Inc. All Rights Reserved. Confidential
MassTransit 6.0 Enterprise Web Configuration for Macintosh OS 10.5 Server
MassTransit 6.0 Enterprise Web Configuration for Macintosh OS 10.5 Server November 6, 2008 Group Logic, Inc. 1100 North Glebe Road, Suite 800 Arlington, VA 22201 Phone: 703-528-1555 Fax: 703-528-3296 E-mail:
Information & Communication Technologies FTP and GroupWise Archives Wilfrid Laurier University
Instructions for MAC users MAC users have not been joined to the Active Directory Domain. ICT is unable to capture GroupWise archives you may have on your computer automatically or confirm if you have
RecoveryVault Express Client User Manual
For Linux distributions Software version 4.1.7 Version 2.0 Disclaimer This document is compiled with the greatest possible care. However, errors might have been introduced caused by human mistakes or by
Updated Since : 2007-02-09
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Avaya S8300 with AudioCodes Mediant 2000 using T1 CAS (In-band DTMF Tones) By : AudioCodes Updated Since : 2007-02-09 READ THIS
VOIP (Voice Over Internet Protocol) Hacking-Fake Calling
VOIP (Voice Over Internet Protocol) Hacking-Fake Calling Author: Avinash Singh Co-Author: Akash Shukla Avinash Singh Corporate Trainer (Virscent Technologies Pvt. Ltd.) Appin Certified Ethical Hacker (ACEH)
How-To Feature Guide. SIP Peering
How-To Feature Guide SIP Peering What is SIP Peering? Sometimes called SIP Trunking SIP Peering allows us to deliver your 2talk services to your SIP-based private branch exchange (IP-PBX) and Unified Communications
TekSIP Proxy frontend for Asterisk PBX
TekSIP Proxy frontend for Asterisk PBX What you need: IP PBX--AsteriskNow 3.x Asterisk 1.8 Asterisk 10.x Asterisk 11.x and Proxy--TekSIP 3.4.7 TekSIP 3.4.8 Install Asterisk with or without SRTP support
Cloud Control Panel (CCP) Installation Guide
Cloud Control Panel (CCP) Installation Guide Version 3.2.0: 17.05.12 Copyright 2012 DNS Europe Ltd. All rights reserved. Cloud Control Panel (CCP) Installation Guide v3.2.0 Table of Contents Table of Contents
Basic Exchange Setup Guide
Basic Exchange Setup Guide The following document and screenshots are provided for a single Microsoft Exchange Small Business Server 2003 or Exchange Server 2007 setup. These instructions are not provided
LoadMaster SSL Certificate Quickstart Guide
LoadMaster SSL Certificate Quickstart Guide for the LM-1500, LM-2460, LM-2860, LM-3620, SM-1020 This guide serves as a complement to the LoadMaster documentation, and is not a replacement for the full
How To Set Up An Onnet Fax On Ancienta.Com (Sagemcom) On A Pc Or Macodeo (Sagecom) With A Cell Phone Or Ipo (Omf) On An Ipo 2 (
Configuration Guide BUSINESS TALK IP SIP SAGEMCOM XMediusFax Server Enterprise 7.0.0.264 Version Version du 11/03/2013 1 of 21 Sommaire Configuration Guide Business Talk IP 1 Document objectives............3
Updated Since : 2007-02-09
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Siemens HiPath3550 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-09 READ
Online Backup Linux Client User Manual
Online Backup Linux Client User Manual Software version 4.0.x For Linux distributions August 2011 Version 1.0 Disclaimer This document is compiled with the greatest possible care. However, errors might
Online Backup Client User Manual Linux
Online Backup Client User Manual Linux 1. Product Information Product: Online Backup Client for Linux Version: 4.1.7 1.1 System Requirements Operating System Linux (RedHat, SuSE, Debian and Debian based
Setting up Hyper-V for 2X VirtualDesktopServer Manual
Setting up Hyper-V for 2X VirtualDesktopServer Manual URL: www.2x.com E-mail: [email protected] Information in this document is subject to change without notice. Companies, names, and data used in examples herein
Online Backup Client User Manual
For Linux distributions Software version 4.1.7 Version 2.0 Disclaimer This document is compiled with the greatest possible care. However, errors might have been introduced caused by human mistakes or by
Shellshock Security Patch for X86
Shellshock Security Patch for X86 Guide for Using the FFPS Update Manager October 2014 Version 1.0. Page 1 Page 2 This page is intentionally blank Table of Contents 1.0 OVERVIEW - SHELLSHOCK/BASH SHELL
LifeSize UVC Multipoint Deployment Guide
LifeSize UVC Multipoint Deployment Guide May 2014 LifeSize UVC Multipoint Deployment Guide 2 LifeSize UVC Multipoint LifeSize UVC Multipoint is a software MCU optimized for conferences that mix high definition
Internet telephony Asterisk system.
Internet telephony Asterisk system. Until recently, only large institutions were able to afford their own telephone exchange. The commercial solutions that were available were based on closed proprietary
This works very well for situations where all computers are within the same LAN and can access both the SQL server and the network shares.
AircastDB Server A networked AircastDB setup involves two types of servers: An SQL server (PostgreSQL, MSSQL) to hold the metadata for the audio files and scheduling information (library, playlists) One
Clearswift Information Governance
Clearswift Information Governance Implementing the CLEARSWIFT SECURE Encryption Portal on the CLEARSWIFT SECURE Email Gateway Version 1.10 02/09/13 Contents 1 Introduction... 3 2 How it Works... 4 3 Configuration
General Guidelines for SIP Trunking Installations
General Guidelines for SIP Trunking Installations 1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: IP Authentication (IP address) or Digest Authentication
IBM WebSphere Application Server Communications Enabled Applications Setup guide
Copyright IBM Corporation 2009, 2011 All rights reserved IBM WebSphere Application Server Communications Enabled Applications Setup guide What this exercise is about... 1 Lab requirements... 2 What you
Practical Guide. How to setup VoIP Infrastructure using AsteriskNOW
Practical Guide How to setup VoIP Infrastructure using AsteriskNOW Table of Contents 1. Background...1 2. The VoIP scenarios...2 3. Before getting started...3 3.1 Training Kits...3 3.2 Software requirements...3
Quick Start Guide v1.0
Quick Start Guide v1.0 Table of contents : 01. Quick Start Guide...03 O2. Configuring your VoIPOffice appliance...14 03. Adding a VoIPtalk trunk...21 04. Configuring UADs for use with VoIPOffice...25 05.
Table 28-1. Example 1-basic settings in Vigor 3300V and 2900V. WAN IP Port Number Phone Number Proxy Codec
28. VoIP Example 1 (Basic Configuration and Registration) There are many different kinds of applications about VoIP function, most of VoIP callings must be via a VoIP Server by registering, except we can
Abstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP IP Telephony Using Avaya 4600 Series IP Telephones, Avaya one-x Desktop Edition, and Asterisk Business Edition PBX Issue
F-Secure Messaging Security Gateway. Deployment Guide
F-Secure Messaging Security Gateway Deployment Guide TOC F-Secure Messaging Security Gateway Contents Chapter 1: Deploying F-Secure Messaging Security Gateway...3 1.1 The typical product deployment model...4
1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by:
1) How do I setup my SIP trunk for inbound/outbound calling? We authenticate IP-PBX SIP Trunking traffic by: IP Authentication (IP address) or Digest Authentication (account and SIP password) After you
Unicorn60x0 IP ANALOG GATEWAY ASTERISK CONFIGURATION
Unicorn60x0 IP ANALOG GATEWAY ASTERISK CONFIGURATION BASIC CONFIGURATION OF THE Unicorn60x0 WITH ASTERISK Due to the various deployment possibilities of the Unicorn60x0 and Asterisk, this configuration
NAS 323 Using Your NAS as a VPN Server
NAS 323 Using Your NAS as a VPN Server Use your NAS as a VPN Server and connect to it using Windows and Mac A S U S T O R C O L L E G E COURSE OBJECTIVES Upon completion of this course you should be able
Updated Since : 2007-02-18
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Panasonic KX-TDA200 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-18 READ
Overview of Asterisk (*) Jeff Gunther
Overview of Asterisk (*) Jeff Gunther Agenda Background Introduction to Asterisk and review the core components of it s architecture. Exploration of Asterisk s telephony and call features. Review some
Installing and Configuring vcloud Connector
Installing and Configuring vcloud Connector vcloud Connector 2.7.0 This document supports the version of each product listed and supports all subsequent versions until the document is replaced by a new
NovaBACKUP. Storage Server. NovaStor / May 2011
NovaBACKUP Storage Server NovaStor / May 2011 2011 NovaStor, all rights reserved. All trademarks are the property of their respective owners. Features and specifications are subject to change without notice.
PortGo 6.0 for Wndows User Guide
PortGo 6.0 for Wndows User Guide PortSIP Solutions, Inc. [email protected] http:// @May 20, 2010 PortSIP Solutions, Inc. All rights reserved. This User guide for PortGo Softphone 6.0. 1 Table of Contents
Customer Tips. Xerox Network Scanning HTTP/HTTPS Configuration using Microsoft IIS. for the user. Purpose. Background
Xerox Multifunction Devices Customer Tips June 5, 2007 This document applies to these Xerox products: X WC Pro 232/238/245/ 255/265/275 for the user Xerox Network Scanning HTTP/HTTPS Configuration using
How To Enable A Websphere To Communicate With Ssl On An Ipad From Aaya One X Portal 1.1.3 On A Pc Or Macbook Or Ipad (For Acedo) On A Network With A Password Protected (
Avaya one X Portal 1.1.3 Lightweight Directory Access Protocol (LDAP) over Secure Socket Layer (SSL) Configuration This document provides configuration steps for Avaya one X Portal s 1.1.3 communication
Online Backup Client User Manual
Online Backup Client User Manual Software version 3.21 For Linux distributions January 2011 Version 2.0 Disclaimer This document is compiled with the greatest possible care. However, errors might have
MyNetFone Virtual Fax. Virtual Fax Installation
Table of Contents MyNetFone Virtual Fax MyNetFone Virtual Fax Installation... 1 Changing the SIP endpoint details for the fax driver... 11 Uninstalling Virtual Fax... 13 Virtual Fax Installation Follow
Setting up VMware ESXi for 2X VirtualDesktopServer Manual
Setting up VMware ESXi for 2X VirtualDesktopServer Manual URL: www.2x.com E-mail: [email protected] Information in this document is subject to change without notice. Companies, names, and data used in examples
Installing Proview on an Windows XP machine
Installing Proview on an Windows XP machine This is a guide for the installation of Proview on an WindowsXP machine using VirtualBox. VirtualBox makes it possible to create virtual computers and allows
Updated Since : 2007-02-18
Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Panasonic KX-TES824 with AudioCodes MP-11x FXO using Analog lines (In-band DTMF) By : AudioCodes Updated Since : 2007-02-18 READ
USING SSL/TLS WITH TERMINAL EMULATION
USING SSL/TLS WITH TERMINAL EMULATION This document describes how to install and configure SSL or TLS support and verification certificates for the Wavelink Terminal Emulation (TE) Client. SSL/TLS support
1 SIP Carriers. 1.1 Tele2. 1.1.1 Warnings. 1.1.2 Vendor Contact. 1.1.3 Versions Verified Interaction Center 2015 R2 Patch1. 1.1.
1 SIP Carriers 1.1 Tele2 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
GlobalProtect Configuration for IPsec Client on Apple ios Devices
GlobalProtect Configuration for IPsec Client on Apple ios Devices Tech Note PAN-OS 4.1 Revision D 2013, Palo Alto Networks, Inc. www.paloaltonetworks.com CONTENTS OVERVIEW... 3 PREREQUISITES... 3 GLOBALPROTECT
Integrating Asterisk FreePBX with Lync Server 2010
1 Integrating Asterisk FreePBX with Lync Server 2010 Author: Baaskar R 1 www.baaskarcharles.com 2 Integrating Asterisk FreePBX with Lync Server 2010... 1 AsteriskNow package Source... 3 Installing AsteriskNow...
