SlideShare a Scribd company logo
1 of 18
Documentation: Implementing Xivo and a light VOIP
Infrastructure
1. Prerequisites and affected components…………………………p.2
2. Running the Wizard………………………………………………..p.2
3. Configuration……………………………………………………….p.3
4. Switch HV-SW06 Configuration………………………………….p.5
5. Configuration of Autoprov for the Cisco SPA512G…………….p.5
6. Create line, users and endpoints………………………………...p.6
7. Group creation……………………………………………………..p.8
8. Creation of Voicemail……………………………………………..p.8
9. Creation of conference room……………………………………..p.9
10. Push address book to endpoints………………………………..p.9
11. Configure call interception………………………………………p.11
12. Planning a Schedule…………………………………………….p.13
13. IVR Creation with text-to-speech………………………………p.14
14. Advanced user configuration…………………………………...p.15
15. Advanced phone provisioning via configuration files…………p.17
16. Security: Disable some features in the phone………………...p.18
1
1. Prerequisites and affected components.
- A virtual or physical machine able to meet the requirements of Xivo Server.
- A VOIP Phone compatible SIP
- A compatible switch that accept voice Vlan and that support PoE concerned by the
implementation (HV-SW06), here HP-2530-48G-PoEP
• Download the ISO image. (latest version) (all versions)
• Boot from the ISO image, select Install and follow the instructions. You must select a
locale with charset UTF-8.
• At the end of the installation, you can continue by running the configuration wizard.
2. Running the Wizard
After the system installation, you must go through the wizard before being able to use your
XiVO. Browse to your server’s IP address to start the configuration wizard.
We choosed http://10.182.3.68)
Language
-Choose English
License
-You then have to accept the GPLv3 License under which XiVO is distributed.
2
Accept the license
3. Configuration
3
Basic configuration
1. Enter the hostname: srv-voip
Enter the domain name: dinant.be
2. Enter the password for the root user of the web interface: root
3. Configure the IP address and gateway used by your XiVO (by default it pre-fills the
fields with the current IP and gateway of the network interface on which you are
connected if and only if network interface has a default gateway).
4. Finally, modify the DNS server information if needed.
Entities and Contexts
Contexts are used for managing various phone numbers that are used by your system.
1. Enter the entity name: dinant.be
2. Enter the number interval for you internal context. The interval will define the users’s
phone numbers for your system: 100-900
3. Enter the DID range and DID length for your system: none here because we don’t
have a SIP line for external calls.
4. You may change the name of your outgoing calls context but unchanged for us
5. Validate
4
4.Switch HV-SW06 Configuration
5.Configuration of Autoprov for the Cisco SPA512G
1. Go to the Configuration tab
2. Provisionning > Plugins >
3. Update the plugins list
4. Install the plugins and the language files corresponding to your VOIP Phone, here for
SPA512G:
5. If the plugin matching your phone is not in the list you must add a deposit through Xivo
Interface.
Identify the plugin needed by your phone: you’ll find a list here :
http://documentation.xivo.fr/production/administration/provisioning/basic_configuration.html#
alternative-plugins-repository
Change the deposit URL via Configuration tab > Provisionning > General.
5
6. Update one more time the plugins list and install concerned plugin
7. Add a network interface to distinguish voice from data in Network> Interface section
and to add a gateway to data interface.
8. Configure DHCP range to automatically assign IP Addresses for provisioning the
phone.
 Here: 10.182.150.1  10.182.150.200
9. In some case you may modify DHCP Configuration to approve the phone to get its IP
address by DHCP.
open terminal and goto: /etc/xivo/custom-templates/dhcp/etc/dhcp, modify the file
dhcpd_subnet.conf.middle and approve the phone : here we approved Cisco SPA 512G
6.Create line, users and endpoints
To autoinstall endpoint you must, plug the phone into the switch, wait a minute
so autoprovisionning will begin.
This one must appear in Xivo in the IPBX Settings > Endpoints
IP Address has been automatically assigned to it.
6
For the line :
Go to Service tab then choose IPBX
Go to IPBX Settings > Line …
Add a SIP line (+)
Choose the context, here we’ve only made configuration of internal call
Note the ID and the password in case of using a Softphone application like X-Lite for
Example
For the Users in the same menu, choose users
Add and fill some field
In the same menu go to Lines , Protocol: SIP, Context: Internal calls, choose a number to be
called, add an endpoint if you want the user line being associated to the endpoint.
Choose an endpoint thanks to its MAC Address
7.Group creation
7
Create a call group, it will contain several users, so a same number will be used to call 2
users at the same time.
Here we’ve created 2 groups : Group 1 (1000) et Group 2 (2000)
Group 1 contains: Julien and Tel Cisco “users”
Group 2 contains : Alain and Johann users
8.Creation of voicemail
Create the voice mail, choose a name, here : Tel Cisco , a number (here 103), a context
(internal call in our case ) and eventually password (none selected in our case),
8
9.Creation of conference room
Create the conference room, choose a name, here: “conferenceroom”, a number (here
250),a PIN code (here 1111, it will be used to ask a code when user want to join the
conference) a context (internal call in our case ) and the maximum number of participants
10.Push address book to endpoint
IPBX > Address book
Add some hosts to the list, in our case we filled the field with entire subnet.
9
11.Configure call interception
Go to: Call management > Call interception…> Add (+)
Enter a name for the interception group
10
Configure interceptors. You can add users or a call group
11
Configure intercepted. You can add users or call group
12
12.Planning a schedule
IPBX > Call management>Schedules
Choose the entity: here dinant.be
The time zone: Here Europe/Brussels
In our case
Worked periods:
Periods not worked:
13
13.IVR Creation (Interactive Voice Response)
with text-to-speech
Google AGI Installation for TTS :
Run bash and download packages below for installation of TTS
- apt-get install perl libwww-perl sox mpg123
- cd /var/lib/asterisk/agi-bin
- wget --no-check-certificate https://raw.github.com/zaf/asterisk-
googletts/master/googletts.agi
- chmod +x googletts.agi
Go to IPBX Configuration > Configuration Files… and edit the file xivo-extrafeatures.conf like
below:
14
14.Advanced user configuration (call forwarding, voicemail
assignment…)
Go to IPBX Settings > Users and then non-response tab.
Here you have for possibilities: What to do in case of, no reply, if a user is busy, if line is
saturated, or if a call failed.
Here I have configured user Alain. If someone calls him, in case of non-response after 5
seconds, the call is automatically forwarded to a group 1. Alain and Johann are members of
this group, so if someone does not have some response from Alain. Alain and Johann will
ring at the same time
Forward a call after 5 sec of non-response to a group of users
15
Here I have configured the account of the cisco phone named “Tel Cisco”.
If the line is busy a WAV files is played. You can upload this file via menu Services IPBX>
sounds file.
You can also find them via terminal by typing : cd /var/lib/xivo/sounds
 Be Careful in order to do that the WAV files must be 8000KHZ,16bit,mono format
PCM encoded
 See also documentation on :
http://documentation.xivo.io/production/administration/sound_files.html
Play a WAV file if user is busy, note that the call not continue to ring anymore if user hang
up.
And much more …
16
15.Advanced phone provisioning via configuration files.
In our case you can access the main config file OF THE PHONE (spaconfig.xml) through the
web interface
In Xivo you can edit a similary file to push its configuration to the phone.
Here is its location : /var/lib/xivo-provd/plugins/xivo-cisco-spa-
7.5.5/var/tftpboot/spa-phone.xml
You could add and edit each fields as you want. All the settings configured will be
automatically provisionned to all the phone at the next resynchronisation.
There is another file with some letters and numbers, it is corresponding to MAC adress of the
phone, you can edit it if you want the settings be pushed specifically and only to this phone:
/var/lib/xivo-provd/plugins/xivo-cisco-spa-
7.5.5/var/tftpboot/a4934cfea742.xml
If you don’t find the field you want to push to the phone you will find it via web configuration
file of the phone like above: in our case http://10.182.250.1/admin/spaconfig.xml
Note: You can also retrieve keywords for configuration parameters by browsing the Xivo web
interface by pointing mouse on an interrogation point next to some fields
17
16.Security: Disable some features in the phone
For security reasons it could be interesting to disable some features in the VOIP Phone, here
Cisco SPA512 (it may change from a phone to another).
In order to do that you must modify one of the configuration file of Xivo that is pushed to the
phone (p.17)
To have the possibility to do that you must first add the fields:
<Phone-UI-readonly>Yes</Phone-UI-readonly>
<Phone-UI-user-mode>Yes</Phone-UI-user-mode>
You can restrict some features by adding the ua= parameters to field followed by its value
into quotation marks: ua=”[value]”
There are 3 values, ro for readonly, na for not available, rw for read-writeable
ro means that the user will see the feature but will not be able to change it
na means that the user won’t see the feature and will not be able to change it
rw means that the user will see the feature and will be able to change it
Attention : To make the possibility to hide and entire menu in the phone, you must apply the
na parameters to all submenu first.
In the example below we decided to hide most of the network parameters
18

More Related Content

What's hot

Ccnp collaboration plus module 1 chapter 5 cisco unified communication express
Ccnp collaboration plus module 1   chapter 5 cisco unified communication expressCcnp collaboration plus module 1   chapter 5 cisco unified communication express
Ccnp collaboration plus module 1 chapter 5 cisco unified communication expressFaisal Khan
 
Using Asterisk in a SIP softswitch
Using Asterisk in a SIP softswitchUsing Asterisk in a SIP softswitch
Using Asterisk in a SIP softswitchMonica McArthur
 
CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overview
CCIE Collaboration Lecture Chapter 4.4 voice gateway   cucm sip overviewCCIE Collaboration Lecture Chapter 4.4 voice gateway   cucm sip overview
CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overviewFaisal Khan
 
Bonrix Bulk voice call - Voice SMS Marketing Web Based Panel
Bonrix Bulk voice call - Voice SMS Marketing Web Based PanelBonrix Bulk voice call - Voice SMS Marketing Web Based Panel
Bonrix Bulk voice call - Voice SMS Marketing Web Based PanelRenish Ladani
 
#1 Basic VoIP Drupal Hands On Experience Webinar
#1 Basic VoIP Drupal Hands On Experience Webinar#1 Basic VoIP Drupal Hands On Experience Webinar
#1 Basic VoIP Drupal Hands On Experience WebinarMicky Metts
 
Ch02 TCP/IP Concepts Review
Ch02 TCP/IP Concepts ReviewCh02 TCP/IP Concepts Review
Ch02 TCP/IP Concepts Reviewphanleson
 
Flashmedia gateway docs_quickstart
Flashmedia gateway docs_quickstartFlashmedia gateway docs_quickstart
Flashmedia gateway docs_quickstartSerge Florov
 
Telephony Service Development on Asterisk Platform
Telephony Service Development on Asterisk PlatformTelephony Service Development on Asterisk Platform
Telephony Service Development on Asterisk PlatformHamid Fadishei
 
Introduction to VoIP using SIP
Introduction to VoIP using SIPIntroduction to VoIP using SIP
Introduction to VoIP using SIPKundan Singh
 
Apple’s facetime protocol
Apple’s facetime protocolApple’s facetime protocol
Apple’s facetime protocolIMTC
 
BlackHat Hacking - Hacking VoIP.
BlackHat Hacking - Hacking VoIP.BlackHat Hacking - Hacking VoIP.
BlackHat Hacking - Hacking VoIP.Sumutiu Marius
 
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونی
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونیاسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونی
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونیMohammad Reza Kamalifard
 
Apple Facetime Protocol
Apple Facetime ProtocolApple Facetime Protocol
Apple Facetime Protocolkshitijmehta23
 
Information on Basic Web Host Manager Setup
Information on Basic Web Host Manager SetupInformation on Basic Web Host Manager Setup
Information on Basic Web Host Manager SetupHTS Hosting
 

What's hot (20)

Introduction To SIP
Introduction  To  SIPIntroduction  To  SIP
Introduction To SIP
 
POSITECH 4 Line Telephone Call Recording Device
POSITECH 4 Line Telephone Call Recording DevicePOSITECH 4 Line Telephone Call Recording Device
POSITECH 4 Line Telephone Call Recording Device
 
Ccnp collaboration plus module 1 chapter 5 cisco unified communication express
Ccnp collaboration plus module 1   chapter 5 cisco unified communication expressCcnp collaboration plus module 1   chapter 5 cisco unified communication express
Ccnp collaboration plus module 1 chapter 5 cisco unified communication express
 
Using Asterisk in a SIP softswitch
Using Asterisk in a SIP softswitchUsing Asterisk in a SIP softswitch
Using Asterisk in a SIP softswitch
 
CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overview
CCIE Collaboration Lecture Chapter 4.4 voice gateway   cucm sip overviewCCIE Collaboration Lecture Chapter 4.4 voice gateway   cucm sip overview
CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overview
 
No More Fraud Cluecon2014
No More Fraud Cluecon2014No More Fraud Cluecon2014
No More Fraud Cluecon2014
 
Bonrix Bulk voice call - Voice SMS Marketing Web Based Panel
Bonrix Bulk voice call - Voice SMS Marketing Web Based PanelBonrix Bulk voice call - Voice SMS Marketing Web Based Panel
Bonrix Bulk voice call - Voice SMS Marketing Web Based Panel
 
#1 Basic VoIP Drupal Hands On Experience Webinar
#1 Basic VoIP Drupal Hands On Experience Webinar#1 Basic VoIP Drupal Hands On Experience Webinar
#1 Basic VoIP Drupal Hands On Experience Webinar
 
Ch02 TCP/IP Concepts Review
Ch02 TCP/IP Concepts ReviewCh02 TCP/IP Concepts Review
Ch02 TCP/IP Concepts Review
 
Flashmedia gateway docs_quickstart
Flashmedia gateway docs_quickstartFlashmedia gateway docs_quickstart
Flashmedia gateway docs_quickstart
 
Introduction to SIP
Introduction to SIP  Introduction to SIP
Introduction to SIP
 
Telephony Service Development on Asterisk Platform
Telephony Service Development on Asterisk PlatformTelephony Service Development on Asterisk Platform
Telephony Service Development on Asterisk Platform
 
Introduction to VoIP using SIP
Introduction to VoIP using SIPIntroduction to VoIP using SIP
Introduction to VoIP using SIP
 
Asterisk Phone Systems
Asterisk Phone SystemsAsterisk Phone Systems
Asterisk Phone Systems
 
Apple’s facetime protocol
Apple’s facetime protocolApple’s facetime protocol
Apple’s facetime protocol
 
BlackHat Hacking - Hacking VoIP.
BlackHat Hacking - Hacking VoIP.BlackHat Hacking - Hacking VoIP.
BlackHat Hacking - Hacking VoIP.
 
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونی
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونیاسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونی
اسلاید اول جلسه یازدهم کلاس پایتون برای هکرهای قانونی
 
SIPob Manual
SIPob ManualSIPob Manual
SIPob Manual
 
Apple Facetime Protocol
Apple Facetime ProtocolApple Facetime Protocol
Apple Facetime Protocol
 
Information on Basic Web Host Manager Setup
Information on Basic Web Host Manager SetupInformation on Basic Web Host Manager Setup
Information on Basic Web Host Manager Setup
 

Viewers also liked

Tournaments Provide Tradition and Opportunity (A Future for the Sport)
Tournaments Provide Tradition and Opportunity (A Future for the Sport)Tournaments Provide Tradition and Opportunity (A Future for the Sport)
Tournaments Provide Tradition and Opportunity (A Future for the Sport)United States Bowling Congress
 
Customer Testimonial: Fingrid
Customer Testimonial: FingridCustomer Testimonial: Fingrid
Customer Testimonial: FingridNordSafety
 
Doss assistive technology
Doss assistive technologyDoss assistive technology
Doss assistive technologyrobertsong326
 
Water and Waste Water Treatment - EN - 140716 - webreduced
Water and Waste Water Treatment - EN - 140716 - webreducedWater and Waste Water Treatment - EN - 140716 - webreduced
Water and Waste Water Treatment - EN - 140716 - webreducedRenan Norbiate de Melo
 
Austin Dow-Smith Design Portfolio-Reduced PDF
Austin Dow-Smith Design Portfolio-Reduced PDFAustin Dow-Smith Design Portfolio-Reduced PDF
Austin Dow-Smith Design Portfolio-Reduced PDFAustin Dow-Smith
 
Alkalizirajte ili umrite
Alkalizirajte ili umriteAlkalizirajte ili umrite
Alkalizirajte ili umriteseki2012
 
Первый этап работы по исследовательским темам Международного Бакалавриата
Первый этап работы по исследовательским темам Международного БакалавриатаПервый этап работы по исследовательским темам Международного Бакалавриата
Первый этап работы по исследовательским темам Международного БакалавриатаIT1811
 
GEF4 - Continue to Share and Enjoy!
GEF4 - Continue to Share and Enjoy!GEF4 - Continue to Share and Enjoy!
GEF4 - Continue to Share and Enjoy!Alexander Nyßen
 
Нормативные документы по пожарной безопасности
Нормативные документы по пожарной безопасностиНормативные документы по пожарной безопасности
Нормативные документы по пожарной безопасностиIT1811
 

Viewers also liked (20)

5 AWARDS - Cover
5 AWARDS - Cover5 AWARDS - Cover
5 AWARDS - Cover
 
Update Portfolio De 2017
Update Portfolio De 2017Update Portfolio De 2017
Update Portfolio De 2017
 
Tournaments Provide Tradition and Opportunity (A Future for the Sport)
Tournaments Provide Tradition and Opportunity (A Future for the Sport)Tournaments Provide Tradition and Opportunity (A Future for the Sport)
Tournaments Provide Tradition and Opportunity (A Future for the Sport)
 
cv_ARIB
cv_ARIBcv_ARIB
cv_ARIB
 
14PtsdCrp
14PtsdCrp14PtsdCrp
14PtsdCrp
 
Customer Testimonial: Fingrid
Customer Testimonial: FingridCustomer Testimonial: Fingrid
Customer Testimonial: Fingrid
 
Doss assistive technology
Doss assistive technologyDoss assistive technology
Doss assistive technology
 
Paper 1-Senthil
Paper 1-SenthilPaper 1-Senthil
Paper 1-Senthil
 
Water and Waste Water Treatment - EN - 140716 - webreduced
Water and Waste Water Treatment - EN - 140716 - webreducedWater and Waste Water Treatment - EN - 140716 - webreduced
Water and Waste Water Treatment - EN - 140716 - webreduced
 
Austin Dow-Smith Design Portfolio-Reduced PDF
Austin Dow-Smith Design Portfolio-Reduced PDFAustin Dow-Smith Design Portfolio-Reduced PDF
Austin Dow-Smith Design Portfolio-Reduced PDF
 
NEW PPT 3.18.2015 FINAL
NEW PPT 3.18.2015 FINALNEW PPT 3.18.2015 FINAL
NEW PPT 3.18.2015 FINAL
 
Arvo Viltrop - African Swine Fever
Arvo Viltrop - African Swine FeverArvo Viltrop - African Swine Fever
Arvo Viltrop - African Swine Fever
 
Alkalizirajte ili umrite
Alkalizirajte ili umriteAlkalizirajte ili umrite
Alkalizirajte ili umrite
 
Eläinlääkäripalvelujen saatavuus ja kustannukset, Luonnonvarakeskus
Eläinlääkäripalvelujen saatavuus ja kustannukset, LuonnonvarakeskusEläinlääkäripalvelujen saatavuus ja kustannukset, Luonnonvarakeskus
Eläinlääkäripalvelujen saatavuus ja kustannukset, Luonnonvarakeskus
 
Первый этап работы по исследовательским темам Международного Бакалавриата
Первый этап работы по исследовательским темам Международного БакалавриатаПервый этап работы по исследовательским темам Международного Бакалавриата
Первый этап работы по исследовательским темам Международного Бакалавриата
 
GEF4 - Continue to Share and Enjoy!
GEF4 - Continue to Share and Enjoy!GEF4 - Continue to Share and Enjoy!
GEF4 - Continue to Share and Enjoy!
 
anwar_Cv
anwar_Cvanwar_Cv
anwar_Cv
 
Нормативные документы по пожарной безопасности
Нормативные документы по пожарной безопасностиНормативные документы по пожарной безопасности
Нормативные документы по пожарной безопасности
 
Tank Mixing Systems - EN - webreduced
Tank Mixing Systems - EN - webreducedTank Mixing Systems - EN - webreduced
Tank Mixing Systems - EN - webreduced
 
Issue 33
Issue 33Issue 33
Issue 33
 

Similar to Implementing Xivo VOIP and a light infrastructure

Setup VoIP System and Interconnection with LTE network
Setup VoIP System and Interconnection with LTE networkSetup VoIP System and Interconnection with LTE network
Setup VoIP System and Interconnection with LTE networkNazmul Hossain Rakib
 
3CX Basic Notes
3CX Basic Notes3CX Basic Notes
3CX Basic Noteskriz5
 
DLink-655 Router Configuration Guide for VoIP
DLink-655 Router Configuration Guide for VoIPDLink-655 Router Configuration Guide for VoIP
DLink-655 Router Configuration Guide for VoIPMyOwn Telco
 
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1Ozeki Informatics Ltd.
 
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docx
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docxNETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docx
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docxrosemarybdodson23141
 
Fanvil configuration guides_en
Fanvil configuration guides_enFanvil configuration guides_en
Fanvil configuration guides_enZIZI Yahia
 
Escene es620 series ip phone user manual en
Escene es620 series ip phone user manual enEscene es620 series ip phone user manual en
Escene es620 series ip phone user manual enEmre Ozcan
 
Yeastar Certified Technician S-Series Handouts
Yeastar Certified Technician S-Series HandoutsYeastar Certified Technician S-Series Handouts
Yeastar Certified Technician S-Series HandoutsDemeu Ltd.
 
VOIP Design & Implementation
VOIP Design & ImplementationVOIP Design & Implementation
VOIP Design & ImplementationAhmed A. Arefin
 
3CX Phone Admin Manual for Version 12
3CX Phone Admin Manual for Version 123CX Phone Admin Manual for Version 12
3CX Phone Admin Manual for Version 12Dave Norris
 
Voice Primer Lab.pdf
Voice Primer Lab.pdfVoice Primer Lab.pdf
Voice Primer Lab.pdfacaldere
 
First steps after free pbx installation
First steps after free pbx installationFirst steps after free pbx installation
First steps after free pbx installationvincent david
 
Ip атс grand stream ucm6102 functional overview and testing-eng
Ip атс grand stream ucm6102 functional overview and testing-engIp атс grand stream ucm6102 functional overview and testing-eng
Ip атс grand stream ucm6102 functional overview and testing-engVladimir Dudchenko
 
How to Use GSM/3G/4G in Embedded Linux Systems
How to Use GSM/3G/4G in Embedded Linux SystemsHow to Use GSM/3G/4G in Embedded Linux Systems
How to Use GSM/3G/4G in Embedded Linux SystemsToradex
 

Similar to Implementing Xivo VOIP and a light infrastructure (20)

Setup VoIP System and Interconnection with LTE network
Setup VoIP System and Interconnection with LTE networkSetup VoIP System and Interconnection with LTE network
Setup VoIP System and Interconnection with LTE network
 
3CX Basic Notes
3CX Basic Notes3CX Basic Notes
3CX Basic Notes
 
DLink-655 Router Configuration Guide for VoIP
DLink-655 Router Configuration Guide for VoIPDLink-655 Router Configuration Guide for VoIP
DLink-655 Router Configuration Guide for VoIP
 
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1
#1 How to develop a VoIP softphone in C# by using Ozeki VoIP SIP SDK - Part 1
 
Asterisk sip trunksetting
Asterisk sip trunksettingAsterisk sip trunksetting
Asterisk sip trunksetting
 
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docx
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docxNETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docx
NETW250 Week 2 iLab Avaya IP Office Phone SystemIntroduction.docx
 
Presentation
PresentationPresentation
Presentation
 
Fanvil configuration guides_en
Fanvil configuration guides_enFanvil configuration guides_en
Fanvil configuration guides_en
 
Escene es620 series ip phone user manual en
Escene es620 series ip phone user manual enEscene es620 series ip phone user manual en
Escene es620 series ip phone user manual en
 
Yeastar Certified Technician S-Series Handouts
Yeastar Certified Technician S-Series HandoutsYeastar Certified Technician S-Series Handouts
Yeastar Certified Technician S-Series Handouts
 
TekFAX Manual
TekFAX ManualTekFAX Manual
TekFAX Manual
 
Tekaba Manual
Tekaba ManualTekaba Manual
Tekaba Manual
 
VOIP Design & Implementation
VOIP Design & ImplementationVOIP Design & Implementation
VOIP Design & Implementation
 
3CX Phone Admin Manual for Version 12
3CX Phone Admin Manual for Version 123CX Phone Admin Manual for Version 12
3CX Phone Admin Manual for Version 12
 
Voice Primer Lab.pdf
Voice Primer Lab.pdfVoice Primer Lab.pdf
Voice Primer Lab.pdf
 
First steps after free pbx installation
First steps after free pbx installationFirst steps after free pbx installation
First steps after free pbx installation
 
Project Pt1
Project Pt1Project Pt1
Project Pt1
 
voip_en
voip_envoip_en
voip_en
 
Ip атс grand stream ucm6102 functional overview and testing-eng
Ip атс grand stream ucm6102 functional overview and testing-engIp атс grand stream ucm6102 functional overview and testing-eng
Ip атс grand stream ucm6102 functional overview and testing-eng
 
How to Use GSM/3G/4G in Embedded Linux Systems
How to Use GSM/3G/4G in Embedded Linux SystemsHow to Use GSM/3G/4G in Embedded Linux Systems
How to Use GSM/3G/4G in Embedded Linux Systems
 

Implementing Xivo VOIP and a light infrastructure

  • 1. Documentation: Implementing Xivo and a light VOIP Infrastructure 1. Prerequisites and affected components…………………………p.2 2. Running the Wizard………………………………………………..p.2 3. Configuration……………………………………………………….p.3 4. Switch HV-SW06 Configuration………………………………….p.5 5. Configuration of Autoprov for the Cisco SPA512G…………….p.5 6. Create line, users and endpoints………………………………...p.6 7. Group creation……………………………………………………..p.8 8. Creation of Voicemail……………………………………………..p.8 9. Creation of conference room……………………………………..p.9 10. Push address book to endpoints………………………………..p.9 11. Configure call interception………………………………………p.11 12. Planning a Schedule…………………………………………….p.13 13. IVR Creation with text-to-speech………………………………p.14 14. Advanced user configuration…………………………………...p.15 15. Advanced phone provisioning via configuration files…………p.17 16. Security: Disable some features in the phone………………...p.18 1
  • 2. 1. Prerequisites and affected components. - A virtual or physical machine able to meet the requirements of Xivo Server. - A VOIP Phone compatible SIP - A compatible switch that accept voice Vlan and that support PoE concerned by the implementation (HV-SW06), here HP-2530-48G-PoEP • Download the ISO image. (latest version) (all versions) • Boot from the ISO image, select Install and follow the instructions. You must select a locale with charset UTF-8. • At the end of the installation, you can continue by running the configuration wizard. 2. Running the Wizard After the system installation, you must go through the wizard before being able to use your XiVO. Browse to your server’s IP address to start the configuration wizard. We choosed http://10.182.3.68) Language -Choose English License -You then have to accept the GPLv3 License under which XiVO is distributed. 2
  • 3. Accept the license 3. Configuration 3
  • 4. Basic configuration 1. Enter the hostname: srv-voip Enter the domain name: dinant.be 2. Enter the password for the root user of the web interface: root 3. Configure the IP address and gateway used by your XiVO (by default it pre-fills the fields with the current IP and gateway of the network interface on which you are connected if and only if network interface has a default gateway). 4. Finally, modify the DNS server information if needed. Entities and Contexts Contexts are used for managing various phone numbers that are used by your system. 1. Enter the entity name: dinant.be 2. Enter the number interval for you internal context. The interval will define the users’s phone numbers for your system: 100-900 3. Enter the DID range and DID length for your system: none here because we don’t have a SIP line for external calls. 4. You may change the name of your outgoing calls context but unchanged for us 5. Validate 4
  • 5. 4.Switch HV-SW06 Configuration 5.Configuration of Autoprov for the Cisco SPA512G 1. Go to the Configuration tab 2. Provisionning > Plugins > 3. Update the plugins list 4. Install the plugins and the language files corresponding to your VOIP Phone, here for SPA512G: 5. If the plugin matching your phone is not in the list you must add a deposit through Xivo Interface. Identify the plugin needed by your phone: you’ll find a list here : http://documentation.xivo.fr/production/administration/provisioning/basic_configuration.html# alternative-plugins-repository Change the deposit URL via Configuration tab > Provisionning > General. 5
  • 6. 6. Update one more time the plugins list and install concerned plugin 7. Add a network interface to distinguish voice from data in Network> Interface section and to add a gateway to data interface. 8. Configure DHCP range to automatically assign IP Addresses for provisioning the phone.  Here: 10.182.150.1  10.182.150.200 9. In some case you may modify DHCP Configuration to approve the phone to get its IP address by DHCP. open terminal and goto: /etc/xivo/custom-templates/dhcp/etc/dhcp, modify the file dhcpd_subnet.conf.middle and approve the phone : here we approved Cisco SPA 512G 6.Create line, users and endpoints To autoinstall endpoint you must, plug the phone into the switch, wait a minute so autoprovisionning will begin. This one must appear in Xivo in the IPBX Settings > Endpoints IP Address has been automatically assigned to it. 6
  • 7. For the line : Go to Service tab then choose IPBX Go to IPBX Settings > Line … Add a SIP line (+) Choose the context, here we’ve only made configuration of internal call Note the ID and the password in case of using a Softphone application like X-Lite for Example For the Users in the same menu, choose users Add and fill some field In the same menu go to Lines , Protocol: SIP, Context: Internal calls, choose a number to be called, add an endpoint if you want the user line being associated to the endpoint. Choose an endpoint thanks to its MAC Address 7.Group creation 7
  • 8. Create a call group, it will contain several users, so a same number will be used to call 2 users at the same time. Here we’ve created 2 groups : Group 1 (1000) et Group 2 (2000) Group 1 contains: Julien and Tel Cisco “users” Group 2 contains : Alain and Johann users 8.Creation of voicemail Create the voice mail, choose a name, here : Tel Cisco , a number (here 103), a context (internal call in our case ) and eventually password (none selected in our case), 8
  • 9. 9.Creation of conference room Create the conference room, choose a name, here: “conferenceroom”, a number (here 250),a PIN code (here 1111, it will be used to ask a code when user want to join the conference) a context (internal call in our case ) and the maximum number of participants 10.Push address book to endpoint IPBX > Address book Add some hosts to the list, in our case we filled the field with entire subnet. 9
  • 10. 11.Configure call interception Go to: Call management > Call interception…> Add (+) Enter a name for the interception group 10
  • 11. Configure interceptors. You can add users or a call group 11
  • 12. Configure intercepted. You can add users or call group 12
  • 13. 12.Planning a schedule IPBX > Call management>Schedules Choose the entity: here dinant.be The time zone: Here Europe/Brussels In our case Worked periods: Periods not worked: 13
  • 14. 13.IVR Creation (Interactive Voice Response) with text-to-speech Google AGI Installation for TTS : Run bash and download packages below for installation of TTS - apt-get install perl libwww-perl sox mpg123 - cd /var/lib/asterisk/agi-bin - wget --no-check-certificate https://raw.github.com/zaf/asterisk- googletts/master/googletts.agi - chmod +x googletts.agi Go to IPBX Configuration > Configuration Files… and edit the file xivo-extrafeatures.conf like below: 14
  • 15. 14.Advanced user configuration (call forwarding, voicemail assignment…) Go to IPBX Settings > Users and then non-response tab. Here you have for possibilities: What to do in case of, no reply, if a user is busy, if line is saturated, or if a call failed. Here I have configured user Alain. If someone calls him, in case of non-response after 5 seconds, the call is automatically forwarded to a group 1. Alain and Johann are members of this group, so if someone does not have some response from Alain. Alain and Johann will ring at the same time Forward a call after 5 sec of non-response to a group of users 15
  • 16. Here I have configured the account of the cisco phone named “Tel Cisco”. If the line is busy a WAV files is played. You can upload this file via menu Services IPBX> sounds file. You can also find them via terminal by typing : cd /var/lib/xivo/sounds  Be Careful in order to do that the WAV files must be 8000KHZ,16bit,mono format PCM encoded  See also documentation on : http://documentation.xivo.io/production/administration/sound_files.html Play a WAV file if user is busy, note that the call not continue to ring anymore if user hang up. And much more … 16
  • 17. 15.Advanced phone provisioning via configuration files. In our case you can access the main config file OF THE PHONE (spaconfig.xml) through the web interface In Xivo you can edit a similary file to push its configuration to the phone. Here is its location : /var/lib/xivo-provd/plugins/xivo-cisco-spa- 7.5.5/var/tftpboot/spa-phone.xml You could add and edit each fields as you want. All the settings configured will be automatically provisionned to all the phone at the next resynchronisation. There is another file with some letters and numbers, it is corresponding to MAC adress of the phone, you can edit it if you want the settings be pushed specifically and only to this phone: /var/lib/xivo-provd/plugins/xivo-cisco-spa- 7.5.5/var/tftpboot/a4934cfea742.xml If you don’t find the field you want to push to the phone you will find it via web configuration file of the phone like above: in our case http://10.182.250.1/admin/spaconfig.xml Note: You can also retrieve keywords for configuration parameters by browsing the Xivo web interface by pointing mouse on an interrogation point next to some fields 17
  • 18. 16.Security: Disable some features in the phone For security reasons it could be interesting to disable some features in the VOIP Phone, here Cisco SPA512 (it may change from a phone to another). In order to do that you must modify one of the configuration file of Xivo that is pushed to the phone (p.17) To have the possibility to do that you must first add the fields: <Phone-UI-readonly>Yes</Phone-UI-readonly> <Phone-UI-user-mode>Yes</Phone-UI-user-mode> You can restrict some features by adding the ua= parameters to field followed by its value into quotation marks: ua=”[value]” There are 3 values, ro for readonly, na for not available, rw for read-writeable ro means that the user will see the feature but will not be able to change it na means that the user won’t see the feature and will not be able to change it rw means that the user will see the feature and will be able to change it Attention : To make the possibility to hide and entire menu in the phone, you must apply the na parameters to all submenu first. In the example below we decided to hide most of the network parameters 18