VoIP/AsteriskOnFedora - Historique des versions

Didier le 20 septembre 2013 à 08:12

2013-09-20T08:12:05Z

← Version précédente		Version du 20 septembre 2013 à 08:12
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	Open source telephony: a Fedora-based VoIP server with Asterisk		'''Open source telephony: a Fedora-based VoIP server with Asterisk'''
	~~{{author\|W. Michael Petullo}}~~

	Voice over Internet Protocol (VoIP) has emerged as a popular technology for modern voice communications. Many organizations have replaced their analog or proprietary digital telephone systems with VoIP-based solutions. This allows the consolidation of telephone services into an existing IP infrastructure. In addition, using IP to host voice services lets the organization leverage existing expertise–while retaining all of the network’s management advantages. Though not without its disadvantages, VoIP provides a compelling option to those looking for a telephone solution.		Voice over Internet Protocol (VoIP) has emerged as a popular technology for modern voice communications. Many organizations have replaced their analog or proprietary digital telephone systems with VoIP-based solutions. This allows the consolidation of telephone services into an existing IP infrastructure. In addition, using IP to host voice services lets the organization leverage existing expertise–while retaining all of the network’s management advantages. Though not without its disadvantages, VoIP provides a compelling option to those looking for a telephone solution.
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	This entry was posted by The editorial team on Thursday, July 24th, 2008 at 5:37 pm and is filed under [http://magazine.redhat.com/category/technical/ Fedora technical]. You can follow any responses to this entry through the RSS 2.0 feed. Both comments and pings are currently closed.		This entry was posted by The editorial team on Thursday, July 24th, 2008 at 5:37 pm and is filed under [http://magazine.redhat.com/category/technical/ Fedora technical]. You can follow any responses to this entry through the RSS 2.0 feed. Both comments and pings are currently closed.
			{{author\|W. Michael Petullo}}

Didier : /* Configuring the Ekiga VoIP client */

2013-09-20T08:01:03Z

Configuring the Ekiga VoIP client

← Version précédente		Version du 20 septembre 2013 à 08:01
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	Ekiga is a software VoIP application that supports SIP. Its configuration is straightforward. The first time Ekiga is executed by a user, it provides a wizard that allows one to configure the user’s full name,		Ekiga is a software VoIP application that supports SIP. Its configuration is straightforward. The first time Ekiga is executed by a user, it provides a wizard that allows one to configure the user’s full name,
	ekiga.net account (this is not required for the purpose of this article), connection type, audio devices, and video devices. Once the software is configured, setup a SIP account by clicking on Edit→Accounts and selecting add. Figure 1 displays an account configured for our SIP user identified as user1.		ekiga.net account (this is not required for the purpose of this article), connection type, audio devices, and video devices. Once the software is configured, setup a SIP account by clicking on '''Edit→Accounts''' and selecting add. Figure 1 displays an account configured for our SIP user identified as user1.

	~~asterisk-ldap-figure1-en~~		[[Fichier:EkigaSetupAsterisk1.png\|Ekiga Account Configuration]]
	~~Figure 1~~. Ekiga Account Configuration

	After an account is set up, it must be activated. Figure 2 shows that the account I just configured is displayed in Ekiga’s list. Click the checkbox to activate the account.		After an account is set up, it must be activated. Figure 2 shows that the account I just configured is displayed in Ekiga’s list. Click the checkbox to activate the account.

	~~asterisk-ldap-figure2-en~~		[[Fichier:EkigaSetupAsterisk2.png\|Ekiga Account Configuration]]
	~~Figure 2~~. Ekiga Account ~~Activation~~

	Figure 3 shows Ekiga’s main interface window. Here, you can see how many accounts have been registered. In order to test your configuration, make a SIP connection to extension 600 and perform an echo test.		Figure 3 shows Ekiga’s main interface window. Here, you can see how many accounts have been registered. In order to test your configuration, make a SIP connection to extension 600 and perform an echo test.

	~~asterisk-ldap-figure3-en~~		[[Fichier:EkigaSetupAsterisk3.png\|Ekiga Primary Interface]]
	~~Figure 3~~. Ekiga Primary Interface
	== Configuring a Cisco VoIP telephone ==		== Configuring a Cisco VoIP telephone ==

Didier le 20 septembre 2013 à 07:57

2013-09-20T07:57:27Z

← Version précédente		Version du 20 septembre 2013 à 07:57
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	This article demonstrated how to install Asterisk on a Fedora system and configure SIP users, extensions, and voicemail. We installed all user data into an LDAP database in order to keep it consolidated.		This article demonstrated how to install Asterisk on a Fedora system and configure SIP users, extensions, and voicemail. We installed all user data into an LDAP database in order to keep it consolidated.

	Asterisk has far too many features to cover in one article. The VoIP Wiki has a section dedicated to Asterisk and is a great place to learn more. The VoIP Wiki covers topics such as hooking Asterisk up to the standard public-switched telephone network using direct inward dialing and automatically emailing voicemail messages to users. You can even find information on how to route calls to a user based on his Jabber presence. Clearly, Asterisk is capable of a wide range of applications. And, perhaps best of all, it’s entirely open source!		Asterisk has far too many features to cover in one article. The VoIP Wiki has a [http://www.voip-info.org/wiki/view/Asterisk section dedicated to Asterisk] and is a great place to learn more. The VoIP Wiki covers topics such as hooking Asterisk up to the standard public-switched telephone network using direct inward dialing and automatically emailing voicemail messages to users. You can even find information on how to route calls to a user based on his Jabber presence. Clearly, Asterisk is capable of a wide range of applications. And, perhaps best of all, it’s entirely open source!

	This entry was posted by The editorial team on Thursday, July 24th, 2008 at 5:37 pm and is filed under Fedora, technical. You can follow any responses to this entry through the RSS 2.0 feed. Both comments and pings are currently closed.		This entry was posted by The editorial team on Thursday, July 24th, 2008 at 5:37 pm and is filed under [http://magazine.redhat.com/category/technical/ Fedora technical]. You can follow any responses to this entry through the RSS 2.0 feed. Both comments and pings are currently closed.

Didier : /* Configuring voicemail */

2013-09-20T07:55:43Z

Configuring voicemail

← Version précédente		Version du 20 septembre 2013 à 07:55
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	configuration engine, and the real-time LDAP back-end. After the configuration files are setup, we will add information to the LDAP database.		configuration engine, and the real-time LDAP back-end. After the configuration files are setup, we will add information to the LDAP database.

	/etc/asterisk/voicemail.conf is similar to sip.conf and		<path>/etc/asterisk/voicemail.conf</path> is similar to <path>sip.conf</path> and
	extensions.conf. This file configures the voicemail module. As with SIP users and extensions, we will configure global parameters in a file, leaving extension-specific parameters to be configured in LDAP:		<path>extensions.conf</path>. This file configures the voicemail module. As with SIP users and extensions, we will configure global parameters in a file, leaving extension-specific parameters to be configured in LDAP:
			<syntaxhighlight lang="ini">
	[general]		[general]
	searchcontexts=yes		searchcontexts=yes
			</syntaxhighlight>
	Add the following line to /etc/asterisk/extconfig.conf in order to activate the real-time LDAP back-end for voicemail:		Add the following line to <path>/etc/asterisk/extconfig.conf</path> in order to activate the real-time LDAP back-end for voicemail:
			<syntaxhighlight lang="ini">
	voicemail => ldap,"dc=example,dc=com",voicemail		voicemail => ldap,"dc=example,dc=com",voicemail
			</syntaxhighlight>

	~~Map Asterisk voicemail configuration options to LDAP attributes by adding the following to~~ /~~etc/asterisk/res_ldap.conf:~~

			Map Asterisk voicemail configuration options to LDAP attributes by adding the following to <path>/etc/asterisk/res_ldap.conf</path>:
			<syntaxhighlight lang="ini">
	[voicemail]		[voicemail]
	context = AstVMContext		context = AstVMContext
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	stamp = AstVMStamp		stamp = AstVMStamp
	additionalFilter = (objectClass=AsteriskVoicemail)		additionalFilter = (objectClass=AsteriskVoicemail)
			</syntaxhighlight>
	Now we will move to updating our LDAP database. First, we must add to the LDAP schema that we installed earlier. The following will add support for the attributes required to provide basic voicemail:		Now we will move to updating our LDAP database. First, we must add to the LDAP schema that we installed earlier. The following will add support for the attributes required to provide basic voicemail:
			<syntaxhighlight lang="perl">
	attributeTypes: (		attributeTypes: (
	NAME 'AstVMMailbox'		NAME 'AstVMMailbox'
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	$ AstVMEmail )		$ AstVMEmail )
	)		)
			</syntaxhighlight>
	Add the following LDIF to the LDAP database using ldapadd in order to add voicemail support for the extensions 2001 and 2002:		Add the following LDIF to the LDAP database using <app>ldapadd</app> in order to add voicemail support for the extensions 2001 and 2002:
			<syntaxhighlight lang="perl">
	dn: ou=voicemail,dc=example,dc=com		dn: ou=voicemail,dc=example,dc=com
	ou: voicemail		ou: voicemail
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	AstApplication: Voicemail		AstApplication: Voicemail
	AstApplicationData: 2002,u		AstApplicationData: 2002,u
			</syntaxhighlight>
	The second entry defines a mailbox for extension 2001. It uses the following options:		The second entry defines a mailbox for extension 2001. It uses the following options:

	AstContext		* '''AstContext''': The context in which this mailbox exists.
	The context in which this mailbox exists.		* '''AstVMMailboxt''': The extension that this mailbox corresponds to.
	~~AstVMMailbox~~		* '''AstVMPasswordt''': The password used to access the mailbox.
	The extension that this mailbox corresponds to.		* '''AstVMFullnamet''': The full name of the mailbox owner.
	~~AstVMPassword~~		* '''AstVMEmailt''': The mailbox owner’s email address.
	The password used to access the mailbox.
	~~AstVMFullname~~
	The full name of the mailbox owner.
	~~AstVMEmail~~
	The mailbox owner’s email address.

	Entity four defines an extension, 2000, that will provide access to the voicemail system when dialed.		Entity four defines an extension, 2000, that will provide access to the voicemail system when dialed.
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	Entity five adds a command to extension 2001 that executes the Voicemail and provides access to the voicemail system. This command will execute if the Dial command defined earlier is not answered within 20 seconds.		Entity five adds a command to extension 2001 that executes the Voicemail and provides access to the voicemail system. This command will execute if the Dial command defined earlier is not answered within 20 seconds.

	To test the voicemail system, use Ekiga to dial 2002. If you don’t answer the call on the other end, then you should be directed to Asterisk’s voicemail system. To listen to new voicemail, dial 2000.		To test the voicemail system, use Ekiga to dial '''2002'''. If you don’t answer the call on the other end, then you should be directed to Asterisk’s voicemail system. To listen to new voicemail, dial '''2000'''.

	== Conclusion ==		== Conclusion ==

Didier : /* Configuring a Cisco VoIP telephone */

2013-09-20T07:49:41Z

Configuring a Cisco VoIP telephone

← Version précédente		Version du 20 septembre 2013 à 07:49
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	As mentioned before, using a Cisco phone to connect to Asterisk using SIP requires Cisco’s SIP software. When a Cisco 7900 Series telephone boots, it will try to configure itself using TFTP. We will place some firmware and configuration files on our server so that the phone will boot properly for our installation.		As mentioned before, using a Cisco phone to connect to Asterisk using SIP requires Cisco’s SIP software. When a Cisco 7900 Series telephone boots, it will try to configure itself using TFTP. We will place some firmware and configuration files on our server so that the phone will boot properly for our installation.

	In order to enable TFTP on the Asterisk server, edit /etc/xinetd.d/tftp and set disable to no. The TFTP daemon is configured on Fedora to serve files out of /var/lib/tftpboot. After you have modified its configuration, restart xinetd with the command service xinetd restart. Cisco’s SIP software package provides the files P0S3-08-2-00.sb2, P0S3-08-2-00.sbn and P0S3-08-2-00.loads. Copy these files to /var/lib/tftpboot.		In order to enable TFTP on the Asterisk server, edit <path>/etc/xinetd.d/tftp</path> and set disable to no. The TFTP daemon is configured on Fedora to serve files out of <path>/var/lib/tftpboot</path>. After you have modified its configuration, restart xinetd with the command
			service xinetd restart
			Cisco’s SIP software package provides the files P0S3-08-2-00.sb2, P0S3-08-2-00.sbn and P0S3-08-2-00.loads. Copy these files to <path>/var/lib/tftpboot</path>.

	{Admon/note\|\|The filenames used by your Cisco firmware may be slightly different. Cisco’s naming convention is P0S3-xxx-y-zz, where x is the major version, y is the minor version, and z is the subversion. P0S3 indicates SIP firmware, P003 indicates SCCP firmware, and P0M3 indicates MGCP firmware.}}		{{Admon/note\|\|The filenames used by your Cisco firmware may be slightly different. Cisco’s naming convention is P0S3-xxx-y-zz, where x is the major version, y is the minor version, and z is the subversion. P0S3 indicates SIP firmware, P003 indicates SCCP firmware, and P0M3 indicates MGCP firmware.}}

	When booting SIP firmware, the telephone will look for three configuration files: OS79XX.TXT, SIPDefault.cnf, and SIPMAC-ADDRESS.cnf. These files should also		When booting SIP firmware, the telephone will look for three configuration files: OS79XX.TXT, SIPDefault.cnf, and SIPMAC-ADDRESS.cnf. These files should also
	be placed in /var/lib/tftpboot.		be placed in <path>/var/lib/tftpboot</path>.

	OS79XX.TXT contains one line: the firmware version to load. Mine contains P0S3-08-2-00.		OS79XX.TXT contains one line: the firmware version to load. Mine contains P0S3-08-2-00.

	SIPDefault.cnf contains configurations common to all SIP telephones. The following example lists the image version and the address of our Asterisk server, and instructs the telephones to register with Asterisk:		SIPDefault.cnf contains configurations common to all SIP telephones. The following example lists the image version and the address of our Asterisk server, and instructs the telephones to register with Asterisk:
			<pre>
	image_version: P0S3-08-2-00 ;		image_version: P0S3-08-2-00 ;
	proxy1_address: 192.168.0.10 ;		proxy1_address: 192.168.0.10 ;
	proxy_register: "1" ;		proxy_register: "1" ;
			</pre>
	Per-telephone configurations are stored in a file named after the telephone’s MAC address. The telephone’s MAC address may be found in its configuration menu. This configuration will correspond with the SIP information found in LDAP and should be saved as SIPMAC-ADDRESS.cnf:		Per-telephone configurations are stored in a file named after the telephone’s MAC address. The telephone’s MAC address may be found in its configuration menu. This configuration will correspond with the SIP information found in LDAP and should be saved as SIPMAC-ADDRESS.cnf:
			<pre>
	line1_name : user2		line1_name : user2
	line1_authname : user2		line1_authname : user2
	line1_password : PLAINTEXT-PASSWORD		line1_password : PLAINTEXT-PASSWORD
			</pre>
			The first step in preparing the telephone is to clear any existing configuration. This may be done by pressing '''<nowiki>**# settings 3</nowiki>''' to access an unlocked Network
			Configuration menu. Here, find the Erase Configuration menu item and select it. Restart the phone and return to the Network Configuration menu. Ensure DHCP Enabled is set. Reboot the phone.

	The ~~first step in preparing the telephone is~~ to ~~clear any existing configuration~~. ~~This may be done by pressing **# settings 3~~ to ~~access an unlocked~~ Network		The phone will pull its network configuration from DHCP and should now respond to pings. Return to the Network Configuration menu and deactivate DHCP. Select the TFTP Server 1 option and set it to your Asterisk / TFTP server. Do not disable DHCP or manually set a TFTP server if your DHCP server provides your TFTP server’s IP address. Reboot the telephone and it should load the SIP software and configuration. As the phone boots, you may see it request files using TFTP by watching the Asterisk server’s <path>/var/log/messages log</path>.
	Configuration menu. ~~Here, find~~ the ~~Erase Configuration menu item~~ and ~~select~~ it~~. Restart the phone and return~~ to ~~the Network Configuration menu~~. ~~Ensure~~ DHCP ~~Enabled is~~ set. Reboot the phone.

	~~The phone will pull~~ its ~~network~~ configuration ~~from DHCP and should now respond to pings~~. ~~Return to the Network Configuration menu and deactivate DHCP~~. ~~Select~~ the TFTP Server 1 option and set it to your Asterisk / TFTP server. Do not disable DHCP or manually set a TFTP server if your DHCP server provides your TFTP server’s IP address. Reboot the telephone and it should load the SIP software and configuration. As the phone boots, you may ~~see it request files using TFTP~~ by ~~watching the Asterisk server’s /var/log/messages log~~.		Once the telephone boots, test its configuration by dialing '''600'''. This will execute an echo test. After the echo test, you may make a Cisco telephone to Ekiga call by dialing 2001.

	~~Once the telephone boots, test its configuration by dialing 600. This will execute an echo test. After the echo test, you may make a Cisco telephone to Ekiga call by dialing 2001.~~
	== Configuring voicemail ==		== Configuring voicemail ==

Didier : /* Configuring LDAP */

2013-09-19T09:47:47Z

Configuring LDAP

Didier : /* Configuring the Asterisk PBX */

2013-09-19T09:38:54Z

Configuring the Asterisk PBX

← Version précédente		Version du 19 septembre 2013 à 09:38
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	The Asterisk PBX has many features and is very flexible. In this section, I will focus on configuring four things: the SIP module, the extensions module, the external configuration engine, and the realtime LDAP backend. Once Asterisk is configured, I will demonstrate how to add SIP users and extensions to your LDAP database.		The Asterisk PBX has many features and is very flexible. In this section, I will focus on configuring four things: the SIP module, the extensions module, the external configuration engine, and the realtime LDAP backend. Once Asterisk is configured, I will demonstrate how to add SIP users and extensions to your LDAP database.

	Asterisk provides a module to support the SIP protocol. SIP is used to set up and tear down a VoIP phone call. Though individual SIP users will be stored in an LDAP database, some global parameters will exist in the following configuration file. Save this configuration as /etc/asterisk/sip.conf:		Asterisk provides a module to support the SIP protocol. SIP is used to set up and tear down a VoIP phone call. Though individual SIP users will be stored in an LDAP database, some global parameters will exist in the following configuration file. Save this configuration as <path>/etc/asterisk/sip.conf</path>:
	<syntaxhighlight lang="ini">		<syntaxhighlight lang="ini">
	[general]		[general]
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	We will now move on to define extensions. In many cases, an extension is simply a telephone number that will be associated with a SIP user. Individual extensions, like SIP users, will be stored in an		We will now move on to define extensions. In many cases, an extension is simply a telephone number that will be associated with a SIP user. Individual extensions, like SIP users, will be stored in an
	LDAP database. However, as with the SIP configuration, global parameters will be configured in a file,		LDAP database. However, as with the SIP configuration, global parameters will be configured in a file,
	/etc/asterisk/extensions.conf. The following example supports the LDAP entries I will use in this article:		<path>/etc/asterisk/extensions.conf</path>. The following example supports the LDAP entries I will use in this article:
	<syntaxhighlight lang="ini">		<syntaxhighlight lang="ini">
	[users]		[users]
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	include => demo		include => demo
	</syntaxhighlight>		</syntaxhighlight>
	This example defines three contexts: default, users, and demo. The line switch => Realtime/@ states that the extensions in a context will be defined using one of Asterisk’s realtime back-ends (in our case, LDAP). Each individual extension will be assigned to one of these contexts. As a collection of available extensions, contexts determine what happens when a user dials a phone number.		This example defines three contexts: default, users, and demo. The line '''<nowiki>switch => Realtime/@</nowiki>''' states that the extensions in a context will be defined using one of Asterisk’s realtime back-ends (in our case, LDAP). Each individual extension will be assigned to one of these contexts. As a collection of available extensions, contexts determine what happens when a user dials a phone number.

	Asterisk’s external configuration engines are activated in /etc/asterisk/extconfig.conf. The following example configures Asterisk to pull SIP and extension information using its real-time LDAP back-end:		Asterisk’s external configuration engines are activated in <path>/etc/asterisk/extconfig.conf</path>. The following example configures Asterisk to pull SIP and extension information using its real-time LDAP back-end:
	<syntaxhighlight lang="ini">		<syntaxhighlight lang="ini">
	[settings]		[settings]
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	This example causes Asterisk to activate its real-time LDAP back-end look for SIP users, SIP peers, and extensions in the LDAP database. In addition, the base DN used for queries is specified.		This example causes Asterisk to activate its real-time LDAP back-end look for SIP users, SIP peers, and extensions in the LDAP database. In addition, the base DN used for queries is specified.

	{{Admon/note\|\|All of the examples in this article will use dc=example,dc=com as the LDAP database’s base DN. This should be replaced with your organization’s base DN.}}		{{Admon/note\|\|All of the examples in this article will use <nowiki>dc=example,dc=com</nowiki> as the LDAP database’s base DN. This should be replaced with your organization’s base DN.}}

	The final configuration file we will write configures Asterisk’s real-time LDAP back-end and should be saved as /etc/asterisk/res_ldap.conf.		The final configuration file we will write configures Asterisk’s real-time LDAP back-end and should be saved as <path>/etc/asterisk/res_ldap.conf</path>.
	<syntaxhighlight lang="ini">		<syntaxhighlight lang="ini">
	[_general]		[_general]

Didier : /* Prerequisites */

2013-09-19T09:34:03Z

Prerequisites

← Version précédente		Version du 19 septembre 2013 à 09:34
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	== Prerequisites ==		== Prerequisites ==

	{{Admon/note\|\|Asterisk 1.6 Beta 10 has not yet been released. This version will fix a bug in Asterisk’s realtime LDAP code that causes the server to crash. In the meantime, one may compile the upcoming realtime LDAP module and install it for use with Fedora’s packages. In order to do this, follow the following instructions after installing the Fedora Asterisk packages:		{{Admon/note\|Asterisk version\|Asterisk 1.6 Beta 10 has not yet been released. This version will fix a bug in Asterisk’s realtime LDAP code that causes the server to crash. In the meantime, one may compile the upcoming realtime LDAP module and install it for use with Fedora’s packages. In order to do this, follow the following instructions after installing the Fedora Asterisk packages:

	# Execute “yum install openldap-devel”		# Execute “yum install openldap-devel”
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	{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}		{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}

	Install the Asterisk components by executing the command <syntaxhighlight lang="bash" enclose="~~none~~">yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain</syntaxhighlight>		Install the Asterisk components by executing the command <syntaxhighlight lang="bash" enclose="div">yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain</syntaxhighlight>
	In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute <syntaxhighlight lang="bash" enclose="~~none~~">yum install tftp-server</syntaxhighlight>. have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.		In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute <syntaxhighlight lang="bash" enclose="div">yum install tftp-server</syntaxhighlight>, have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.

	{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}		{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}

Didier : /* Prerequisites */

2013-09-19T09:30:44Z

Prerequisites

← Version précédente		Version du 19 septembre 2013 à 09:30
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	{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}		{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}

	Install the Asterisk components by executing the command <syntaxhighlight lang="bash" enclose="~~div~~">yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain</syntaxhighlight>		Install the Asterisk components by executing the command <syntaxhighlight lang="bash" enclose="none">yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain</syntaxhighlight>
	In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute <syntaxhighlight lang="bash" enclose="~~div~~">yum install tftp-server</syntaxhighlight>. have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.		In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute <syntaxhighlight lang="bash" enclose="none">yum install tftp-server</syntaxhighlight>. have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.

	{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}		{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}

Didier : /* Prerequisites */

2013-09-19T09:30:04Z

Prerequisites

← Version précédente		Version du 19 septembre 2013 à 09:30
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	}}		}}

	There are a few prerequisites to discuss before covering the configuration of Asterisk. First, my instructions assume that you have already installed Fedora 9. Second, you must have an existing LDAP server. The LDAP schema I use in this article is compatible with Fedora Directory Server. I recommend using the FreeIPA system to manage FDS. Fedora 9 provides packages for both FDS (fedora-ds-base) and FreeIPA (ipa-server). Finally, the machine hosting Asterisk must allow incoming UDP connections to ports 5060 (Session Initiation Protocol) and 69 (TFTP) to pass through its firewall.		There are a few prerequisites to discuss before covering the configuration of Asterisk. First, my instructions assume that you have already installed Fedora 9. Second, you must have an existing LDAP server. The LDAP schema I use in this article is compatible with Fedora Directory Server. I recommend using the [http://www.freeipa.org/ FreeIPA] system to manage FDS. Fedora 9 provides packages for both FDS (fedora-ds-base) and FreeIPA (ipa-server). Finally, the machine hosting Asterisk must allow incoming UDP connections to ports 5060 (Session Initiation Protocol) and 69 (TFTP) to pass through its firewall.

	{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}		{{Admon/note\|\|In general, the LDAP database is optional. All configurations may be stored in flat files on the Asterisk server’s disk. However, I have chosen to store my Asterisk users and extensions in an LDAP database. This is beneficial for organizations that wish to consolidate user information in LDAP. In addition, Asterisk’s realtime LDAP backend is new to 1.6. I hope to help fill the gap in existing documentation with this article.}}

	Install the Asterisk components by executing the command yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain. In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute yum install tftp-server, have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.		Install the Asterisk components by executing the command <syntaxhighlight lang="bash" enclose="div">yum install asterisk asterisk-ldap asterisk-alsa asterisk-voicemail asterisk-voicemail-plain</syntaxhighlight>
			In order to configure a Cisco 7900 Series VoIP telephone, you will also need to execute <syntaxhighlight lang="bash" enclose="div">yum install tftp-server</syntaxhighlight>. have an existing DHCP server (the one on an inexpensive wireless router will work fine) and a copy of Cisco’s Session Initiation Protocol (SIP) software for the phone.

	{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}		{{Admon/note\|\|Depending on how you buy a Cisco telephone, you may need to purchase the rights to Cisco’s SIP software. Cisco telephones are typically sold with support for the proprietary SCCP protocol. In order to use the telephone with SIP, one must load a SIP flash image onto the telephone. This image may not have been included in the original telephone purchase. One way to obtain a license for the software is to purchase a Cisco SmartNet support package. Support for a 7900 Series telephone costs less than $10 and should be available from the vendor that sold you the telephone. I will cover installing the SIP software later in the article.}}

	The Fedora project does not yet package Asterisk’s audio files because their copyright holder has not yet assigned a license to them (see Red Hat Bugzilla bug #428832). These audio files are required to implement features like an echo test and voicemail. Fortunately, it is quite simple to install them. Visit the Asterisk website and download the latest Asterisk 1.6 series release. Inside the Asterisk archive is another, sounds/asterisk-core-sounds-en-gsm-1.4.11.tar.gz. Extract this archive file into /usr/share/asterisk/sounds.		The Fedora project does not yet package Asterisk’s audio files because their copyright holder has not yet assigned a license to them (see [https://bugzilla.redhat.com/show_bug.cgi?id=428832 Red Hat Bugzilla bug #428832]). These audio files are required to implement features like an echo test and voicemail. Fortunately, it is quite simple to install them. Visit the [http://www.asterisk.org/ Asterisk website] and download the latest Asterisk 1.6 series release. Inside the Asterisk archive is another, sounds/asterisk-core-sounds-en-gsm-1.4.11.tar.gz. Extract this archive file into /usr/share/asterisk/sounds.

	== Configuring the Asterisk PBX ==		== Configuring the Asterisk PBX ==

← Version précédente		Version du 19 septembre 2013 à 09:47
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	== Configuring LDAP ==		== Configuring LDAP ==

	With the exception of actual users and extensions, our simple Asterisk installation is fully configured. We will now finish our installation by adding users and extensions to our LDAP database. First, we need to install an Asterisk schema for LDAP at /etc/dirsrv/slapd-DOMAIN/schema/99asterisk.ldif. After the schema has been installed, restart FDS using service dirsrv restart.		With the exception of actual users and extensions, our simple Asterisk installation is fully configured. We will now finish our installation by adding users and extensions to our LDAP database. First, we need to install an Asterisk schema for LDAP at <path>/etc/dirsrv/slapd-DOMAIN/schema/99asterisk.ldif</path>. After the schema has been installed, restart FDS using <syntaxhighlight lang="bash">service dirsrv restart</syntaxhighlight>
	<syntaxhighlight lang="python">		<syntaxhighlight lang="python">
	dn: cn=schema		dn: cn=schema
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	)		)
	</syntaxhighlight>		</syntaxhighlight>
	We may now create two SIP users, one for our Ekiga installation and the other for our Cisco telephone. If you write the following LDIF into a file, you may load it into the LDAP database using ldapadd -x -D "cn=Directory Manager" -W -f PATH-TO-LDIF.		We may now create two SIP users, one for our Ekiga installation and the other for our Cisco telephone. If you write the following LDIF into a file, you may load it into the LDAP database using <syntaxhighlight lang="bash">ldapadd -x -D "cn=Directory Manager" -W -f PATH-TO-LDIF</syntaxhighlight>.
	<syntaxhighlight lang="python">		<syntaxhighlight lang="python">
	dn: ou=sippeers,dc=example,dc=com		dn: ou=sippeers,dc=example,dc=com
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	* '''AstAccountType''': Defines the type of client. Asterisk sends calls to peers (i.e., a SIP provider). The user type may place calls only. Friends are both peers and users.		* '''AstAccountType''': Defines the type of client. Asterisk sends calls to peers (i.e., a SIP provider). The user type may place calls only. Friends are both peers and users.

	Once a user is defined, one may test Asterisk. Start the Asterisk service by executing service asterisk start as root. Execute an Asterisk shell using the command asterisk -rv. At Asterisk’s prompt, execute sip show peer user1 load. Asterisk should print all of the information for user1.		Once a user is defined, one may test Asterisk. Start the Asterisk service by executing <syntaxhighlight lang="bash">service asterisk start</syntaxhighlight> as root. Execute an Asterisk shell using the command <syntaxhighlight lang="bash">asterisk -rv</syntaxhighlight>. At Asterisk’s prompt, execute <syntaxhighlight lang="bash">sip show peer user1 load</syntaxhighlight>. Asterisk should print all of the information for user1.

	Once you have defined a user, you may start writing your dial plan. First, we will associate a phone number with each user. Use ldapadd to add the following to your LDAP database:		Once you have defined a user, you may start writing your dial plan. First, we will associate a phone number with each user. Use ldapadd to add the following to your LDAP database:
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	* '''AstApplicationData''': Arguments, separated with commas (without spaces), and passed to a command. The Dial has two arguments. The first is the user that the call will be connected to. The second is the number of seconds to wait before quitting.		* '''AstApplicationData''': Arguments, separated with commas (without spaces), and passed to a command. The Dial has two arguments. The first is the user that the call will be connected to. The second is the number of seconds to wait before quitting.

	In order to facilitate testing our installation, we will define the following additional extension. This extension executes three commands to implement an echo test that may be reached by dialing ~~’600~~.’		In order to facilitate testing our installation, we will define the following additional extension. This extension executes three commands to implement an echo test that may be reached by dialing '''600'''.
	<syntaxhighlight lang="python">		<syntaxhighlight lang="python">
	dn: cn=600-1,ou=extensions,dc=example,dc=com		dn: cn=600-1,ou=extensions,dc=example,dc=com
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	AstApplicationData: demo-echodone		AstApplicationData: demo-echodone
	</syntaxhighlight>		</syntaxhighlight>
	After schema edits, FDS must be restarted using service dirsrv restart.		After schema edits, FDS must be restarted using <syntaxhighlight lang="bash">service dirsrv restart</syntaxhighlight>

	The basic Asterisk configuration is now complete. Restart the Asterisk service using service asterisk restart.		The basic Asterisk configuration is now complete. Restart the Asterisk service using <syntaxhighlight lang="bash">service asterisk restart</syntaxhighlight>

	== Configuring the Ekiga VoIP client ==		== Configuring the Ekiga VoIP client ==

VoIP/AsteriskOnFedora - Historique des versions

Didier le 20 septembre 2013 à 08:12

Didier : /* Configuring the Ekiga VoIP client */

Didier le 20 septembre 2013 à 07:57

Didier : /* Configuring voicemail */

Didier : /* Configuring a Cisco VoIP telephone */

Didier : /* Configuring LDAP */

Didier : /* Configuring the Asterisk PBX */

Didier : /* Prerequisites */

Didier : /* Prerequisites */

Didier : /* Prerequisites */

Didier : /* Configuring the Ekiga VoIP client */

Didier : /* Configuring voicemail */

Didier : /* Configuring a Cisco VoIP telephone */

Didier : /* Configuring LDAP */

Didier : /* Configuring the Asterisk PBX */

Didier : /* Prerequisites */

Didier : /* Prerequisites */

Didier : /* Prerequisites */