Recommendations for configuring the Asterisk system - История изменений

N.Skrypnik в 17:38, 29 марта 2024

2024-03-29T17:38:30Z

← Предыдущая версия		Версия от 17:38, 29 марта 2024
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	'''Basic Asterisk settings that the [[Usm_asterisk_EN\|usm_asterisk]] module is designed for'''		'''Basic Asterisk settings that the [[Usm_asterisk_EN\|usm_asterisk]] module is designed for'''

	This article provides a typical basic contact center configuration, which allows you to receive incoming calls from a SIP line and transfer them to a queue to which ~~operators~~ are connected.		This article provides a typical basic contact center configuration, which allows you to receive incoming calls from a SIP line and transfer them to a queue to which users are connected.

	Follow the instructions for installing the telephone system: http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14		Follow the instructions for installing the telephone system: http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14
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	== Input data for the demo configuration ==		== Input data for the demo configuration ==

	* one external SIP line [SIPOPOPERATOR], from the side of which incoming phone calls are received. Phone number 0551234567, password qwerty123456, ~~operator~~ server IP address 11.22.33.44		* one external SIP line [SIPOPOPERATOR], from the side of which incoming phone calls are received. Phone number 0551234567, password qwerty123456, user server IP address 11.22.33.44
	* one serial search group (call queue)		* one serial search group (call queue)
	* four internal SIP lines on which contact centre ~~operators~~ work - all of them are included in one queue		* four internal SIP lines on which contact centre users work - all of them are included in one queue

	== sip.conf ==		== sip.conf ==
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	register=0551234567:qwerty123456@11.22.33.44/0551234567		register=0551234567:qwerty123456@11.22.33.44/0551234567

	; next, the settings for the external ~~operator~~ that provides the phone number		; next, the settings for the external user that provides the phone number
	[SIPOPERATOR]		[SIPOPERATOR]
	type=friend		type=friend
	host=11.22.33.44		host=11.22.33.44
	port=5060		port=5060
	defaultip=~~here_ip_address_on_interface_to_the_operator_side~~		defaultip=here_ip_address_on_interface_to_the_user_side
	fromdomain=~~here_domain_name_of_the_operator_softswitch_name~~		fromdomain=here_domain_name_of_the_user_softswitch_name
	fromuser=0551234567		fromuser=0551234567
	username=0551234567		username=0551234567
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	; here are the general settings for all internal sip lines		; here are the general settings for all internal sip lines
	[CallCenter](!)		[CallCenter](!)
	call-limit=1 ; Limit simultaneous calls (~~operators~~ receive only one call)		call-limit=1 ; Limit simultaneous calls (users receive only one call)
	qualify = no		qualify = no
	dtmfmode=rfc2833		dtmfmode=rfc2833
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	allow=alaw		allow=alaw
	allow=gsm		allow=gsm
	context=from-internal-sip ; this context will be used in routing for calls from contact centre ~~operator~~ lines		context=from-internal-sip ; this context will be used in routing for calls from contact centre user lines

	The 4 internal telephone lines with common settings are described below		The 4 internal telephone lines with common settings are described below
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	nat=no		nat=no
	secret=here_password		secret=here_password
	callerid=~~operator~~ 500 <500>		callerid=user 500 <500>

	[501](CallCenter)		[501](CallCenter)
Строка 85:		Строка 85:
	nat=no		nat=no
	secret=here_password		secret=here_password
	callerid=~~operator~~ 501 <501>		callerid=user 501 <501>

	[502](CallCenter)		[502](CallCenter)
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	nat=no		nat=no
	secret=here_password		secret=here_password
	callerid=~~operator~~ 502 <502>		callerid=user 502 <502>

	[503](CallCenter)		[503](CallCenter)
Строка 97:		Строка 97:
	nat=no		nat=no
	secret=here_password		secret=here_password
	callerid=~~operator~~ 503 <503>		callerid=user 503 <503>

	== queues.conf ==		== queues.conf ==
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	[callcenter-queue]		[callcenter-queue]
	monitor-type = MixMonitor		monitor-type = MixMonitor
	music = file_name ; name of the file without extension, which will be played to the user in the ~~operator~~'s answer waiting mode.		music = file_name ; name of the file without extension, which will be played to the user in the user's answer waiting mode.
	strategy = ringall ; serial search strategy - call all users at once		strategy = ringall ; serial search strategy - call all users at once
	timeout = 30		timeout = 30
	retry = 2		retry = 2
	joinempty = yes ; a call can join the queue even if no ~~operator~~ is connected to it		joinempty = yes ; a call can join the queue even if no user is connected to it
	ringinuse = no ; do not accept calls to people who are already on a call		ringinuse = no ; do not accept calls to people who are already on a call
	leavewhenempty = yes		leavewhenempty = yes
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	exten => s,n,Hangup() ; direct the caller to the hangup queue		exten => s,n,Hangup() ; direct the caller to the hangup queue

	; outgoing calls - those that originate from the contact centre ~~operators~~ to the PSTN		; outgoing calls - those that originate from the contact centre users to the PSTN
	[from-internal-sip]		[from-internal-sip]
	exten => _X.,1,Set(CALLERID(num)=380551234567) ; Change the Caller_ID before sending the call to the PSTN to the outside line number		exten => _X.,1,Set(CALLERID(num)=380551234567) ; Change the Caller_ID before sending the call to the PSTN to the outside line number

N.Skrypnik в 14:57, 3 октября 2023

2023-10-03T14:57:46Z

← Предыдущая версия		Версия от 14:57, 3 октября 2023
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	exten => s,n,Queue(callcenter-queue) ; direct the call to the "callcenter-queue" queue		exten => s,n,Queue(callcenter-queue) ; direct the call to the "callcenter-queue" queue
	exten => s,n,Hangup() ; direct the caller to the hangup queue		exten => s,n,Hangup() ; direct the caller to the hangup queue

	; outgoing calls - those that originate from the contact centre operators to the PSTN		; outgoing calls - those that originate from the contact centre operators to the PSTN
	[from-internal-sip]		[from-internal-sip]

N.Skrypnik в 14:57, 3 октября 2023

2023-10-03T14:57:04Z

← Предыдущая версия		Версия от 14:57, 3 октября 2023
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	; here you need to list all external sip lines in the specified format		; here you need to list all external sip lines in the specified format
	register=0551234567:qwerty123456@11.22.33.44/0551234567		register=0551234567:qwerty123456@11.22.33.44/0551234567

	; next, the settings for the external operator that provides the phone number		; next, the settings for the external operator that provides the phone number
	[SIPOPERATOR]		[SIPOPERATOR]
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	insecure=port,invite		insecure=port,invite
	context=from-external-sip ; this context will be used in routing when calling from an external sip line		context=from-external-sip ; this context will be used in routing when calling from an external sip line

	; here are the general settings for all internal sip lines		; here are the general settings for all internal sip lines
	[CallCenter](!)		[CallCenter](!)
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	allow=gsm		allow=gsm
	context=from-internal-sip ; this context will be used in routing for calls from contact centre operator lines		context=from-internal-sip ; this context will be used in routing for calls from contact centre operator lines

	The 4 internal telephone lines with common settings are described below		The 4 internal telephone lines with common settings are described below
	[500](CallCenter)		[500](CallCenter)
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	secret=here_password		secret=here_password
	callerid=operator 501 <501>		callerid=operator 501 <501>

	[502](CallCenter)		[502](CallCenter)
	username=502		username=502

N.Skrypnik в 14:54, 3 октября 2023

2023-10-03T14:54:35Z

← Предыдущая версия		Версия от 14:54, 3 октября 2023
Строка 1:		Строка 1:
	[[Recommendations_for_configuring_the_Asterisk_system\|en]] \| [[Рекомендации_по_настройке_системы_Asterisk\|ru]]		[[Recommendations_for_configuring_the_Asterisk_system\|en]] \| [[Рекомендации_по_настройке_системы_Asterisk\|ru]]

	'''Basic Asterisk settings that the [[usm_asterisk]] module is designed for'''		'''Basic Asterisk settings that the [[Usm_asterisk_EN\|usm_asterisk]] module is designed for'''

	This article provides a typical basic contact center configuration, which allows you to receive incoming calls from a SIP line and transfer them to a queue to which operators are connected.		This article provides a typical basic contact center configuration, which allows you to receive incoming calls from a SIP line and transfer them to a queue to which operators are connected.

N.Skrypnik в 14:54, 3 октября 2023

2023-10-03T14:54:03Z

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2023-10-02T15:33:01Z

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