Recommendations for configuring the Asterisk system
Basic Asterisk settings that the 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 users are connected.
Follow the instructions for installing the telephone system: http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14
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, user server IP address 11.22.33.44
- one serial search group (call queue)
- four internal SIP lines on which contact centre users work - all of them are included in one queue
sip.conf
Contents of the file responsible for SIP lines
; general settings [general] language = ru bindport=5060 callcounter = yes limitonpeers = yes dtmf=rfc2833 compensate=yes defaultexpirey=3600 disallow=all allow=alaw registertimeout=3600 ; here you need to list all external sip lines in the specified format register=0551234567:qwerty123456@11.22.33.44/0551234567 ; next, the settings for the external user that provides the phone number [SIPOPERATOR] type=friend host=11.22.33.44 port=5060 defaultip=here_ip_address_on_interface_to_the_user_side fromdomain=here_domain_name_of_the_user_softswitch_name fromuser=0551234567 username=0551234567 secret=qwerty123456 registertimeout=3600 defaultexpirey=3600 nat=no canreinvite=no disallow=all allow=alaw qualify=yes dtmfmode=inband insecure=port,invite 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 [CallCenter](!) call-limit=1 ; Limit simultaneous calls (users receive only one call) qualify = no dtmfmode=rfc2833 canreinvite=no pickupgroup=1 callgroup=1 host=dynamic type=friend port=5060 qualify=yes deny=0.0.0.0/0.0.0.0 permit=0.0.0.0.0/0.0.0.0.0 ; allowed to connect from anywhere. Change if needed otherwise callcounter=yes faxdetect=no disallow=all allow=alaw allow=gsm 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 [500](CallCenter) username=500 nat=no secret=here_password callerid=user 500 <500> [501](CallCenter) username=501 nat=no secret=here_password callerid=user 501 <501> [502](CallCenter) username=502 nat=no secret=here_password callerid=user 502 <502> [503](CallCenter) username=503 nat=no secret=here_password callerid=user 503 <503>
queues.conf
The group in the configuration file with the settings of serial search groups (queues) is discussed below
[callcenter-queue] monitor-type = MixMonitor 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 timeout = 30 retry = 2 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 leavewhenempty = yes announce-frequency = 30 announce-holdtime = no ; the members of the group are then listed member => SIP/500 member => SIP/501 member => SIP/502 member => SIP/503
extensions.conf
The following file configures routing. A fragment that routes incoming calls to the group and outgoing calls through the external SIP line is considered
; incoming calls are those that come from the external sip line towards the contact centre
[from-external-sip]
exten => s,1,Ringing() ; if required, send a ringing signal to the caller (if not required, delete the line).
exten => s,n,Wait(2) ; wait two seconds for the caller to hear the first beep after dialling the number, if you answer immediately without beeping - the caller gets scared :)
exten => s,n,Answer(300) ; answer the call
exten => s,n,Playback(hello) ; play the hello file (file name without extension).
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
; outgoing calls - those that originate from the contact centre users to the PSTN
[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.,n,Dial(SIP/SIPOPOPERATOR/${EXTEN},300) ; Direct the call towards SIPOPERATOR
Do not make any other settings if you do not know what they are for.
If you configure the system through various web-interfaces (trixbox, elastix, freepbx, etc.), then it is impossible to guarantee the work of usm_asterisk module, because the logic of Asterisk operation after such configuration is often unpredictable and it is impossible to trace the call passing through the standard module.