![]() ![]() ![]() We’ve been posting tutorials regularly on Asterisk PBX and VoIP network design for SOHO to Enterprise. Open /etc/asterisk/extensions.Exten=>_xxxx,1,Dial(SIP/$,10,T) ![]() I recommend an seperate folder ~ # mkdir ~ # ffmpeg -i /tmp/mySpeech.mp3 -ar 8000 -ac 1 -ab 64 ~ # ffmpeg -i /tmp/mySpeech.mp3 -ar 8000 -ac 1 -ab 64 -f mulaw /var/lib/asterisk/sounds/my/mySpeech.pcm -map 0:0 -map 0:0 It has to convert in the gsm specific format. *CLI> NOTICE: pbx_spool.c:463 attempt_thread: Call completed to can also use your own speech files. When you want to transfer the call to an agent, you have to. If the call would accepted a hello world greeting is ~ # asterisk -c an example of an Asterisk extension number to which calls from Aimylogic are transferred. To test the setup, copy the mycall.call file to ~ cp /root/mycall.call /var/spool/asterisk/outgoing Importend is the Asterisk Ready ~ # asterisk -c asterisk-phones is the context and 10 is the extension refered from the mycall.call file.įor test purposes start asterisk in foreground. Move all default extension ~ # mkdir ~ # mv /etc/asterisk/extensions.* /etc/asterisk/extensions_orgĪnd create a /etc/asterisk/nf with sequences what to do if the mycall.call file is used. Replace 012345678with the number you want to call. Then create a call file /root/mycall.call. At the register directive adjust the internal phone number (here 623) noted previously. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. It runs on Linux, BSD, Windows and macOS and provides all of the features you would expect from a PBX and more. And change the localnet and permit properties to your subnet. Asterisk is a complete PBX (private branch exchange) in software. For incoming calls, Sectionname must match the register directiveĪdjust the host and fromdomain variables with the IP Address of your Fritzbox. Register => virt asterisk SIP phone for outgoing calls Rename the default /etc/asterisk/sip.conf file an create a new one Then install ~ # sudo ~ #apt-get install asterik ffmpeg Note the username and the internal phone number (here 623). The new phone in Phonelist New Phone overview ![]() Deselect Numbers asterisk shouldn’t answer Open the Fritz!Box GUI, then open the Phonelist Phonelistĭeselect Phone numbers for which asterisk should not receive incoming calls (this would also work). In this example I will use a Fritz!box as the upstream SIP Server for asterisk.įirst create an IP Phone and an corrosponding User Account at the Fritz!box. Asterisk is a complete VoIP/SIP solution but can also be used as a SIP client to send a prerecorded message. ![]()
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