There are lots of bug in this version, you can try with stable version.ĭnrvsoymn (svzcsfegvwef54gbhhi at gmail dot com)Ĭialis vs viagra reviews cialis free samples buy cialis cheap buy cialis online ’ As stated, we will review softphones in a later in the tutorial.Ĭall soft corporation (Info at call-soft dot com) The users and extensions are now registered on Asterisk but the users must also be registered on a SIP or IAX client softphone.
When any change is made in conf files from /etc/asterisk/ or changes that relate with some of these files, you must type 'reload' in the Asterisk Command Line Interface (CLI) to make the changes effective. You have completed the registration of 4 users(2 SIP/2 IAX) and 4 extensions. Note that the Dial command when using IAX2 protocol is : call the second registered user, create extension 2222. To call the first user 'ivan_iax' dial 1111.
Now register a second IAX user following the same steps. Secret equals your chosen password, host equals 'dynamic IP' and context is 'tutorial'. The user is 'ivan_iax' and type is 'friend' again (Inbound/Oubound calls allowed). Set the host IP to dynamic and create a password as described previously. More detailed configuration information for a series of phones can be found here:Ĭreate user 'ivan_iax' with the same username and join it to the tutorial context. For now, just make sure you have registered the users and extensions. However, softphones will be reviewed later. The final step is to register the user to a compatible softphone. Now when user 'ivan' or any other user from the tutorial context dials 4321, the user 'test' will be called.ģ.
Register the extension(4321) in /etc/asterisk/nf in the same context = tutorial. Start by registering the second user in the same way in /etc/asterisk/sip.conf Follow this same process to register another SIP user and extension in order to place test calls. We now have a registered SIP user and extension on Asterisk. The priority determines the sequence in which the extensions will be executed. The command is : exten => number, priority, Dial(protocol/user). When dialing number 1234, Asterisk will first Dial the user xlite through SIP protocol.
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