1/29/2024 0 Comments Add x lite softphone to freepbx![]() Thus, to test your X-Lite soft phone you can simply call yourself, and the call will loop. As stated, we will review softphones in a later in the tutorial. This means they can accept multiple incoming calls at the same time. 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'. I know we can configure xlite with freepbx. In reality the exact process for connecting will depend on the network topology of your client. At its most basic, you need the Freepbx server name, extension and secret. The user is 'ivan_iax' and type is 'friend' again (Inbound/Oubound calls allowed). Yes, you can register Xlite softphone endpoints to FreePBX extensions. 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. Sangoma FreePBX Softphones unifies the user experience with audio and video calling, screen-sharing and chat, from any device, where ever you are. More detailed configuration information for a series of phones can be found here: 01. As stated, we will review softphones in a later in the tutorial. However, softphones will be reviewed later. FreePBX Softphones (20 User Package) 299 for a 1 year license. Create user ivaniax with the same username and join it to the tutorial context. 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. Calling Out) When I try calling out from the softphone, the mess. I have even placed the computer in the DMZ of my router to minimize firewall issues, but I still cannot make or receive calls. The softphone shows a correct login with the PBX. Register the extension(4321) in /etc/asterisk/nf in the same context = tutorial. FreePBX is setup on my network (ISO from PBXinaFlash) and the computer is able to ping internet sites. 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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