No need to edit nf, or ur asterisk is version 12 For the sake of terminology, it is useful to note that though we have this SIP configuration configured with "type=friend", most people refer to this as configuring a SIP peer.īackup and edit a new blank nf, just like you did with nf. For this example to work, just make sure you have everything exactly as written above. Then add the following to your sip.conf file:īasic configuration will be explained in more detail in other sections of the wiki. ? Asterisk 12 and beyond: You'll probably want to use chan_pjsip (the newest driver), but you still have the option of using chan_sip as wellįollow the instructions below for the channel driver you chose.īackup and edit a new blank sip.conf, just like you did with nf. ? Asterisk 11 and previous: chan_sip is the primary option. You'll have to pick one to use for the example. When a phone dials extension 100, we are telling Asterisk to Answer the call, Wait one second, then Play (Playback) a sound file (hello-world) to the channel and Hangup.ĭepending on the version of Asterisk in use, you may have the option of more than one SIP channel driver. A dialplan is simply instructions telling Asterisk what to do with a call.Įdit your blank nf to reflect the following: We are going to use a very simple dialplan. I'm assuming you use the VI/VIM editor here, after all, it is the best. To get started, go ahead and move to the /etc/asterisk/ directory where the files are located.īackup the sample nf and create a new one You can use the defaults for nf and nf, we'll only need to modify nf and sip.conf or nf. If you have no configuration files in /etc/asterisk/ then grab the sample config files from the by navigating to it and running "make samples". You should have already run "make samples" if you installed from source, otherwise you may have the sample config files if you installed from packages. For example if you want to use chan_pjsip, then make sure you followed the guide. ? When you built Asterisk, you should have made sure to build the SIP channel driver you wanted to use, which may imply other requirements. ? If you use your own hardware phone, we assume both the phone and Asterisk can reach each other and are on the same subnet. Zoiper is suitable for users who require advanced features and customization capabilities.? You have a SIP phone plugged into the same LAN where the Asterisk server is plugged in, or can install the Zoiper softphone used in the example It also offers more customization options, allowing users to personalize the user interface and settings according to their preferences. Zoiper provides advanced call handling features such as call transfer, call recording, voicemail, and conferencing. It supports multiple protocols, including SIP, IAX, and others, making it compatible with various VoIP services. Zoiper, on the other hand, is a feature-rich softphone that offers a wide range of functionalities. However, it may have limited advanced features and customization options compared to more comprehensive softphones. MicroSIP supports standard SIP protocols and allows users to make voice and video calls, send instant messages, and manage contacts. It offers a minimalistic user interface and is designed to be efficient and resource-friendly. MicroSIP is a lightweight and simple SIP client that focuses on providing basic calling functionality. MicroSIP and Zoiper are both Voice over IP (VoIP) softphones, but they have distinct differences in their features and focus.
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