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Asterisk – CLI commands

Agent commands agent logoff  - Sets an agent offline agent show  - Show status of agents agent show online  - Show all online agents AGI commands agi dump html  - Dumps a list of AGI commands in HTML format agi exec  - Add AGI command to a channel in Async AGI agi set debug [on|off]  - Enable/Disable AGI debugging agi show commands [topic]  - List AGI commands or specific help dnsmgr refresh  - Performs an immediate refresh dnsmgr reload  - Reloads the DNS manager configuration dnsmgr status  - Display the DNS manager status Calendar commands calendar dump sched  - Dump calendar sched context calendar show calendar  - Display information about a calendar calendar show calendars  - Show registered calendars Channel commands channel originate  - Originate a call channel redirect  - Redirect a call channel request hangup  - Request a hangup on a given channel Cli commands cli check permissions  - Try a permissions config for a user cli reload permi

Making video calls – Asterisk tutorial

sip.conf To be able to send video during a call, codec h263 and video support must be enabled. This is done by adding three lines in the sip.conf file (Location: /etc/asterisk/sip.conf). Add the following lines in the [general] tab of the file. videosupport = yes ; Enable video allow = h263 ; H.263 is our video codec allow = h263p ; H.263p is the enhanced video codec Reload Reload the sip.conf file by running the following command in the CLI console:  reload Config in Softphone  Click “Softphone”  Click “Preferences”    Click “Video Codecs”  Verify that h263 and h263+ are selected Click “OK”  Click the video (Webcam) icon to display the video  Dial the other sip phone (number 1002)  Click “Show Video” on the two sip phones. http://asteriskvoipsystem.org/making-video-calls-asterisk-tutorial/

Step by step: Configure call recording - Asterisk tutorials

This article will cover enabling asterisk to record calls. You may want this to interview people over the phone, podcast, or some other purpose. In features.conf, under: [featuremap] uncomment the line that looks like this: automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor — Make sure to set the X and/or x option in the Dial() or    Queue() app call! Then, enable the X option for Dial() in your dialplan in extensions.conf: PLEASE NOTE:  change you need to make – add the X  your dial rule make look different with me. exten => s,n,Dial(SIP/100,60) make it this instead: exten => s,n,Dial(SIP/100,60,X) The X is what tells Asterisk to allow callers to dial *3 during a call to enable or disable recording. From the asterisk console (run asterisk -r), you should see a line like this appear when the user starts a recording: – User hit ‘*3′ to record call. filename: auto-xxxxx-EXTENSION-DIALEDNUMBER When the recording ends, un