Skip to main content

Setup external calls (Outbound Route) - FreePBX tutorials

Before going to this step, you have completed Part 1 (Setup SIP trunk) and Part2 (Setup an extension) We have configured our trunk and our extension and now we need to tell FreePBX using that trunk when someone dials a number. To do that, you need to create an Outbound Routes Creating an outbound route Choose tab Connectivity => Outbound Routes outbound route   Create a name for route, using whatever name you want. i had used "ext-calls" outbound route   Then select the trunk you had created, my siptrunk was name "voip", so i picked it. After that, click submit changes. outbound route http://asteriskvoipsystem.org/setup-external-calls-freepbx-tutorials/

Comments

  1. 먹튀사이트 잡는 고릴라의 먹튀검증 통과 메이저토토사이트 안전놀이터 추천 훌륭한 먹튀검증을 통한 안전토토사이트를 추천합니다 고릴라만의 검증 시스템은 특별합니다 전세계적인 먹튀검증을 인전받은 최고의 메이저사이트 추천을 합니다 자세한 내용은 내 웹 사이트를 방문하십시오 검증사이트.

    ReplyDelete
  2. I can set up my new idea from this post. It gives in depth information. Thanks for this valuable information for all. FreePBX Exploit Phone Home

    ReplyDelete
  3. I truly discovered this an excessive amount of data. It is the thing that I was looking for. I might want to recommend you that please continue sharing such sort of information. Cloud PBX Houston

    ReplyDelete
  4. I will suggest reading this article because it will really help those who need this information. Thanks for the information which you have shared here. If anyone looking for the virtual pbx service then visit on dls.net

    ReplyDelete

Post a Comment

Popular posts from this blog

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

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

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/