Skip to main content

Terms and Conditions

Welcome to asteriskvoipsystem.blogspot.com. If you continue to browse and use this website you are agreeing to comply with and be bound by the following terms and conditions of use, which together with our privacy policy govern asteriskvoipsystem.blogspot.com's relationship with you in relation to this website.

The term asteriskvoipsystem.blogspot.com or 'us' or 'we' refers to the owner of the website. The term 'you' refers to the user or viewer of our website. The use of this website is subject to the following terms of use:

The content of the pages of this website is for your general information and use only. It is subject to change without notice.
Neither we nor any third parties provide any warranty or guarantee as to the accuracy, timeliness, performance, completeness or suitability of the information and materials found or offered on this website for any particular purpose. You acknowledge that such information and materials may contain inaccuracies or errors and we expressly exclude liability for any such inaccuracies or errors to the fullest extent permitted by law.
Your use of any information or materials on this website is entirely at your own risk, for which we shall not be liable. It shall be your own responsibility to ensure that any products, services or information available through this website meet your specific requirements.
This website contains material which is owned by or licensed to us. This material includes, but is not limited to, the design, layout, look, appearance and graphics. Reproduction is prohibited other than in accordance with the copyright notice, which forms part of these terms and conditions.
All trademarks reproduced in this website, which are not the property of, or licensed to the operator, are acknowledged on the website.
Unauthorised use of this website may give rise to a claim for damages and/or be a criminal offence.
From time to time this website may also include links to other websites. These links are provided for your convenience to provide further information. They do not signify that we endorse the website(s). We have no responsibility for the content of the linked website(s).
You may not create a link to this website from another website or document without asteriskvoipsystem.blogspot.com's prior written consent.

Comments

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
5 comments
Read more

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
16 comments
Read more

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/
10 comments
Read more