Configuring an Asterisk server. If you want to set up Calculate Directory Server as an IP dial system, you should use Asterisk, a software implementation of a telephone PBX released under the GPL licence, that supports various Vo. IP protocols. To configure Asterisk, you will need to edit files /etc/asterisk. This article describes how to configure a regular Asterisk server. Problem specification. Linksys SPA2. 10. Incoming calls must go to the first available line. If the main phone line is busy, it goes to the fax line. Calls from a Moscow server to Saint- Petersburg (area code 8. Saint- Petersburg. All the settings below were tested on and apply to the following versions of packages. Install the Asterisk server. With the recent release of Asterisk 11 I thought I’d put together an install tutorial for Asterisk 11 and Centos 6. Run the make samples command to install the default configuration files. Here is a brief set of “install from source” instructions to install Asterisk 13. Changes to consider: new SIP channel driver powered by PJSIP SIP stack. My Asterisk PBX installation and configuration guide. Copy them to the asterisk configuration directory. Asterisk Default Install Directory Not FoundFor other packages, add support for all available codecs. Installing the required packages. It is now time for setup. Basic setup of SIP. To do this, edit the file /etc/asterisk/sip. Add a template for the SIP phones' common settings. The template is set with an exclamation point in brackets put near the section name . To mend this, you will have to add an outcoming SIP to /etc/asterisk/extensions. Configuring internal numbers. Freepbx on Ubuntu (Ubuntu v12, Asterisk v11. Install Asterisk/FreePBX required. If this has already been changed from the default/var/www directory then edit. Asterisk Default Install Directory OpencartThis page shows installation of Asterisk 11.0.0 on CentOS 6. It's been assumed that you have already installed CentOS 6 on your machine. Index of /pub/telephony/asterisk. Name Last modified Size Description; Parent Directory - misc/ 04-Nov-2013 13:45 - old-releases/ 09-Sep-2016 12:55 - releases/. How to install Asterisk on Debian, get it up and running in 15 minutes! Simple walkthrough with commands, video and the definitive guide to Asterisk. This guide covers the installation of Asterisk and Freepbx from source on Debian v7. Although Debian includes Asterisk deb packages they are not used in this guide. Asterisk is the #1 open source communications toolkit. Asterisk powers IP PBX systems, VoIP gateways, conference servers, and is used by SMBs, enterprises, call. Launch the console manager of your Asterisk server by running asterisk - r and reload; this will make Asterisk re- read all settings. Or you can also re- read them one by one, by running sip reload to restart SIP settings and dialplan reload to restart the dialing scheme. If you execute sip show peers, you will see the list of available users on the server, which of them are registered now and what their IPs are. Configure the connections of the two servers and calls passed between them. To allow this, you have to set up a connection between your two Asterisk servers. Setting up a connection between two Asterisk servers through the SIP protocol. Accordingly, this server will register on the Moscow server, 1. Now all our users in Moscow as well as in Saint- Petersburg (both servers running Calculate Directory Server with Asterisk) can join each other. Set Digium AEX8. 04. E and configure the dialing scheme. You must configure your interface card to allow them to. Setting up the system to work with the interface card. To do this, you will edit the /etc/dahdi/system. Your card has a module trunk including 4 FXO ports, with channels numbered from 5 to 8 (if you look on your card, these will be 4 left ports. Channels are numbered from right to left; consequently, the eighth channel is at the extreme left and the first channel is at the extreme right.)Append the following to the configuration file. The lines above mean that channels from 5 to 8 use fxsks signaling. Explaining FXO and FXS concepts. To do so, edit the /etc/asterisk/chan. The main line is on channel 7, while channel 8 is used for faxing. The group=1 parameter, set for both channels, puts them together in one group. This will permit you to make calls through the first available line. Configuring the dialing scheme for receiving calls. In the welcome queue, on the other had, the greeting message will be played, as we will describe later. Goto. If. Time(1. Ring: default) verifies the condition: if it is from 7. All phone sets will thus ring when the majority of staff are not in office, and in working hours the normal order is respected. Then, under the default tag, your call will go to the secretary for 6 seconds, then, for 1. Moscow and, finally, for 6. Saint- Petersburg. The r parameter in a Queue() call tells the program to play ringing instead of MOH. You will also need a context for the second line; create the section . This is done in the /etc/asterisk/queues. A record will be created for each member. To handle extensions in the welcome queue, you have specified the context . A record must be done accordingly in /etc/asterisk/extensions. During 1. 2 seconds, the person who called will hear the greeting message and, since you have specified the n parameter, will be able to dial an extension number that will be handled by the all- local- sip context. When 1. 2 seconds have elapsed and if no extension has been dialed, the call will be discarded from the queue and handled according to the dialing scheme. When the function Queue(msk- manager. Since you have specified the r parameter, the person who is calling will hear buzzing while they are waiting. Meanwhile, the call will be handled according to the strategy you specified - in this case, ringall (i. If any member in the queue takes the phone, the connection is established. Otherwise, if nobody takes the phone within 1. Customizing the Music On Hold (MOH) message. MOH is described in /etc/asterisk/musiconhold. When installing Asterisk, this file contains this. The recorded message is put in this directory. To decrease server load, you should first convert the message with ffmpeg@ (the package media- video/ffmpeg is pre- installed in Calculate Linux Desktop) to the most commonly used codecs: . When there is a call to a standard number (7 digits), it will be extended to 1. Moscow, where your server is located), while complete 1. The complete syntax is $. However, those are only main settings, that allow issuing and receiving calls. The Asterisk server provides unlimited telephony solutions for configuring and running an office IP- PBX. We hope that this article helped you understand how Asterisk works; now you can devise your own configuration. Asterisk Installation & Configuration . Similar configuration should also work for Asterisk 1. System Setup. Asterisk and SIP. Required Packages. Install the following dependencies: wgetgccgcc- c++ncurses- devellibxml. Using YUM, all dependencies can be installed with: yum install wget gcc gcc- c++ ncurses- devel libxml. Install libsrtp. First try installing libsrtp from the repo. If libsrtp is not available in the repo install it from source. CFLAGS=- f. PICmake & & make install. Install Asteriskcd /usr/local/src/. Download Asterisk with wget http: //downloads. Extract Asterisk: tar zxvf asterisk*. Enter the Asterisk directory: cd /usr/local/src/asterisk*. Run the Asterisk configure script: ./configure - -libdir=/usr/lib. Run the Asterisk menuselect tool: make menuselect. In the menuselect, go to the resources option and ensure that res. If there are 3 x’s next to res. Save the configuration (press x). Compile and install Asterisk: make & & make install. If you need the sample configs you can run make samples to install the sample configs. If you need to install the Asterisk startup script you can run make config. Setup DTLS Certificatesmkdir /etc/asterisk/keys. Enter the Asterisk scripts directory: cd /usr/local/src/asterisk*/contrib/scripts. Create the DTLS certificates (replace pbx. My Super Company with your company name): ./ast. The global settings do not flow down into the peer settings very well. Here you will set up two peers, one for a Web. RTC client and one for a non- Web. RTC SIP client. The Web. RTC peer requires encryption, avpf, and icesupport to be enabled. In most cases, directmedia should be disabled. Also under the Web. RTC client, the transport needs to be listed as . All of these config lines should be under the peer itself; setting these config lines globally might not work. Introduced in Asterisk 1. Tell Asterisk to enable DTLS for this peer. Tell Asterisk to not verify your DTLS certs. Tell Asterisk where your DTLS cert file is. Tell Asterisk where your DTLS private key is. Tell Asterisk to use actpass SDP parameter when setting up DTLS. When creating a UA, add the configuration parameter hack. Ip. In. Contact. If you are missing this property you will be able to make calls from Web. RTC, but not receive calls through Asterisk will fail. Additionally this guide will only work with audio calls, Asterisk will reject video calls. The following configuration example creates a UA for the Asterisk configuration above. Replace the values with the values from your config. This is new as of commit 3. SIP. js Master branch. Troubleshooting. Firefox 3. SIP. js 0. 6. 4 or later to interop with Free. SWITCH or Asterisk. This forum post on troubleshooting Web. RTC issues is a great guide for trouble shooting problems with Asterisk. Asterisk Secure Calling Guide can help you setup dtls certificates.
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