Jul 31, 2017 - DDNS should be set up in order to configure SSL for your NAS. (the asterisk stands for the subdomains that will be covered by the SSL). Email: provide your email address. This file contains server.csr which should be used for SSL activation and server.key which is necessary for SSL installation. Apr 24, 2013 I installed the 'official' Synology Asterisk package, and it works quite well. However, I would prefer to have FreePBX as a configuration tool on top of that, otherwise configuration and handling is rather complicated. Install FreePBX on top of Asterisk package. Quote; Unread post by davetunnicliff » Wed Apr 24, 2013 10:08 am Hi Derek.
Performance expectations The slug's IXP420 should have enough horse power for a home PBX with up to 4 lines, when less CPU intensive codecs (like GSM, G711u, G711a or G726) are used. The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible to rewrite them using the DSP extended instruction set supported by the IXP4xx. The Intel(R) IXP4XX DSP Software Library contains efficient implementations of all codecs including G729 and other VoIP goodies, but it looks that it cannot be used by asterisk. /opt/sbin/asterisk -r on the same computer on which Asterisk is running. More than one console CLI can connect to Asterisk simultaneously.
You can list all the available CLI commands by entering 'help', or get information on a particular command by entering 'help '. To start asterisk at boot time, create a script whose name starts with Snumbernumber in /opt/etc/init.d/ that executes asterisk: /opt/etc/init.d # cat S99asterisk #!/bin/sh if -f /opt/var/run/asterisk.pid ; then kill `cat /opt/var/run/asterisk.pid` else killall asterisk fi rm -f /opt/var/run/asterisk.pid umask 077 /opt/sbin/asterisk. How to configure music on hold Playing MP3 on the slug will not work. You will have to convert your MP3 files to native format GSM and/or ULAW (using for example the free sound conversion software from ) and configure asterisk to use the native format. Your musiconhold.conf file should look like this:; Music on hold class definitions;;native-random default mode=files directory=/opt/var/lib/asterisk/moh-native; Change to path of actual files random=yes; Play the files in a random order No volume or other sound adjustments are available (but you can use the WavePad sound editor from to do that or add effects). If the file is available in the same format as the channel's codec, then it will be played without transcoding. Files can be present in as many formats as you wish, and the 'best' format will be chosen at playback time.
NOTE: If you are not using 'autoload' in modules.conf, then you must ensure that the format modules for any formats you wish to use are loaded before resmusiconhold. If you do not do this, resmusiconhold will skip the files it is not able to understand when it loads.
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To transcode to ULAW (for example) using the 'switch' sound conversion software:. set the output format to.raw. in the encoder setings select:.
Format: G711 ULAW. Sample: 8000. Channels 1 - Mono. put the transcoded files in the directory specified in musiconhold.conf. change the.raw extension to.ulaw. How to configure the voicemail system to send messages by email I was not able to make Asterisk to work with the email client that came with the Linksys firmware (the voicemail message showed up appended to the text, instead of being attached as.wav file).
I've installed esmtp which has a sendmail compatible syntax: ipkg install esmtp Then, I created /opt/etc/esmtprc where I configured esmtp to use my ISP outgoing email server: hostname=smtp.myoutgoingmailserver.net:25 username= yourusername password= yourpassword. note - the username/password should be the same account as used in the serveremail entry in the voicemail.conf file In /opt/etc/asterisk/voicemail.conf I configured the following:.
in general section I configured the 1st recording format to be wav49 because it can be played by windows media player. Format=wav49. enabled voicemail to send messages as email attachment attach=yes. the serveremail line forms the 'From' part of the email header and will (most likely) be matched by your ISP against the username and password in the esmtprc file.
(anti-spam etc) serveremail= youusername@youremaildomain. the fromstring line forms the display portion of the 'From' email address - and as such an email from 'you' to 'you' could still bear the display name of 'myvm', and thus be sortable/filterable etc. Fromstring= emailfromdisplayname.
configured the command used to send email mailcmd=/opt/sbin/sendmail -t. note: the -t option allows esmtp to retreive the mailing information from the headers within the body (provided to esmtp by asterisk). added the email address to each mailbox 400 = 1234,John Smith,[email protected]. Call your favorite IM client from asterisk Here is an example of how to setup asterisk to be able to call yahoo buddies:.
define the following peer in sip.conf yahoo-proxy-out type=peer host=yahoo.com outboundproxy=yahoo.gtalk2voip.com fromuser=YourYahooID fromdomain=yahoo.com nat=yes canreinvite=no disallow=all allow=ulaw allow=gsm dtmfmode=rfc2833. create an extension for every yahoo ID you want to be able to call in your extensions.conf: exten = ExtensionForYahooBuddy,1,Dial(SIP/YahooBuddyID@yahoo-proxy-out,120,T). Asterisk and FreePBX For (Pogoplug V1, V2, Pro, Biz, Seagate Dockstar, Goflex) the instructions previously, actually, work pretty well. There may be some step I forgot but there was always a work around (e.g. Let ln -s be your friend), check permissions Don't do this unless you feel like thrashing a system and spending a lot of time. I can only give the highlights.
I may have missed some steps. But Plug Computers definitely can run Asterisk and FreePBX using Optware, my setup has been rock solid for 3 months straight. The setup I was able to try involved Asterisk 1.6, FreePBX 2.8, using Lighttpd Make sure that you have a fair sized drive, I use a minimum of a 4 Gb USB flash.
Basically, follow the instructions previously and Here is a summary from that article: Install lighttpd with PHP and Modules: php, phpmyadmin, php-fcgi, php-pear, php-mysql, fcgi, mysql. Two things about the php package in Optware: 1. You will get errors if you do not initialize pear 'pear install DB'.
Make sure to place '/opt/share/pear' in your php.ini includepath. Note: I get the question that when you run info.php, you get pear not installed.
That's ok just as long as you downloaded pear-php. More irritating was that that the repository php did not have gettext (can now download from nslu2-asterisk yahoo group file section). This was actually easy to solve but was tedious: Download the version of php source that you are using and build for just that module:./configure -with-gettext=shared,/opt/bin make cp modules/gettext.so /opt/lib/php/extensions/ in php.ini extension section extension=gettext.so Install Asterisk16 and Asterisk16-addons. Also the Asterisk14 core and extra sounds with alaw and ulaw. Install esmtp as above. Create an asterisk user and group.
Download FreePBX 'Extract the file and go into the subdirectory mysqladmin -p create asterisk mysqladmin -p create asteriskcdrdb mysql -p asterisk GRANT ALL PRIVILEGES ON asteriskcdrdb. TO asteriskuser@localhost IDENTIFIED BY 'secret'; mysql GRANT ALL PRIVILEGES ON asterisk. TO asteriskuser@localhost IDENTIFIED BY 'secret'; mysql flush privileges; mysql q./installamp -username=asteriskuser -password=secret Follow the prompts and use the values in the previous part of this page (above), remember the file structure of this system is based on /opt. Wherever you have asterisk or freepbx, make sure the files and directory are owned by asterisk user and group. Make sure you have this start script in /opt/etc/init.d with a name like S99asterisk (just like above with a few modifications). #!/bin/sh mount -o rw,remount / if -f /opt/var/run/asterisk.pid ; then kill `cat /opt/var/run/asterisk.pid` else killall asterisk fi rm -f /opt/var/run/asterisk.pid umask 077 #/opt/sbin/asterisk /opt/sbin/amportal start mount -o ro,remount / In '/etc', you need to make links to /opt/etc/asterisk and amportal.conf.
If you haven't done it, you need to change your var link from /tmp/var to /opt/var. There should be links in /usr/sbin for amportal and safeasterisk. Also there should be a link from /opt/include to /usr/include. So restart your system and make sure that asterisk starts. Typing 'ps -w' should get you: 602 root 2344 S /bin/sh /usr/sbin/safeasterisk -U asterisk -G asterisk 609 asterisk 35840 S /opt/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c You should see if you can load the FreePBX web page. Assuming everything went well, If you want to upgrade you FreePBX modules, now is the time before you set up your trunks and routes. FOP tends to take up some memory so you might need to disable it in amportal.conf with FOPDISABLE=true You can save some memory (somewhat necessary with a Dockstar) by shutting down my.pogoplug.com with a script in rcS that includes: /usr/bin/killall hbwd /usr/bin/killall udhcpc /usr/bin/killall hbplug /usr/bin/killall dropbear (Make sure you install openssh if you do this.) You can run 'asterisk -r' to run cli (Asterisk Command Line Interface).
Finally, if you want to install only Asterisk and not FreePBX, it is a lot quicker and fits nicely into a Dockstar (Remember, above, they did it with an NSLU2 with only 32Mb instead of the Dockstar's 128Mb or Pogoplug with 256Mb. FreePBX and Asterisk with all the trimmings is about 94 Mb). If you are a good Google jockey, you can actually search for a script and howto to send mp3 recordings of messages. A final caveat is to make sure your router firewall allows port 5060 to 5080 and port forwarding. These instructions also works with a Pogoplug Pro with Asterisk 1.8 and FreePBX 2.8. In manager.conf, make sure that enable=yes. There may be symlink errors that can be fixed manually.
Use Asterisk 1.8 in place of previous versions. Add-ons are included already. Use ulaw and alaw files from previous version.