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April 16, 2011, at 06:54 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 337 from:

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.

to:

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.

January 26, 2011, at 08:17 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 252 from:

For PlugComputers the instructions previously, actually, work pretty well.\\

to:

For PlugComputers (Pogoplug V1, V2, Pro, Biz, Seagate Dockstar, Goflex) the instructions previously, actually, work pretty well.\\

January 17, 2011, at 10:14 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed lines 264-265 from:
 Note: I get the question that when you run info.php, you get pear not installed.  Thats ok just as long as you downloaded pear-php.
2. 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:\\
to:
 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.
2. 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:\\
January 17, 2011, at 10:12 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 265 from:
 2.  More irritating was that that the repository php did not have gettext.  This was actually easy to solve but was tedious:\\
to:
 2.  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:\\
December 29, 2010, at 12:31 PM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed lines 336-338 from:

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.

to:

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.

December 29, 2010, at 12:27 PM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Added line 316:

Also there should be a link from /opt/include to /usr/include.\\

Changed lines 334-338 from:

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.

to:

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.

December 23, 2010, at 06:42 AM by OddballHero -- Plug guide for FreePBX and Asterisk.
Changed line 263 from:
 1.  You will get errors if you do not initialize pear "pear install DB".  Make sure to place "/opt/share/pear" in your PATH.\\
to:
 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 include_path.\\
October 17, 2010, at 10:11 PM by OddballHero -- Plug guide for FreePBX and Asterisk.
Changed line 263 from:
 1.  You will get errors if you do not initialize pear "pear install DB"\\
to:
 1.  You will get errors if you do not initialize pear "pear install DB".  Make sure to place "/opt/share/pear" in your PATH.\\
October 15, 2010, at 09:06 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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.

to:

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.

October 15, 2010, at 09:04 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 292 from:

Follow the prompts and use the values in the previous part of this page, remember the file structure of this system is based in /opt.\\

to:

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.\\

October 15, 2010, at 09:00 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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.

to:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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.

October 15, 2010, at 08:59 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed lines 330-331 from:
 /usr/bin/killall dropbear (make sure you install openssh if you do this.)
to:
 /usr/bin/killall dropbear (Make sure you install openssh if you do this.)
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). 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.

to:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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.

October 15, 2010, at 08:53 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). 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 firewall allows port 5060 to 5080 and port forwarding.

to:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). 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.

October 15, 2010, at 08:52 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). If you are a good Google jockey, you can actually search for a script and howto to send mp3 recordings of messages.

to:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). 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 firewall allows port 5060 to 5080 and port forwarding.

October 15, 2010, at 08:47 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 333 from:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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).

to:

Finally, if you want to install only Asterisk and not FreePBX, it is a lot easier 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). If you are a good Google jockey, you can actually search for a script and howto to send mp3 recordings of messages.

October 15, 2010, at 08:39 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed lines 325-326 from:

You can save some memory (somewhat necessary) by shutting down my.pogoplug.com with a script in rcS that includes:

to:

You can save some memory (somewhat necessary with a Dockstar) by shutting down my.pogoplug.com with a script in rcS that includes:

October 15, 2010, at 08:37 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 315 from:

There should be links in /usr/sbin for amportal and safe_sterisk.\\

to:

There should be links in /usr/sbin for amportal and safe_asterisk.\\

Changed lines 325-326 from:

You can save some memory by shutting down my.pogoplug.com with a script in rcS that includes:

to:

You can save some memory (somewhat necessary) by shutting down my.pogoplug.com with a script in rcS that includes:

Changed lines 332-333 from:

You can run "asterisk -r" to run cli (Asterisk Command Line Interface).

to:

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 easier 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).

October 15, 2010, at 08:25 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed lines 261-262 from:

Two things about the php package in Optware:

to:

Two things about the php package in Optware:

Changed lines 279-280 from:

Extract the file and go into the subdirectory

to:

Extract the file and go into the subdirectory

Changed lines 294-295 from:

Make sure you have this start script in /opt/etc/init.d with a name like S99asterisk (just like above with a few modifications).

to:

Make sure you have this start script in /opt/etc/init.d with a name like S99asterisk (just like above with a few modifications).

Changed line 316 from:

So restart your system and make sure that asterisk starts.\\

to:

So restart your system and make sure that asterisk starts.\\\

Changed lines 325-326 from:

You can save some memory by shutting down my.pogoplug.com with a script in rcS that includes:

to:

You can save some memory by shutting down my.pogoplug.com with a script in rcS that includes:

Changed lines 330-331 from:
 /usr/bin/killall dropbear make sure you install openssh if you do this.
to:
 /usr/bin/killall dropbear (make sure you install openssh if you do this.)
October 15, 2010, at 08:22 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 254 from:

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.\\

to:

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.\\

Changed lines 272-273 from:
 extension=gettext.so

to:
 extension=gettext.so
Changed lines 294-295 from:

Make sure you have this start script in /opt/etc/init.d with a name like S99asterisk.

to:

Make sure you have this start script in /opt/etc/init.d with a name like S99asterisk (just like above with a few modifications).

Changed line 332 from:

You can run "asterisk -r" to run cli (Asterisk Command Line Interface.

to:

You can run "asterisk -r" to run cli (Asterisk Command Line Interface).

October 15, 2010, at 08:17 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Changed line 251 from:

PlugComputers - Asterisk and FreePBX?

to:

PlugComputers - Asterisk and FreePBX

Changed line 255 from:

The setup I was able to try involved Asterisk 1.6, FreePBX? 2.8, using Lighttpd\\

to:

The setup I was able to try involved Asterisk 1.6, FreePBX 2.8, using Lighttpd\\

Added line 262:
Added line 273:
Changed line 278 from:

Download FreePBX? "http://mirror.freepbx.org/freepbx-2.8.0.tar.gz"\\

to:

Download FreePBX "http://mirror.freepbx.org/freepbx-2.8.0.tar.gz"\\

Added line 280:
Added line 291:
Changed line 293 from:

Wherever you have asterisk or freepbx, make sure the files and directory are owned by asterisk user and group.\\

to:

Wherever you have asterisk or freepbx, make sure the files and directory are owned by asterisk user and group.\\

Added line 295:
Added line 312:
Added line 318:
Changed lines 321-322 from:

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.\\

to:

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.\\

Added line 326:
Added line 331:
October 15, 2010, at 08:11 AM by OddballHero -- PlugComputers, Asterisk, and FreePBX with Lighttpd.
Added lines 250-322:

PlugComputers - Asterisk and FreePBX?

For PlugComputers 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 http://blog.hoopycat.com/2009/08/asterisk-freepbx-ubuntu-lighttpd-linode
Here is a summary from that article:
Install lighttpd with PHP and MySQL
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"
Note: I get the question that when you run info.php, you get pear not installed. Thats ok just as long as you downloaded pear-php.
2. More irritating was that that the repository php did not have gettext. 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? "http://mirror.freepbx.org/freepbx-2.8.0.tar.gz"
Extract the file and go into the subdirectory
mysqladmin -p create asterisk
mysqladmin -p create asteriskcdrdb
mysql -p asterisk < SQL/newinstall.sql
mysql -p asteriskcdrdb < SQL/cdr_mysql_table.sql
mysql -p
mysql> 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
./install_amp --username=asteriskuser --password=secret
Follow the prompts and use the values in the previous part of this page, remember the file structure of this system is based in /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.
#!/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 safe_sterisk.
So restart your system and make sure that asterisk starts.
Typing "ps -w" should get you:
602 root 2344 S /bin/sh /usr/sbin/safe_asterisk -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 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.
November 06, 2009, at 12:18 AM by cdoban -- updated asterisk book link
Changed lines 5-6 from:

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

to:

http://astbook.asteriskdocs.org/

January 03, 2009, at 08:11 PM by cdoban -- added : Missed Call Email Notification
Changed lines 169-170 from:
to:
November 09, 2008, at 06:51 PM by pa -- remove spam
Changed lines 1-248 from:

http://rollyo.com/search.html?q=xboxoffer.com&sid=web

to:

The Open Source VoIP PBX System

http://www.asterisk.org/

Download the O'Reilly book "Asterisk: The Future of Telephony" http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

Installation

From the root prompt, type:

ipkg install asterisk

Optionally install the additional sound package:

ipkg -force-overwrite install asterisk-sounds

Configuration:

The original sample configuration files are in /opt/etc/asterisk/sample

Take a look at it, consult the voip-info.org Asterisk wiki and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32MB of RAM I'll recommend you to use a slim configuration (modules.conf).

I have tested it with the second Asterisk slim configuration with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

You have to configure the path to the various asterisk component in asterisk.conf:

 [directories]
 astetcdir => /opt/etc/asterisk
 astmoddir => /opt/lib/asterisk/modules
 astvarlibdir => /opt/var/lib/asterisk
 astagidir => /opt/var/lib/asterisk/agi-bin
 astspooldir => /opt/var/spool/asterisk
 astrundir => /opt/var/run
 astlogdir => /opt/var/log/asterisk

Use the voip-info.org Asterisk wiki to find out how to configure:

 extensions.conf
 iax.conf
 sip.conf
 voicemail.conf

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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Flash installation

To install on a USB flash disk, 128Mb or more is recommended to allow room for voicemail files etc.. See Ext3flash. It has been reported to run on 64Mb.

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

Starting and stopping Asterisk

If you have just installed and configured Asterisk, you can try running it for the first time in console mode with some debugging applied with this command:

   /opt/sbin/asterisk -vvvc 

Use the command "stop now" to shut down Asterisk from the CLI console.

If run with no arguments, Asterisk is launched as a daemon process:

   /opt/sbin/asterisk 

You can get a CLI console to an already-running daemon by typing:

   /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 <command>".

To start asterisk at boot time, create a script whose name starts with S[number][number] 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

Asterisk GUI

There is a simple Asterisk GUI for Unslung (Optware): http://www.fivn.com/products/asterisk.html

How to connect a standard phone and to a PSTN phone line

An Analog Telephone Adaper (ATA) like Sipura SPA-3000 can be used to connect a standard analog phone and to connect Asterisk to a PSTN phone line.

How to use a Gizmo Project account with 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 http://www.nch.com.au/switch ) 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 http://www.nch.com.au 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_ res_musiconhold. If you do not do this, res_musiconhold will skip the files it is not able to understand when it loads.

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.my_outgoing_mail_server.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,my_email@address.com

Useful dialplan macros

Here are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

Useful features

Here are the recipes for some useful features:

Provisioning a Cisco 79XX series IP phone

The TFTP and HTML server capability of the NSLU2 can be used in conjunction with Asterisk to provision a Cisco 79XX series IP phone. For further information see: http://www.ambor.com/public/home_pabx/home_pabx.html

How to connect a YeaLink USB phone

This article describes how to connect and use a YeaLink USB-P1K handset with the NSLU2 as a standalone SIP VoIP phone.

How to make SIP work if NAT firewall is involved

  • in sip.conf, set nat=yes to the client definition:

[xlite1]
type=friend
regexten=401
username=xlite1
secret=passwd
context=default
callerid="John Smith" <401>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
mailbox=401@default

  • in the general section of sip.conf, provide either your domain or external IP if you don't have a domain:

externhost=yourdomain.net
externip = 200.201.202.203

  • configure your NAT router to forward the following ports to your NSLU2:

UDP 5060 for SIP (signalling)
UDP 10000-20000 for RTP (voice)

Make calls to/from IM clients

Call your sip address from your favorite IM client

Now thanks to gtalk2voip you can call your SIP address from your favorite IM client (yahoo messenger, windows live messenger, gtalk, or gizmoproject):

  • Setup:
    • from the gtalk2voip web page send yourself an invitation to join
    • add the gtalk2voip buddy to your buddies list
  • Make a call
    • open a chat window with the gtalk2voip buddy and send the following IM to it: CALL sip_address
    • gtalk2voip will first call the messenger (originating party) and after you accept the call it will call the sip address.

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)

gTalk

Here is an example of how to use asterisk 1.4 with Google Talk.

app_notify

Starting with asterisk14_1.4.13-2 app_notify is available (it can send notifications over the network to announce the callers name and telephone number to a desktop PC). For how to configure, check out [http://www.mezzo.net/asterisk/app_notify.html].

nslu2-asterisk group

For more information on using Asterisk on NSLU2 join the nslu2-asterisk group:
http://groups.yahoo.com/group/nslu2-asterisk/

November 09, 2008, at 12:24 PM by httprollyocomsearchhtmlqxboxoffercomsidweb -- <a href=\" http://www.exalead.fr/search/results?q=site%3Axboxoffer.comSearchs=web&t=0 \">xbox console</a> <a href=\" http://www.snap.com/classicsearch.php?query=site%3Axboxoffer.com.comm&go=Searchs=web&t=0 \">xbox 360 guitar hero</a> <a href=\" http://www.xomreviews.com/xboxoffer.com \">xbox</a>
Changed lines 1-248 from:

The Open Source VoIP PBX System

http://www.asterisk.org/

Download the O'Reilly book "Asterisk: The Future of Telephony" http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

Installation

From the root prompt, type:

ipkg install asterisk

Optionally install the additional sound package:

ipkg -force-overwrite install asterisk-sounds

Configuration:

The original sample configuration files are in /opt/etc/asterisk/sample

Take a look at it, consult the voip-info.org Asterisk wiki and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32MB of RAM I'll recommend you to use a slim configuration (modules.conf).

I have tested it with the second Asterisk slim configuration with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

You have to configure the path to the various asterisk component in asterisk.conf:

 [directories]
 astetcdir => /opt/etc/asterisk
 astmoddir => /opt/lib/asterisk/modules
 astvarlibdir => /opt/var/lib/asterisk
 astagidir => /opt/var/lib/asterisk/agi-bin
 astspooldir => /opt/var/spool/asterisk
 astrundir => /opt/var/run
 astlogdir => /opt/var/log/asterisk

Use the voip-info.org Asterisk wiki to find out how to configure:

 extensions.conf
 iax.conf
 sip.conf
 voicemail.conf

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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Flash installation

To install on a USB flash disk, 128Mb or more is recommended to allow room for voicemail files etc.. See Ext3flash. It has been reported to run on 64Mb.

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

Starting and stopping Asterisk

If you have just installed and configured Asterisk, you can try running it for the first time in console mode with some debugging applied with this command:

   /opt/sbin/asterisk -vvvc 

Use the command "stop now" to shut down Asterisk from the CLI console.

If run with no arguments, Asterisk is launched as a daemon process:

   /opt/sbin/asterisk 

You can get a CLI console to an already-running daemon by typing:

   /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 <command>".

To start asterisk at boot time, create a script whose name starts with S[number][number] 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

Asterisk GUI

There is a simple Asterisk GUI for Unslung (Optware): http://www.fivn.com/products/asterisk.html

How to connect a standard phone and to a PSTN phone line

An Analog Telephone Adaper (ATA) like Sipura SPA-3000 can be used to connect a standard analog phone and to connect Asterisk to a PSTN phone line.

How to use a Gizmo Project account with 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 http://www.nch.com.au/switch ) 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 http://www.nch.com.au 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_ res_musiconhold. If you do not do this, res_musiconhold will skip the files it is not able to understand when it loads.

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.my_outgoing_mail_server.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,my_email@address.com

Useful dialplan macros

Here are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

Useful features

Here are the recipes for some useful features:

Provisioning a Cisco 79XX series IP phone

The TFTP and HTML server capability of the NSLU2 can be used in conjunction with Asterisk to provision a Cisco 79XX series IP phone. For further information see: http://www.ambor.com/public/home_pabx/home_pabx.html

How to connect a YeaLink USB phone

This article describes how to connect and use a YeaLink USB-P1K handset with the NSLU2 as a standalone SIP VoIP phone.

How to make SIP work if NAT firewall is involved

  • in sip.conf, set nat=yes to the client definition:

[xlite1]
type=friend
regexten=401
username=xlite1
secret=passwd
context=default
callerid="John Smith" <401>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
mailbox=401@default

  • in the general section of sip.conf, provide either your domain or external IP if you don't have a domain:

externhost=yourdomain.net
externip = 200.201.202.203

  • configure your NAT router to forward the following ports to your NSLU2:

UDP 5060 for SIP (signalling)
UDP 10000-20000 for RTP (voice)

Make calls to/from IM clients

Call your sip address from your favorite IM client

Now thanks to gtalk2voip you can call your SIP address from your favorite IM client (yahoo messenger, windows live messenger, gtalk, or gizmoproject):

  • Setup:
    • from the gtalk2voip web page send yourself an invitation to join
    • add the gtalk2voip buddy to your buddies list
  • Make a call
    • open a chat window with the gtalk2voip buddy and send the following IM to it: CALL sip_address
    • gtalk2voip will first call the messenger (originating party) and after you accept the call it will call the sip address.

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)

gTalk

Here is an example of how to use asterisk 1.4 with Google Talk.

app_notify

Starting with asterisk14_1.4.13-2 app_notify is available (it can send notifications over the network to announce the callers name and telephone number to a desktop PC). For how to configure, check out [http://www.mezzo.net/asterisk/app_notify.html].

nslu2-asterisk group

For more information on using Asterisk on NSLU2 join the nslu2-asterisk group:
http://groups.yahoo.com/group/nslu2-asterisk/

to:

http://rollyo.com/search.html?q=xboxoffer.com&sid=web

February 22, 2008, at 07:33 AM by Loc Nguyen -- Update FIVN link
Changed lines 98-99 from:

There is a simple Asterisk GUI for Unslung (Optware): http://www.fivn.com/index.php?page=software&menu=asterisk

to:

There is a simple Asterisk GUI for Unslung (Optware): http://www.fivn.com/products/asterisk.html

November 14, 2007, at 03:55 AM by osas -- add app_notify info
Added lines 242-245:

app_notify

Starting with asterisk14_1.4.13-2 app_notify is available (it can send notifications over the network to announce the callers name and telephone number to a desktop PC). For how to configure, check out [http://www.mezzo.net/asterisk/app_notify.html].

October 14, 2007, at 05:26 AM by cdoban --
Changed lines 240-241 from:

Here is an example of how to use asterisk 1.4 with *Google Talk

to:

Here is an example of how to use asterisk 1.4 with Google Talk.

October 14, 2007, at 05:25 AM by cdoban -- added gTalk config example - posted by \"dickmars\"
Added lines 238-241:

gTalk

Here is an example of how to use asterisk 1.4 with *Google Talk

September 27, 2007, at 04:03 AM by Dusan maletic -- Corrected boot time asterisk script. Original one didn\'t work properly.
Added lines 82-92:
   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
Added line 94:
June 27, 2007, at 09:22 PM by lImbus -- typo
Changed line 50 from:

The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible to rewrite them using the DSP exetended 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:

to:

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:

March 24, 2007, at 12:45 AM by cdoban -- call to / from IM clients
Changed line 195 from:

call your sip address from your favorite IM client

to:

Call your sip address from your favorite IM client

Changed line 205 from:

call your favorite IM client from asterisk

to:

Call your favorite IM client from asterisk

March 24, 2007, at 12:42 AM by cdoban -- call to / from IM clients
Changed lines 193-194 from:

call to / from IM clients

to:

Make calls to/from IM clients

Changed line 198 from:
  • SETUP:
to:
  • Setup:
Changed lines 201-203 from:
  • MAKE A CALL
    • open a chat window with the gtalk2voip buddy and send the following IM to it:

CALL sip_address

to:
  • Make a call
    • open a chat window with the gtalk2voip buddy and send the following IM to it: CALL sip_address
Added lines 205-225:

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)
March 24, 2007, at 12:29 AM by cdoban -- call to / from IM clients
Changed lines 192-205 from:
to:

call to / from IM clients

call your sip address from your favorite IM client

Now thanks to gtalk2voip you can call your SIP address from your favorite IM client (yahoo messenger, windows live messenger, gtalk, or gizmoproject):

  • SETUP:
    • from the gtalk2voip web page send yourself an invitation to join
    • add the gtalk2voip buddy to your buddies list
  • MAKE A CALL
    • open a chat window with the gtalk2voip buddy and send the following IM to it:

CALL sip_address

  • gtalk2voip will first call the messenger (originating party) and after you accept the call it will call the sip address.
March 18, 2007, at 04:18 PM by osas -- moved gizmo section into a new page
Changed lines 92-128 from:

Add the following in sip.conf and replace YourGizmoSIPnumber and YourGizmoPassword with your 11 digits gizmo SIP number and password:

; register to gizmo
register => YourGizmoSIPnumber:YourGizmoPassword@proxy01.sipphone.com/YourGizmoSIPnumber

[gizmo]
type=friend
insecure=very
context=from-gizmo
username=YourGizmoSIPnumber
secret=YourGizmoPassword
host=proxy01.sipphone.com
fromuser=YourGizmoSIPnumber
fromdomain=proxy01.sipphone.com
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
dtmfmode=rfc2833

Add the following in extensions.conf:

; outgoing gizmo
exten => _9.,1,SetCallerID("your name" <YourGizmoSIPnumber>)
exten => _9.,2,Dial(SIP/${EXTEN:1}@gizmo,120,T)
exten => _9.,3,Congestion
exten => _9.,103,Congestion

[from-gizmo]
exten => YourGizmoSIPnumber,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => YourGizmoSIPnumber,2,Hangup

Note:
This will allow you to make outgoing calls to Gizmo by preceding the called number with 9. The stdexten macro assumes that the variables INRINGSEXT, INRINGSDEV are defined in your dial plan and represents the extension and device to ring when incoming calls are received from Gizmo.

to:
January 22, 2007, at 03:01 AM by cdoban -- added callbackDISA feature done by Jian
Added lines 191-194:

Useful features

Here are the recipes for some useful features:

January 20, 2007, at 03:38 AM by cdoban -- added callback macro
Changed lines 86-87 from:

There is available Asterisk GUI for Unslung (Optware). It is simple and easy which can maintain Asterisk without need to login a server. http://www.fivn.com/index.php?page=software&menu=asterisk

to:

There is a simple Asterisk GUI for Unslung (Optware): http://www.fivn.com/index.php?page=software&menu=asterisk

Changed lines 189-190 from:
to:
January 02, 2007, at 06:48 AM by Loc Nguyen -- Added Asterisk GUI
Changed lines 85-87 from:
to:

Asterisk GUI

There is available Asterisk GUI for Unslung (Optware). It is simple and easy which can maintain Asterisk without need to login a server. http://www.fivn.com/index.php?page=software&menu=asterisk

October 28, 2006, at 05:54 PM by cdoban -- How to make SIP work if NAT firewall is involved
Changed line 195 from:
  • set nat=yes to the client definition:
to:
  • in sip.conf, set nat=yes to the client definition:
Changed line 211 from:
  • in the general section provide either your domain or external IP if you don't have a domain:
to:
  • in the general section of sip.conf, provide either your domain or external IP if you don't have a domain:
October 28, 2006, at 05:51 PM by cdoban -- How to make SIP work if NAT firewall is involved
October 28, 2006, at 05:49 PM by cdoban --
Changed line 194 from:

How to make SIP work if Asterisk is behind NAT firewall

to:

How to make SIP work if NAT firewall is involved

Changed lines 196-209 from:

[xlite1] type=friend regexten=401 username=xlite1 secret=passwd context=default callerid="John Smith" <401> host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no disallow=all allow=ulaw allow=gsm

to:

[xlite1]
type=friend
regexten=401
username=xlite1
secret=passwd
context=default
callerid="John Smith" <401>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no
disallow=all
allow=ulaw
allow=gsm\\

Changed line 212 from:

externhost=yourdomain.net

to:

externhost=yourdomain.net\\

Changed line 216 from:

UDP 5060 for SIP (signalling)

to:

UDP 5060 for SIP (signalling)\\

October 28, 2006, at 05:45 PM by cdoban --
Added lines 194-219:

How to make SIP work if Asterisk is behind NAT firewall

  • set nat=yes to the client definition:

[xlite1] type=friend regexten=401 username=xlite1 secret=passwd context=default callerid="John Smith" <401> host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no disallow=all allow=ulaw allow=gsm mailbox=401@default

  • in the general section provide either your domain or external IP if you don't have a domain:

externhost=yourdomain.net externip = 200.201.202.203

  • configure your NAT router to forward the following ports to your NSLU2:

UDP 5060 for SIP (signalling) UDP 10000-20000 for RTP (voice)

October 16, 2006, at 09:57 PM by cdoban -- added link to \"How to connect a YeaLink USB phone\" page
Changed lines 191-193 from:

How to connect a YeaLink? USB phone

This article describes how to connect and use a YeaLink? USB-P1K? handset with the NSLU2 as a standalone SIP VoIP? phone.

to:

How to connect a YeaLink USB phone

This article describes how to connect and use a YeaLink USB-P1K handset with the NSLU2 as a standalone SIP VoIP phone.

October 16, 2006, at 09:55 PM by cdoban -- link to How to connect a YeaLink USB phone
Changed lines 191-193 from:

How to connect a Yeaphone USB phone

Here? are instructions for connecting a Yeaphone directly to the NSLU2 to be used as a SIP phone with asterisk.

to:

How to connect a YeaLink? USB phone

This article describes how to connect and use a YeaLink? USB-P1K? handset with the NSLU2 as a standalone SIP VoIP? phone.

October 16, 2006, at 09:50 PM by cdoban -- link to ConnectUSBPhone page
Added lines 191-193:

How to connect a Yeaphone USB phone

Here? are instructions for connecting a Yeaphone directly to the NSLU2 to be used as a SIP phone with asterisk.

August 18, 2006, at 08:47 PM by cdoban --
Changed lines 173-174 from:

serveremail=youusername@youremaildomain

to:

serveremail=youusername@youremaildomain

Deleted line 175:
Deleted line 178:
August 18, 2006, at 11:25 AM by Ian Watt -- addition of email display name detail
Changed lines 172-174 from:
  • the serveremail line forms the 'From' part of the email header and will (most likely) be match by your ISP against the username and password in the esmtprc file. (anti-spam etc)

serveremail=youusername@youremaildomain fromstring=EmailFromDisplayName?

to:
  • 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

Added line 181:
August 18, 2006, at 10:57 AM by Ian Watt -- added the display name for \'From\' in voicemail.conf
Added line 174:
August 18, 2006, at 04:33 AM by cdoban --
Changed lines 175-177 from:
  • the -t option allows esmtp to retreive the mailing information from the headers within the body (provided to esmtp by asterisk)

mailcmd=/opt/sbin/sendmail -t

  • added the same email address to each mailbox
to:

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
August 18, 2006, at 01:44 AM by Ian Watt -- small edits...
Changed line 175 from:
  • the -t option allows esmtp to retreive the mailing information from the headers within the body (provided to esmtp by asterisk)
to:
  • the -t option allows esmtp to retreive the mailing information from the headers within the body (provided to esmtp by asterisk)
Deleted lines 179-180:

(I'm not sure, but I remember having difficulties when I tried to configure distinct email addresses for each mailbox.) This is fixed when using the -t option

August 17, 2006, at 10:33 PM by Ian Watt -- emstp voicemail/email option fixed
Changed lines 162-163 from:

hostname=smtp.my_outgoing_mail_server.net:25

to:

hostname=smtp.my_outgoing_mail_server.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

Added lines 172-173:
  • the serveremail line forms the 'From' part of the email header and will (most likely) be match by your ISP against the username and password in the esmtprc file. (anti-spam etc)

serveremail=youusername@youremaildomain

Changed lines 175-176 from:

mailcmd=/opt/sbin/sendmail -f asterisk@my_domain.net my_email@address.com

to:
  • the -t option allows esmtp to retreive the mailing information from the headers within the body (provided to esmtp by asterisk)

mailcmd=/opt/sbin/sendmail -t

Changed lines 180-181 from:

I'm not sure, but I remember having difficulties when I tried to configure distinct email addresses for each mailbox.

to:

(I'm not sure, but I remember having difficulties when I tried to configure distinct email addresses for each mailbox.) This is fixed when using the -t option

July 13, 2006, at 12:10 PM by ambanmba -- Added Cisco 7940 section + some typo fixes
Changed line 177 from:

Hare are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

to:

Here are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

Added lines 182-184:

Provisioning a Cisco 79XX series IP phone

The TFTP and HTML server capability of the NSLU2 can be used in conjunction with Asterisk to provision a Cisco 79XX series IP phone. For further information see: http://www.ambor.com/public/home_pabx/home_pabx.html

July 12, 2006, at 05:04 AM by cdoban -- startup sctipt
Changed lines 78-79 from:

To start asterisk at boot time, create a script whose name starts with S[number][number] in /opt/etc/init.d/ that executes asterisk.

to:

To start asterisk at boot time, create a script whose name starts with S[number][number] in /opt/etc/init.d/ that executes asterisk:

   /opt/etc/init.d # cat S99asterisk
   #!/bin/sh
   /opt/sbin/asterisk
June 23, 2006, at 02:11 PM by JimmyFergus -- mention flash disk installation
Changed lines 54-55 from:

To install on a USB flash disk, 128Mb or more is recommended to allow room for voicemail files etc.. See Ext3flash. It has been reported to run on 64Mb.

to:

To install on a USB flash disk, 128Mb or more is recommended to allow room for voicemail files etc.. See Ext3flash. It has been reported to run on 64Mb.

June 23, 2006, at 02:10 PM by JimmyFergus -- mention flash disk installation
Added lines 53-55:

Flash installation

To install on a USB flash disk, 128Mb or more is recommended to allow room for voicemail files etc.. See Ext3flash. It has been reported to run on 64Mb.

June 11, 2006, at 06:12 AM by cdoban -- how to make asterisk VM to send emails on the slug
Changed line 147 from:

How to configure the voicemail system to send the messages by email

to:

How to configure the voicemail system to send messages by email

Changed lines 150-152 from:
  1. ipkg install esmtp

Then I created /opt/etc/esmtprc where I configured esmtp to use my ISP outgoing email server:

to:

> ipkg install esmtp

Then, I created /opt/etc/esmtprc where I configured esmtp to use my ISP outgoing email server:

Deleted line 166:
June 11, 2006, at 06:07 AM by cdban -- how to make asterisk VM send emails on the slug
Added lines 147-167:

How to configure the voicemail system to send the 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:

  1. ipkg install esmtp

Then I created /opt/etc/esmtprc where I configured esmtp to use my ISP outgoing email server: hostname=smtp.my_outgoing_mail_server.net:25

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

  • configured the command used to send email

mailcmd=/opt/sbin/sendmail -f asterisk@my_domain.net my_email@address.com

  • added the same email address to each mailbox

400 => 1234,John Smith,my_email@address.com

I'm not sure, but I remember having difficulties when I tried to configure distinct email addresses for each mailbox.

June 11, 2006, at 05:31 AM by cdban --
Deleted lines 72-73:

To start asterisk at boot time, create a script to launch asterisk whose name starts with S[number][number] to /opt/etc/init.d/.

Added lines 75-76:

To start asterisk at boot time, create a script whose name starts with S[number][number] in /opt/etc/init.d/ that executes asterisk.

June 10, 2006, at 08:20 PM by henry -- Add hint about starting asterisk when it boots up
Added lines 73-74:

To start asterisk at boot time, create a script to launch asterisk whose name starts with S[number][number] to /opt/etc/init.d/.

May 30, 2006, at 08:12 AM by gda -- restored
Changed lines 1-7 from:

leggey http://brin.to.p <a href="http://brin.to.pl/">buy phentermine</a> <a href="http://pechen.to.pl/">buy fioricet</a> <a href="http://karcer.to.pl/">buy diazepam</a> <a href="http://bugaboo.to.pl/">buy hydrocodone</a> <a href="http://zerro.to.pl/">buy cialis</a> <a href="http://pikachu.to.pl/">buy carisoprodol</a> <a href="http://gendalf.to.pl/">buy tramadol</a>

to:

The Open Source VoIP PBX System

http://www.asterisk.org/

Download the O'Reilly book "Asterisk: The Future of Telephony" http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

Installation

From the root prompt, type:

ipkg install asterisk

Optionally install the additional sound package:

ipkg -force-overwrite install asterisk-sounds

Configuration:

The original sample configuration files are in /opt/etc/asterisk/sample

Take a look at it, consult the voip-info.org Asterisk wiki and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32MB of RAM I'll recommend you to use a slim configuration (modules.conf).

I have tested it with the second Asterisk slim configuration with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

You have to configure the path to the various asterisk component in asterisk.conf:

 [directories]
 astetcdir => /opt/etc/asterisk
 astmoddir => /opt/lib/asterisk/modules
 astvarlibdir => /opt/var/lib/asterisk
 astagidir => /opt/var/lib/asterisk/agi-bin
 astspooldir => /opt/var/spool/asterisk
 astrundir => /opt/var/run
 astlogdir => /opt/var/log/asterisk

Use the voip-info.org Asterisk wiki to find out how to configure:

 extensions.conf
 iax.conf
 sip.conf
 voicemail.conf

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 exetended 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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

Starting and stopping Asterisk

If you have just installed and configured Asterisk, you can try running it for the first time in console mode with some debugging applied with this command:

   /opt/sbin/asterisk -vvvc 

Use the command "stop now" to shut down Asterisk from the CLI console.

If run with no arguments, Asterisk is launched as a daemon process:

   /opt/sbin/asterisk 

You can get a CLI console to an already-running daemon by typing:

   /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 <command>".

How to connect a standard phone and to a PSTN phone line

An Analog Telephone Adaper (ATA) like Sipura SPA-3000 can be used to connect a standard analog phone and to connect Asterisk to a PSTN phone line.

How to use a Gizmo Project account with asterisk

Add the following in sip.conf and replace YourGizmoSIPnumber and YourGizmoPassword with your 11 digits gizmo SIP number and password:

; register to gizmo
register => YourGizmoSIPnumber:YourGizmoPassword@proxy01.sipphone.com/YourGizmoSIPnumber

[gizmo]
type=friend
insecure=very
context=from-gizmo
username=YourGizmoSIPnumber
secret=YourGizmoPassword
host=proxy01.sipphone.com
fromuser=YourGizmoSIPnumber
fromdomain=proxy01.sipphone.com
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
dtmfmode=rfc2833

Add the following in extensions.conf:

; outgoing gizmo
exten => _9.,1,SetCallerID("your name" <YourGizmoSIPnumber>)
exten => _9.,2,Dial(SIP/${EXTEN:1}@gizmo,120,T)
exten => _9.,3,Congestion
exten => _9.,103,Congestion

[from-gizmo]
exten => YourGizmoSIPnumber,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => YourGizmoSIPnumber,2,Hangup

Note:
This will allow you to make outgoing calls to Gizmo by preceding the called number with 9. The stdexten macro assumes that the variables INRINGSEXT, INRINGSDEV are defined in your dial plan and represents the extension and device to ring when incoming calls are received from Gizmo.

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 http://www.nch.com.au/switch ) 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 http://www.nch.com.au 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_ res_musiconhold. If you do not do this, res_musiconhold will skip the files it is not able to understand when it loads.

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

Useful dialplan macros

Hare are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

nslu2-asterisk group

For more information on using Asterisk on NSLU2 join the nslu2-asterisk group:
http://groups.yahoo.com/group/nslu2-asterisk/

May 30, 2006, at 06:09 AM by legioner --
Changed lines 1-153 from:

The Open Source VoIP PBX System

http://www.asterisk.org/

Download the O'Reilly book "Asterisk: The Future of Telephony" http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

Installation

From the root prompt, type:

ipkg install asterisk

Optionally install the additional sound package:

ipkg -force-overwrite install asterisk-sounds

Configuration:

The original sample configuration files are in /opt/etc/asterisk/sample

Take a look at it, consult the voip-info.org Asterisk wiki and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32MB of RAM I'll recommend you to use a slim configuration (modules.conf).

I have tested it with the second Asterisk slim configuration with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

You have to configure the path to the various asterisk component in asterisk.conf:

 [directories]
 astetcdir => /opt/etc/asterisk
 astmoddir => /opt/lib/asterisk/modules
 astvarlibdir => /opt/var/lib/asterisk
 astagidir => /opt/var/lib/asterisk/agi-bin
 astspooldir => /opt/var/spool/asterisk
 astrundir => /opt/var/run
 astlogdir => /opt/var/log/asterisk

Use the voip-info.org Asterisk wiki to find out how to configure:

 extensions.conf
 iax.conf
 sip.conf
 voicemail.conf

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 exetended 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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

Starting and stopping Asterisk

If you have just installed and configured Asterisk, you can try running it for the first time in console mode with some debugging applied with this command:

   /opt/sbin/asterisk -vvvc 

Use the command "stop now" to shut down Asterisk from the CLI console.

If run with no arguments, Asterisk is launched as a daemon process:

   /opt/sbin/asterisk 

You can get a CLI console to an already-running daemon by typing:

   /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 <command>".

How to connect a standard phone and to a PSTN phone line

An Analog Telephone Adaper (ATA) like Sipura SPA-3000 can be used to connect a standard analog phone and to connect Asterisk to a PSTN phone line.

How to use a Gizmo Project account with asterisk

Add the following in sip.conf and replace YourGizmoSIPnumber and YourGizmoPassword with your 11 digits gizmo SIP number and password:

; register to gizmo
register => YourGizmoSIPnumber:YourGizmoPassword@proxy01.sipphone.com/YourGizmoSIPnumber

[gizmo]
type=friend
insecure=very
context=from-gizmo
username=YourGizmoSIPnumber
secret=YourGizmoPassword
host=proxy01.sipphone.com
fromuser=YourGizmoSIPnumber
fromdomain=proxy01.sipphone.com
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
dtmfmode=rfc2833

Add the following in extensions.conf:

; outgoing gizmo
exten => _9.,1,SetCallerID("your name" <YourGizmoSIPnumber>)
exten => _9.,2,Dial(SIP/${EXTEN:1}@gizmo,120,T)
exten => _9.,3,Congestion
exten => _9.,103,Congestion

[from-gizmo]
exten => YourGizmoSIPnumber,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => YourGizmoSIPnumber,2,Hangup

Note:
This will allow you to make outgoing calls to Gizmo by preceding the called number with 9. The stdexten macro assumes that the variables INRINGSEXT, INRINGSDEV are defined in your dial plan and represents the extension and device to ring when incoming calls are received from Gizmo.

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 http://www.nch.com.au/switch ) 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 http://www.nch.com.au 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_ res_musiconhold. If you do not do this, res_musiconhold will skip the files it is not able to understand when it loads.

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

Useful dialplan macros

Hare are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

nslu2-asterisk group

For more information on using Asterisk on NSLU2 join the nslu2-asterisk group:
http://groups.yahoo.com/group/nslu2-asterisk/

to:

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May 13, 2006, at 02:20 AM by osas -- fix the example config for music on hold
Changed lines 122-123 from:

[native-random]\\

to:

;[native-random]
[default]\\

May 09, 2006, at 04:03 AM by cdoban -- added link to nslu2-asterisk group
Added lines 150-152:

nslu2-asterisk group

For more information on using Asterisk on NSLU2 join the nslu2-asterisk group:
http://groups.yahoo.com/group/nslu2-asterisk/

May 07, 2006, at 10:30 PM by cdoban --
Changed lines 117-118 from:

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 convertion software from http://www.nch.com.au/switch ) and configure asterisk to use the native format.

to:

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 http://www.nch.com.au/switch ) and configure asterisk to use the native format.

April 26, 2006, at 05:19 AM by cdoban --
Changed line 124 from:

directory=/opt/lib/asterisk/moh-native ; Change to path of actual files\\

to:

directory=/opt/var/lib/asterisk/moh-native ; Change to path of actual files\\

April 13, 2006, at 12:05 AM by buggy -- Changed Music on hold to relect new format
Changed lines 119-124 from:

Your musiconhold.conf file should look like that:
[classes]
[moh_files]
default => /opt/var/lib/asterisk/moh-native,r ;change it to the actually path to your files

The 'r' option at the end cause the files to be played in random order.\\

to:

Your musiconhold.conf file should look like this:
; Music on hold class definitions
;
[native-random]
mode=files
directory=/opt/lib/asterisk/moh-native ; Change to path of actual files
random=yes ; Play the files in a random order
\\

March 09, 2006, at 03:58 AM by cdoban -- start/stop commands
Added lines 56-74:

Starting and stopping Asterisk

If you have just installed and configured Asterisk, you can try running it for the first time in console mode with some debugging applied with this command:

   /opt/sbin/asterisk -vvvc 

Use the command "stop now" to shut down Asterisk from the CLI console.

If run with no arguments, Asterisk is launched as a daemon process:

   /opt/sbin/asterisk 

You can get a CLI console to an already-running daemon by typing:

   /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 <command>".

February 13, 2006, at 11:36 PM by cdoban -- link to o\\
Added lines 4-6:

Download the O'Reilly book "Asterisk: The Future of Telephony" http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

February 10, 2006, at 05:26 PM by cdoban -- added useful dialplan macros
Changed lines 122-125 from:
to:
February 10, 2006, at 05:12 PM by cdoban -- added useful dialplan macros
Added lines 120-125:

Useful dialplan macros

Hare are some useful asterisk dialplan macros I create based on examples posted on www.voip-info.org:

February 06, 2006, at 10:16 PM by cdoban -- gizmo project config
Changed lines 57-58 from:

Add the following in sip.conf and replace <your gizmo SIP number> and <your gizmo SIP number> with your gizmo credentials:

to:

Add the following in sip.conf and replace YourGizmoSIPnumber and YourGizmoPassword with your 11 digits gizmo SIP number and password:

Changed lines 60-61 from:

register => <your gizmo SIP number>:<your gizmo password>@proxy01.sipphone.com/<your gizmo SIP number>

to:

register => YourGizmoSIPnumber:YourGizmoPassword@proxy01.sipphone.com/YourGizmoSIPnumber

Changed lines 66-67 from:

username=<your gizmo SIP number>
secret=<your gizmo password>\\

to:

username=YourGizmoSIPnumber
secret=YourGizmoPassword\\

Changed line 69 from:

fromuser=<your gizmo SIP number>\\

to:

fromuser=YourGizmoSIPnumber\\

Changed line 81 from:

exten => _9.,1,SetCallerID("your name" <your gizmo SIP number>)\\

to:

exten => _9.,1,SetCallerID("your name" <YourGizmoSIPnumber>)\\

Changed lines 87-89 from:

exten => <your gizmo SIP number>,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => <your gizmo SIP number>,2,Hangup

to:

exten => YourGizmoSIPnumber,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => YourGizmoSIPnumber,2,Hangup

February 06, 2006, at 02:22 AM by cdoban -- added config for gizmo project connection
Changed line 53 from:

Connect a standard phone and to a PSTN phone line

to:

How to connect a standard phone and to a PSTN phone line

Changed line 56 from:

How use a Gizmo Project account with asterisk

to:

How to use a Gizmo Project account with asterisk

Changed line 58 from:

\\

to:
Changed line 61 from:

\\

to:
Changed line 77 from:

\\

to:
Changed line 79 from:

\\

to:
Changed line 81 from:

exten => _9.,1,SetCallerID?("your name" <your gizmo SIP number>)\\

to:

exten => _9.,1,SetCallerID("your name" <your gizmo SIP number>)\\

Changed line 85 from:

\\

to:
Changed line 89 from:

\\

to:
February 06, 2006, at 02:18 AM by cdoban -- added config to connect to gizmo project
Added lines 56-93:

How use a Gizmo Project account with asterisk

Add the following in sip.conf and replace <your gizmo SIP number> and <your gizmo SIP number> with your gizmo credentials:
; register to gizmo
register => <your gizmo SIP number>:<your gizmo password>@proxy01.sipphone.com/<your gizmo SIP number>
[gizmo]
type=friend
insecure=very
context=from-gizmo
username=<your gizmo SIP number>
secret=<your gizmo password>
host=proxy01.sipphone.com
fromuser=<your gizmo SIP number>
fromdomain=proxy01.sipphone.com
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
dtmfmode=rfc2833
Add the following in extensions.conf:
; outgoing gizmo
exten => _9.,1,SetCallerID?("your name" <your gizmo SIP number>)
exten => _9.,2,Dial(SIP/${EXTEN:1}@gizmo,120,T)
exten => _9.,3,Congestion
exten => _9.,103,Congestion
[from-gizmo]
exten => <your gizmo SIP number>,1,Macro(stdexten,${INRINGSEXT},${INRINGSDEV})
exten => <your gizmo SIP number>,2,Hangup
Note:
This will allow you to make outgoing calls to Gizmo by preceding the called number with 9. The stdexten macro assumes that the variables INRINGSEXT, INRINGSDEV are defined in your dial plan and represents the extension and device to ring when incoming calls are received from Gizmo.

February 06, 2006, at 01:36 AM by cdoban --
Changed lines 44-53 from:

Connect a standard phone and to a PSTN phone line

to:

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 exetended 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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

Connect a standard phone and to a PSTN phone line

Changed line 56 from:

How to configure music on hold

to:

How to configure music on hold

Deleted lines 81-88:

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 exetended 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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug

If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

October 15, 2005, at 05:21 PM by cdoban -- music on hold configuration
Changed lines 53-55 from:

default => /opt/var/lib/asterisk/moh-native,r ;change it to the actually path to your files
\\

to:

default => /opt/var/lib/asterisk/moh-native,r ;change it to the actually path to your files

Changed lines 58-60 from:

Files can be present in as many formats as you wish, and the 'best' format will be chosen at playback time.
\\

to:

Files can be present in as many formats as you wish, and the 'best' format will be chosen at playback time.

Changed lines 64-72 from:

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

to:

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
October 15, 2005, at 07:55 AM by cdoban -- music on hold configuration
Added lines 47-74:

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 convertion software from http://www.nch.com.au/switch ) and configure asterisk to use the native format.

Your musiconhold.conf file should look like that:
[classes]
[moh_files]
default => /opt/var/lib/asterisk/moh-native,r ;change it to the actually path to your files

The 'r' option at the end cause the files to be played in random order.
No volume or other sound adjustments are available (but you can use the WavePad sound editor from http://www.nch.com.au 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_ res_musiconhold. If you do not do this, res_musiconhold will skip the files it is not able to understand when it loads.

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

October 09, 2005, at 09:29 PM by cdoban --
Changed lines 23-24 from:

I have tested it with the second Asterisk slim configuration guide with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

to:

I have tested it with the second Asterisk slim configuration with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

Changed lines 48-51 from:

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 and 711u) are used.

The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible (with a little effort) to use the Intel(R) IXP4XX DSP Software Library which contains efficient implementations of all codecs including G729. http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

to:

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 exetended 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: http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

June 27, 2005, at 05:01 AM by cdoban --
Changed line 44 from:

Connecting a standard phone and to the phone line

to:

Connect a standard phone and to a PSTN phone line

June 27, 2005, at 04:41 AM by cdoban -- connect a regular phone and phone line to the slug
Added lines 44-46:

Connecting a standard phone and to the phone line

An Analog Telephone Adaper (ATA) like Sipura SPA-3000 can be used to connect a standard analog phone and to connect Asterisk to a PSTN phone line.

June 13, 2005, at 08:46 PM by tman --
Changed lines 23-24 from:

I have tested it with the second Asterisk slim configuration guide with the iLBC codec commented out (requires a floating point unit)

to:

I have tested it with the second Asterisk slim configuration guide with the iLBC codec disabled as it requires a floating point unit which isn't present on the IXP420.

June 13, 2005, at 08:45 PM by tman --
Changed lines 18-19 from:

Take a look at it, consult the voip-info.org Aterisk wiki and create your configuration files in /opt/etc/asterisk

to:

Take a look at it, consult the voip-info.org Asterisk wiki and create your configuration files in /opt/etc/asterisk

June 13, 2005, at 08:44 PM by tman -- Cleaned up formatting
Changed line 1 from:

The Open Source VoIP PBX System \\

to:

The Open Source VoIP PBX System

Changed lines 4-5 from:

Installation \\

to:

Installation

Changed lines 7-9 from:
ipkg install asterisk
to:
ipkg install asterisk
Changed lines 11-24 from:
ipkg -force-overwrite install asterisk-sounds


Configuration:
The original sample configuration files are in /opt/etc/asterisk/sample
Take a look at it, consult the wiki page ( http://www.voip-info.org/wiki-Asterisk ) and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32Mb of RAM I'll recommend you to use a slim configuration (modules.conf).
I have tested it with the second slim configuration described here:
http://www.voip-info.org/wiki-Asterisk+Slimming
with the iLBC codec commented out (requires a float coprocessor).
\\

to:
ipkg -force-overwrite install asterisk-sounds

Configuration:

The original sample configuration files are in /opt/etc/asterisk/sample

Take a look at it, consult the voip-info.org Aterisk wiki and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32MB of RAM I'll recommend you to use a slim configuration (modules.conf).

I have tested it with the second Asterisk slim configuration guide with the iLBC codec commented out (requires a floating point unit)

Changed lines 26-49 from:

asterisk.conf:
cat asterisk.conf
[directories]
astetcdir => /opt/etc/asterisk
astmoddir => /opt/lib/asterisk/modules
astvarlibdir => /opt/var/lib/asterisk
astagidir => /opt/var/lib/asterisk/agi-bin
astspooldir => /opt/var/spool/asterisk
astrundir => /opt/var/run
astlogdir => /opt/var/log/asterisk

Use the wiki to find out how to configure:

extensions.conf
iax.conf
sip.conf
voicemail.conf


Performance? expectations:
The slug's IXP400 should have enough horse power for a home PBX with up to 4 lines, when less CPU intensive codecs (like GSM and 711u) are used.
The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible (with a little effort) to use the Intel(R) IXP4XX DSP Software Library which contains efficient implementations of all codecs including G729.
http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug \\

to:

asterisk.conf:

 [directories]
 astetcdir => /opt/etc/asterisk
 astmoddir => /opt/lib/asterisk/modules
 astvarlibdir => /opt/var/lib/asterisk
 astagidir => /opt/var/lib/asterisk/agi-bin
 astspooldir => /opt/var/spool/asterisk
 astrundir => /opt/var/run
 astlogdir => /opt/var/log/asterisk

Use the voip-info.org Asterisk wiki to find out how to configure:

 extensions.conf
 iax.conf
 sip.conf
 voicemail.conf

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 and 711u) are used.

The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible (with a little effort) to use the Intel(R) IXP4XX DSP Software Library which contains efficient implementations of all codecs including G729. http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm

Asterisk sample configuration for Slug

Deleted line 50:

\\

June 04, 2005, at 04:35 PM by cdoban --
Changed lines 11-12 from:

The original sample configuration files are in:
/opt/etc/asterisk/sample \\

to:

The original sample configuration files are in /opt/etc/asterisk/sample \\

Changed lines 34-38 from:

Use the wiki to find out how to configure:
extensions.conf
iax.conf
sip.conf
voicemail.conf \\

to:

Use the wiki to find out how to configure:

extensions.conf
iax.conf
sip.conf
voicemail.conf
June 04, 2005, at 04:32 PM by cdoban --
Changed lines 5-8 from:

From the root prompt, type:
-> ipkg install asterisk
optionally install the additional sound package:
-> ipkg -force-overwrite install asterisk-sounds \\

to:

From the root prompt, type:

ipkg install asterisk

Optionally install the additional sound package:

ipkg -force-overwrite install asterisk-sounds
Changed line 41 from:

Performance? expectation: \\

to:

Performance? expectations: \\

Deleted line 44:

http://www.intel.com/design/network/manuals/273811_v_1_1.htm \\

Changed lines 47-48 from:
to:

Asterisk sample configuration for Slug
If you want to try the Asterisk VoIP PBX without going trough the hassle of configuring it from the scratch, you can start with this sample configuration and you will have Asterisk running on the Slug in minutes.

June 04, 2005, at 04:24 PM by cdoban --
Added lines 4-9:

Installation
From the root prompt, type:
-> ipkg install asterisk
optionally install the additional sound package:
-> ipkg -force-overwrite install asterisk-sounds
\\

June 01, 2005, at 05:03 PM by cdoban -- added Asterisk sample configuration for Slug
Added lines 41-43:
June 01, 2005, at 05:09 AM by cdoban --
Changed line 38 from:

(with a little effort) to use the Intel(R) IXP4XX DSP Software Library Release 1.1 which contains efficient implementations of all codecs including G729. \\

to:

(with a little effort) to use the Intel(R) IXP4XX DSP Software Library which contains efficient implementations of all codecs including G729. \\

June 01, 2005, at 12:06 AM by Dietmar Zlabinger -- added link to performance
Changed line 35 from:

Performance expectation: \\

to:

Performance? expectation: \\

May 30, 2005, at 09:40 PM by cdoban -- asterisk performance expectation
Changed lines 34-40 from:
to:


Performance expectation:
The slug's IXP400 should have enough horse power for a home PBX with up to 4 lines, when less CPU intensive codecs (like GSM and 711u) are used.
The CPU intensive codecs (iLBC, G729, Speex) are not working, but it should be possible (with a little effort) to use the Intel(R) IXP4XX DSP Software Library Release 1.1 which contains efficient implementations of all codecs including G729.
http://www.intel.com/design/network/manuals/273811_v_1_1.htm
http://www.intel.com/design/network/products/npfamily/ixp425swr1.htm \\

May 30, 2005, at 08:43 PM by cdoban --
Changed line 1 from:

Asterisk - The Open Source VoIP PBX System \\

to:

The Open Source VoIP PBX System \\

Added line 34:
May 30, 2005, at 08:40 PM by cdoban -- added asterisk page
Added lines 1-33:

Asterisk - The Open Source VoIP PBX System
http://www.asterisk.org/

Configuration:
The original sample configuration files are in:
/opt/etc/asterisk/sample
Take a look at it, consult the wiki page ( http://www.voip-info.org/wiki-Asterisk ) and create your configuration files in /opt/etc/asterisk

Because the NSLU has only 32Mb of RAM I'll recommend you to use a slim configuration (modules.conf).
I have tested it with the second slim configuration described here:
http://www.voip-info.org/wiki-Asterisk+Slimming
with the iLBC codec commented out (requires a float coprocessor).

You have to configure the path to the various asterisk component in asterisk.conf:
cat asterisk.conf
[directories]
astetcdir => /opt/etc/asterisk
astmoddir => /opt/lib/asterisk/modules
astvarlibdir => /opt/var/lib/asterisk
astagidir => /opt/var/lib/asterisk/agi-bin
astspooldir => /opt/var/spool/asterisk
astrundir => /opt/var/run
astlogdir => /opt/var/log/asterisk

Use the wiki to find out how to configure:
extensions.conf
iax.conf
sip.conf
voicemail.conf \\

Page last modified on April 16, 2011, at 06:54 AM