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Home Last modified: 17-12-2013

Set up your own PBX with Asterisk

Introduction

Important: To log stuff to the console, either use Verbose(), or use NoOp() but the latter will only work if you set "verbosity" to at least 3 (in the console, type "set verbose 3").

Why choose Asterisk to build a PBX over other open-source solutions?

"There are several SIP implementations that are OSS, but they are primarily what are known as "call proxies" instead of more full-featured PBX applications. This means that they

function only to connect two endpoints together, and are basically just large, fast, directory servers. Examples of SIP Proxies are ser and Vocal."

Is Asterisk the only PBX that can rewrite CID name on the fly? Check Freeswitch

Which environment to choose?

To set up Asterisk, several solutions are available:

Install a bare Linux distro, and install the whole shebang from source code (recommended)

Install a bare Linux distribution that supports RPM or other packagers, and install the required components through this package in binary form. Red Hat RPM packages for Asterisk

and the driver modules can be obtained from http://www.macvoip.com or from http://www.n2net.net

"Engineered by Digium in conjunction with rPath, Pound Key includes all the Linux components necessary to run, debug and build Asterisk, and only those components. You no

longer have to worry about kernel versions and package dependencies. Unlike other Linux distributions used to deploy Asterisk, no unnecessary components that might

compromise security or performance are included." AstLinux is also a bundled version of Linux and Asterisk, but optimized for small format hardware platforms

(Important: You might want to take a deep look at whether you want to use a commercial Asterisk) Use the TrixBox distribution (previously Asterisk@Home) that aims to do the

same thing as Pound Key. Ideal for newbies, but requires a second host since, on purpose, it must be managed through a web interface but doesn't have X (Lynx could do the

job, I guess). I don't like A@H because it hides the internals (hence, not a good tool to learn how Asterisk works), and installs a lot of stuff that is probably useless for a home

solution (SugarCRM, etc.)

How to connect Asterisk to the POTS/PSTN (ie. regular, analog phone line)

There are two solutions:

A stand-alone PSTN gateway like the Linksys 3102 or the Minitar MVA11A

A PCI card (don't use the cheap X100P/X101P cards: Get quality hardware from brands like Sangoma, Mediatrix, Welltech, Patton, Audiocodes, etc.)

Which branch to use?

As of Feb 2012, four branches are available: 1.4, 1.6, 1.8, and 10. I don't know of a good source to know what the major changes are in each branch and make an informed choice

before upgrading, but some information are available in UPGRADE*.txt files in tarballs along with the CHANGES file.

Installing from packages

Here's how to install Asterisk on Ubuntu from packages.

Asterisk and dahdi

1. apt-get install asterisk

2. apt-get install asterisk-config

Installing "asterisk" takes care of installing configuration files and Dahdi

Sound files

Those files are encoded in GSM.

1. apt-get install asterisk-sounds-main

2. apt-get install asterisk-sounds-extra

Localized Sound Files

Voice prompts are available for a few languages:

asterisk-prompt-fr-armelle - French voice prompts for Asterisk by Armelle Desjardins

asterisk-prompt-fr-proformatique - French voice prompts for Asterisk

http://downloads.asterisk.org/pub/telephony/sounds/

Post-install tweaking

The following changes included some needed to use Dahdi, ie. with a PCI interface.

1. cd /etc/asterisk

2. cp modules.conf modules.conf.orig

3. vi modules:

noload => pbx_ael.so

noload => pbx_lua.so

noload => app_cdr.so

noload => app_fax.so

noload => app_festival.so

noload => app_followme.so

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noload => app_forkcdr.so

noload => app_mp3.so

noload => app_meetme.so

noload => res_ldap.conf

noload => res_phoneprov.so

4. mv features.conf features.conf.orig

5. mv users.conf users.conf.orig

6. mv sip.conf sip.conf.orig

7. mv extensions.conf extension.conf.orig

8. mv say.conf say.conf.orig

9. vi chan_dahdi.conf:

language=fr

10. vi cdr.conf: enable=no

11. cd /etc/dahdi

12. vi modules

13. vi system.conf

14. vi etc/modprobe.d/dahdi.conf: options wctdm opermode=FRANCE

15. /etc/init.d/asterisk restart

Basic Asterisk server with Dahdi from source

Dependencies

1. Update the host with the latest of installed apps + kernel/kernel-dev. If the kernel was updated, reboot to use the latest version

2. Download the dependencies:

ncurses ncurses-devel

openssl openssl-devel

libssl-dev

zlib zlib-devel zlib1g-dev

bison bison-devel

libnewt-dev

initrd-tools

procps

wget

libusb-dev (to avoid "waitfor_xpds: Missing astribank_is_starting")

For CentOS:

yum install kernel-devel kernel bison openssl-devel gcc gcc-c++ libtermcap libtermcap-devel ncurses ncurses-devel zlib zlib-devel newt newt-devel wget

On a Debian/Ubuntu server, I've seen this recommended:

apt-get install build-essential libncurses5-dev libcurl3-dev libvorbis-dev libspeex-dev unixodbc unixodbc-dev libiksemel-dev linux-headers-`uname -r` libnewt-dev wget

Asterisk

1. cd /usr/src

2. wget -c http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz

3. tar xzvf asterisk-1.4-current.tar.gz

4. cd asterisk-1.4.????

5. ./configure

6. make menuselect (UI OK from Putty, BAD from SecureCRT) > Core Sound Packages, Music On Hold File Packages, and Extras Sound Packages

7. make

8. make install

9. make samples

10. make config

Asterisk add-on's

Includes MySQL support for call detail records and MP3 support for MOH.

1. cd /usr/src

2. wget -c http://downloads.digium.com/pub/asterisk/asterisk-addons-1.4-current.tar.gz

3. tar xzvf asterisk-addons-1.4-current.tar.gz

4. cd asterisk-addons-1.4.11

5. ./configure

6. make

7. make install

8. make samples

Dadhi

1. lspci -vv (Check that the hardware is detected)

2. cd /usr/src

3. wget -c http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz

4. tar xzvf dahdi-linux-complete-current.tar.gz

5. cd dahdi-linux-complete-2.3.0.1+2.3.0

6. make

7. make install

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8. make config

9. (Not needed) vi /etc/modprobe.conf

10. vi /etc/modprobe.d/dahdi.conf ("You should place any module parameters for your DAHDI modules here")

options wctdm opermode=FRANCE

11. (optional) vi /etc/modprobe.d/blacklist ("blacklist all the drivers by default in order to ensure that /etc/init.d/dahdi installs them in the correct order so that the spans are ordered

consistently.")

1. Comment out modules you do NOT want to blacklist

2. Add "blacklist netjet"

3. Reboot to get rid of the NetJet module that is loaded instead of the Wctdm module, and run "lsmod | grep -i netjet" to check that netjet/ISDN are gone

12. vi /etc/dahdi/modules

wctdm

13. vi /etc/dahdi/system.conf

loadzone = fr

defaultzone = fr

fxsks = 1

echocanceller=mg2,1

14. vi /etc/asterisk/indications.conf

[general]

country=fr

[fr]

description = France

; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

ringcadence = 1500,3500

; Dialtone can also be 440+330

dial = 440

busy = 440/500,0/500

ring = 440/1500,0/3500

; CONGESTION - not specified

congestion = 440/250,0/250

callwait = 440/300,0/10000

; DIALRECALL - not specified

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

; RECORDTONE - not specified

record = 1400/500,0/15000

info = !950/330,!1400/330,!1800/330

stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440

15. vi /etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]

signalling = fxs_ks

usecallerid = yes

hidecallerid = no

callwaiting = yes

callwaitingcallerid = yes

threewaycalling = yes

transfer = yes

canpark = yes

cancallforward = yes

callreturn = yes

echocancel = yes

echocancelwhenbridged = yes

relaxdtmf = yes

rxgain = 0.0

txgain = 0.0

busydetect=yes

;busycount=6

answeronpolarityswitch=yes

hanguponpolarityswitch=yes

context = from-pstn

channel => 1

1. vi extensions.conf

[from-pstn]

exten => s,1,NoOp(Incoming call)

2. /etc/init.d/dahdi start, tail /var/log/messages, and check LED on PCI card

3. Restart Asterisk

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In case of trouble, try to put the card into another slot, or even in a difference computer. Take a look at OpenVox's Troubleshooting of Analog cards.

If you get "driver should be 'wctdm' but is actually 'netjet'" when starting dahdi:

Commands

dahdi_cfg -vv to configure

dahdi_hardware to check hardware

dahdi_scan to display channels

CLI> dahdi show status

Troubleshooting

# dmesg

# lspci -vv

# dahdi_hardware

# dahdi_cfg -vv

SIP accounts

1. Create SIP accounts and a basic dialplan:

cd /etc/asterisk

mkdir orig

mv sip.conf ./orig

mv extensions.conf ./orig

vim sip.conf:

[general]

port = 5060

bindaddr = 0.0.0.0

context = others

disallow=all

allow=ulaw

allow=alaw

allow=gsm

nat=no

qualify=yes

host=dynamic

[2000]

type=friend

context=my-phones

secret=1234

[2001]

type=friend

context=my-phones

secret=1234

vim extensions.conf:

[others]

[my-phones]

exten => 2000,1,Dial(SIP/2000)

exten => 2001,1,Dial(SIP/2001)

exten => s,1,Verbose(Yes!)

exten => 9999,1,Goto(s,1)

2. Launch Asterisk, and connect to its console:

safe_asterisk

asterisk -r ("quit" to exit)

3. Configure two SIP phones to connect to Asterisk with the above accounts, and use one phone to ring the other

Once installed, Asterisk files can be located in the following directories:

/etc/asterisk/

/etc/zaptel.conf (not part of Asterisk, so located outside /etc/asterisk/)

/usr/lib/asterisk/modules/

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/var/lib/asterisk/

/var/spool/asterisk/

/var/run/

/var/log/asterisk/

Modules are located under /lib/modules/'uname -r'/misc (eg. wcfxo.o, zaptel.o, ztdummy.o, etc.)

By default, Asterisk loads a lot of stuff, and must be told explicitely not to load them through /etc/asterisk/modules.conf, using eg. noload => pbx_ael.so. Modules usually live under

/usr/lib/asterisk/modules/.

IAX accounts

Compiling Asterisk for Windows

1. Install Cygwin. You may need to manually install/upgrade tools like autoconf, automake etc depending on your Cygwin installation

2. Install build essentials in Cygwin

3. Download Asterisk source (I used 1.4.x) and unzip it using tar (You may need to install tar manually as it is missing in some Cygwin default installations. Don't use windows unzip for

as it will create some abnormal character in source and will make unexpected compile time errors)

4. Run bootstrap it will report any missing or lower version libs, prerequisite or tools

5. You manually need to download and compile termcap, ncurses

6. Run configure

7. Make menuselect and disable all non-required modules as it will save to resolve lot of not needed dependencies

8. Run make

9. Resolve any missing reported by make

10. After successful make run make install

11. Once make install okey you can run asterisk on Cygwin console and also directly run by double clicking on asterisk.exe in c:/Cygwin/usr/sbin/.

Once you have compiled it you can copy asterisk.exe to any other system not having Cygwin installed but you have to care about following:

You must have to create required directories structure like Cygwin on system drive.

You must need to copy required Cygwin DLLs to new systems \windows\system32\ folder. You can identify required DLLs by trying to run asterisk.exe and it will report missing

DLLs one by one.

I did just for my experiment and fun and was able to make successful SIP calls using static files configuration. However I suggest to use SIPx, Yate or FreeSWITCH if you want to stick

with windows as that have native windows ports and have all required features you need in a PABX or VoIP switch.

One more thing previously there was a project named as AstWin which was maintaining asterisk's port to windows and providing an installable package of Asterisk for windows. I am not

aware about current state of project but, I have installation package of Asterisk for windows version 1.2. If anyone need it contact me direct at email imfanee@gmail.com I will send the

software as attachment.

Tips to compile Asterisk

Since Asterisk is often updated, packages found on the Net are usually a bit stale, and it 's better to learn how to compile it yourself. Here are some tips I gathered:

Set a hostname, eg. "sip" or "sip.acme.com" (but make sure it can actually be resolved)

Install Linux, including development tools (GCC, ncurses, openssl, zlib, bison, kernel sources)

Libpri is only needed if you have a PCI ISDN card

Zaptel is only needed if you have a PCI Analog card, ie. to connect the PC to an analog telephone line through an FXO port

Check that your motherboard supports at least PCI 2.2, and that it lets you assign an IRQ to a given PCI slot. These cards generate huge amounts of interrupts during use, and

any conflict with other devices will result in jittery voice and overall poor performance (Why is my card getting an IRQ miss?)

Look in /proc/interrupts to ensure that wcfxo has an IRQ all to itself. If it is sharing an IRQ, move the card to a different PCI slot and see if that resolves the conflict. Read Asterisk

PCI bus Troubleshooting

Once the board is installed, boot up, and run "dmesg" and "lspci -v" to check that Linux detected it

The Zapata channel module, chan_zap.so, is used by to communicate with the Linux kernel, where the drivers for the hardware are loaded. The Zaptel interface is a kernel

loadable module that presents an abstraction layer between the hardware drivers (eg. wctdm, wcfxo) and the Zapata module in Asterisk (ie. chan_zap.so -> Zaptel module

/dev/zap -> wcfxo -> FXO card.) While Asterisk itself compiles on a variety of platforms, the Zaptel drivers are Linux-specific — they are written to interface directly with the Linux

kernel

Newer distros rely on devfs and the udev daemon to create device nodes dynamically. If yours uses those tools, check its documentation on how to add Zaptel (eg. editing

/etc/udev/rules.d/50-udev.rules)

cat /proc/zaptel/*

ztcfg reads /etc/zaptel.conf to configure channel(s)

to load zaptel, you just need to edit zaptel.conf, load wcfxo, and run ztcfg : modprobe wcfxo ; ztcfg -vv

wcfxo and zaptel modules: The order of execution of these commands is important, because voice channels are numbered in the same order their interface cards are enabled. For

instance, if you had two X100P cards installed or a TDM400P with two FXO modules, you would have two logical voice channels. To check that those two modules were loaded

successfully, run "lsmod | grep (zaptel|wcfxo)" (or wctdm instead of wcfxo).

zaptel drivers are installed in /lib/modules/'uname -r'/misc (eg. tor2.o torisa.o wcfxo.o wcfxs.o wcfxsusb.o wct1xxp.o wcusb.o zaptel.o ztd-eth.o ztdummy.o ztdynamic.o).

"One other package you may want to install is asterisk-sounds. While Asterisk comes with many sound prompts in the main source distribution, the asterisk-sounds package will give

you even more. If you would like to expand the number of professionally recorded prompts for use with your Asterisk system, this package is essential."

The asterisk-addons package contains code to allow the storage of Call Detail Records (CDRs) to a MySQL database and to natively play MP3s, as well as an interpreter for loading

Perl code into memory for the life of an Asterisk process. Programs are placed into asterisk-addons when there are licensing issues preventing them from being implemented directly

into the Asterisk source code, or when they are not yet ready for primetime.

/var/log/asterisk/messages

"zap show channels" in the Asterisk console

Reloading after making changes:

zaptel.conf: run "ztcfg -vv"

zapata.conf: "reload" from the Asterisk console (or use asterisk -rx "reload" to execute the command directly)

sip.conf and iax.conf: "reload chan_sip.so" and "reload chan_iax2.so" from the Asterisk console

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extensions.conf: "reload" from the Asterisk console

If you have problems using SIP phones, make sure that your SIP client is using the G.711 codec (either alaw or ulaw) as that is a codec that is known to work with Asterisk

"The s extension will only be selected when a call enters a context without a target extension defined. Imagine a ZAP channel (regular phone line), it rings, asterisk answers but it

has no idea what the target extension will be so it throws it into s. Why doesn't this work with SIP? When a SIP packet comes in, it has the target extension in the packet so

instead of sending it to s it tries to send the call to the target extension. If the target extension can't be matched it sends a not found back to the SIP phone, basically telling you

you dialed a wrong number."

In extensions.conf, add NoOp() in context to know where it fails

Optimization can be done through the Zaptel and Asterisk configuration files (eg. CONFIG_ZAPTEL_MMX is not AMD-friendly, etc.)

Before recompiling Asterisk, remove all files under /usr/lib/asterisk/modules

Trimming it down

Remove unneeded modules: /etc/asterisk/modules.conf

There are different types of modules:

applications

channels

codecs

formats

functions

pbx

resources

In sip.conf, make use of templates:

[common](!)

context=my-phones

type=friend

host=dynamic

qualify=yes

[9000](common)

secret=1234

[9001](common)

secret=1234

Voicemail

1. Add this kind of stuff in voicemail.conf:

[general]

format = wav

[default]

2000 => 4711,Joe Bloggs,joeb@megacorp.biz

2001 => 0815,Darlene Doe

2. Update extensions.conf

[others]

[my-phones]

exten => 2000,1,Dial(SIP/2000,20)

exten => 2000,2,VoiceMail(2000,u)

exten => 2001,1,Dial(SIP/2001,20)

exten => 2001,2,VoiceMail(2001,u)

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)

3. In the console, type "reload"

Adding an FXO card with Zaptel

1. Insert the PCI card, boot up, and type 'lspci -v' to check that Linux did detect it (eg. "Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface"). It

should not share an IRQ with another card

2. Download the Linux source code and headers for the kernel version you are currently using (cat /proc/version)

3. Download and untar the Zaptel source code:

wget http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz

tar xzvf zaptel-1.4-current.tar.gz

4. Compile the Zaptel module:

cd zaptel-1.4.5.1

make clean

./configure

# make menuselect

make

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make install

make config

5. Edit /etc/default/zaptel to match your hardware

6. Create a user "asterisk" to run Zaptel, or you'll get "udevd : lookup_user : specified user 'asterisk' unknown" when rebooting:

useradd asterisk

edit /etc/password and /etc/shadow to disable login for this system account

7. Create Zaptel's configuration file /etc/zaptel.conf:

fxsks=1

loadzone=fr

defaultzone=fr

Alternatively use the script genzaptelconf to generate a zaptel.conf that should work with your system

8. Activate Zaptel: ztcfg -vv

9. Edit /etc/asterisk/zapata.conf:

[channels]

language=fr

context=my-phones ;Must match section in extensions.conf

usecallerid=yes

hidecallerid=no

immediate=no

signalling=fxs_ks

echocancel=yes

echocancelwhenbridged=yes

group=1

channel=>1 ;Must match channel # in zaptel.conf

10. /etc/rc.d/init.d/zaptel start

11. lsmod if you want to check that the modules were indeed loaded

12. Run zttool: If an analog line is plugged into the card and the card was configured with ztcfg, zttool should say OK; Otherwise, it should "Unconfigured"

13. Recompile and reinstall Asterisk

14. Edit extensions.conf so that Asterisk knows what to do when a call comes in from the PSTN on the FXO card (context=my-phones above).

Here's an example that just plays back what you say in the phone (Note: Must add other stuff for a complete extensions.conf)

[my-phones]

; incoming calls from the FXO port are directed to this context from zapata.conf

;Asterisk's parser is so brain-dead that you can't use a comma, even with quotes

;BAD exten => s,1,Verbose("Hello, World!")

exten => s,1,Verbose(Hello World!)

libpri even when not using an ISDN board? "Libpri provides the libraries required for using Primary Rate ISDN (PRI) trunks, as well as a number of other telephony interfaces. Even if we do

not have a PRI line at this time, it is a good idea to install it, as it will not create any conflicts. Parts of the Asterisk code depend on the libraries included in the libpri package. Therefore,

any time we install libpri, we should recompile Asterisk."

If our system is configured to start the Zaptel hardware at boot time, we can accomplish this by running:

$ /etc/init.d/zaptel stop

$ /etc/init.d/zaptel start

If, however, we elected not to start Zaptel interfaces at boot time, we can implement our changes as we go by running:

$ ztcfg -vvv

Remember: Changes to the file will not take effect until we have zaptel.confrestarted the drivers.

Zapata.conf is read by Asterisk. Therefore, to read changes made to this file, we can issue a reload in the Asterisk console. Zaptel will NOT have to be restarted to apply any changes we

make in zapata.conf.

To test the card, run "zttest -c 10"

NEEDED? Load modules wcfxo (zaptel loaded automagically?)

NEEDED? echo "ztdummy" >> /etc/modules : "Zaptel "ticks" once per millisecond (1000 times per second). On each tick every active zaptel channel reads and 8 bytes of data. Asterisk

also uses this for timing, through a zaptel pseudo channel it opens.

However, not all PBX systems are connected to a telephony provider via a T1 or similar connection. With an analog connection you are not synced to the other party. And some systems

don't have Zaptel hardware at all. Even a digital card may be used for other uses or is simply not connected to a provider. Zaptel cards are also capable of providing timing from a clock on

card. Cheap x100P clone cards are sometimes used for that pupose.

If all the above fail, you can use the module ztdummy to provide timing alone without needing any zaptel hardware. It will work with most systems and kernels.

You can check the zaptel timing source with zttest, which is a small utility that is included with zaptel. It runs in cycles. In each such cycle it tries to read 8192 bytes, and sees how long it

takes. If zaptel is not loaded or you don't have the device files, it will fail immedietly. If you lack a timing device it will hang forever in the first cycle. Eitherwise it will just give you in each

cycle the percent of how close it was. Also try running it with the option -v for a verbose output."

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NEEDED? modprobe ztdummy (modprobe = insmod, rmmod)

Do you actually have any zaptel kernel modules loaded ?

lsmod

how to unload/reload zaptel? rmmod?

ubuntu*CLI> zap show channels

No such command 'zap show' (type 'help' for help)

zap show status

ztmonitor

The main method to configure Zaptel devices is using the utility *ztcfg*. ztcfg reads data from the configuration file /etc/zaptel.conf , figures out what configuration to send to channels,

and send it.

is ztdummy automatically loaded when loading either zaptel or wcfxo? Zaptel timers for Asterisk, How to compile ztdummy

if ztcfg -vv = 0 channels configured. -> /etc/zaptel.conf

Asterisk Installation

Practical Asterisk

Asterisk (voip-info.org)

Asterisk Installation Guide

How to install asterisk from source on Debian

What do I do if I can't compile Zaptel package on my system ?

What is fxotune and how do I use it?

Asterisk behind a NAT firewall

Since Asterisk (1.8) still doesn't really support STUN, at the very least, you must open a range of UDP ports on your firewall and forward them to Asterisk to match RTP ports in rtp.conf.

In addition, if you intend to receive calls other than from a VoIP provider, you must also forward UDP5060 which is the standard SIP port. The reason you don't need this to

register/receive calls to/from your VoIP provider, is that in this case, Asterisk is just an SIP client and the use of "qualify" will keep the port open.

Here's how to configure sip.conf when Asterisk and an XLite client when both are located on a private LAN and need to make/receive calls from the Internet:

[general]

;Basic protection

context=invalid

;Default for clients

bindport=5060

bindaddr=0.0.0.0

srvlookup=yes

;Your public IP address

externip=1.2.3.4

externrefresh=10

fromdomain=test.localhost.com

;rewrite source IP address in SDP packets

nat=yes

localnet=192.168.0.0/24

;keep firewall ports open

qualify=yes

;RTP packets must all through Asterisk server

; Asterisk by default tries to redirect

canreinvite=no

dtmfmode=auto

disallow=all

allow=ulaw

allow=alaw

register => mylogin:test@voip.com

[VoIPProvider]

nat=yes

qualify=yes

[200]

;username=200

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type=friend

secret=test

qualify=yes

nat=no

host=dynamic

context=myinternal

How to Configure SIP and NAT By Sean Walberg

Testing Asterisk and NAT

IP used in REGISTER

When the Asterisk server and the SIP clients are all located on the same LAN (with non-routable IP's), it appears that SIP clients are smart enough to send their LAN IP instead of the

WAN IP even when set to use STUN when REGISTERing to the SIP server (Asterisk).

Opening SIP and RTP ports on NAT

Can Asterisk do it? If yes, 1.4, 1.6, 1.8?

Direct media mode and NAT

Can RTP packets flow directly between SIP clients when there are on either side of the NAT (ie. one in the LAN, one on the Net)? If yes, are some features lost?

How many ports does RTP need? 1 for RTP and 1 for RTSP?

How to use a port other than UDP 5060 on the router?

To avoid breaking attempts while still allowing remote SIP clients to REGISTER and remote SIP clients/servers to INVITE: Create SRV record in DNS?

How to scan UDP ports from the Net?

How to monitor hacking attemps?

/tmp/asterisk/log/event_log is empty

/var/log/messages?

Hung up from remote Ekiga: XLite doesn't detect call ended

OK when XLite hangs up.

OK when ZoIPer hangs up.

-> Ekiga issue.

Using an Atcom AG-188N PSTN Gateway

http://www.atcom.cn/AG188N.html

Adding a Linksys 3102 VoIP gateway

See this.

Using a GSM cellphone with Asterisk

"OpenBTS is an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface to standard GSM handset and uses the Asterisk

software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and

operated at substantially lower cost than existing technologies in greenfields in the developing world.

In plain language, we are working on a new kind of cellular network that can be installed and operated at about 1/10 the cost of current technologies, but that will still be compatible

with most of the handsets that are already in the market. This technology can also be used in private network applications (wireless PBX, rapid deployment, etc.) at much lower cost and

complexity than conventional GSM."

Zaptel and call progress

Ideally, you should connect Asterisk to the POTS through either an ISDN line or SIP + VoIP provider, since, unlike analog on lines, they separate data and signaling, which makes eg.

disconnect supervision very easy.

If you must use an analog line, professional-grade, PSTN gateways are usually more reliable than entry-level Digium knock-offs and the Zaptel/Dahdi driver + Zapata.

If you're lucky, your telco reverses line polarity or drops battery ("kewlstart", an extension of "loopstart") to indicate that the remote end has answered/hung up. To indicate a hangup,

the "loopstart" solution can reassert the dial-tone or give a busy/fast busy signal.

Otherwise, if you are relying on Zaptel+PCI or the Linksys 3102 gateway, Asterisk may not report that the callee is either BUSY or has picked up the phone.

And even if Asterisk/Zaptel does detect an answer, automated calls can't tell if the remote party is a human being or an answering machine.

So the only reliable solution if you need to automate calls (ie. "robocall") is to loop through a voice message asking the callee to hit any key on their dialpad to confirm that they did

answer the call, and use a time-out if no DTMF has been typed within a certain time-frame.

As for detecting that the callee has hung up, try "busydetect=yes" in zapata.conf or chan_dahdi.conf.

Telco provides answer/hangup signal

If you are lucky, your telco uses the easier method of signalling answer/hangup, through polarity reversal or open loop disconnect a.k.a. Calling Party Control. In this case, edit zapata.conf

thusly:

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;tells chan_zap to monitor that line continuously for eg. pre-ring CID

;polarityevents=no

;how long (ms) to ignore Polarity Switch events after we answer a call

;polarityonanswerdelay=1

answeronpolarityswitch = yes

hanguponpolarityswitch = yes

If you are semi-lucky, you live in the US and Zaptel can be told to analyze tones:

callprogress=yes

progzone=us

If none of the above applies, while Zaptel can detect a BUSY signal (which should be sent when the callee is already online or has hung up your call), it can't detect an offhook. Strangely

enough, asterisk/indications.conf is only used to play tones, not analyze them.

ChanIsAvail

When called with the name of the channel, ChanIsAvail() returns the status in the AVAILORIGCHAN variable (AVAILSTATUS isn't reliable). Here, we dial out from the CLI, and check the

variable while the line is engaged.

First, extensions.conf where the call will be handled:

;Loop until Zap/1 is available or INDEX < 10

exten => 1111,1,Set(INDEX=0)

exten => 1111,n,While(1)

exten => 1111,n,ChanIsAvail(Zap/1)

exten => 1111,n,GotoIf($["${AVAILORIGCHAN}" != "" | ${INDEX} > 10]?exit)

exten => 1111,n,Wait(5)

exten => 1111,n,Set(INDEX=$[${INDEX} + 1])

exten => 1111,n,EndWhile()

;how did we exit loop?

exten => 1111,n(exit),GotoIf($["${AVAILORIGCHAN}" = ""]?na:ok)

exten => 1111,n(na),NoOp(Channel still N.A.)

exten => 1111,n,Goto(end)

exten => 1111,n(ok),NoOp(Channel OK)

;give */Zaptel time to recover

exten => 1111,n,Wait(5)

exten => 1111,n,Hangup()

exten => 1111,n(end),Hangup

Other ways to check the content of AVAILORIGCHAN:

exten => 7777,n,GotoIf($[!${ISNULL(${AVAILORIGCHAN})}]?available:not_available)

exten => 7777,n,GotoIf($[${EXISTS(${AVAILORIGCHAN})}]?available:not_available)

Next, place a call from the CLI:

originate Zap/1/5551234 extension 7777@internal

An alternative way to check if a Zaptel port is available and wait until the channel is available:

exten => 7777,1,Set(INDEX=0)

;j = Add 101 to priority to jump when N.A.

exten => 7777,n(check),ChanIsAvail(Zap/1,j)

exten => 7777,n,NoOp(Channel is available)

exten => 7777,n,Hangup

exten => 7777,103,NoOp(Channel N.A.)

exten => 7777,n,Wait(5)

exten => 7777,n,Set(INDEX=$[${INDEX} + 1])

exten => 7777,n,GotoIf($[${INDEX} < 10]?check)

exten => 7777,n,NoOp(Giving up)

exten => 7777,n,Hangup

Note that ChanIsAvail simply says if the port is available: In case you're trying to dial a remote end through a landline, the remote line might (still) be busy.

Also note that DEVICE_STATE (1.6, backported to 1.4) doesn't work with FXO modules:

;Dahdi doesn't support DEVICE_STATE

;CLI> core show channeltypes

exten => 2222,1,NoOp(${DEVICE_STATE(Dahdi/1)})

exten => 2222,n,Hangup()

Play message and wait for confirmation

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As a work-around, instead of just using Wait() and start playing a message blindly, you can play a message asking the callee to hit a DTFM to confirm that they're ready to proceed:

[callback]

exten => s,1,Wait(2)

exten => s,n,Answer

;expects 1 key,4 tries, wait 5 seconds between tries

exten => s,n(read),Read(key,please-type,1,,4,5)

exten => s,n,GotoIf($[${LEN(${key})} == 0]?end)

;callee is alive

exten => s,n,Playback(auth-thankyou)

exten => s,n(end),Wait(1)

exten => s,n,Hangup()

CHANNEL()

Theoretically, using CHANNEL(), it should be possible to use a While loop to force Asterisk to pause until the callee has answered, but it doesn't work:

;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing Offhook, Pre-ring, Unknown

exten => start,n,Set(INDEX=0)

exten => start,n,While($["${CHANNEL(state)}" != "OffHook" & ${INDEX} < 10])

exten => start,n,NoOp(Channel still ringing: ${CHANNEL(state)})

exten => start,n,Wait(1)

exten => start,n,Set(INDEX=$[${INDEX} + 1])

exten => start,n,EndWhile()

Detecting a remote BUSY

When calling out, this is how to detect that the remote line is already engaged:

Make a call automatically through a callfile

Use Dial() manually and read the DIALSTATUS variable to check how the call went. The problem with Dial() is that it 's a blocking application, so it won't work if you want to

perform tasks after the callee has answered. (try g/G)

Callfile

A callfile simply dials out and jumps to the context:

[callback]

exten => start,1,Answer()

exten => start,n,Playback(beep)

exten => start,n,Hangup

;0 - Failed (not busy or congested)

;1 - Hung up

;3 - Ring timeout

;5 - Busy

;8 - Congestion

exten => failed,1,NoOp(Reason call file failed is ${REASON})

An alternative:

[callback]

exten => start,1,GotoIf($[ "${DIALSTATUS}"="BUSY"]?end)

exten => start,n,Wait(2)

exten => start,n,Answer()

exten => start,n,Playback(some-important-message)

exten => start,n(end),HangUp

Dial()

exten => 123,1,Dial(Zap/1/5551234,10)

;just hangs up and doesn't proceed?

exten => 123,n,NoOp(Called ended with ${DIALSTATUS})

exten => 123,n,Hangup()

exten => h,1,NoOp(Called ended with ${DIALSTATUS})

Another example about the DIALSTATUS variable:

exten => s,n,Dial(Zap/1/5551234)

exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-ANSWER,1,Hangup

exten => s-CANCEL,1,Hangup

exten => s-NOANSWER,1,Hangup

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exten => s-BUSY,1,Busy ;Only works with SIP calls

exten => s-CHANUNAVAIL,1,Verbose(Not available)

exten => s-CONGESTION,1,Congestion

exten => _s-.,1,Congestion

exten => s-,1,Congestion

RetryDialing

Doesn't work if remote end is busy: Simply hangs up and ends there:

[redial]

;No difference

exten => start,1,Progress()

exten => start,n,RetryDial(pls-hold-while-try,10,3,Zap/1/5551234,30)

exten => start,n,NoOp(Result is ${DIALSTATUS})

exten => h,1,NoOp(In h, result is ${DIALSTATUS}

;no difference

exten => failed,1,NoOp(Reason call file failed is ${REASON})

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Hangup

exten => h,1,Goto(redial,start,1)

Testing

As a test, here's what to put in extensions.conf...

[callback]

exten => start,1,Answer()

exten => start,n,Wait(1)

exten => start,n,Playback(tt-monkeysintro)

exten => start,n,Wait(1)

exten => start,n,Hangup

... and here's how to make an outgoing call and check that Asterisk doesn't jump to the [callback] context until the call has been answered:

CLI> originate SIP/voip-out/5551234 extension start@callback

CLI> originate Zap/1/5551234 extension start@callback

Note that "originate" isn't very reliable, and sometimes drops a call for no reason.

Files common to Zaptel and Dahdi

/etc/asterisk/indications.conf (Telephone line signalling; call progress tones played to the callers with Playtones() once they are within Asterisk)

CHECK /etc/modprobe.conf

Files specific to Zaptel

/etc/zaptel.conf

/etc/asterisk/zapata.conf

Files specific to Dahdi

/etc/dahdi/modules

/etc/dahdi/system.conf

/etc/asterisk/chan_dahdi.conf

modprobe.conf

options wctdm opermode=FRANCE

;options wctdm opermode=FRANCE debug=1

To actually see data in /var/log/messages, you must also edit /etc/asterisk/logger.conf.

asterisk/indications.conf

[general]

country=fr

[fr]

description = France

...

zaptel.conf

loadzone = fr

defaultzone=fr

fxsks=1

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asterisk/zapata.conf

First, include your locale, eg.

language=fr

Next, tell Asterisk which signaling to use to match the one used in zaptel.conf:

fxs_ks=1

The better way to detect that the remote end has been disconnected is when your telco performs a temporary polarity switch, ie. kewlstart/fxsks:

answeronpolarityswitch=yes

hanguponpolarityswitch=yes

;tells chan_zap to monitor that line continuously for eg. pre-ring CID

;polarityevents=yes

;how long (ms) to ignore Polarity Switch events after we answer a call

;polarityonanswerdelay=1

If this doesn't work, it could mean that your telco indicates a hangup with a loop disonnect, a.k.a. Calling Party Control. I didn't find any settings in any file to enable this.

If this is not available either from your telco, it means that your telco plays tones to indicate call progress (eg. Disconnection Tone to signal a hangup). Apparently, the only setting

available in zapata.conf is to detect a BUSY signal:

busydetect=yes

busycount=4

I guess Asterisk knows how to handle other tone through the parameters set in other files above.

"progzone" parameter: "This defines the timing and frequencies for call progress detection, which are buried in the sources in asterisk/dsp.c. This is DIFFERENT than the call progress

timing defined in zaptel/zonedata.c and in /etc/asterisk/indications.conf, and so far only options you can use (defined in dsp.c) are us, ca, br, cr and uk."

"callprogress" parameter: "Highly experimental and can easily detect false answers, so don't count on it being very accurate. Also, it is currently configured only for standard U.S. phone

tones". Enabling this with non-US telcos may prevent Zaptel from working.

"flash" parameter: "used only with (non-PRI) T1 lines".

After editing zapata.conf, remember to stop Asterisk/restart Zaptel/start Asterisk, or type the following in the CLI: module reload chan_zap.so

dahdi/modules

dahdi/system.conf

asterisk/chan_dahdi.conf

Q&A

Sometimes, "originate" doesn't actually ring a remote number

ip04*CLI> originate Zap/1/5551234 extension 8888@internal

Try with x86 Linux and check

Why dont call files trigger the "failed" extension?

If calling out through Zaptel+PCI or the Linksys 3102, try with a VoIP provider.

Tips & tricks

Tones

Here is a list of the different call-progress tones, including Ringback, Ringtone, and Busy. International line settings can be found at www.3amsystems.com.

Closing channel

If you need to hang up a Zaptel channel in the Asterisk console: "soft hangup Zap/1-1" or "soft hangup Dahdi/1-1"

Reloading Zaptel/Dahdi

If you changed a channel's configuration in zapata.conf/chan_dahdi.conf, type "module reload chan_zap.so" or "module reload chan_dahdi.so", respectively.

Wait() in h

If you need to wait in the "h" extension, Wait() won't work. A work-around is to use system(/bin/sleep 10) to use the system's command instead.

Repeating a phone number

Here's how to have Asterisk repeat a phone number the French way:

1. Put localized sound files in /var/lib/asterisk/sounds/fr/

2. Edit /etc/asterisk/asterisk.conf thusly:

[options]

;Layout requires Asterisk 1.4+

languageprefix = yes

3. Edit /etc/asterisk/say.conf:

[general]

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; method for playing numbers and dates

; old - using asterisk core function

; new - using this configuration file

mode=new

...

[fr](date-base,digit-base)

;BAD _[n]um:0. => num:${SAY:1}

_[n]um:X => digits/${SAY}

_[n]um:1X => digits/${SAY}

_[n]um:[2-9]0 => digits/${SAY}

_[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}

_[n]um:71 => digits/60, vm-and, num:1${SAY:1}

_[n]um:7X => digits/60, num:1${SAY:1}

_[n]um:9X => digits/80, num:1${SAY:1}

_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}

_[n]um:100 => digits/hundred

_[n]um:1XX => digits/hundred, num:${SAY:1}

_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred

_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}

_[n]um:1000 => digits/thousand

_[n]um:1XXX => digits/thousand, num:${SAY:1}

_[n]um:[2-9]000 => num:${SAY:0:1}, digits/thousand

_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}

_[n]um:XX000 => num:${SAY:0:2}, digits/thousand

_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}

_[n]um:XXX000 => num:${SAY:0:3}, digits/thousand

_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}

_[n]um:1000000 => num:${SAY:0:1}, digits/million

_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}

_[n]um:[2-9]000000 => num:${SAY:0:1}, digits/million

_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}

_[n]um:XX000000 => num:${SAY:0:2}, digits/million

_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}

_[n]um:XXX000000 => num:${SAY:0:3}, digits/million

_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}

_datetime::. => date:AdBY 'digits/at' H 'hours' M 'perc':${SAY}

_date::. => date:AdBY:${SAY}

_time::. => date:H 'hours' M 'perc':${SAY}

;800 numbers

_pho[n]e:08XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:3},num:${SAY:4:2}, num:${SAY:6:2},num:${SAY:8:2}

_pho[n]e:XXXX => num:${SAY:0:2}, num:${SAY:2:2}

_pho[n]e:0[1-9]XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}

_pho[n]e:. => digit:${SAY}

4. Edit /etc/asterisk/extensions.conf:

exten => 1000,1,Playback(phone:0123456789,say)

Recording a message

Asterisk only supports WAV files encoded in 16-bit, 8000Hz, mono. Here's how to call Asterisk from XLite, and record a message in low- and high-quality formats (source):

[context_for_my_handset]

exten => 101,1,Playback(vm-intro)

exten => 101,n,Record(maingreeting.wav)

exten => 101,n,Wait(2)

exten => 101,n,Playback(maingreeting)

exten => 101,n,Hangup

Alternatively, you can record a message in a more compact format such as uLaw, which is a better option if this is the codec that is likely to be used for incoming calls, as this will spare

Asterisk from having to convert messages to another format:

[context_for_my_handset]

exten => 101,1,Playback(vm-intro)

exten => 101,n,Record(maingreeting.ulaw)

exten => 101,n,Wait(2)

exten => 101,n,Playback(maingreeting)

exten => 101,n,Hangup

CLI> core show translation

Improving sound quality

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To keep jitter/latency as low as possible, make sure the host running Asterisk has enough hardware (CPU, RAM, network), only runs the minimal software required, and has as few

peripherals connected as possible.

Make sure all the routers within your control are configured to use QoS so that VoIP traffic is favored over non-isochronous traffic.

Wireshark is a useful tool, as it can trace VoIP traffic and compute delay.

VAD (Voice Activity Detection)

VLAN, QoS

Codecs

Make sure all devices going through Asterisk use the same codec, so that Asterisk doesn't have to perform transcoding.

How to get sound infos on a live conversation?

show translation

sip show channels

sip show channel <Call ID>

I'd recommend you get a hardware timer card for Asterisk. This will greatly improve your audio quality since your test hardware is not powerful enough for the software based timer to do

it's job properly

Sangoma VoiceTime USB stick

Level too low

Choppy sound

Provided the issue occurs even with a single call, ie. bandwidth is plenty, check the latency and round-trip time (RTT) between the two end-points.

Echo issues

Echo, heard either at your end or the remote end, can have two causes:

Impedance mismatch between the device (the IP-to-PSTN gateway), the local loop to the telco, and the remote hybrid that turns this two-wire cable into a four-wire cable

before the voice signal is digitized by the telco.

This mismatch causes some of your voice to be sent back to you: The delay is too short for echo to be noticeable when using analog equipments (so you'll just think of this as

side-tone), but noticeable because of the delay that VoIP adds to the process (analog-to-digital conversion in the IP phone, DAC before setting your voice to the telco through

the POTS line, and back.)

Solution: If the device that you use to connect to the POTS has some impedance settings, play with this setting. Also check its in/out gain settings. If it still fails, try its own echo

canceller, if any, or a software echo canceller like the OSLEC if your IP phone or IP PBX supports it

Cheap or badly-configured telephone where the voice coming from the earpiece is picked up by the microphone, causing a loop-back.

Things to try: Lower the incoming volume on your IP phone so the microphone doesn't pick up the signal coming from the earpiece. Try a different IP phone. Use a headset

More information:

Echo and Soft VoIP PBX Systems By David Mandelstam

Echo Cancellation Demystified By Alexey Frunze

Echo Elimination & DTMF Problem Tech Bulletin

Asterisk echo cancellation

The Echo "Phenomenon", Causes and Solutions

http://en.wikipedia.org/wiki/Echo_cancellation

Echo Analysis for Voice over IP

Possible Fix for line echo Prob-SPA3102

VoIP Echo and how to correct it

Steps to reduce echo in Asterisk

High performance echo canceller for Asterisk VoIP systems by Scott Kurtz

Writing dialplans

The meat of Asterisk resides in extensions.conf, ie. the dialplan.

Application vs. function?

"An application is something that performs an action on a channel (such as playing a sound prompt, gathering DTMF input, putting the call into a call queue, etc.). A function, on the

other hand, is used to get or set values, and doesn't directly manipulate the channel. These values *might* have something to do with the channel (such as is the case with the CDR

function), but don't necessarily have to (such as is the case with the CUT and LEN functions).

You could also think of it as the difference between a procedure and a function. A procedure does something and returns nothing. A function may or may not be doing something, but

its primary function is to return a value. Unlike other languages, in Asterisk, the return value of a function may not be directly ignored (i.e. you HAVE to get it, even if you do nothing

with it). Of course, setting a dialplan function completely ruins this nice dichotomy. ;-)"

"An application is a "command" executed by a dialplan priority, such as Record, Verbose, TrySystem, etc. A function needs to be evaluated inside ${ } and returns a string value that is

substitued in place of the ${ }."

"In addition to dialplan applications, which have been part of Asterisk almost from the very beginning, Asterisk also supports functions as of Asterisk 1.2. This is part of a long-standing

effort to make Asterisk behave more like a programming environment.

In contrast to applications, functions may not be called directly. Instead, they are called inside applications and return a value, or -- in a departure from the classical definition of a function -

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- they may even be written to using the application Set() (see the section called “Set()”). Function names are always written in uppercase letters. Surprisingly, functions are written in the

same way as variables, inside curly braces and preceded by a $ character ( ${} ). This is necessary because strings are not always bounded by quotation marks."

Functions

BLACKLIST BLACKLIST() Check if the callerid is on the blacklist

CUT CUT(<varname>,<char-delim>,<range-s Slices and dices strings, based upon a named delimiter.

DB DB(<family>/<key>) Read from or write to the Asterisk database

DB_DELETE DB_DELETE(<family>/<key>) Return a value from the database and delete it

DB_EXISTS DB_EXISTS(<family>/<key>) Check to see if a key exists in the Asterisk database

ENV ENV(<envname>) Gets or sets the environment variable specified

EVAL EVAL(<variable>) Evaluate stored variables.

EXISTS EXISTS(<data>) Existence Test: Returns 1 if exists, 0 otherwise

FIELDQTY FIELDQTY(<varname>|<delim>) Count the fields, with an arbitrary delimiter

FILTER FILTER(<allowed-chars>|<string>) Filter the string to include only the allowed characters

GLOBAL GLOBAL(<varname>) Gets or sets the global variable specified

IF IF(<expr>?[<true>][:<false>]) Conditional: Returns the data following '?' if true else the data following ':'

ISNULL ISNULL(<data>) NULL Test: Returns 1 if NULL or 0 otherwise

LANGUAGE LANGUAGE() Gets or sets the channel's language.

LEN LEN(<string>) Returns the length of the argument given

MATH MATH(<number1><op><number 2>[,<type Performs Mathematical Functions

MD5 MD5(<data>) Computes an MD5 digest

QUOTE QUOTE(<string>) Quotes a given string, escaping embedded quotes as necessary

RAND RAND([min][|max]) Choose a random number in a range

REGEX REGEX("<regular expression>" <data> Regular Expression

SET SET(<varname>=[<value>]) SET assigns a value to a channel variable

SHA1 SHA1(<data>) Computes a SHA1 digest

SORT SORT(key1:val1[...][,keyN:valN]) Sorts a list of key/vals into a list of keys, based upon the vals

SPRINTF SPRINTF(<format>|<arg1>[|...<argN>] Format a variable according to a format string

STAT STAT(<flag>,<filename>) Does a check on the specified file

STRFTIME STRFTIME([<epoch>][|[timezone][|for Returns the current date/time in a specified format.

STRPTIME STRPTIME(<datetime>|<timezone>|<for Returns the epoch of the arbitrary date/time string structured as described in the format.

TIMEOUT TIMEOUT(timeouttype) Gets or sets timeouts on the channel.

Applications

AGI: Executes an AGI compliant application

Answer: Answer a channel if ringing

BackGround: Play an audio file while waiting for digits of an extension to go to.

BackgroundDetect: Background a file with talk detect

Busy: Indicate the Busy condition

Congestion: Indicate the Congestion condition

ContinueWhile: Restart a While loop

DBdel: Delete a key from the database

DBdeltree: Delete a family or keytree from the database

Dial: Place a call and connect to the current channel

Echo: Echo audio, video, or DTMF back to the calling party

EndWhile: End a while loop

Exec: Executes dialplan application

ExecIf: Executes dialplan application, conditionally

ExitWhile: End a While loop

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ExtenSpy: Listen to a channel, and optionally whisper into it

FollowMe: Find-Me/Follow-Me application

Gosub: Jump to label, saving return address

GosubIf: Conditionally jump to label, saving return address

Goto: Jump to a particular priority, extension, or context

GotoIf: Conditional goto

Hangup: Hang up the calling channel

HasNewVoicemail: Conditionally branches to priority + 101 with the right options set

HasVoicemail: Conditionally branches to priority + 101 with the right options set

LookupBlacklist: Look up Caller*ID name/number from blacklist database

LookupCIDName: Look up CallerID Name from local database

Macro: Macro Implementation

MacroExclusive: Exclusive Macro Implementation

MacroExit: Exit From Macro

MacroIf: Conditional Macro Implementation

MailboxExists: Check to see if Voicemail mailbox exists

NoOp: Do Nothing

Playback: Play a file

PrivacyManager: Require phone number to be entered, if no CallerID sent

Random: Conditionally branches, based upon a probability

Read: Read a variable

ReadFile: ReadFile(varname=file,length)

Record: Record to a file

Return: Return from gosub routine

Set: Set channel variable(s) or function value(s)

SetCallerID: Set CallerID

SetCallerPres: Set CallerID Presentation

SetGlobalVar: Set a global variable to a given value

System: Execute a system command

TryExec: Executes dialplan application, always returning

TrySystem: Try executing a system command

Verbose: Send arbitrary text to verbose output

VoiceMail: Leave a Voicemail message

VoiceMailMain: Check Voicemail messages

Wait: Waits for some time

WaitExten: Waits for an extension to be entered

WaitForSilence: Waits for a specified amount of silence

While: Start a while loop

Handling busy/N.A.

(Source)

u causes the "unavailable" message to be played. The pathname for this message is /var/lib/asterisk/sounds/vm-isunavail.gsm[16]

b causes the "busy" to be played. The pathname for this message is /var/lib/asterisk/sounds/vm-rec-busy.gsm.

s suppresses playback of the "unavailable" or "busy" notifications, plays a beep, and begins recording.

If the caller presses "0" while listening to the prompt, the application will jump to extension "o" (the small letter o) in the specified context.

If the caller presses "*" while listening to the prompt, the application will jump to extension "a" (the small letter a) in the specified context.

Looping

Note: For a long time, priority jumping was a standard way of moving a call about the dialplan. Specific applications (e.g. Dial()) would elevate the priority by 101 under certain

circumstances. This feature is now officially deprecated. Basically, this means that while it is currently still supported, eventually it will be removed. Anyone who continues to use it is

making their dialplans vulnerable to failure after an upgrade.

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Here's an example of While/EndWhile:

exten => 1013,n,Set(i=1)

exten => 1013,n,While($[${i} < 10])

exten => 1013,n,SayNumber(${i})

exten => 1013,n,Wait(1)

exten => 1013,n,Set(i=$[${i} + 1])

exten => 1013,n,EndWhile()

Also look at ContinueWhile()

exten => 1015,1,Gosub(cid-set)

exten => 1015,n,Dial(SIP/${EXTEN})

exten => 1015,n(cid-set),Set(CALLERID(all)=Apfelmus GmbH <012345678>)

exten => 1015,n,Return()

Here's how to check if a CID number is in the database, and branch to the right location depending on the result:

exten => 888,1,Set(CALLERIDNUM=111)

exten => 888,n,Verbose(${IF($["${DB(friends/${CALLERIDNUM})}"!=""]?good:bad)})

exten => 888,n,GotoIf($["${DB(friends/${CALLERIDNUM})}"!=""]?fred)

exten => 888,n,Verbose(Bad)

exten => 888,n,Hangup()

exten => 888,n(fred),Verbose(Good)

Here's how to check if a file exists, and branch to the right location:

exten => 888,1,Set(WAV_FILE=/root/asterisk/zapata.conf)

exten => 888,n,Set(WAV_FILE=${IF($["${STAT(e,${WAV_FILE})}" = "1"]?${WAV_FILE}:"")})

exten => 888,n,GotoIf($["${STAT(e,${WAV_FILE})}" = "1"]?next)

exten => 888,n,Verbose(Not found)

exten => 888,n,Hangup

exten => 888,n(next),Verbose(Found)

Calling external scripts with AGI

Important: AGI scripts dump error messages to the Asterisk ’console’ session. If you started your session using normal init scripts, just attaching to it normally won’t show you these

messages. You need to launch asterisk with the command ’asterisk -vvvvvvcr’. For debugging purposes you can type "agi debug" on the CLI.

(from some online source) New in Asterisk v1.2.11: GET VARIABLE can now retrieve global variables (see bug 7609)

New in Asterisk v1.2: CallerID is reported with agi_callerid and agi_calleridname instead of a single parameter holding both. The agi_callerid previously held the value "Name"<Number> and

the agi_calleridname was not present. In v1.2, agi_callerid has Number and the agi_calleridname has Name.

Asterisk communicates with the AGI program over stdin and stdout. The arguments are passed directly to the AGI program at execution time.

The AGI program must be flagged as executable in the filesystem. The path is relative to the Asterisk AGI directory, which is at /var/lib/asterisk/agi-bin/ by default.

Returns -1 on hang-up or if the program requests a hang-up; returns 0 if not.

This application sets the following channel variable upon completion:

AGISTATUS

The status of the attempt to the run the AGI script text string, one of SUCCESS | FAILED | HANGUP

To run AGI programs on inactive channels (as in the case of an h-extension, where the channel is on-hook), used DeadAGI() instead. Should your AGI program need access to the

incoming audio stream, use EAGI() instead of AGI(). The incoming audio stream is provided on file descriptor 3[47]

; run AGI on a hung-up channel:

exten => h,1,DeadAGI(agi-test)

asterisk*CLI> agi show

get variable Gets a channel variable

set callerid Sets callerid for the current channel

a reminder: 0: stdin, 1: stdout, 2:stderr. File descriptor 3 is freely assignable.

As an alternative you may execute PHP scripts using System(). See also. the section called “System()”, the section called “AGI()”

exten => 777,1,Set(CALLERIDNUM=1234567890)

exten => 777,n,ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI,/root/dummy.php,${CALLERIDNUM},param2)

; Retrieve http://example.com/page.php?id=1&action=view :

exten => 123,1,Set(foo=${CURL(http://example.com/page.php?id=1&action=view)})

(As of Asterisk 1.2.8, use a pipe ("|") character instead of commas as a parameter delimiter.)

; set HOME:

exten => 123,1,Set(ENV(HOME)=/myAst)

Here's an example of a Python script that can be called from Asterisk through AGI:

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#! /usr/bin/env python

> import posix

> posix.close(1)

> posix.open("/dev/null", posix.O_WRONLY)

import os

import sys

import time

print os.getpid()

null = os.open(os.devnull,os.O_RDWR)

os.dup2(null, sys.stdin.fileno())

os.dup2(null, sys.stdout.fileno())

os.dup2(null, sys.stderr.fileno())

os.close(null)

print "You won't see this"

print >>sys.stderr, "Or this"

time.sleep(60)

>> import os,sys,time

>> print "pre:", os.getpid()

>> sys.stdout = open(os.devnull, 'w')

>> print "post:", os.getpid()

>> time.sleep(60)

sys.stdout = open(os.devnull, 'w')

if os.fork():

sys.exit(0)

Here's an example of a command-line PHP script (the -q switch tells PHP not to return HTML headers) using an SQLite database and being called from Asterisk through the AGI application:

#!/usr/bin/php -q

<?php

//myscript.php 123

try {

$dbh = new PDO("sqlite:mydata.db");

$dbh->exec("CREATE table customers (id INTEGER NOT NULL PRIMARY KEY, tel CHAR(10))");

$dbh->exec("INSERT INTO customers VALUES (NULL, '" . $argv[1] . "')");

$sql = 'SELECT tel FROM customers WHERE id=1';

$result = $dbh->query($sql);

$return = $result->fetch();

echo "Result = " . $return['tel'];

$dbh = null;

}

catch(PDOException $e)

{

die($e->getMessage());

#How to tell AGI that it bombed?

}

?>

Calling Lua scripts through AGI

An AGI script must always start with reading all the input sent by Asterisk through stdin:

--Must first empty stdin

while true do

local line = io.read()

if line == "" then break end

-- Without line below, script never ends

io.write("NOOP ",line,"\n")

end

Using pbx_lua

In addition to calling the Lua interpreter through the AGI interface to make it easier to write a diaplan application, Matt Nicholson from Digium added the pbx_lua module

(/usr/lib/asterisk/modules/*so) in 2008 so you can use /etc/asterisk/extensions.lua instead of extensions.conf.

extensions.lua is more readable than extension.conf, but doesn't seem to add more syntax, so calling Lua scripts through AGI seems like a more interesting alternative.

Handling calls with Adhearsion

An alternative is to use Adhearsion, wich is a framework written in Ruby that uses the AGI and AMI to manage calls.

Tips & Tricks

Do not put spaces before and/or after a "=" sign:

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exten => 999,1,Set(CALLERIDNUM=${CALLERID(num)})

;BAD exten => 999,n,Set(CALLERIDNAME = ${DB(cidname/${CALLERIDNUM})})

exten => 999,n,Set(CALLERIDNAME=${DB(cidname/${CALLERIDNUM})})

exten => 999,n,Verbose(CID is ${CALLERIDNAME})

GosubIf/Return: Pay attention to the fact that Return doesn't send us back to the GosubIf() line but rather the line that follows it:

exten => 777,1,GosubIf($["${CALLERID(num)}" = "101"]?setcallerid,1:ok,1)

exten => 777,n,Verbose(This is where Return sends us)

exten => 777,n,Hangup

exten => setcallerid,1,Set(CALLERID(num)=123)

exten => setcallerid,n,Verbose(Before Return)

exten => setcallerid,n,Return

exten => ok,1,Verbose(We never see this...)

Different ways to generate a new filename dynamically

exten => _[1_4],n,Set(RANDNAME=${RAND()});

exten => _[1-4],n,Record(/tmp/msg${RANDNAME}.wav,3,30);

;${RECORDED_FILE} contains full path to file, eg. /tmp/msg1

exten => _[1-4],n,Record(/tmp/msg%d.wav,3,30)

exten => _[1-4],n,Verbose(Fichier = ${RECORDED_FILE}.wav)

exten => _[1-4],n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})

exten => _[1-4],n,Record(/tmp/${CALLTIME}.wav,3,30)

WAV files must be converted to sample rate 8KHz in mono: In Linux, you can use sox to do this:

sox input.wav -r 8000 -c 1 -s -w output.wav resample -ql

Asterisk 1.4 added a CLI command to convert sound files from one format to another:

file convert <file_in.format> <file_out.format>

Shell script equivalent:

#!/bin/bash

# Converts a audio file from alaw to a ulaw

rasterisk -x "file convert /tmp/file_in.alaw /tmp/file_out.ulaw"

Callback

In case you can make free calls from your broadband at home through an RJ11 plug in the modem, you can set up Asterisk to wait for calls from your cellphone, and have Asterisk call you

back through the FXO/Zaptel module before prompting you for the number you want to call before bridging the two channels.

For some reason, Asterisk doesn't support reusing the same channel from which the original call came, either within extensions.conf per se, or even by calling an AGI script, even after

calling Hangup() to make sure the channel is dead, so either use a second channel or use SendDTMF() to dial the second number.

Direct callback

This solution simply waits for the channel to die, and then tries to dial out:

[globals]

CID=""

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

;wait for cellphone to go back on-hook

exten => s,n,Wait(10)

exten => s,n,Hangup

;dial_exec_full: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time

exten => h,1,Dial(Zap/1/${CID})

Local channel

This code will wait for the channel to die, and then jump to another context through a local channel:

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

;wait for cellphone to go back on-hook

exten => s,n,Wait(10)

exten => s,n,Hangup

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;dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time

exten => h,1,Dial(Local/start@callback)

[callback]

exten => start,1,Dial(Zap/1/${CID})

AGI

This code will wait for the channel to die, and then call an AGI script which will create a callfile and move it to Asterisk's outgoing/ directory

extensions.conf

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

;wait for cellphone to go back on-hook

exten => s,n,Wait(10)

exten => s,n,Hangup

;__ast_request_and_dial: Unable to request channel Zap/1/123456

;attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)

exten => h,1,DeadAGI(callback.lua)

callback.lua

-- read input

while true do

local line = io.read()

if line == "" then break end

io.write("NOOP ",line,"\n")

end

-- read number to call back

cid = arg[1]

io.write(string.format("NOOP CID is %s\n",cid))

response=io.read()

-- create and move callfile

file = io.open ("/var/tmp/callback.call","w")

channel = string.format("Channel: Zap/1/%s\n",cid)

file:write(channel)

file:write("Context: callback\n")

file:write("Extension: start\n")

file:close()

os.execute("mv /var/tmp/callback.call /var/tmp/asterisk/outgoing")

AGI + cron

Here, we'll use an AGI script to stop/edit/start cron to load a second script the next minute is up. The second script will take care of stopping/editing/starting cron and creating a callfile.

extensions.conf

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

exten => s,n,Wait(2)

exten => s,n,Hangup

exten => h,1,DeadAGI(/var/tmp/cron.lua,${CID})

cron.lua

#!/var/tmp/lua

CRONTAB="/etc/config/crontab"

CRONTAB_COPY="/etc/config/crontab.orig"

--Must first empty stdin

while true do

local line = io.read()

if line == "" then break end

-- Without line below, script never ends

io.write("NOOP ",line,"\n")

end

-- read CID number from args

cid = arg[1]

io.write("NOOP After CID ",cid,"\n")

line = io.read()

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-- stop cron

os.execute("/etc/init.d/cron stop")

-- cp original crontab, and add script to copy

command = string.format("cp %s %s",CRONTAB,CRONTAB_COPY)

io.write("NOOP ",command,"\n")

line = io.read()

os.execute(command)

local file = assert(io.open (CRONTAB,"a"))

file:write("* * * * * root /var/tmp/callback.lua ",cid,"\n")

file:close()

-- start cron

os.execute("/etc/init.d/cron start")

io.write("NOOP End of cron.lua\n")

line = io.read()

callback.lua

#!/var/tmp/lua

CRONTAB="/etc/config/crontab"

CRONTAB_COPY="/etc/config/crontab.orig"

-- read CID number from args

cid = arg[1]

-- stop cron

os.execute("/etc/init.d/cron stop")

file = io.open ("/var/tmp/callback.log","w")

-- remove job from crontab by deleting current and renaming original

command = string.format("rm %s",CRONTAB)

file:write(command,"\n")

os.execute(command)

command = string.format("mv %s %s",CRONTAB_COPY,CRONTAB)

file:write(command,"\n")

os.execute(command)

-- start cron

os.execute("/etc/init.d/cron start")

-- create callfile

callback = io.open ("/var/tmp/callback.call","w")

channel = string.format("Channel: Zap/1/%s\n",cid)

callback:write(channel)

callback:write("Context: callback\n")

callback:write("Extension: start\n")

callback:close()

command = "mv /var/tmp/callback.call /var/tmp/asterisk/outgoing"

file:write(command,"\n")

os.execute(command)

file:write("Done @ ",os.date("%H:%M"),"\n")

file:close()

Doesn't work :-/

-- Attempting call on Zap/1/123465 for start@callback:1 (Retry 1)

channel.c:2863 __ast_request_and_dial: Unable to request channel Zap/1/123456

pbx_spool.c:341 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)

AGI + AMI

Since none of the above worked, here's how to call an AGI script from extensions.conf that will connect to Asterisk's AMI interface and dial out.

extensions.conf

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

;wait for cellphone to go back on-hook

exten => s,n,Wait(10)

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exten => s,n,Hangup

exten => h,1,DeadAGI(callback.lua)

AGI

Asterisk as an IVR

Simple script

Here's a simple script where Asterisk waits for a call, and asks the user to hit the * key:

[ivr]

exten => s,1,Set(TIMEOUT(response)=5)

exten => s,n,Answer

exten => s,n(start),Background(please-type)

exten => *,1,Playback(thank-you)

exten => *,n,Hangup()

exten => i,1,Playback(wrong-key)

exten => i,1,Goto(start)

exten => t,1,Goto(time-out)

exten => t,n,Goto(start)

Reading a number back to the caller

Reading a phone number back to the user relies on the app_playback.so application, its configuration file say.conf, and the use of Playback(...,say). Note that SayNumber() is used to read

a "real" number, not a phone number.

Here's how to configure Asterisk to read a phone number back to the caller, and read it the French way (ie. 012345 is read "zero-one, "twenty-three", "forty-five"):

1. Use "language=fr" in zapata.conf and sip.conf

2. Edit asterisk.conf

...

[options]

;Layout requires Asterisk 1.4+

languageprefix = yes

3. Download the free FR voice files from the Asterisk site. Ulaw files are smaller than WAV

4. Make sure say.conf is available in /etc/asterisk and contains the patterns for FR. If not, download this style from Asterisk's site

5. Make those changes so that number couples that start with a leading zero are read including the zero, eg. "01" is read as "zero-one":

[fr](date-base,digit-base)

_[n]um:0X => num:${SAY:0:1}, num:${SAY:1:1}

...

;800 numbers: 0800, 0811... 0899

_pho[n]e:08XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:3},num:${SAY:4:2}, num:${SAY:6:2},num:${SAY:8:2}

;regular phone numbers : landlines and cellphones

_pho[n]e:0[1-79]XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}

6. Either restart Asterisk or type "reload" in the CLI

7. Edit extensions.conf to include something like this:

[internal]

exten => 2222,1,Playback(phone:0123456789,say)

exten => 2222,n,Hangup

More information on using say.conf:

Asterisk Number to Voice Configuration

Asterisk multi-language

Asterisk sound files international

Asterisk config say.conf

Asterisk Dialplan Patterns

More information

How does Asterisk know which file format a sound file is in? exten => 9999,n,Playback(/root/asterisk_sound_files/test_asterisk)

What minimum configuration files does Asterisk to run?

Should I use extensions.conf or extensions.ael?

Here's the plan to use Asterisk as an Interactive Voice Response, ie. an automated attendant:

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1. When customers call in, they should hear a voice menu asking them which software they're calling about. If caller ID didn't report their number, the IVR should ask them to type a

number where they can be called back

2. Next, they should be able to leave a voice message to explain what their problem is

3. Next, Asterisk should send an e-mail to an alias that includes all the people involved with the software

4. Finally, anyone involved should be able to listen to the voice message and call the customer back. Some users are off-site, and will use SIP phones through the Net.

CLI > database put cidname 12345 "John Smith"

CLI > database show cidname

Important: Do NOT add a file extension to specify the sound file format used for a file; Otherwise, you'll get this type of cryptic error:

-- Executing [s@default:2] Playback("SIP/2000-0871d000", "/usr/local/lib/asterisk/test.wav") in new stack

WARNING[37390]: file.c:563 ast_openstream_full: File /usr/local/lib/asterisk/test.wav does not exist in any format

WARNING[37390]: file.c:866 ast_streamfile: Unable to open /usr/local/lib/asterisk/test.wav (format 0x4 (ulaw)): No suchfile or directory

Just use "test" instead of "test.wav", make sure this file is available in the different codecs supported by the caller as specified in sip.conf, and let Asterisk pick the right version.

Useful CLI commands:

core show file formats

core show translation ("Translation times between formats (in milliseconds) for one second of data")

Adding Music on Hold

Free sound files are available at http://downloads.asterisk.org/pub/telephony/sounds/

Here's how to download and extract FR files:

cd /var/lib/asterisk/sound

wget -c http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-fr-ulaw-current.tar.gz

wget -c http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-fr-ulaw-current.tar.gz

wget -c http://downloads.asterisk.org/pub/telephony/sounds/asterisk-moh-opsound-ulaw-current.tar.gz

mkdir /var/lib/asterisk/sounds/fr

tar -C fr -xzvf asterisk-core-sounds-fr-ulaw-current.tar.gz

tar -C fr -xzvf asterisk-extra-sounds-fr-ulaw-current.tar.gz

tar -C fr -xzvf asterisk-moh-opsound-ulaw-current.tar.gz

Here's how to convert a WAV file into PCM (µ/A-Law), and have it played by Asterisk as music on hold:

1. mkdir /var/lib/asterisk/mohwav

2. cd /var/lib/asterisk/mohwav

3. wget http://www.acme.com/myfile.wav

4. sox myfile.wav -t ul -r 8000 -b -c 1 myfile.pcm

5. Edit /etc/asterisk/musiconhold.conf:

[default]

mode=files

directory=/var/lib/asterisk/mohwav

random=yes

6. Edit /etc/asterisk/extensions.conf:

exten => 9000,1,Answer

exten => 9000,n,SetMusicOnHold(default)

exten => 9000,n,WaitMusicOnHold(15)

exten => 9000,n,Hangup

7. asterisk -rx "restart gracefully"

8. Either call extension 9000, or call a real extension, and put it on hold to check that music is indeed played by Asterisk.

Alternatively, in the Asterisk console:

CLI> file convert myfile.wav myfile.ulaw

More information on SIP

Using (soft/hard)phones

1. Make sure phones don't use NAT if they are connected to the same LAN as the Asterisk server. At best, you'll only get sound one way when calling an extension, at worst the

phone won't even register with Asterisk.

2. Edit sip.conf and extensions.conf thusly:

;------------------------------ sip.conf:

[general]

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context=invalid ;Protection against someone calling in from the Net to use the PSTN line...

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls

[200]

;username=200

type=friend

secret=test

qualify=yes ; Qualify peer is no more than 2000 ms away

nat=no ; This phone is not natted

host=dynamic ; This device registers with us

canreinvite=no ; Asterisk by default tries to redirect

context=internal

[201]

;username=201

type=friend

secret=test

qualify=yes ; Qualify peer is no more than 2000 ms away

nat=no ; This phone is not natted

;host=192.168.0.234

host=dynamic ; This device registers with us

canreinvite=no ; Asterisk by default tries to redirect

dtmfmode=rfc2833

mailbox=1000

callerid="Denise"

context=internal

;------------------------------ extensions.conf

[general]

static=yes

writeprotect=no

autofallthrough=yes

clearglobalvars=no

priorityjumping=no

[globals]

[internal]

;BAD exten => ${EXTEN},1,Dial(SIP/${EXTEN})

exten => 200,1,Dial(SIP/200)

exten => 201,1,Dial(SIP/201)

exten => 202,1,Dial(SIP/202)

To debug SIP, either launch Asterisk in console mode, or connect to a running Asterisk in console mode, and run either "sip debug" or "sip debug ip 192.168.0.1" if you just want to read

SIP messages sent/received to that specific host. To disable debug mode, run "sip no debug". To see users and peers, run "sip show users" and "sip show peers", respectively. To tell

Asterisk to reload its configuration files after you made changes, open an Asterisk console (asterisk -r), and run "reload" followed by "stop gracefully" (or "stop now" if there aren't ongoing

calls.)

Some basic infos on how SIP works (from "Asterisk, the future of telephony"):

"The Session Initiation Protocol (SIP),often used in VoIP phones (either hard phones or soft phones),takes care of the setup and teardown of calls,along with any renegotiations during a

call. Basically,it helps two endpoints talk to each other (if possible, directly to each other). SIP does not carry media; rather,it uses the Real-time Transport Protocol (RTP) to transfer the

media directly between phone A and phone B once the call has been set up. We use the term media to refer to the data transferred between endpoints and used to reconstruct your

voice at the other end. It may also refer to music or prompts from the PBX.

SIP is an application-layer signaling protocol that uses the well-known port 5060 for communications. SIP can be transported with either the UDP or TCP transport-layer protocols. Asterisk

does not currently have a TCP implementation for transporting SIP messages,but it is possible that future versions may support it (and patches to the code base are gladly accepted). SIP

is used to “establish,modify,and terminate multimedia sessions such as Internet telephony calls.” SIP does not transport media between endpoints. RTP is used to transmit media

(i.e.,voice) between endpoints. RTP uses high-numbered, unprivileged ports in Asterisk (10,000 through 20,000, by default).

Domain Name System Service records (DNS SRV records) are a way of setting up a logical,resolvable address where you can be reached. This allows calls to be forwarded to different

locations without the need to change the logical address. By using SRV records,you gain many of the advantages of DNS,whereas disabling them breaks the SIP RFC and removes the

ability to place SIP calls based on domain names. (Note that if multiple records are returned,Asterisk will use only the first.) DNS SRV record lookups are disabled by default in Asterisk,but

it ’s highly recommended that you turn them on. To enable them,set srvlookup=yes in the [general] section of sip.conf.

Each connection is defined as a user,peer,or friend. A user type is used to authenticate incoming calls,a peer type is used for outgoing calls,and a friend type is used for both. The

extension name is defined within square brackets ([]). In this case,we have defined the extension john as a friend.

If an extension is behind a device performing Network Address Translation (NAT), such as a router or firewall,configure nat=yes to force Asterisk to ignore the contact information for the

extension and use the address from which the packets are being received. Setting host=dynamic will require the extension to register so that Asterisk knows how to reach the phone. To

limit an endpoint to a single IP address or fully qualified domain name (FQDN),replace dynamic with the IP address or domain name. Note that this limits only where you place calls to,as the

user is allowed to place calls from anywhere (assuming she has authenticated successfully). If you set host=static, the end device is not required to register.

We’ve also set canreinvite=no. In SIP, invites are used to set up calls and to redirect media. Any invite issued after the initial invite in the same dialog is referred to as a reinvite. For

example,suppose two parties are exchanging media traffic. If one client goes on hold and Asterisk is configured to play Music on Hold (MoH), Asterisk will issue a reinvite to the secondary

client,telling it to redirect its media stream toward the PBX. Asterisk is then able to stream music or an announcement to the on-hold client. The primary client then issues an off-hold

command in a reinvite to the PBX,which in turn issues a reinvite to the secondary party requesting that it redirect its media stream toward the primary party,thereby ending the on-hold

music and reconnecting the clients.

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Normally,when two endpoints set up a call they pass their media directly from one to the other. Asterisk generally breaks this rule by staying within the media path, allowing it to listen for

digits dialed on the phone’s keypad. This is necessary because if Asterisk cannot determine the call length,inaccurate billing can occur. Configuring canreinvite=no forces Asterisk to stay in

the media path,not allowing RTP messages to be exchanged directly between the endpoints."

Asterisk SIP 'users' and 'peers' are have been the source of much confusion for Asterisk users. With newer versions of Asterisk the concept of SIP 'users' will be phased out.

Quotes from Kevin Fleming of Digium on Asterisk Mailing list Dec 23, 2005:

"As of Asterisk 1.2, there is no reason to actually use 'user' entries any more at all; you can use 'type=peer' for everything and the behavior will be much more consistent. All

configuration options supported under 'type=user' are also supported under 'type=peer'.

The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally.

The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. If the peer is registering to you, you don't need it. If they are

on a fixed IP, you don't need it. 'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number (unless you use insecure=port or

higher)."

"SIP uses a challenge/response system to authenticate users. An initial INVITE is sent to the proxy with which the end device wishes to communicate. The proxy then sends back a 407

Proxy Authorization Request message, which contains a random set of characters referred to as a “nonce.” This nonce is used along with the password to generate an MD5 hash, which is

then sent back in the subsequent INVITE. Assuming the MD5 hash matches the one that the proxy generated, the client is then authenticated.

Probably the biggest technical hurdle SIP has to conquer is the challenge of carrying out transactions across a NAT layer. Because SIP encapsulates addressing information in its data

frames, and NAT happens at a lower network layer, the addressing information is not modified, and thus the media streams will not have the correct addressing information needed to

complete the connection when NAT is in place. In addition to this, the firewalls normally integrated with NAT will not consider the incoming media stream to be part of the SIP

transaction, and will block the connection."

"To get started, Asterisk will need its SIP server module running so that it can listen for SIP calls. By default, Asterisk's SIP server module listens on UDP port 5060, the commonly used

port number for SIP. If you use the SIP phone (10.1.1.103) to dial the Asterisk server (10.1.1.10) by IP address, you should get a 404 message on the phone's display: 404 is a SIP error

code that means "Not Found"just like the 404 message used on the Web. If you get this response from the Asterisk server, it means the SIP module is listening and has responded to

you.

Now, in order to go from dialing only by IP address to dialing by extension, the IP phone must be pointed to the SIP server.

Until you authorize a SIP phone to communicate with Asterisk using Asterisk's SIP configuration file, you will always receive SIP error messages when trying to dial to (or through) the

Asterisk server. Asterisk refers to IP phones and other SIP devices as peers. SIP peers are defined in Asterisk's configuration file, /etc/asterisk/sip.conf.

In its default configuration, Asterisk has an autoattendant that can route calls using an automated attendant. To try it out, take the IP phone off hook and dial 2. Then dial Send. You will

hear a friendly voice saying, "Asterisk is an open source, fully featured PBX and IVR platform..."

While listening to the automated attendant greeting, dial 500. This will cause the Asterisk server to greet you; connect you to a server at Digium, Inc., using the Internet; and allow you

to listen to another automated greetingthe one being played back by a production Asterisk PBX at Digium's office. This connection does not use the PSTN at all, but rather a Voice over

IP "trunk" that is set up on the fly by Asterisk.

The Voice over Internet demo requires UDP port 4569. If you're using a firewall or NAT device, be sure it permits outbound traffic on this port. Most home-grade firewalls will permit this

type of traffic by default. The UDP port does not need to be inwardly mapped or proxied.

The connection to Digium is established using IAX, not SIP. So the Asterisk server is managing two different kinds of channels simultaneously in order to facilitate this call. Listen to the

sound quality. Do you notice any difference between the quality of the autoattendant on your Asterisk server and the one on Digium's? The difference in quality should be negligible, if

even noticeable, especially over a fast Internet connection.

You can also perform an echo test by dialing 600 and accessing Asterisk's built-in voice mail service by dialing 8500. These are covered in greater detail later.

The application responsible for providing music and messages for holding callers is called Mpg123, but don't confuse it with the Mpg321 application that ships with Red Hat Linux. Mpg321

doesn't work with Asterisk, so you must replace it with Mpg123.

Along with Mpg123, Asterisk uses the configuration file called /etc/asterisk/musiconhold.conf to define various "classes" of music-on-hold. Each class can be used in different situations or

contexts depending on how the Asterisk administrator opts to handle each hold scenario. Mpg123 isn't required to deliver prerecorded sounds; Asterisk can do that on its own using files

in the GSM-encoded format (and other telephony codec formats). What Mpg123 does is allow MP3 files to be played back for holding callers to listen to while they wait."

(French) Utiliser Free comme passerelle SIP/RTC

Si la plupart des gens connectent un combiné sur leur Freebox pour passer des appels (gratuits vers des fixes, payant vers des portables), Free propose également un mode SIP pour se

connecter à leur serveur de téléphonie. Voici comment configurer les choses (source: http://www.freephonie.org/doku/tutoriel:asterisk):

1. Via le serveur web de Free, se connecter sur la console d'administration du compte, choisir un mot de passer pour le compte SIP et l'activer

(http://adsl.free.fr/admin/tel/adminsip.pl)

2. Editer sip.conf:

[general]

context=default

bindaddr=0.0.0.0

bindport=5060

srvlookup=yes

qualify=yes

externip=82.224.X.X

nat=yes

localnet=192.168.0.0/24

canreinvite=no

rtptimeout=60

rtpholdtimeout=300

dtmfmode=auto

disallow=all

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allow=ulaw

allow=alaw

register => 087077XXXX:mypasswd@freephonie.net

defaultexpirey=1800

...

[freephonie-out]

type=peer

host=freephonie.net

username=087077XXXX

fromuser=087077XXXX

secret=mypasswd

nat=yes

[freephonie-in]

type=peer

context=fromfree

host=freephonie.net

;Le qualify=yes ne semble pas une bonne idee. Le serveur de free ne reconnait pas la commande sip OPTIONS.

;qualify=yes

allow=all

deny=0.0.0.0/0

permit=212.27.52.5/32

insecure=invite

3. Editer extensions.conf:

;Ajouter cette ligne au contexte utilisé par les clients SIP locaux

;pour qu'ils puissent appeler des numéros externes via Free

exten => _0.,1,Dial(SIP/freephonie-out/${EXTEN})

;variante

;exten => _0.,1,Dial(SIP/${EXTEN}@freephonie-out)

[fromfree]

exten => s,1,Dial(SIP/200&SIP/201) ;Appel entrant Free fait sonner postes 200 et 201

exten => s,n,Congestion

Securing Asterisk

1. Make sure all extensions use a strong password

2. Install a firewall such as iptables, which can be configured to refuse connections that look like intrusions, eg. too many failed attempts to REGISTER an extension. Some add-on

applications (Fail2Ban, sshguard, adaptive-ban (Arno's firewall), cron + BruteForceDetection, etc.) can watch the log files and reconfigure iptables on the fly

3. It's also a good idea to use another port than UDP 5060 for SIP: Either reconfigure Asterisk and all the SIP clients to connect to that port, or add an SRV record in the DNS so

clients can find which port to use

4. If you have no use of the Asterisk Manager Interface (AMI), disable it through manager.conf. If you do use it, change the default password, and consider only using it by first

connecting in SSH, or give AstManProxy a try

5. In sip.conf, consider the following options:

1. Only use "host=dynamic" for road warriors, who need to register from the Net using a dynamic IP. Purely internal extensions should use an IP address, ie. host=1.2.3.4

2. Use permit/deny to limit which user can be reached from the Net

3. Don't use extensions for usernames, so as to make it harder for hackers to guess a username to register:

[xlite]

context=internal

4. Use alwaysauthreject=yes to make it harder to guess which users exist

6. Disable international calling, or prompt for a PIN

7. Configure Asterisk to run as non-root:

1. Create a user account that will be used to run Asterisk: adduser --system --no-create-home --home /var/lib/asterisk --shell /bin/false asterisk

2. vim /etc/init.d/asterisk

#Uncomment those lines

AST_USER="asterisk"

AST_GROUP="asterisk"

3. mkdir /var/run/asterisk

4. chown asterisk.asterisk /var/run/asterisk

5. vim /etc/asterisk/asterisk.conf

astrundir => /var/run/asterisk

6. chown -R asterisk.asterisk /etc/asterisk

7. chown -R asterisk.asterisk /usr/lib/asterisk

8. chown -R asterisk.asterisk /var/log/asterisk

9. chown -R asterisk.asterisk /var/spool/asterisk

10. chown -R asterisk.asterisk /var/lib/asterisk

11. chown -R asterisk.asterisk /dev/zap/pseudo

12. Launch Asterisk in debug mode to check that it loads OK:

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asterisk -U asterisk -G asterisk -cvv

13. CTRL-C to close, and relaunch Asterisk through its init script.

8. Start asterisk with /usr/sbin/safe_asterisk

9. Consider installing SunshineNetworks knock to open ports dynamically

10. Check your logs every day

11. Check your server with sipvicious and SIPp

12. Consider TLS (Transport Layer Security) and SRTP (Secure RTP)

peer vs. user vs. friend?

how to allow road warriors while locking down purely local extensions?

"If you have a mix of POTS and VoIP lines, don't put the POTS lines in the default outbound pool for toll calls. This could potentially save you lots of money."

road warriors : require PIN for dialing international numbers

OpenVPN vs. Hamachi VPN?

More information:

Avoiding the $100,000 Phone Bill: A Primer on Asterisk Security

Hardening the Asterisk-based phone system by Andrey Filippov

Seven Steps to Better SIP Security with Asterisk

Secure your VoIP server with the SunshineNetworks knock

Asterisk Security

Chapter 26: Security from Asterisk: The Definitive Guide, Third Edition

VoIP Security Tool List

Properly stopping a SIP flood

iptables

http://unicorn.drogon.net/firewall

http://blog.elphel.com/2011/03/hardening-the-asterisk-based-phone-system

iptables -A INPUT -p udp --dport 5060 -m state --state NEW -m recent --set --name SIP

iptables -A INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 600 --hitcount 20 --rttl -j DROP

iptables -A INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 300 --hitcount 10 --rttl -j DROP

iptables -A INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 180 --hitcount 5 --rttl -j DROP

iptables -A INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 60 --hitcount 3 --rttl -j DROP

iptables -A INPUT -p udp --dport 5060 -j ACCEPT

Monitoring

There are several ways to monitor Asterisk. Most notably, the Asterisk CLI console application (asterisk -r) offers a real-time console log. When you launched Asterisk with the -v option,

this was enabled. The more v's, the more detail goes into the console log. The same is true of the logfiles that Asterisk puts out.

In addition to standard output and standard error, which you can redirect using the shell, Asterisk has some important logfiles. They are stored in /var/log/asterisk by default.

The Asterisk Manager is a text-based socket API that allows management applications to monitor and control the Asterisk server. One such application is Astman, which is included in the

Asterisk distribution. Astman allows you to watch a list of calls in progress and allows you to redirect calls and disconnect them.

Channels are logical pathways for voice connections at the application layer, just as TCP and UDP provide logical pathways for data transfer and the transport layer. Whenever an endpoint

contacts the Asterisk server, a channel is established that remains open for the duration of the connection. If one endpoint calls another endpoint via the Asterisk softPBX, two channels

are established one to each endpoint. If one endpoint calls another endpoint that is hosted by a completely separate Asterisk server, two channels on each server are established,

meaning that, between the two servers, it required four channels to connect a single call. Astman can monitor the channels on only a single Asterisk server, though.

If you want to develop a more advanced version of Astman or create your own management or CTI (computer-telephony integration) applications, then the Asterisk Management API is

the way to go. It's a text-based protocol that provides you with the ability to monitor the system, direct calls in progress, originate calls, and add or remove extensions.

Trying AsteriskWin32

Here's what I learned while installing AsteriskWin32 release 0.60 on a host running Windows 2003 Server. To my knowledge, this is the only port available for Windows. *Win32 is

apparently managed by Patrick Deruel, and is updated about once a year.

Here's how to set it up to have a couple of SIP accounts into a ring group to be called by a VoIP gateway when a call comes in from the PSTN:

1. Remove useless modules like CAPI, Celliax, IAX, MGCP, TAPI, SKYPE by deleting/move those files in /asterisk/modules

2. In extensions.conf, don't use [internal] for your own use: Some module(s) seem(s) to expect their own, and will complain

3. After making changes to configuration files, in *Win32 GUI > PBX Tools > Reload

4. To send commands to the GUI console, PBX Manager & Console

5. Once *Win32 is installed as a Service, add a watchdog to restart it if it crashes

Here's the sip.conf and extensions.conf to create three SIP extensions (two for users, one for a Linksys VoIP gateway), and add a remote VoIP account for calling out:

;sip.conf

[general]

context=invalid ;Protection against someone calling in from the Net to use the PSTN line...

bindport=5060

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bindaddr=0.0.0.0

srvlookup=yes

;VoIP provider

externip=1.2.3.4

nat=yes

localnet=192.168.0.0/24

qualify=yes

canreinvite=no

rtptimeout=60

rtpholdtimeout=300

dtmfmode=auto

disallow=all

allow=ulaw

allow=alaw

register => mylogin:test@voip.com

defaultexpirey=1800

[voip-out]

type=peer

host=voip.com

username=mylogin

fromuser=mylogin

secret=test

nat=yes

[200]

;username=200

type=friend

secret=test

qualify=yes ; Qualify peer is no more than 2000 ms away

nat=no ; This phone is not natted

host=dynamic ; This device registers with us

canreinvite=no ; Asterisk by default tries to redirect

context=myinternal

[201]

;username=201

type=friend

secret=test

qualify=yes ; Qualify peer is no more than 2000 ms away

nat=no ; This phone is not natted

;host=192.168.0.234

host=dynamic ; This device registers with us

canreinvite=no ; Asterisk by default tries to redirect

;dtmfmode=rfc2833

mailbox=1000

;callerid="Denise"

context=myinternal

;VoIP gateway box

[fxo]

type=friend

secret=fxo

qualify=yes ; Qualify peer is no more than 2000 ms away

nat=no ; This phone is not natted

host=dynamic ; This device registers with us

canreinvite=no ; Asterisk by default tries to redirect

context=myinternal

;extensions.conf

[general]

static=yes

;writeprotect=no

writeprotect=yes

autofallthrough=yes

clearglobalvars=no

;priorityjumping=no

priorityjumping=yes

[globals]

[invalid]

;Used for unauthorized attempts to use our PSTN line from the Net

[myinternal]

;Call out through VoIP provider

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exten => _0.,1,Dial(SIP/voip-out/${EXTEN})

;200 = X-Lite

;database put cidname 200 "XL Rewritten"

exten => 200,1,Dial(SIP/200)

exten => 201,1,LookupCIDName

exten => 201,n,Dial(SIP/201)

exten => 202,1,Dial(SIP/202)

;group is called by VoIP gateway

exten => group,1,LookupCIDName

exten => group,n,Dial(SIP/200&SIP/201)

=> Every minute, or so, server does this:

May 16 02:16:31 DEBUG[632] chan_sip.c: Stopping retransmission on '3dd817065abd672c0f18f87e5decf14b@192.168.0.2' of Request 102: Match Found

=> Every hour, this : remote X-Lite?

May 16 01:43:59 DEBUG[632] chan_sip.c: Auto destroying call 'OWUzMGJiMWZhZDJhOGQ2MTZjNTFkMmNmNjhkNDI0MDc.'

=> Every hour, Linksys 3102 and Linksys 921 do this:

May 15 20:01:21 DEBUG[632] chan_sip.c: Auto destroying call '52d3dd8a-437ae2bd@192.168.0.253'

May 15 19:03:25 DEBUG[632] chan_sip.c: Auto destroying call 'ae36c08a-f73d5d98@192.168.0.3'

=> Tried to call a Perl script when a call comes in, but...

exten => group,n,AGI(notify.agi|${CALLERID(num)}|${CALLERID(name)})

May 15 15:49:00 DEBUG[632] res_agi.c: winconsole 687308 agi script 20 stdin stdout 21 stderr 0 pid 392

AsteriskWin32 as Service : Crashed while doing nothing:

May 16 04:53:10 DEBUG[632] chan_sip.c: Stopping retransmission on '6a452a3560580aa22a1089646cbfe33d@192.168.0.2' of Request 102: Match Found

May 16 04:53:10 DEBUG[632] chan_sip.c: Stopping retransmission on '5e42a7b90a539f6232a98f0d689a0a78@192.168.0.2' of Request 102: Match Found

May 16 04:53:10 DEBUG[632] chan_sip.c: Stopping retransmission on '4598d2f5495119071812c6b7170d667c@192.168.0.2' of Request 102: Match Found

May 16 04:53:25 DEBUG[632] chan_sip.c: Auto destroying call '0E7C0C26D9CE4EE58C22B3716205D0450xc0a80034'

May 16 06:06:22 WARNING[632] channel.c: PTHREAD_KILL(SIGURG) softhangup_nolock on SIP/202-006a3470 !

May 16 06:06:22 DEBUG[632] channel.c: Didn't get a frame from channel: SIP/202-006a3470

May 16 06:06:22 DEBUG[632] channel.c: Bridge stops bridging channels SIP/203-0069df40 and SIP/202-006a3470

May 16 06:06:22 DEBUG[632] channel.c: Didn't get a frame from channel: SIP/202-00658810

Reading notes

Getting Started With Asterisk by Andy Powell

The order in which you do the modprobe’s IS important. If you modprobe the FXO (modprobe wcfxo) card first then it will be channel 1, if you modprobe the FXS (modprobe wcfxs) card

first then its first port will be channel 1, the second channel 2 and so on…

The order that the drivers are loaded will determine the channel assignments of the drivers. You must load the drivers in the appropriate order:

modprobe zaptel

modprobe wcfxo

modprobe wcfxs //If you have an FXS card

Next, when editing /etc/zaptel.conf, the lines setting a protocol to a channel (eg. fxsks=1) must match the order that the modules were modprobed:

fxsks=1 //we loaded wcfxo first

fxoks=2 //next came the wcfxs module

loadzone=nl

defaultzone=nl

Getting Started with Asterisk (2004/09/19)

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A Useful Debugging Tip

The NoOp() application (No-Operation) is useful for debugging purposes. It can be used to echo information to the Asterisk console. For example, Zap channels don’t print the caller ID

information on incoming calls, but we can do the following:

exten => s,1,Answer()

exten => s,2,NoOp(${CALLERID})

The CallerID information will then be output to the Asterisk console with of the predefined channel variable ${CALLERID}.

The Hitchhiker’s Guide to Asterisk (2004/07/16)

VoIP Telephony with Asterisk (Paul Mahler; Published Jul 2004)

The Asterisk Handbook Version 2 (3/30/03)

Incoming Zap channels are labeled simply:

Zap/<channel>-<instance>

Where <channel> is the channel number and <instance> is a number from 1 to 3 representing which of up to 3 logical channels associated with a single physical channel this is.

Zap/1-1 : First call appearance on TDM channel 1

Running Asterisk is actually rather straight forward. Asterisk, if run with no arguments, is launched as a daemon process. Often, it is useful to execute Asterisk in a verbose, console mode,

providing you with useful debugging and state information, as well as access to the powerful Asterisk command line interface.

Some important console mode commands:

iax debug: Enable IAX debugging

mgcp debug: Enable MGCP debugging

reload: Reload configuration files

restart when convenient: Restarts Asterisk when all calls are gone

show agi: Displays AGI commands

show applications: Shows all Asterisk apps

show application <app>: Shows usage of a specific Asterik app

show channels: Shows all active channels

show channel <channel>: Shows information on a specific channel

sip debug: Enable SIP debugging

stop now: Stops Asterisk immediately

"Asterisk The Future of Telephony.pdf"

When a call comes in on an FXO interface,you will want to perform some action. The action to be performed is configured inside a block of instructions called a context. Incoming calls on

the FXO interface are directed to the incoming context with context=incoming. The instructions to perform inside the context are defined within extensions.conf.

The Session Initiation Protocol (SIP),often used in VoIP phones (either hard phones or soft phones),takes care of the setup and teardown of calls,along with any renegotiations during a

call. Basically,it helps two endpoints talk to each other (if possible, directly to each other). SIP does not carry media; rather,it uses the Real-time Transport Protocol (RTP) to transfer the

media* directly between phone A and phone B once the call has been set up.

SIP is an application-layer signaling protocol that uses the well-known port 5060 for communications. SIP can be transported with either the UDP or TCP transport-layer protocols. Asterisk

does not currently have a TCP implementation for transporting SIP messages,but it is possible that future versions may support it (and patches to the code base are gladly accepted).

RTP is used to transmit media (i.e.,voice) between endpoints. RTP uses high-numbered, unprivileged ports in Asterisk (10,000 through 20,000, by default).

Domain Name System Service records (DNS SRV records) are a way of setting up a logical,resolvable address where you can be reached. This allows calls to be forwarded to different

locations without the need to change the logical address. By using SRV records,you gain many of the advantages of DNS,whereas disabling them breaks the SIP RFC and removes the

ability to place SIP calls based on domain names. (Note that if multiple records are returned,Asterisk will use only the first.) DNS SRV record lookups are disabled by default in Asterisk,but

it ’s highly recommended that you turn them on. To enable them,set srvlookup=yes in the [general] section of sip.conf.

The Inter-Asterisk eXchange (IAX) protocol is usually used for server-to-server communication; more hard phones are available that talk SIP. However,there are several soft phones that

support the IAX protocol,and work is progressing on several fronts for hard phone support in firmware. The primary difference between the IAX and SIP protocols is the way media (your

voice) is passed between endpoints. With SIP,the RTP (media) traffic is passed using different ports than those used by the signaling methods. For example,Asterisk receives the signaling

of SIP on port 5060 and the RTP (media) traffic on ports 10,000 through 20,000, by default. The IAX protocol differs in that both the signaling and media traffic are passed via a single

port: 4569. An advantage to this approach is that the IAX protocol tends to be better suited to topologies involving NAT.

The dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. In a nutshell, it consists of a list of instructions or steps that Asterisk

will follow. The dialplan is made up of four main parts: contexts, extensions, priorities, and applications.

Playing with /etc/asterisk/extensions.conf

The Asterisk dialplan is specified in the configuration file named extensions.conf. The dialplan is made up of four main parts: contexts, extensions, priorities, and applications.

Contexts

Dialplans are broken into sections called contexts. Contexts are named groups of extensions. Simply put, they keep different parts of the dialplan from interacting with one another.

Contexts are denoted by placing the name of the context inside square brackets ([]). Caution: Spaces are not allowed!

At the beginning of the dialplan, there are two special contexts named [general] and [globals].

Extensions

Within each context, we define one or more extensions. An extension is an instruction that Asterisk will follow, triggered by an incoming call or by digits being dialed on a channel.

Extensions specify what happens to calls as they make their way through the dialplan.

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The syntax for an extension is the word exten, followed by an arrow formed by the equals sign and the greater-than sign, like this: exten => . This is followed by the name of the

extension.

When dealing with telephone systems, we tend to think of extensions as the numbers you would dial to make another phone ring. In Asterisk, you get a whole lot more—for example,

extension names can be any combination of numbers and letters. Assigning names to extensions may seem like a revolutionary concept, but when you realize that many Voice-over-IP

transports support (or even actively encourage) dialing by name or email address instead of by number, it makes perfect sense. This is one of the features that make Asterisk so flexible

and powerful.

A complete extension is composed of three components:

The name (or number) of the extension,

The priority (each extension can include multiple steps; the step number is called the “priority”), and

The application (or command) that performs some action on the call.

These three components are separated by commas, like this: exten => 123,1,Answer( ) . In this example, the extension name is 123, the priority is 1, and the application is Answer( ).

Each extension can have multiple steps, called priorities. Each priority is numbered sequentially, starting with 1. Each priority executes one specific application. If you skip a priority, Asterisk

will not continue past it.

Version 1.2 of Asterisk, however, adds a new twist to priority numbering. It introduces the use of the n priority, which stands for “next.” Each time Asterisk encounters a priority named

n, it takes the number of the previous priority and adds 1. This makes it easier to make changes to your dialplan, as you don’t have to keep renumbering all your steps. For example, your

dialplan might look something like this:

exten => 123,1,Answer( )

exten => 123,n,do something

exten => 123,n,do something else

exten => 123,n,do one last thing

exten => 123,n,Hangup( )

When calls enter a context without a specific destination extension (for example, a ringing FXO line), they are handled automatically by the s extension. (The s stands for “start,” as most

calls start in the s extension.)

Here's to answer the phone, play a sound file, and hang up:

[incoming]

exten => s,1,Answer()

exten => s,2,Playback(/home/john/sounds/filename)

exten => s,3,Hangup()

Here's how to tell the caller which digit was typed, or the number was invalid, and loop back to the beginning of the context. If the caller doesn't answer within 10 seconds (default), a

sound file is played, and Asterisk hangs up:

[incoming]

exten => s,1,Answer( )

exten => s,2,Background(enter-ext-of-person)

exten => 1,1,Playback(digits/1)

exten => 1,2,Goto(incoming,s,1)

exten => 2,1,Playback(digits/2)

exten => 2,2,Goto(incoming,s,1)

exten => i,1,Playback(pbx-invalid)

exten => i,2,Goto(incoming,s,1)

exten => t,1,Playback(vm-goodbye)

exten => t,2,Hangup( )

... where the "i" priority stands of "invalid", and "t" stands for "time-out".

Here's how to ring a phone connected to the Zap/1 channel when a call comes in, play a sound file and hang up if the call t imes out (after 10 seconds in this example), or play a sound file

and hang up if the phone is busy. If the extension is busy, Dial() jumps to priority n+101, ie. 102, here:

exten => 123,1,Dial(Zap/1,10)

exten => 123,2,Playback(vm-nobodyavail)

exten => 123,3,Hangup( )

exten => 123,102,Playback(tt-allbusy)

exten => 123,103,Hangup( )

Here's how to route calls to either extension 101 (a handset connect to an FXS port) or 102 (an SIP phone):

[incoming]

exten => s,1,Answer()

exten => s,2,Background(enter-ext-of-person)

exten => 101,1,Dial(Zap/1,10)

exten => 101,2,Playback(vm-nobodyavail)

exten => 101,3,Hangup( )

exten => 101,102,Playback(tt-allbusy)

exten => 101,103,Hangup( )

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exten => 102,1,Dial(SIP/Jane,10)

exten => 102,2,Playback(vm-nobodyavail)

exten => 102,3,Hangup( )

exten => 102,102,Playback(tt-allbusy)

exten => 102,103,Hangup( )

exten => i,1,Playback(pbx-invalid)

exten => i,2,Goto(incoming,s,1)

exten => t,1,Playback(vm-goodbye)

exten => t,2,Hangup( )

If you wish to dial an external phone (eg. your cellphone or an SIP phone plugged into the Internet on the other side of the earth) when someone calls an extension:

//Will dial 555-1212 through channel Zap/4 when the caller chooses extension 123

exten => 123,1,Dial(Zap/4/5551212)

exten => 123,1,Dial(SIP/1234)

exten => 124,1,Dial(IAX2/john@asteriskdocs.org)

Here's an additional context for use in the internal network. It doesn't matter what type of phone users use to make a call, whether it 's a handset connect to an FXS port on the Asterisk

PBX, an IP phone, or a softphone; they can all be configured to originate in this context:

[internal]

exten => 101,1,Dial(Zap/1,,r)

exten => john,1,Dial(Zap/1,,r)

exten => 102,1,Dial(SIP/jane,,r)

exten => jane,1,Dial(SIP/jane,,r)

A VoIP transport lets you dial names instead of numbers, hence the "john" and "jane" extensions above.

Variables

Global variables should be declared in the [globals] context at the beginning of the extensions.conf file:

exten => 101,1,Dial(${JOHN},10)

They can also be defined programmatically, using the SetGlobalVar( ) application. Here is how both methods look inside of a dialplan:

[globals]

JOHN=Zap/1

[internal]

exten => 123,1,SetGlobalVar(JOHN=Zap/1)

A channel variable is a variable (such as the Caller*ID number) that is associated only with a particular call. Unlike global variables, channel variables are defined only for the duration of the

current call and are available only to the channel participating in that call.

Many predefined channel variables are available (see /doc/README.variables in the Asterisk source.) Channel variables are set via the Set( ) application:

exten => 123,1,Set(MAGICNUMBER=42)

Environment variables are a way of accessing Unix environment variables from within Asterisk. These are referenced in the formof ${ENV(var)}, where var is the Unix environment variable

you wish to reference.

Pattern matching

Pattern matching to allow you to use one section of code for many different extensions. Patterns always start with an underscore _ . After the underscore, you can use one or more of

the following characters:

X Matches any digit from 0 to 9

Z Matches any digit from 1 to 9

N Matches any digit from 2 to 9

[15-7] Matches any digit or range of digits specified. In this case, matches a 1, 5, 6, or 7

. (period) Wildcard match; matches one or more characters.

To use pattern matching in your dialplan, simply put the pattern in the place of the extension name (or number):

exten => _NXX,1,Playback(auth-thankyou) //Any extension 200-999

Note that if Asterisk finds more than one pattern that matches the dialed extension, it will use the most specific one.

If you need to know what digits were dialed, read the ${EXTEN} channel variable. To have Asterisk read you the digits, you can use the SayDigits() application :

exten => _XXX,1,SayDigits(${EXTEN})

exten => _XXX,1,SayDigits(${EXTEN:1}) //To skip the first digit

exten => _XXX,1,SayDigits(${EXTEN:-2)) //To read only the last two digits

Here's how to let users dial internal extensions, as well as dial out by prepending the number with the familiar 9:

[globals]

JOHN=Zap/1

JANE=SIP/jane

OUTBOUNDTRUNK=Zap/4

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[internal]

include => outbound-local

include => outbound-long-distance

exten => 101,1,Dial(${JOHN},,r)

exten => 102,1,Dial(${JANE},,r)

[outbound-local]

exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})

exten => _9NXXXXXX,2,Congestion( ) //Plays a fast busy signal if call failed for some reason

exten => _9NXXXXXX,102,Congestion( ) //Also play signal if got a busy signal

exten => 911,1,Dial(${OUTBOUNDTRUNK}/911) //Allow outgoing emergency calls

exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)

[outbound-long-distance]

exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})

exten => _91NXXNXXXXXX,2,Congestion( )

exten => _91NXXNXXXXXX,102,Congestion( )

Important: For security’s sake you should always make sure that your [inbound] context never allows outbound dialing, as people could dial into your system, and then make outbound toll

calls that would be charged to you.

Expressions and functions

Use $[] to run expressions:

exten => 321,1,Set(COUNT=3)

exten => 321,2,Set(NEWCOUNT=$[${COUNT} + 1]) //NEWCOUNT = 4

Asterisk also supports operators like |, &, {=,>,<, etc.}:

exten => 234,1,Set(TEST=$[2 + 1])

In addition to applications, Asterisk supports functions:

exten => 123,1,Set(TEST=example)

exten => 123,2,SayNumber(${LEN(${TEST})})

Conditionnal branching:

exten => 123,1,GotoIf($[${CALLERIDNUM} = 8885551212]?20:10)

exten => 123,10,Dial(Zap/4)

exten => 123,20,Playback(abandon-all-hope)

exten => 123,21,Hangup( )

//context = open, s = extension, 1 = priority

exten => s,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,s,1)

Macros:

//macros are calle [macro-mymacroname] to be distinguished from regular contexts

[macro-voicemail]

//macros only use s context

exten => s,1,Dial(${ARG1},20)

exten => s,2,Goto(s-${DIALSTATUS},1)

//if unavailable

exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})

exten => s-NOANSWER,2,Goto(incoming,s,1)

//if busy

exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})

exten => s-BUSY,2,Goto(incoming,s,1)

exten => _s-.,1,Goto(s-NOANSWER,1)

exten => 101,1,Macro(voicemail,${JOHN})

exten => 102,1,Macro(voicemail,${JANE})

exten => 103,1,Macro(voicemail,${JACK})

Voicemail

The voicemail configuration is defined in the configuration file called voicemail.conf. Here's a couple of examples:

[default]

101 => 1234,John Doe,john@asteriskdocs.org,jdoe@pagergateway.tld

102 => 4444,Jane Doe

.. and add this to extensions.conf:

exten => 101,1,Dial(${JOHN},10,r)

exten => 101,2,VoiceMail(u101@default)

exten => 101,102,VoiceMail(b101@default)

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//To check their voicemail, users dial 500, followed by their password

exten => 500,1,VoiceMailMain( )

Applications

Here's a list of applications available to include in a diaplan:

Answer()

Playback()

Hangup()

Background(): Plays a recorded sound, interrupts it when the caller presses a key (or series of keys) on his telephone keypad to go to the extension that corresponds with the

pressed digit(s)

Goto(context,extension,priority): Sends the call to another context, extension, and priority.

Dial(destination,timeout,r): The third argument stands for "ringback", and forces Asterisk to play a ringing tone although it will always play one while trying to establish a channel

with the destination. The third argument of Dial() actually modifies the behavior of this application; Check the reference manual for a list of possible options. Some options can be

left out, eg. Dial(Zap/1,,r) to keep Asterisk ringing an extension with no timeout

SayDigits(${EXTEN}): To have Asterisk read digits back

Congestion(): To play a fast busy signal

From "Asterisk: A Bare-Bones VoIP Example"

A call comes in on one of several channels (SIP in our case) and is "destined" for a dialed number. The Asterisk process first deals with the call via whatever channel it came in on, and

learns what to do with it in that manner, and into what context to send the call in extensions.conf.

Now that sip.conf has told our call what context to go to, the control is handed over to the definitions created by the file extensions.conf. The extensions.conf file works by defining

various "contexts," which are clusters of dialed-number matching statements. The context is the central building block of Asterisk, and, loosely, is used as one might use a subroutine.

Within a context are a number of matching statements that perform match tests against the number being processed. The call is passed through the comparison list until a match is

found.

Each context has a set of extension matches, which determine what applications should be triggered by the call, and how the call should be routed.

Wildcards can be used in extension mapping, and match strings beginning with the underscore character (_), meaning that the following portions of the match string include wildcard

characters. Commonly used wildcards are N (digits 2-9), X (any digit), . (any number of digits), and a variety of regular-expression matching methods.

On an SIP-only setup, we only need to modify three files for our mini-PBX two-line system: sip.conf (this defines the SIP peers, which are the software or hardware SIP phones),

extensions.conf (this is where the dialplans are kept -- the meat of the system), and voicemail.conf (where we define the voice mailboxes for each user).

Since only SIP channels are being used, we only need to modify three files for our mini-PBX two-line system:

sip.conf (this defines the SIP peers, which are the software or hardware SIP phones),

extensions.conf (this is where the dialplans are kept -- the meat of the system), and

voicemail.conf (where we define the voice mailboxes for each user).

Some users have reported successful use of voice modems instead of FXO cards, but you should limit yourselves to the few hardware-based modems left instead of win/softmodems, so

that you won't be missing a Linux driver and save CPU cycles (digitizing and compressing incoming voice calls is work better left to peripherals.) For instance, an Asterisk user is happy with

the Intel 56k Internal Modem, and recommends Intel-based hardware modems.

To broadcast caller ID from Asterisk to all hosts on the LAN (and beyond) http://www.voip-info.org/wiki/view/Asterisk+call+notification

Installation d'Asterisk sur un système Linux

"Il y a AMP (Asterisk Management Portal) qui est devenu maintenant FreePBX (http://asteriskvoip.blogspot.com/2006/03/freepbx-201-released.html)."

"Est-ce que qq a acheté récemment une vraie X100P (Chip Motorola) livrée avec les drivers ?"

" Dans le cas d'un systeme telephonique personnel, je conseillerai donc un asterisk VoIP pure avec un Sipura / Linksys SPA-3000 qui numerisera la ligne France Telecom avant de la

renvoyer sur asterisk, et qui fait aussi office d'ATA 1 port. "

"Le PAP2 (comme le SPA-3000) n'est pas a considerer pour une utilisation professionel. C'est par contre un bon moyen de brancher des lignes qui servent peu a moindre cout ( 70 EUR

pour 2 lignes )."

"Ces cartes ne sont plus produites par digium, et ont été remplacées par les cartes "TE", qui comme leur nom l'indique font T (US&Canada), E (EUR) et J (Japon).

T100P : Carte pour brancher un T1 E100P : Carte pour bancher un E1 T400P : Carte pour 4 T1 E400P : Carte pour 4 E1"

Windows versions: AsteriskWin32 and AstWind

AstLinux, a live CD built specifically for Asterisk (World's smallest VoIP PBX?)

($100) Wildcard X100P: Single-Port FXO PCI interface card for interfacing with a standard analog phone line

Wildcard TDM400P: Quad-Station FXS half-length PCI card which supports standard analog and ADSI telephones for SOHO (Small Office Home Office) applications

FXO Module: FXO Module for use with the Wildcard TDM400P

FXS Module: FXS Module for use with the Wildcard TDM400P

Some kits are available:

Asterisk Developer's Kit (TDM) (Includes one FXO card to connect an analog line, and one single-port card to connect an analog phone; PC must run Linux since Asterisk is Linux-

only)

Asterisk Dev Kit PCI (Same as above, but don't understand difference with the TDM version)

Asterisk Developer's Kit (LITE) (Same as above, except the single-port card is replaced with a USB device)

Softphones are software VoIP clients, while IP phones are stand-alone handsets that connect to the LAN through an RJ45 cable without any need for a PC.

Asterisk consists of the following bits and pieces:

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Asterisk: the core project to turn a PC into a PBX

Zaptel: Kernel interface device drivers for Analog and Digital interface cards. If you are using Digium hardware you will need to download zaptel. ("The Zapata Telephony project led

to the development of open source device drivers for a range of telephony interfaces. Zaptel is Digium's revolutionary Pseudo-TDM engine for Linux") Short for "Zapata

Telephony". Zaptel refers to Jim Dixon's open computer telephony hardware driver API. Zaptel drivers were first released for BSD and Jim's Tormenta series of DIY T1 interface

cards. Digium later produced interface cards from Jim's designs and improved the Zaptel drivers on the Linux platform. Digium then added further drivers also following the Zaptel

API for other telephony hardware.

Asterisk-Addons: various modules and addons

Asterisk-Sounds: Default IVR files and miscellaneous sound files

Libiax: Library for the Inter-Asterisk eXchange (IAX) protocol. Libiax provides a foundation for the development of IAX clients

Libpri: Primary Rate ISDN (PRI) library for T1/E1/J1 interfaces

Compact and/or Solid-state Asterisk

Here are the options I found if you want to build yourself a small form-factor Asterisk server. As for using a Compact Flash card to hold the whole system, ie. even logs and voice

messages: "Just make sure your using an industrial compact flash card. These support 1-2 million cycles where many of the retail cards only support 100,000 cycles. We also greatly limit

the logs being generated. Writing logs files creates many times more write cycles than voicemail ever could. If your concerned about logs use syslog to send them to an external system."

Since those hosts don't have much horse-power, avoid asking Asterisk to convert from one sound format to another. "show translation" shows the time needed to convert sound files,

and gives an idea of performance.

To convert between different codec formats can you use the asterisk CLI command:

file convert <file_in.format> <file_out.format>

To convert from a shell script can you do like this:

#!/bin/bash

# Converts a audio file from alaw to a ulaw

rasterisk -x "file convert /tmp/file_in.alaw /tmp/file_out.ulaw"

DLink DIR-825

http://wiki.openwrt.org/toh/d-link/dir-825

Buffalo WZR-HP-G300h

http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h

Asus WL500GP

http://wiki.openwrt.org/toh/asus/wl500gp

Asus WL600G

http://wiki.openwrt.org/toh/asus/wl600g

Netgear WNR3500

PCEngines Alix

http://www.pcengines.ch/alix.htm

Besides the motherboards, you need a CF as main storage device (it is mounted ro on /). You could also use a USB stick to mount /var in rw mode. Obviously you need even a power

supply (sold by Pcengines).

The 2d3/2d13 are very nice for the price.

Intel D201GLY(2)

Intel D201GLY motherboard

IP04 IP-PBX

"The IP04 is a 4 port IP-PBX that runs Asterisk and uClinux on a powerful embedded Blackfin processor. To build an Asterisk IP-PBX you normally need a x86 PC plus a PCI card for the

analog ports. With the IP04 you get all of that functionality in a tiny, low cost, low power, silent box. uCasterisk is in the process of being deprecated. Astfin is the successor to uCasterisk,

and where most Blackfin Asterisk software development is now focused."

Digium Asterisk Appliance

http://linuxdevices.com/news/NS5862403213.html

http://linuxdevices.com/news/NS6530620277.html

Pika Warp

http://www.linuxdevices.com/news/NS3749020799.html

http://blog.tmcnet.com/blog/tom-keating/asterisk/pika-warp-appliance-for-asterisk-review.asp

PIKA WARP Appliance for Asterisk Developers Kit $550

non-discounted list price is $725

Also works with FreeSWITCH

HP/Compaq T5700 XPE

PCI adaptor kit available for this thin client device.

Linksys WRT54 router

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Linux/Asterisk OpenWRT on a Linksys WRTGS54SL router (or Asus WL-500g series). The exisisting implementations of both run very poorly on a non-fpu cpu's, especialy if clock speed <

400 Mhz I have run asterisk (and still do) on mips,ixp and powerpc (all without fpu's) and i think that without modifications the codecs are not so usable

Apparently you can add a SD card interface to a WRT54g relatively easily. You could also use something like a Linksys NSLU2 network storage device with a USB memory stick in it.

Through USB port, add an SD Card/Memory Key/USB Hard Drive. Check the Mitsubishi Diamond R100 (rebranched Asus WL-500g)

Linksys nslu2 NAS

"I've been using a Linksys nslu2 (slug) as a lightweight asterisk server. It isn't a broadband router, but it is cheap and works well. Add a usb flash/hdd"

http://www.voip-info.org/wiki/view/Asterisk+Linksys+NSLU2

http://en.wikipedia.org/wiki/NSLU2

Soekris board

"I have purchased a couple of the soekris net4801 boards and have asterisk up and running on them fine but they just don't quite cut it in the processing power department. I've been

able to get about 10 simultaneous SIP calls with simple ulaw (no encoding decoding). While this might be OK for a very small business or home I just don't think it leaves a lot of overhead

to do anything else.

Perhaps the new Soekris net5501 that is about to be released will help you?"

http://www.voip-info.org/wiki/view/Asterisk+hardware+Soekris

Gumstix

Asterisk on Gumstix SBC

Mini-ITX Via board

About 500E for a full host (motherboard, RAM, IDE hard disk, DVD drive, case).

Via Epia motherboard: I've built several systems based on this motherboard (the 1GHz fanless one) Compressed codecs are fine - as long as you aren't transcoding ;-) I figured I could push

30 non transcoded calls through one, but I've never had the ability to fully test it out. The max. I had going on one system was 20 calls. 5 calls to music on hold (where it 's transcoding

from the GSM moh file to G711 is causing my R&D box (wich has a 533MHz VIA processor with 64Kb cache) is using between 5 and 12% CPU. I'd expect one of my 1Ghz boxes to hardly

notice this at all.Make sure you compile asterisk in i586 mode - it 's in the Makefile in 1.2.x. It'll crash otherwise as the VIA processors are lacking some vital MMX instructions. Boot it off

flash and have it load an initrd.gz into RAM. Everything will run entirely from RAM - no writes to the flash at all! I can get everything inside a 48MB flash drive, but I use 64MB ones which

gives me space to store configs, etc.. (of-course, I make it sound so simple ;-) but I'd already worked this out some years back for a diskless router project). I keep voicemail on a 2nd

flash IDE device mounted as ext2 (not 3 as ext3 writes regularly!)

Get a low end motherboard, like a VIA EPIA that doesn't use much power and a solid state hard drive. a CompactFlash card can be connected to IDE with a simple adapter and used as a

'drive'. It is recommended to store logs and stuff on another flash drive as flash memory wears down over time, this way you don't lose any config files if/as/when it dies.

http://www.limeylinux.org

AstLinux

Moved here.

Askorzia Appliance

From €225

http://shop.askozia.com

Installing AskoziaPBX on a Compact Flash

AskoziaPBX is derived from m0n0wall, is based on FreeBSD and, as of January 2008, runs Asterisk 1.4. Currently, it 's off-limit to any customization, ie. you can't change the dialplan, and

can't even get a login console when booting the CF card. Pretty much all you're allowed to do is add phone extensions to it.

To modify an AskoziaPBX image, you'll need a FreeBSD system to decompress and mount distribution images, and reflash the CF card.

Here's how to install Askozia on a Compact Flash card:

1. Check which drive+partition the CF card uses: fdisk -l

2. # wget http://askozia.com/downloads/pbx-generic-pc-pb12.2.img

3. (On Linux) To copy the image to the CF card, run the following:

gunzip -c pbx-platform-xxx.img | dd of=/dev/hd[x] bs=16k

Note: [x] is either the whole card if you don't have a boot loader (eg. /dev/sdb), or just a partition on the card (eg. /dev/sdb1). If you get a "Boot error", it may mean that you

copied the image to a partition but don't have a boot loader on the card to actually boot the image

4. You can now boot the host with the CF card

5. In the Askozia menu, assign an Ethernet interface, and let the system reboot

6. Back in the Askozia menu, if the IP address doesn't match your network (192.168.1.0/24), assign an IP address and subnet mask so you can access the embedded web server and

configure Asterisk with a browser

7. Aim your browser at http://askozia/, and logon as admin/askozia

8. The first sections you want to check are System > General Setup, System > Interfaces, Phones > SIP

Extra Stuff

Booting Linux from Compact Flash

Step by Step Guide to Creating an Embedded Asterisk Server By Victor Hoodicoff

Configuring a tiny Asterisk system

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www.astlinux.org is designed with running from Compact Flash memory in mind. However, it can also be installed to a regular hard drive like other distributions ("We need some

assistance with documentation"; Page on voip-info.org). Its mailing list is at https://lists.sourceforge.net/lists/listinfo/astlinux-users

LimeyLinux ("A Linux distribution tailored to run Asterisk on VIA Mini-ITX boards and which is small enough to fit on a 128MB or 512MB, or 1GB compact flash card and 512MB of

RAM")

Embedded Asterisk Project - Aussie Made! http://forums.whirlpool.net.au/forum-replies.cfm?t=736924

Asterisk - PXE boot Edition (based on asterisk/zaptel/libpri 1.0.7)

Xorcom, a live CD based on Debian Live

Buy complete fanless system flash card ready unit with four ethernet interfaces: http://www.ibt.ca/v2/items/fwa7204/index.html It is very small, in an aluminium extrusion case,

very robust. Install Astlinux on 128 MB flash card - http://www.asterisk.org Voila! - flash based firewall/asterisk machine capable of running 20+ extension office. I have several

installations like this. About $350.

You might want to check these guys out as well. Their communication appliance boxes are nice and reasonably priced. I don't have any affiliation with them.

http://www.portwell.com

http://www.allo.com/products/micropbx.php ($550; No FXO port)

This guy ran asterisk on a board with a blackfin processor. http://www.rowetel.com . Free Telephony Project

getting an embedded linux chip and choosing flash ram and storage

Linux on some mini macs, eg. 4S Newcom's Blue4S

Resonance networks sell asterisk boxes preconfigured and budled with G 729 codecs. (As of May 2007: "Please check back shortly for products")

LinuxDevices.com

Step by Step Guide to Creating an Embedded Asterisk Server By Victor Hoodicoff

Booting Linux from Compact Flash

miniBSD - reducing FreeBSD (last update: 08/25/2005)

Xorcom

FreeBSD on the Soekris net4501 By Michael R. Brumm

Installing FreeBSD 6.2 on a Soekris net5501

Troubleshooting

If you're having problems with Asterisk, here are some tricks to try:

Running a syslog-ng server, and have Asterisk and all SIP devices send debug information to it

Running a packet sniffer on the Asterisk server: WireShark (ex-Ethereal) is available in both GUI and text-mode versions, and undestands SIP, TCPDump is a text-mode sniffer and

can save dumps that can be analyzed by Wireshark

#192.168.0.253 is a VoIP gateway

tcpdump host 192.168.0.253 not port 56051 #to filter out connections to syslog server

Launch the Asterisk console, and type "sip debug". To turn it off, "sip no debug"

If you're having problems calling into Asterisk from a remote location with firewalls in the way, add an extension that will echo back what you say:

exten => 111,1,Answer()

exten => 111,2,Playback(welcome)

exten => 111,3,Playback(demo-echotest)

exten => 111,4,Echo()

exten => 111,5,Playback(demo-echodone)

exten => 111,6,Playback(vm-goodbye)

exten => 111,7,Hangup()

If you can't hear the voice, make sure that either the IP phone is either set to use STUN, or that you forced it to use a specific UDP port (eg. 5004, and 5005 for a GrandStream

phone) and opened those ports on the router.

Strings to look for when running "sip debug":

Peer audio RTP

We're at

m=audio

"IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by ATA for firewall or NAT penetration in different network environment. If

symmetric NAT is detected, STUN will not work and ONLY outbound proxy will provide solution for it."

Launch asterisk with maximum verbosity "asterisk -vvvvvgd &"

Commands you might need to investigate:

Either launch Asterisk in verbose mode with "-vvvvvvvvvvvv", or type " core set verbose 3" in the CLI

If you have problems with a TDM card, "cat /proc/zaptel/*", and "zap show channels" in the CLI

"lspci -v" to list available PCI cards

"lsmod" to list loaded modules: TDM cards need zaptel, wctdm, and wcfxo

AMI requires two tries to login

Start by hitting the carriage return once, then type the following, and end with another carriage return:

action: Login

username: admin

secret: test

A phone is marked as UNREACHABLE

If Asterisk is sitting behind a NAT router, and the phone is living on the outside, make sure sip.conf tells Asterisk that it 's set up in a private network, and that UDP5060 is statically open

on the router to allow remote phones to connect to Asterisk:

[general]

...

externip = the.router.s.public.address

localnet=192.168.0.0/255.255.0.0

nat=yes

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canreinvite=yes

Hardware not detected

For PCI cards, make sure they're not sharing IRQs with other cards

Check that the power supply is strong enough

Start in runlevel 3, ie. no X

Recompile zaptel and asterisk

Use a more recent PC

Check length of cable between FXO and wall plug

Check signal level: ztmonitor <channel number> -v ("gives you a visual representation of the sound strengths and makes it easy to see if the receive or transmit signals are too high

or out of balance")

try turning off call progress detection

try fxotune

adjust gain levels

Fails compiling zaptel

Check that the kernel sources match the kernel you are running (cat /proc/version to check the kernel version; next, go to /usr/src/ to check which version of the kernel sources

are available, if any)

Check that the version of gcc installed is the same that was used to compile the kernel (cat /proc/version to check which version was used to compile it; gcc -v to check which

version is installed)

Check that all the required dependencies are installed, and can be found

"You may have to do a "make prepare" in the kernel source directory before you compile zaptel"

Fedora 5 ships with a bug in the kernel sources. You should run "yum search kernel", and download (yum update) the latest kernel and kernel sources for your architecture. Make

sure both are for the same architecture: If you mix them (eg. i586 and i686, you'll get an error when loading the zaptel module: "FATAL: Error inserting zaptel

(/lib/modules/2.6.17-1.2139_FC5/misc/zaptel.ko): Invalid module format", along with this error in dmesg: "zaptel: disagrees about version of symbol struct_module".) Reboot with

the new kernel

Next, you'll need to update the /lib/modules/<kernel version>build symlink to point to the new kernel sources, eg. "ln -s /usr/src/kernels/2.6 /lib/modules/`uname -r`build". If you

don't, you'll get an error when compiling zaptel.

More information: http://forums.fedoraforum.org/showthread.php?t=115233, http://forums.digium.com/viewtopic.php?t=7061 and Asterisk Zaptel Installation

Asterisk doesn't hang up FXO calls

Resolving hangup detection problems with fxo cards

Too much echo using an FXO card

"Hint: if you are experiencing problems with echo on your analog calls, you may wish to uncomment the KFLAGS+=-DAGGRESSIVE_SUPPRESSOR line and run make clean; make; make

install -- this enforces a more rapid echo interceptor for analog circuits."

"Failed to initailize [sic] DAA, giving up" with an FXO card

If modeprobing wcfxo says something like..

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

FATAL: Error running install command for wcfxo

... and dmesg says:

Zapata Telephony Interface Registered on major 196

Zaptel Version: 1.2.5 Echo Canceller: MG2

Failed to initailize DAA, giving up...

wcfxo: probe of 0000:00:0e.0 failed with error -5

... this is most likely because the FXO card shares its interrupt with another PCI card. Go into the BIOS, and configure the slot so that it has its own IRQ. Run "lspci -v" to check if another

card is sharing the IRQ.

Besides playing with the BIOS, you can launch Linux with the "pci=noacpi" switch ("interrupt problems are more likely on Linux 2.6 than 2.4 since the ACPI interface is more fully utilised.

Try booting with acpi=off".) When OK, the FXO card should show up in /proc/zaptel/1 (or 2, etc.)

Zaptel can't access the card

If using a Linux distro that uses the udevd daemon to dynamically populate /dev with device nodes, you must add some rules to /etc/udev/rules.d/50-udev.rules.

Freshmaker failed register test

This shows in "dmesg" when trying to use a PCI card + Zaptel. Make sure the PCI card is OK hardware-wise, its golden edge is clean, and the card is well seated in its slot.

If still NOK, try the card on another motherboard.

Caller ID

Note that you need to restart Asterisk to re-configure the Zap channel using "reload chan_zap.so". This will reload the configuration file, but not all configuration options are re-configured

during a reload.

Crappy sound when running echo test

Q&A

Freephonie: How to get rid of "determine_firstline_parts: Bad request protocol Packet"?

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This warning showing in the logs is due to some incompatibility between Asterisk 1.4 and Cirpack telephony servers. There are different ways to solve this:

Upgrade to Asterisk 1.6/1.8

Edit channels/chan_sip.c to comment this part out:

/*

if (strcasecmp(e, "SIP/2.0") ) {

ast_log(LOG_WARNING, "Bad request protocol %s\n", e);

return -1;

}

*/

Use iptables with advanced Netfilter options in the kernel:

# freephonie.net = 212.27.52.5

iptables -I INPUT -p udp --src 212.27.52.5 --dport 5060 -m string --algo bm --string "Cirpack KeepAlive Packet" -j DROP

Edit asterisk/logger.conf thusly:

console => notice,error

Flash() doesn't work

By default, the hook flash is too long for European telcos. You'll need to edit some include files before recompiling Zaptel/Dahdi.

Important: In case you already compiled Zaptel/Dahdi, make sure you start from a clean plate by removing all object files (rm -f *\.o) from its source directory.

Zaptel

zaptel.h

#define ZT_DEFAULT_FLASHTIME 100

zconfig.h

#define SHORT_FLASH_TIME

Dahdi

kernel.h

DAHDI_DEFAULT_FLASHTIME 100

dahdi_config.h.

#define SHORT_FLASH_TIME

After restarting the new Zaptel/Dadhi, run either "ztcfg -vv" or "dahdi_cfg -vv" and "dmesg" to check that it loads OK.

How to reload Zaptela after editing zapata.conf?

CLI> module reload chan_zap.so

Why can't make callback?

Can't figure out why I had to use AGI to use a script to make a callback instead of using this simpler way:

[from_fxo]

exten => s,1,Wait(2)

exten => s,n,Set(CID=${CALLERID(num)})

exten => s,n,Hangup()

;Wait for RINGs to stop

exten => h,1,Wait(10)

;DOESN'T WORK

;exten => h,n,Dial(Zap/1/${CID})

exten => h,n,Dial(Local/start@callback)

[callback]

exten => start,1,Verbose(In callback, CID is ${CID})

exten => start,n,Dial(Zap/1/${CID})

exten => start,n,Hangup

[sip.conf] Peer vs. friend?

This is used to handle incoming calls. The difference between the two is that a "peer" is matched on its IP address (and possibly, its port number) while a friend is identified by its entry

name in sip.conf.

Note that confusingly, "sip show peers" in the CLI shows both peers and friends, and that the third option "user" is pretty much deprecated.

www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

What's the difference between a B2BUA and a register/proxy server?

www.voip-info.org/wiki/view/Asterisk+SIP+not-proxy

OpenSIPS and Asterisk Compared

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Dialing SIP URI's with a keyboard-less telephone?

Calling SIP URI's (ie. sip:user@domain) is easy when using an SIP application running on a computer, but not very practical with regular IP phones.

A solution is to map the URI with a number from eg. Freenum.

Connecting to PSTN/ISDN: PCI or appliance?

There are two options when connecting Asterisk directly to the telephone network, whether through an analog line or an ISDN line:

PCI card inside the Asterisk server: It requires care when upgrading Asterisk, but has the advantage of being open-source so issues can more easily be investigated and fixed

An SIP-capable appliance that connects to Asterisk through the LAN: Makes it easier to upgrade Asterisk, but has the disadvantage of being closed-source

The main brands are: Digium, Patton, Audiocodes, Rhino Equipment, and Sangoma.

How to trim the list of modules that Asterisk loads?

As specified in /etc/asterisk/asterisk.conf, modules are loaded from /etc/asterisk/modules.conf, but the binary might have been compiled to either look for modules elsewhere or (CHECK)

have modules compiled directly in the asterisk binary.

Here's a working modules.conf:

[modules]

autoload=no

; Used applications

load => app_dial.so

load => app_playback.so

load => app_read.so

; Channel drivers

; probably you don't need chan_local, but it 's nice

load => chan_local.so

load => chan_sip.so

; Codecs

load => codec_gsm.so

load => codec_ulaw.so

load => codec_alaw.so

load => codec_a_mu.so

load => format_gsm.so

load => format_pcm.so

; PBX - choose between AEL and .conf files

; load => pbx_config.so

load => pbx_ael.so

; use pbx_spool if you want to drop .call files

load => pbx_spool.so

; Functions

load => func_strings.so

load => func_rand.so

load => func_logic.so

load => func_cut.so

load => func_callerid.so

; Resources

; if you don't have agi, you may remove this

load => res_agi.so

load => res_musiconhold.so

; res_features is required by chan_sip

load => res_features.so

Can Asterisk fetch configuration files dynamically?

Yes, by compiling it with support for Curl:

[general]

static=yes

writeprotect=no

clearglobalvars=yes

[globals]

MYPHONE=SIP/2000

#exec /usr/bin/curl -s http://server/dialplan.txt

This can also be used to retrieve voicemail records. It's easier than using a database, reloading app_voicemail.so, or when removing a voicemail box. All the passwords are updated via the

web server, so there is no need to update / write to the voicemail.conf file.

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In extensions.conf, a script fails to fork

;BAD exten => _[1-4],n,AGI(netcid.py|${CALLERID(num)}|${CALLERID(name)})

exten => _[1-4],n,System(netcid.py ${CALLERID(num)} ${CALLERID(name)})

What are recommended IAX (soft|hard)phones?

Softphones

ZoIPer (ex-Idefisk)

iaxComm ("Update 6 jun 2006")

YakaPhone

DIAX ("Latest update on 22-Jun-2005")

Kiax (Uses GTK?)

Hardphones

AT-530

Can I use SQLite instead of BerkeleyDB?

As of Jan 2008, either use the func_odbc() application, call a script with AGI(), or try the recent Asterisk cmd MYSQL add-on.

Playback(/tmp/myfile.wav) fails

WARNING[622]: file.c:568 ast_openstream_full: File /tmp/myfile.wav does not exist in any format

WARNING[622]: file.c:871 ast_streamfile: Unable to open /tmp/myfile.wav (format 0x4 (ulaw)): No such file or directory

WARNING[622]: app_playback.c:437 playback_exec: ast_streamfile failed on SIP/2000-0868d000 for /tmp/myfile.wav

Remove the file extension:

exten => 1,1,Playback(/tmp/myfile)

Is ztdummy still needed?

Looks like recent versions of Linux no longer need the ztdummy kernel module to provide a software timing source in the absence of a hardware timing source.

In case it 's still needed, and provided the "make install" or "make config" didn't already take care of business:

//to load the thing right away

modprobe ztdummy

//to have it be loaded up at boot time

echo "ztdummy" >> /etc/modules

Which version am I running?

asterisk -V

Record's ${RECORDED_FILE} doesn't work

This channel variable is set by Record() only if you use the %d trick to have it generate a new, sequential filename dynamically:

exten => 888,1,Playback(/root/asterisk/leave_msg)

;BAD exten => 888,n,Record(/tmp/wrong.wav,3,30)

;BAD exten => 888,n,Verbose(Recorded is ${RECORDED_FILE})

exten => 888,n,Record(/tmp/test%d.wav,3,30)

exten => 888,n,Verbose(Recorded is ${RECORDED_FILE})

exten => 888,n,Hangup()

NoOp() is not displaying anything

It apparently depends on the verbosity level. Use Verbose() instead.

WARNING: chan_zap.c: process_zap: Ignoring signalling

This is just due to the fact that chan_zap cannot change the signalling of a channel when typing "reload" in the CLI. This parameter is ignored on reload.

How to force using specific ports for RTP?

1.2.2.: rtp.conf

rtpstart=

Takes a numeric value, which is the first port of the port range that can be used by asterisk to send and receive RTP.

rtpend=

Takes a numeric value, which is the last port of the port range that can be used by asterisk to send and receive RTP.

How can all PC's be notified of a call?

This is useful when either using an IP hardphone that only displays CID numbers (eg. the GrandStream BudgeTone 101), or when users simply don't have an extension and simply want to

see who's calling. A good and free that does just is NetCID, the client-side of the great Identify caller ID application for Windows.

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Here's how to configure Asterisk to broadcast a message when a call comes in, so NetCID can pick it up through its default UDP 42685 (more information):

1. cd /var/lib/asterisk/agi-bin

2. vi ncid.agi, and copy the contents shown above

3. chmod 755 ./ncid.agi

4. Add a reference to it your dialplan, eg. exten => cid,n,AGI(ncid.agi|${CALLERIDNUM}|${CALLERIDNAME})

5. From a remote PSTN phone, call into Asterisk, and check that NetCID pops up

The only issue I found is when using characters > 128, ie. non-basic ASCII characters: Show up OK in NetCID, but not in eg.X-Lite, but it 's an issue with code pages when importing data

into Asterisk to rewrite CID data on the fly through LookupCID().

Here's the Python version of the above Perl ncid.agi script:

#TODO

# - Unicast to list of remote users

# - STDOUT and STDERR

# - Add Perl' open STDOUT, '>/dev/null'; fork and exit; to avoid waiting

#!/usr/bin/python

import socket,sys,time,os

def sendstuff(data):

s.sendto(data,(ipaddr,portnum))

return

#BAD?

#import posix

#posix.close(1)

#posix.open("/dev/null", posix.O_WRONLY)

#BAD?

sys.stdout = open(os.devnull, 'w')

if os.fork():

sys.exit(0)

try:

cidnum = sys.argv[1]

except:

print "Format: netcid.py cidnum cidname"

sys.exit(1)

try:

cidname = sys.argv[2]

except:

print "Format: netcid.py cidnum cidname"

sys.exit(1)

now = time.localtime(time.time())

dateandtime = time.strftime("%d/%m/%y %H:%M", now)

myarray = []

myarray.append("STAT Rings: 1")

myarray.append("RING")

myarray.append("NAME " + cidname)

myarray.append("TTSN Call from " + cidname)

myarray.append("NMBR " + cidnum)

myarray.append("TYPE K")

#First, let's broadcast to the LAN

s = socket.socket(socket.AF_INET,socket.SOCK_DGRAM)

s.setsockopt(socket.SOL_SOCKET,socket.SO_BROADCAST,True)

portnum = 42685

#ipaddr = "192.168.0.255"

ipaddr = "localhost"

for i in myarray:

sendstuff(i)

#Must pause, and send IDLE for dialog box to close and call to be logged in

time.sleep(5)

sendstuff("IDLE " + dateandtime)

#Next, let's unicast to the following remote hosts on the Net

How to (re)write caller ID name on the fly?

1. Add items in Asterisk's embedded database to match a name to a number. If you only have a few entries, you can connect to the Asterisk server in console mode and type eg.

"database put cidname 123456 "My cellphone". If you have a lot of entries, create an executable script to run this type of instructions:

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asterisk -rx 'database put cidname 1234567 "My cellphone"'

asterisk -rx 'database put cidname 7896543 "My home"'

Note: Whatch out for embedded characters in the CID name such as ', ?, !, and extendend ASCII characters like é, è, etc. (non-ASCII chars showed OK in NetCID, but not X-Lite)

2. Edit the context in extensions.conf that handles incoming calls:

[incoming]

exten => group,1,LookupCIDName

exten => group,n,Dial(SIP/200&SIP/201&SIP/202)

3. Reload Asterisk, and call into it.

This works because LookupCIDName looks for numbers in families cidname (and blacklisted numbers from the blacklist family). More information on using the database here.

Here are some useful commands that you can use while in an Asterisk console:

to display all entries for a family: database show cidname

to display the value that matches a key: database get cidname 1234567

to remove all entries in one "family" (ie. group): database deltree cidname

How to update a list of calls on the web?

Here's how to call a PHP script from an AGI Perl script each time a call comes in:

#!/usr/bin/perl

#Save this script as /var/lib/asterisk/agi-bin/web.agi

#use LWP::Simple;

use URI::Escape;

use LWP 5.64;

open STDOUT, '>/dev/null';

#Let Asterisk go back to work and let the script run its life

fork and exit;

my $cidnum = $ARGV[0];

my $cidname = $ARGV[1];

#CID name may contain spaces and other no-no characters

$safe_cidname = uri_escape($cidname);

my $browser = LWP::UserAgent->new;

my $url = "http://www.acme.com/input.php?";

$url .= "name=" . $safe_cidname . "&";

$url .= "number=" . $cidnum . "&";

($min, $hrs, $day, $month, $year) = (localtime) [1,2,3,4,5];

$currentdate = sprintf("%02d/%02d/%02d", $day, $month+1, $year);

$currenttime = sprintf("%02d:%02d", $hrs,$min);

$url .= "date=" . $currentdate . "&";

$url .= "time=" . $currenttime;

my $response = $browser->get( $url );

die "Can't get $url -- ", $response->status_line unless $response->is_success;

And here's how to tell Asterisk to call the script:

exten => group,1,LookupCIDName

;Important: The script must be called before dialing extensions

exten => group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)})

exten => group,n,Dial(${EXT200}&${EXT201})

How to leave console mode without stopping Asterisk?

Just hit CTRL-C or type "exit". "stop now" stops the server as well.

Why two files, zaptel.conf and zapata.conf?

Because Zaptel cards actually predate Asterisk, so are not specifically built to run with this software. Once configured through zaptel.conf, a Zaptel card can be used by any application,

not just Asterisk. Hence the need for zapata.conf to act as a glue between the Zaptel card and Asterisk.

Zaptel hardware vs. Zapata hardware?

"Asterisk uses the zapata.conf file to determine the settings and configuration for telephony hardware installed in the system. The zapata.conf file also controls the various features and

functionality associated with the hardware channels,such as Caller ID, call waiting, echo cancellation, and a myriad of other options.

When you configure zaptel.conf and load the modules,Asterisk is not aware of anything you’ve configured. The hardware doesn’t have to be used by Asterisk; it could very well be used

by another piece of software that interfaces with the Zaptel modules. You tell Asterisk about the hardware and control the associated features via zapata.conf."

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What are devfs and udev?

"In the early days of Linux, the system’s /dev/ directory was populated with a list of devices with which the system could potentially interact. At the time, nearly 18,000 devices were

listed. That all changed when devfs was released, allowing dynamic creation of devices that are active within the system. Some of the recently released distributions have incorporated the

udev daemon into their systems to dynamically populate /dev/ with device nodes." (from "Asterisk, the future of telephony")

Which FXO card?

So-called Zaptel cards, ie. either the X100P or X101P cards, are cheaper because they essentially are so-called voice winmodems, ie. they need a driver to perform most of the work that

real modems normally do themselves through an on-board CPU. They're cheaper, and usable for a personal server, but shouldn't be considered for a professional server.

"The older X100P card used a Motorola chipset,and the X101P (which Digium sold before completely switching to the TDM400P) is based on the Ambient/Intel MD3200 chipset. These

cards are modems with drivers adapted to utilize the card as a single FXO device (the telephone interface cannot be used as an FXS port). Support for the X101P card has been dropped

in favor of the TDM series of cards. Use of these cards (or their clones) is not recommended in production environments."

"If you have a new 3.3v only motherboard then make very sure that the brand that you buy supports this or your system will refuse to boot with the card inserted. A lot of X100P clone

cards have the 3.3v notch in their PCI interface, but do not support 3.3v operation."

Here are the latest infos I gathered on the X100P/X101P cards to explain why a lot of people have problems with this hardware:

The Zaptel driver generates about 1,000 interrupts per second, so it should not be used on slow hardware, and you must make sure the card doesn't share its interrupt with

another PCI card

Analog line is a delicate environment where impedance, tension, and line quality are some of the critical components. The original X100P made by Digium as well as cheap knock-

offs on eBay often use the Silicon Labs DAA chips Si3012/Si3035 which is only good for use in countries that use the FCC standard; Countries that use the CTR21 standard

(Europe, and others) require the Si3014/Si3034 chips, which support global line standards (more information on www.x100p.com)

Echo issue: Using the the open-source OSLEC echo canceller and a bit of hacking could solve this problem. A user reportes: "The OSLEC benchmark tell me it can run 14

concurrent instances on a 550MHz VIA C3 processor. On a 1.6GHz Atom, it tells me it can do 80 concurrent instances"

Stand-alone cards like those from Digium cost ten times more but are much better quality, but even those don't necessarily handle call information OK (caller ID, call progress), at

least in countries other than the US. OpenVox makes cheaper clones of Digium cards.

Most of the clone cards don't support far-end disconnect supervision, so you'll have problems where Asterisk can't tell that the other party has hung up the phone.

FXO? FXS?

"An FXO interface is thus named because it connects to an Office , where as an FXS interface, connects to a Station. The terms "FXO" and "FXS" have their origins in an old telephone

service called Foreign eXchange (FX). The original purpose of an FX circuit was to allow an analog phone at a remote location to be connected to a PBX somewhere else. An FX circuit has

two ends (the Station end, where the telephone is, and the Office end, where the PBX is)."

STUN

STUN (Simple Traversal of UDP over NATs): Used so clients can tell on which ports they're listening behind a NAT firewall. Can be implemented in the firewall itself, or you can build a

STUN server (which just echos back this information to the client).

IAX vs. SIP

"The Inter-Asterisk eXchange (IAX) protocol is usually used for server-to-server communication; more hard phones are available that talk SIP. However,there are several soft phones that

support the IAX protocol,and work is progressing on several fronts for hard phone support in firmware.

The primary difference between the IAX and SIP protocols is the way media (your voice) is passed between endpoints. With SIP,the RTP (media) traffic is passed using different ports

than those used by the signaling methods. For example,Asterisk receives the signaling of SIP on port 5060 and the RTP (media) traffic on ports 10,000 through 20,000, by default. The

IAX protocol differs in that both the signaling and media traffic are passed via a single port: 4569. An advantage to this approach is that the IAX protocol tends to be better suited to

topologies involving NAT."

Resources

Getting help

Asterisk forums (@ Asterisk)

Asterisk mailing-list (@ gmane.org)

Asterisk forums (@ AsteriskGuru)

(French) Asterisk-France

IRC: You can connect to the irc.freenode.net server (Port: 6667) and talk to Asterik users over the #asterisk channel. Note that this Asterisk channel now requires that your nick

be registered with the Freenode Nickerv in order to participate. To do this, type "/msg NickServ register mypassword" in your IRC client to register your password.

Tools

SmartCID ("A php script to replace callerid information with lookup from a local mysql table, and if that fails then reverse phone number lookup from 411.com.")

NetCID

Visual Diaplan

Frequency Analyzer

"WaveSurfer is an open source tool for sound visualization and manipulation. Typical applications are speech/sound analysis and sound annotation/transcription."

Monitoring Asterisk: Nagios, Munin

Positron Telecommunication Systems' "Asterisk PBX on a card" V-114

Books

Asterisk: The Future of Telephony, Second Edition (2007)

Asterisk: The Future of Telephony (2005)

Asterisk Gateway Interface 1.4 and 1.6 Programming

Asterisk 1.4 - The Professional's Guide (sequel to Building Telephony Systems with Asterisk)

Practical Asterisk 1.4 and 1.6: From Beginner to Expert

Sites and articles

Practical Asterisk (2007)

1/22/2014 Set up your own PBX with Asterisk

http://www.fredshack.com/docs/asterisk.html 46/46

The Little Asterisk Handbook by Andy Powell (2005)

Getting Started With Asterisk by Andy Powell (2003)

The Asterisk Handbook (2003)

Asterisk Cookbook (as of Jan 2008, the book as been postponed at O'Reilly, but the site offers some useful tips)

A $10 Linux Answering Machine By Bob Smith

History of Zapata Telephony and how it relates to Asterisk PBX By Jim Dixon

Asterisk Handbook Documentation

(French) Votre propre PBX sous GNU/Linux + Asterisk

Flash Operator Panel (FOP; "Flash Operator Panel is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display

information about your PBX activity in real time.")

Asterisk: A Bare-Bones VoIP Example by John Todd

VoIP and POTS Integration with Asterisk by John Todd

List of standard Asterisk applications

Installation d'Asterisk CVS avec deux cartes X100P par Serge Carpentier et Frédéric Marti

(French) FreePhonie et Asterisk, Installer Asterisk sous Linux

Bunch of e-books on Asterisk and VoIP

http://kb.digium.com (including How do I install the Zaptel drivers on a system running Fedora Core?, What do I do if I can't compile Zaptel package on my system ?)

"MozIAX is a Firefox VoIP extension, a cross platform software IAX2 phone (softphone) to be used with Asterisk, the Open Source PBX. Like Asterisk, MozIAX is free software."

Asterisk The software phone company By John Goerzen

How to set up VoIP calls in your Cybercafé with Xorcom Rapid, Asterisk PBX and *starShop-OSS by sylverboss

Fedora Core 5 Installation Guide

Fedora Core 5 Linux Installation Notes

Initial investigations on hooking up Skype to SIP with Asterisk: NCH Uplink, ChanSkype, and PSGw

Various tones used in national networks (according to itu-t recommendation e.180)(03/1998)

Free, SIP et Asterisk by Samuel Tardieu

OpenVox (telephony cards)

ZapMicro (telephony cards)

SIP and NAT - what is the problem, really?

Asterisk PCI bus Troubleshooting

Asterisk Gateway Interface Scripting with PHP

Temp

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf

http://www.automated.it/guidetoasterisk.htm#_Toc49248767

http://mobile.newsforge.com/print.pl?sid=06/04/06/1950254

http://faq.sipbroker.com/tiki-index.php?page=Asterisk%20Configuration

http://forum.gizmoproject.com/viewtopic.php?t=1059&start=0&postdays=0&postorder=asc&highlight=

http://www.onlamp.com/lpt/a/3956

http://www.vonmag.com/issue/2004/julaug/columns/isenberg.asp

http://www.voip-forum.com/?p=131&more=1

VoIPowering Your Office with Asterisk – Part 1 and 2: Introduction

VoIPowering your Office with Asterisk—Building a Test Lab By Carla Schroder: Part 1, 2, and 3

ISP-In-A-Box: Installing a Free Asterisk PBX Phone System (Part I)

Nerd Vittles (lots of tutorials)

TriBox (ex-Asterisk@Home)

Install a SIP softphone, and dial out through Zap/2

http://www.experts-exchange.com/Networking/VoIP_Voice_over_IP/Q_21820311.html

http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html

CLI > zap show channels

<lesouvage> joe_acme: it must be something like exten => s,n,DIAL(ZAP/g1/mobilephonenumber,15,r) in the extensions that handles the incoming call. You have to group the two fxo

ports into one group to make this work.

ring groups

hunt groups

To see errors produced by the modprobe command, use the command dmesg. Other helpful error related information is avalable in any of the files created in the directory /proc/zaptel.

Thiscommand, an these files, can help you diagnose errors in the zaptel configuration process, for example boards tha have not been provided with power or drivers that are loading in the

wrong order.

The program ztcfg reads the configuration information in zaptel.conf and configures the drivers.You must run ztcfg each time zaptel driver are loaded, for example after booting the

machine.You can run ztcfg after you have made any changes to zaptel.conf to reconfigure the drivers.

http://forums.digium.com/viewtopic.php?t=7356