Software Phonepatch for the Asterisk PBX
Arnau Sánchez
EHAS Foundation
Copyright © 2006 Arnau Sanchez
29 April 2006
Revision History
Revision 0.1	25/05/2006 Introduction		
	
Revision 0.2	31/05/2006 Hardware, Asterisk configuration, Phonepatch configuration, Tests		

Table of Contents
1. Introduction	000
2. Hardware	000
3. Configuration	000
4. Outcall daemon	000
5. Testing the phonepatch	000
A. GNU Free Documentation License	000
Permission is granted to copy, distribute and/or modify this document under the terms of the GNU Free Documentation License, Version 1.2 or any later version published by the Free Software Foundation; with no Invariant Sections, no Front-Cover Texts, and no Back-Cover Texts. A copy of the license is included in the section entitled "GNU Free Documentation License".
1. Introduction
A phonepatch is a device that allows radio stations to make phone calls. On the past, that could only be achieved with a hardware device, connected to the PSTN (Public Switched Telephone Network) and with a radio transceiver. Nowadays, with the arise of software PBX, the versatility of a phonepatch is not constrained to hardware, and can be integrated as part of the software. This radio-phonepatch has been designed to work together with the Open-Source Asterisk PBX on a GNU/Linux operating system.
Figure 1. Hardware phonepatch

Figure 2. Software Asterisk-phonepatch

The phonepatch has been designed to work on centralized networks, so the phone link can be used for more than one radio station. The only limitation is, of course, that just one simultaneous talk is possible. In those networks, each radio-user will have its own Asterisk extension and will be able to receive and make (anoymously or not) phone calls
Figure 3. Typical network using an Asterisk phonepatch

1.1. Features
•	Incoming calls (from phoneline or VoIP links) to radio users.
•	Outgoing calls (from radio to phoneline or VoIP links)
•	Multiple phonepatch instances on the same computer (using each one a different sound card, interface board and radio transceiver)
•	Audio language configurable, available english and spanish.
•	Use festival for text-to-speech.
•	Phone, Radio and Festival audio-gain configurable.
•	Radio users can control phonepatch with DTMF tones.
•	CTCSS authentification
•	VOX (Voice Operated Transmitter) processing for automatic PTT keying (phoneline->radio)
•	Compatible with HF and VHF/UHF transceivers (with carrier detection if available)
•	Very tolerant DTMF decoding.
•	PTT control configurable: minimum on time, maximum on time, penalty for maximum_on_time reached.
•	Uses soundcard to send and receive voice.
•	Serial/parallel port (plus extenal apps) to interface PTT and carrier detection.
•	Phonepatch audio messages (incoming, outcoming call, errors...) are configurable. All audio formats (WAV, AU, GSM, PCM, ...) can be used.
•	Non-anonymous outgoing calls
1.2. Asterisk interface
There are several mechanism to interface an application with Asterisk. Probably the most obvious is to create a new Asterisk channel library (chan_phonepatch.so). This solution garranties full-integration, but needs to patch the Asterisk code directly, which means lots for maintenance work and incompatability issues between versions.
Another option is that the phonepatch replicates a SIP Phone. This option is interesting, but not easy to implement. SIP protocol is not trivial, although there are some SIP frameworks available. For example, for Python we have the (abandoned?) shttom project (http://shtoom.divmod.org/) Interesting for future implementations.
The easiest mechanism to interface Asterisk, and the used in the present version of the Asterisk-Phonepatch, is the AGI (Asterisk Gateway Interface). However, the AGI do not provide directly the bidirectional audio we need: the special EAGI mode (Extended AGI) opens the file descriptor number 3 with the running API, wheren application can read the received audio directly in raw format (8000sps/16 bits/little-endian), but it does not provide a direct way to send audio to the asterisk link. This issue is solved with a simple trick, using the AGI command GET DATA (STREAM_FILE works fine also) to play an audio file. This audio file is not a normal one, but a pipe that where the phonepatch writes the audio to send. Asterisk didn't notice the difference and works without any apreciable latency.
2. Hardware
Although the phonepatch is esentially software, the interface between radio transceiver and the computer requires still some hardware, which involves sound, PTT (Pust-to-Talk) activation and carrier detection (used only for VHF and UHF transceivers).
2.1. Audio
The easiest way to interface voice with a computer is undoubtely using the soundcard as cheap D/A & A/D converter. It has no special needings, except for, of course, that must be compatible with GNU/Linux. The 2.6.x kernel series brings drivers from the ALSA project (hyyp://www.alsa-project.org), an excellent software that covers eventually any existing soundcard. Refer to its web for more details.
Check the soundcard you have installed, and its id-number:
# cat /proc/asound/cards

0 [ICH            ]: ICH4 - Intel ICH
                     Intel ICH with STAC9752/53 at 0xdfebfe00, irq 16
The phonepatch uses the OSS (Open Sound System) library to access the soundcard, so make sure you have also the module snd-pcm-oss loaded, apart from the ones needed for your soundcard:
modprobe snd-pcm-oss
With the OSS library, soundcards are accesed via device files c /dev/dsp (device 0), /dev/dsp1 (device 1), /dev/dsp2 (device 2), and so on. Before trying to run the phonepatch, make sure that your soundcard is not used by any other process:
# lsof /dev/dsp
On the radio port (usually at the back) you must be find the transmission (Tx) and received (Rx) audio pins, and connect them to, respectively, soundcard's speaker and line-in (or microphone).
Figure 4. Soundcard to radio connection

Make sure that the mixer values are correctly configured (use the alsamixer or rexima): Line In (or Mic) must be selected for recording. Volume recording depends on line InGain. Test that the soundcard output and recording work with utilities that use OSS devices, as rawplay and rawrec (both on rawrec debian package)
2.2. Radio control
2.2.1. PTT
By default, radios are in receiving state, so you have to indicate when to transmit. Note that this is also true for full-duplex radio (which can transmit and receive at the same time). The line responsible for this task is the PTT (Push-to-Talk), usually activated at low-level (Ground) and deactivated at high-impedancy.
There are many ways to interface the PTT, the phonepatch provides applications for the two most common alternatives:
•	Serial
Use a serial port (pin RTS) to set/unset PTT.
Figure 5. PTT signalling using a serial port

•	Parallel
Use the parallel port (pin DATA0) to set/unset PTT.
Figure 6. PTT signalling using a parallel port

These schemes have been taken from the Soundmodem (http://www.baycom.org/~tom/pcf/ptt_circ/ptt.html) page. Soundmodem is an interesenting project that uses soundcards as modem for digital transmission, so the hardware involved is similar to what we need for building the phonepatch hardware.
You must find the PTT (located usually at the rear part of the radio) and connect it to your serial or parallel port depending on what solution is best for you.
After that, you should test that PTT works well, using the utility radiocontrol:
radiocontrol -v -m serial -d /dev/ttyS0 -p on
radiocontrol -v -m parallel -d /dev/parport0 -p off
radiocontrol -v -m command -d eboard -o "unset ptt, set ptt,," -p on
2.2.2. Carrier detection
UHF/VHF transceivers work usually with FM modulations, which means that they have a way to determine if a radio is receiving audio or not: that's called carrier detection. When using such transceivers, apart from the PTT signalling the phonepatch must poll the state of that signal, to decide whether the audio received comes from a remote perr or not. If detected, this audio will be sent to the phone link.
As for the PTT, the carrier detection is made with the serial and parallel port:
•	Serial
Use a serial port (pin DCD) to get carrier detection state
Figure 7. Carrier detection using a serial port

•	Parallel
Use a parallel port (pin Busy) to get carrier detection state
Figure 8. PTT signalling using a parallel port

After that, you should test that carrier detection works well, using as well the utility radiocontrol:
radiocontrol -v -m serial -d /dev/ttyS0 -c
radiocontrol -v -m parallel -d /dev/parport0 -c
radiocontrol -v -m command -d eboard -o ",,get carrier,get carrier: (1|0)" -c

2.2.3. Device lines fully configurable
Although lines used by default on these board are pretty common (RTS, DCD, DATA0, ...), your cable may use others for PTT keying and carrier detection (and power needed for carrier detection). For this reason, all lines (both serial and parallel port) are fully configurable with the option --device-lines pttline,carrierdetectionline,powerline. In this example we use DTR for PTT, DCD for carrier detection and RTS for power:
radiocontrol -v -m serial -d /dev/ttyS0 --device-lines dtr,dcd,rts -p on
radiocontrol -v -m parallel -d /dev/parport0 --device-lines data,paperout,autofeed -c

3. Configuration
3.1. Asterisk modules
Make sure that the modules res_agi.so and format_sln.so are enabled on /etc/asterisk/modules.conf:
...
; Formats
load => format_sln.so
...
; Channels
load => res_agi.so
...
3.2. Groups permissions
Normally, Asterisk is run as asterisk user. Then, the phonepatch must also set its UID to asterisk, in other case it cannot be stopped and continued (using signals) from the asterisk process. As the phonepatch will load as asterisk user, we must assure that it has access to all hardware involved: soundcard, serial/parallel ports:
# ls /dev/dsp* /dev/ttyS* /dev/parport*
crw-rw----  1 root audio 14,  3 2006-07-04 09:33 /dev/dsp
crw-rw----  1 root dialout 4, 64 2006-07-04 11:32 /dev/ttyS0
crw-rw----  1 root lp 4, 64 2006-07-04 11:32 /dev/parport0

In that case, you should do:
usermod -G asterisk,audio,dialout,lp asterisk
3.3. Asterisk extensions
On the Asterisk extensions file configuration (/etc/asterisk/extensions.conf) we have to add all the radio stations for which we want to enable the phonepatch. Although not really necessary, it's recomendable that yout create a new context (for example: radio) for these special users.
On this example, we will asign extensions for PuertoSaija (151) and Munichis (152).
[radio]

; Phonepatch
exten => 151,1,Answer()
exten => 151,2,EAGI(phonepatch.agi|puertosaija)
exten => 151,3,Hangup

exten => 152,1,Answer()
exten => 152,2,EAGI(phonepatch.agi|munichis)
exten => 152,3,Hangup

This extensions are used in both ingoing and outgoing calls. If you plan to use anonymous calls (where the extension is only for ingoing calls), it's not necessary to add another extension. For simplicity, use the same context that you choose for the incoming calls:
[radio]

...

; Static outgoing extension
exten => 160,1,Answer()
exten => 160,2,EAGI(phonepatch.agi|outcall)
exten => 160,3,Hangup
3.4. Phonepatch configuration
Configuration file is place by default at /etc/asterisk/phonepatch. Global parameters and included in [general] section, options regarding a phonepatch (soundcard, PTT/Carrier) must be included in a [phonepatchname] section. Numeric sections are reserved to the phonepatch extensions.
When the phonepatch needs to get the value of a parameter, it searchs the configuration file in that order:
•	Extension
•	Phonepatch definition
•	General
•	Default values (defined by template: /usr/share/asterisk-phonepatch/phonepatch.conf.template)
If you need to define more than one phonepatch, you have to specify another soundcard and PTT/Carrier interface. Example using 2 phonepatchs:
; Asterisk Phonepatch configuration

[phonepatch1]

soundcard_device=/dev/dsp
soundcard_samplerate=8000
soundcard_latency=0.01
ptt=off
carrier_detection=off
radio_control=serial[dtr,dsr,rts]:/dev/ttyUSB0
outcall_daemon=enabled

[phonepatch2]

outcall_daemon=off
soundcard_device=/dev/dsp1
soundcard_samplerate=8000
soundcard_latency=0.05
ptt=off
carrier_detection=off
radio_control=serial[dtr,dsr,rts]:/dev/ttyUSB1

[general]

language="es"
sounds_dir="/usr/share/asterisk-phonepatch/sounds/"
spool_dir="/var/spool/asterisk/outgoing/"

dtmf_noisy_mode_button=
dtmf_sensibility=1.0

festival_audio_gain=5.0
telephony_audio_gain=1.0
radio_audio_gain=1.0

carrier_polling_time=0.5
carrier_threshold_signal=0.1
carrier_tail_time=2
carrier_max_time=20
carrier_wait_time=5
full_duplex=off

ptt_threshold_signal=0.01
ptt_tail_time=2
ptt_max_time=30
ptt_wait_time=5
ptt_txdelay=0.5

ctcss_tx_amplitude=0.001

hangup_button=#
end_audio=php_endcall.ogg
call_limit=600

incall_report_audio=@%d
incall_report_audio_wait=5
incall_report_timeout=60
incall_report_timeout_audio=php_notanswered.ogg

incall_answer_mode=dtmf
incall_answer_button=*

outcall_channel = IAX2/guest:violin@localhost

outcall_context=metadistro
outcall_timeout=10
outcall_priority=1

askfortone_button="*"
tone_audio=php_tone.ogg
tone_audio_time=3
tone_timeout=20
tone_timeout_audio=php_timeoutcall.ogg
outcall_button="*"
clear_button="#"

ring_audio=php_ring.ogg
ring_audio_time=1
ring_audio_wait=3
ring_timeout_audio=php_ringtimeout.ogg

ctcss_tx=
ctcss_rx=
outcall=on
incall=on
name=default
phonepatch=phonepatch1

outcall_askfortone_mode=dtmf
outcall_dmtf_extension_mode=on

[puertosaija]
name="estacion puertosaija"
ctcss_tx=1Z
ctcss_tx_amplitude=0.02
outcall_dtmf_id=1
outcall_extension=151
incall_answer_mode=ctcss
phonepatch=phonepatch2

[munichis]
name="estacion munichis"
outcall_dtmf_id=2
outcall_extension=152
ctcss_rx=200.0
ctcss_tx_amplitude=0.02
ctcss_tx=150.0
As base configuration file, you can also use the example located at /usr/share/asterisk-phonepatch/phonepatch.conf.example. Copy it to /etc/asterisk/phonepatch.conf and costumize it. Take a look on the manual page (man phonepatch.conf) to see a detailed explanation of all the variables. Default values are good for most situations, but take care on the following ones:
•	ptt
On most cases you will have to enable this options to turn PTT on when sending audio to the radio.
•	radio_control
Use serial:device, parallel:device or command:external_command. Valid values are for example serial:/dev/ttyS0, parallel:/dev/parport0 or command: /usr/sbin/external-app-to-interface-board.
When using parallel port, make sure that the module ppdev is loaded.
The command option runs an external command which is supposed to accept commands from the stdin and returning its output through stdout. When using the command option, you have to fill ptt_on, ptt_off, get_carrier and get_carrier_response parameters. The first three are the strings that would be sent to process to turn ptt of, turn it off, and get carrier state. The last one, get_carrier_response, expects a special regular expression: let's imagine that the external application returns "get carrier: on" to say there is carrier and "get carrier: off" if there is not. In that case you would have to set get_carrier_response parameter to "get carrier: (on|off)".
If your board don't use the standard lines, you can set the ones you have that way (device_name[ptt,carrierdetection,power]:device). Examples:
serial[dtr,dcd,rts]:/dev/ttyS0
parallel[data0,paperout,autofeed]:/dev/parpot0
Use the negative prefix ("-") to logically negate a line.
•	carrier_detection
It prevents PTT activation when the carrier is detected. Must be enabled for VHF/UHF transceivers, but of course only if your interface board is able to read the carrier_detection state, otherwise the phonepatch won't work. Sometimes enabling carrier_detection is optional, but in other cases is compulsory, because the radio outputs noise on the Rx audio signal (noise that the phone user will listen!). Motorola transceivers are known to behave that way.
If carrier_detection is enabled, the carrier_polling_time applies. This is the time that the phonepatch waits between succesive looks to carrier state. Use small values carefully as it is likely to load the CPU. 0.5 seconds should be enough for most cases.
•	full_duplex
You should enable this parameter only if you have a full duplex transceiver (notice that it will only work if carrier_detection is also enabled). Notice that most transceivers are half-duplex and this options should remain disabled.
•	outcall_extension
Here you specify the asterisk extension used for outgoing calls. For anonymous calls set outcall_extension_mode=static and simply give the extension used. For outcall_extension_mode=ctcss, write the specifica asterisk extension inside the station's configuration. For outcall_extension_mode=dtmf, add a outcall_dtmf_id (different for each extension) to allow the phonepatch to identify the radio user. If you want to append this id to the final asterisk extension, enable the outcall_extension_append_id option inside the phonepatch extension's section).
•	ptt_threshold_signal
Phones don't have a PTT, so we need to guess when the phone user is talking or not, and activate the PTT automatically (this is called a VOX mechanism). The ptt_threshold_signal is the parameter you need to adjust depending on your phoneline and soundcard levels. Maximum received power is the unity (1.0), so values around 0.05-0.2 should work in most scenarios.
•	dtmf_noisy_mode_button
It has been implemented for very noisy links (i.e. HF) where a tone can be decoded accidentally more than once. With this DTMF button you indicate that you are going to push next number. Example (using #): user pressed "1#2#3#4#", but the number was decoded, due to noise, as 1111#2#333333###4444##; however, this will be correctly interpreted as 1234. Therefore, you can repeat as many times as you want a number to assure it's decoded, and then press noisy_mode_button (also so the times you want).
•	outcall_channel
That's the channel used on outgoing calls. Example: IAX2/user:password@IP
4. Outcall daemon
Once configured, incoming calls should work without doing anything else, since Asterisk will launch automatically the EAGI script when a call for a phonepatch extension is received. However, to enable outgoing calls we need that the phonepatch listens to the channel, detect DTMF tones from the user and make calls when necessary. To enable the daemon mode, edit the main configuration file /etc/asterisk/phonepatch.conf and set outcall_daemon=on inside all the phonepatches sections you want to enable.
After a /etc/init.d/asterisk-phonepatch restart, you sould see the process on a ps and the pidfile for them:
# init.d/asterisk-phonepatch start
Starting Asterisk Phonepatch daemons: asterisk-phonepatch

# ps awx |grep phonepatch
19119 ?        S      0:00 /usr/bin/python /usr/sbin/asterisk-pho ....
19412 ?        S      0:00 /usr/bin/python /usr/sbin/asterisk-pho ....

# cat /var/run/asterisk/phonepatch.pid
19119

# cat /var/run/asterisk/phonepatch2.pid
19412
Daemon info is written to /var/log/daemon.log, check this file to see what is its state:
# tail -f /var/log/daemon.log
Jun  9 14:41:41 localhost asterisk-phonepatch[6561]: soundcard opened: /dev/dsp1 (8000 sps)
Jun  9 14:41:41 localhost asterisk-phonepatch[6561]: radio control opened: command:eboard -p
Jun  9 14:41:41 localhost asterisk-phonepatch[6564]: creating pidfile: /var/run/asterisk-phonepatch.pid
Jun  9 14:41:41 localhost asterisk-phonepatch[6564]: waiting for askfortone_button
When debugging, you can start phonepatch daemon directly from the console (in this mode only one can be active, select it with the -p option).
# init.d/asterisk-phonepatch stop
Stopping Asterisk Phonepatch daemons:
 phonepatch: done

# asterisk-phonepatch -p phonepatch1
phonepatch -- daemon -  -- start of state: daemon
phonepatch -- daemon -  -- soundcard opened: /dev/dsp (8000 sps)
phonepatch -- daemon -  -- radio control opened: command:eboard -ep
phonepatch -- daemon -  -- creating pidfile: /var/run/asterisk/phonepatch1.pid
5. Testing the phonepatch
Connect to the running Asterisk with with a high-level verbose:
asterisk -dr -vvvvvvvvvvvvvvv
5.1. Incoming calls
•	From a telephone, call a extension defined in the phonepatch.
•	On the telephone link and the user radio, a message like "Radio call for _station_" (defined on report_call_audio) will be heard. Station name is syntetized with Festival. You can create your own audio files to hear a more friendly voice.
•	Press answer_button on the microphone of the user radio.
•	Now it should be possible to talk in both directions
5.2. Outgoing calls
•	From the user radio press the button askfortone_button, a tone should be heard for tone_audio_time seconds.
•	When the tone audio finishes, dial the destination number and finally press call_button, the call should now start..
•	While asterisk is trying to stablish the call, a ring audio will be listened (ring_audio) on the radio.
•	If the call is not answered, the (ring_timeout_audio) will be listened and the communication will be closed.
•	If the call is answered, the phonepatch opens the bridge, and it should be possible to talk in both directions
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GNU FDL Modification Conditions
	A.	Use in the Title Page (and on the covers, if any) a title distinct from that of the Document, and from those of previous versions (which should, if there were any, be listed in the History section of the Document). You may use the same title as a previous version if the original publisher of that version gives permission.
	B.	List on the Title Page, as authors, one or more persons or entities responsible for authorship of the modifications in the Modified Version, together with at least five of the principal authors of the Document (all of its principal authors, if it has fewer than five), unless they release you from this requirement.
	C.	State on the Title page the name of the publisher of the Modified Version, as the publisher.
	D.	Preserve all the copyright notices of the Document.
	E.	Add an appropriate copyright notice for your modifications adjacent to the other copyright notices.
	F.	Include, immediately after the copyright notices, a license notice giving the public permission to use the Modified Version under the terms of this License, in the form shown in the Addendum below.
	G.	Preserve in that license notice the full lists of Invariant Sections and required Cover Texts given in the Document's license notice.
	H.	Include an unaltered copy of this License.
	I.	Preserve the section Entitled "History", Preserve its Title, and add to it an item stating at least the title, year, new authors, and publisher of the Modified Version as given on the Title Page. If there is no section Entitled "History" in the Document, create one stating the title, year, authors, and publisher of the Document as given on its Title Page, then add an item describing the Modified Version as stated in the previous sentence.
	J.	Preserve the network location, if any, given in the Document for public access to a Transparent copy of the Document, and likewise the network locations given in the Document for previous versions it was based on. These may be placed in the "History" section. You may omit a network location for a work that was published at least four years before the Document itself, or if the original publisher of the version it refers to gives permission.
	K.	For any section Entitled "Acknowledgements" or  "Dedications", Preserve the Title of the section, and preserve in the section all the substance and tone of each of the contributor acknowledgements and/or dedications given therein.
	L.	Preserve all the Invariant Sections of the Document, unaltered in their text and in their titles. Section numbers or the equivalent are not considered part of the section titles.
	M.	Delete any section Entitled "Endorsements". Such a section may not be included in the Modified Version.
	N.	Do not retitle any existing section to be Entitled "Endorsements" or to conflict in title with any Invariant Section.
	O.	Preserve any Warranty Disclaimers.
If the Modified Version includes new front-matter sections or appendices that qualify as Secondary Sections and contain no material copied from the Document, you may at your option designate some or all of these sections as invariant. To do this, add their titles to the list of Invariant Sections in the Modified Version's license notice. These titles must be distinct from any other section titles.
You may add a section Entitled "Endorsements", provided it contains nothing but endorsements of your Modified Version by various parties--for example, statements of peer review or that the text has been approved by an organization as the authoritative definition of a standard.
You may add a passage of up to five words as a Front-Cover Text, and a passage of up to 25 words as a Back-Cover Text, to the end of the list of Cover Texts in the Modified Version. Only one passage of Front-Cover Text and one of Back-Cover Text may be added by (or through arrangements made by) any one entity. If the Document already includes a cover text for the same cover, previously added by you or by arrangement made by the same entity you are acting on behalf of, you may not add another; but you may replace the old one, on explicit permission from the previous publisher that added the old one.
The author(s) and publisher(s) of the Document do not by this License give permission to use their names for publicity for or to assert or imply endorsement of any Modified Version.
A.6. COMBINING DOCUMENTS
You may combine the Document with other documents released under this License, under the terms defined in section 4 above for modified versions, provided that you include in the combination all of the Invariant Sections of all of the original documents, unmodified, and list them all as Invariant Sections of your combined work in its license notice, and that you preserve all their Warranty Disclaimers.
The combined work need only contain one copy of this License, and multiple identical Invariant Sections may be replaced with a single copy. If there are multiple Invariant Sections with the same name but different contents, make the title of each such section unique by adding at the end of it, in parentheses, the name of the original author or publisher of that section if known, or else a unique number. Make the same adjustment to the section titles in the list of Invariant Sections in the license notice of the combined work.
In the combination, you must combine any sections Entitled "History" in the various original documents, forming one section Entitled "History"; likewise combine any sections Entitled "Acknowledgements", and any sections Entitled "Dedications". You must delete all sections Entitled "Endorsements".
A.7. COLLECTIONS OF DOCUMENTS
You may make a collection consisting of the Document and other documents released under this License, and replace the individual copies of this License in the various documents with a single copy that is included in the collection, provided that you follow the rules of this License for verbatim copying of each of the documents in all other respects.
You may extract a single document from such a collection, and distribute it individually under this License, provided you insert a copy of this License into the extracted document, and follow this License in all other respects regarding verbatim copying of that document.
A.8. AGGREGATION WITH INDEPENDENT WORKS
A compilation of the Document or its derivatives with other separate and independent documents or works, in or on a volume of a storage or distribution medium, is called an "aggregate" if the copyright resulting from the compilation is not used to limit the legal rights of the compilation's users beyond what the individual works permit. When the Document is included in an aggregate, this License does not apply to the other works in the aggregate which are not themselves derivative works of the Document.
If the Cover Text requirement of section 3 is applicable to these copies of the Document, then if the Document is less than one half of the entire aggregate, the Document's Cover Texts may be placed on covers that bracket the Document within the aggregate, or the electronic equivalent of covers if the Document is in electronic form. Otherwise they must appear on printed covers that bracket the whole aggregate.
A.9. TRANSLATION
Translation is considered a kind of modification, so you may distribute translations of the Document under the terms of section 4. Replacing Invariant Sections with translations requires special permission from their copyright holders, but you may include translations of some or all Invariant Sections in addition to the original versions of these Invariant Sections. You may include a translation of this License, and all the license notices in the Document, and any Warranty Disclaimers, provided that you also include the original English version of this License and the original versions of those notices and disclaimers. In case of a disagreement between the translation and the original version of this License or a notice or disclaimer, the original version will prevail.
If a section in the Document is Entitled "Acknowledgements", "Dedications", or "History", the requirement (section 4) to Preserve its Title (section 1) will typically require changing the actual title.
A.10. TERMINATION
You may not copy, modify, sublicense, or distribute the Document except as expressly provided for under this License. Any other attempt to copy, modify, sublicense or distribute the Document is void, and will automatically terminate your rights under this License. However, parties who have received copies, or rights, from you under this License will not have their licenses terminated so long as such parties remain in full compliance.
A.11. FUTURE REVISIONS OF THIS LICENSE
The Free Software Foundation may publish new, revised versions of the GNU Free Documentation License from time to time. Such new versions will be similar in spirit to the present version, but may differ in detail to address new problems or concerns. See http://www.gnu.org/copyleft/.
Each version of the License is given a distinguishing version number. If the Document specifies that a particular numbered version of this License "or any later version" applies to it, you have the option of following the terms and conditions either of that specified version or of any later version that has been published (not as a draft) by the Free Software Foundation. If the Document does not specify a version number of this License, you may choose any version ever published (not as a draft) by the Free Software Foundation.
A.12. ADDENDUM: How to use this License for your documents
To use this License in a document you have written, include a copy of the License in the document and put the following copyright and license notices just after the title page:
Sample Invariant Sections list
Copyright (c) 2006 Arnau Sanchez Permission is granted to copy, distribute and/or modify this document under the terms of the GNU Free Documentation License, Version 1.2 or any later version published by the Free Software Foundation; with no Invariant Sections, no Front-Cover Texts, and no Back-Cover Texts. A copy of the license is included in the section entitled "GNU Free Documentation License".
If you have Invariant Sections, Front-Cover Texts and Back-Cover Texts, replace the "with...Texts." line with this:
Sample Invariant Sections list
with the Invariant Sections being LIST THEIR TITLES, with the Front-Cover Texts being LIST, and with the Back-Cover Texts being LIST.
If you have Invariant Sections without Cover Texts, or some other combination of the three, merge those two alternatives to suit the situation.
If your document contains nontrivial examples of program code, we recommend releasing these examples in parallel under your choice of free software license, such as the GNU General Public License, to permit their use in free software.