[asterisk-commits] branch group/autoconf_and_menuselect r18300 -
/team/group/autoconf_and_menuse...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Apr 7 08:40:13 MST 2006
Author: russell
Date: Fri Apr 7 10:40:08 2006
New Revision: 18300
URL: http://svn.digium.com/view/asterisk?rev=18300&view=rev
Log:
add doc files that didn't get automatically merged into this branch so that they
don't get removed when the branch is merged into the trunk.
Added:
team/group/autoconf_and_menuselect/doc/ael.txt (with props)
team/group/autoconf_and_menuselect/doc/app-sms.txt (with props)
team/group/autoconf_and_menuselect/doc/asterisk-conf.txt (with props)
team/group/autoconf_and_menuselect/doc/backtrace.txt (with props)
team/group/autoconf_and_menuselect/doc/billing.txt (with props)
team/group/autoconf_and_menuselect/doc/callingpres.txt (with props)
team/group/autoconf_and_menuselect/doc/cdrdriver.txt (with props)
team/group/autoconf_and_menuselect/doc/chaniax.txt (with props)
team/group/autoconf_and_menuselect/doc/channels.txt (with props)
team/group/autoconf_and_menuselect/doc/channelvariables.txt (with props)
team/group/autoconf_and_menuselect/doc/cliprompt.txt (with props)
team/group/autoconf_and_menuselect/doc/configuration.txt (with props)
team/group/autoconf_and_menuselect/doc/cygwin.txt (with props)
team/group/autoconf_and_menuselect/doc/dundi.txt (with props)
team/group/autoconf_and_menuselect/doc/enum.txt (with props)
team/group/autoconf_and_menuselect/doc/extconfig.txt (with props)
team/group/autoconf_and_menuselect/doc/externalivr.txt (with props)
team/group/autoconf_and_menuselect/doc/freetds.txt (with props)
team/group/autoconf_and_menuselect/doc/h323.txt (with props)
team/group/autoconf_and_menuselect/doc/hardware.txt (with props)
team/group/autoconf_and_menuselect/doc/ices.txt (with props)
team/group/autoconf_and_menuselect/doc/jitterbuffer.txt (with props)
team/group/autoconf_and_menuselect/doc/linkedlists.txt (with props)
team/group/autoconf_and_menuselect/doc/math.txt (with props)
team/group/autoconf_and_menuselect/doc/misdn.txt (with props)
team/group/autoconf_and_menuselect/doc/mp3.txt (with props)
team/group/autoconf_and_menuselect/doc/musiconhold-fpm.txt (with props)
team/group/autoconf_and_menuselect/doc/mysql.txt (with props)
team/group/autoconf_and_menuselect/doc/odbcstorage.txt (with props)
team/group/autoconf_and_menuselect/doc/privacy.txt (with props)
team/group/autoconf_and_menuselect/doc/realtime.txt (with props)
team/group/autoconf_and_menuselect/doc/security.txt (with props)
team/group/autoconf_and_menuselect/doc/smdi.txt (with props)
team/group/autoconf_and_menuselect/doc/sms.txt (with props)
Added: team/group/autoconf_and_menuselect/doc/ael.txt
URL: http://svn.digium.com/view/asterisk/team/group/autoconf_and_menuselect/doc/ael.txt?rev=18300&view=auto
==============================================================================
--- team/group/autoconf_and_menuselect/doc/ael.txt (added)
+++ team/group/autoconf_and_menuselect/doc/ael.txt Fri Apr 7 10:40:08 2006
@@ -1,0 +1,273 @@
+The Asterisk Extension Language
+===================================
+
+Over time, people have been pushing to add features to extensions.conf to make
+it more like a programming language. AEL is intended to provide an actual
+programming language that can be used to write an Asterisk dialplan.
+
+Getting Started
+-------------------------
+The AEL parser (pbx_ael.so) is completely separate from the module
+that parses extensions.conf (pbx_config.so). To use AEL, the only thing that
+has to be done is the module pbx_ael.so must be loaded by Asterisk. This will
+be done automatically if using 'autoload=yes' in /etc/asterisk/modules.conf.
+When the module is loaded, it will look for 'extensions.ael' in /etc/asterisk/.
+Both extensions.conf and extensions.ael can be used in conjunction with each
+other if that is what is desired. Some users may want to keep extensions.conf
+for the features that are configured in the 'general' section of
+extensions.conf.
+
+
+Reloading extensions.ael
+-------------------------
+To reload extensions.ael, the following command can be issued at the CLI.
+
+ *CLI> reload pbx_ael.so
+
+
+Contexts
+-------------------------
+Contexts in AEL represent a set of extensions in the same way that they do
+in extensions.conf.
+
+context default {
+
+};
+
+
+Extensions
+-------------------------
+To specify an extension in a context, the following syntax is used. If more
+than one application is be called in an extension, they can be listed in order
+inside of a block.
+
+context default {
+ 1234 => Playback(tt-monkeys);
+ 8000 => {
+ NoOp(one);
+ NoOp(two);
+ NoOp(three);
+ };
+ _5XXX => NoOp(it's a pattern!);
+};
+
+
+Includes
+-------------------------
+Contexts can be included in other contexts. All included contexts are listed
+within a single block.
+
+context default {
+ includes {
+ local;
+ longdistance;
+ international;
+ };
+};
+
+
+Dialplan Switches
+-------------------------
+Switches are listed in their own block within a context.
+
+context default {
+ switches {
+ DUNDi/e164;
+ IAX2/box5;
+ };
+ eswitches {
+ IAX2/context@${CURSERVER};
+ };
+};
+
+
+Ignorepat
+-------------------------
+ignorepat can be used to instruct channel drivers to not cancel dialtone upon
+receipt of a particular pattern. The most commonly used example is '9'.
+
+context outgoing {
+ ignorepat => 9;
+};
+
+
+Variables
+-------------------------
+Variables in Asterisk do not have a type, so to define a variable, it just has
+to be specified with a value.
+
+Global variables are set in their own block.
+
+globals {
+ CONSOLE=Console/dsp;
+ TRUNK=Zap/g2;
+};
+
+Variables can be set within extensions as well.
+
+context foo {
+ 555 => {
+ x=5;
+ y=blah;
+ NoOp(x is ${x} and y is ${y} !);
+ };
+};
+
+Writing to a dialplan function is treated the same as writing to a variable.
+
+context blah {
+ s => {
+ CALLERID(name)=ChickenMan;
+ NoOp(My name is ${CALLERID(name)} !);
+ };
+};
+
+
+Loops
+-------------------------
+AEL has implementations of 'for' and 'while' loops.
+
+context loops {
+ 1 => {
+ for (x=0; ${x} < 3; x=${x} + 1) {
+ Verbose(x is ${x} !);
+ };
+ };
+ 2 => {
+ y=10;
+ while (${y} >= 0) {
+ Verbose(y is ${y} !);
+ y=${y}-1;
+ };
+ };
+};
+
+
+Conditionals
+-------------------------
+AEL supports if and switch statements. Note that if you have an else
+clause, you MUST place braces around the non-else portion of the if
+statement.
+
+context conditional {
+ _8XXX => {
+ Dial(SIP/${EXTEN});
+ if (${DIALSTATUS} = "BUSY") {
+ Voicemail(${EXTEN}|b);
+ } else
+ Voicemail(${EXTEN}|u);
+ };
+ _777X => {
+ switch (${EXTEN}) {
+ case 7771:
+ NoOp(You called 7771!);
+ break;
+ case 7772:
+ NoOp(You called 7772!);
+ break;
+ case 7773:
+ NoOp(You called 7773!);
+ // fall through
+ default:
+ NoOp(In the default clause!);
+ };
+ };
+};
+
+
+goto and labels
+-------------------------
+This is an example of how to do a goto in AEL.
+
+context gotoexample {
+ s => {
+begin:
+ NoOp(Infinite Loop! yay!);
+ Wait(1);
+ goto begin;
+ };
+};
+
+
+Macros
+-------------------------
+A macro is defined in its own block like this. The arguments to the macro are
+specified with the name of the macro. They are then reffered to by that same
+name. A catch block can be specified to catch special extensions.
+
+macro std-exten( ext , dev ) {
+ Dial(${dev}/${ext},20);
+ switch(${DIALSTATUS) {
+ case BUSY:
+ Voicemail(b${ext});
+ break;
+ default:
+ Voicemail(u${ext});
+ };
+ catch a {
+ VoiceMailMain(${ext});
+ return;
+ };
+};
+
+A macro is then called by preceeding the macro name with an ampersand.
+
+context example {
+ _5XXX => &std-exten(${EXTEN}, "IAX2");
+};
+
+
+Examples
+------------------------
+
+context demo {
+ s => {
+ Wait(1);
+ Answer();
+ TIMEOUT(digit)=5;
+ TIMEOUT(response)=10;
+restart:
+ Background(demo-congrats);
+instructions:
+ for (x=0; ${x} < 3; x=${x} + 1) {
+ Background(demo-instruct);
+ WaitExten();
+ };
+ };
+ 2 => {
+ Background(demo-moreinfo);
+ goto s|instructions;
+ };
+ 3 => {
+ LANGUAGE()=fr;
+ goto s|restart;
+ };
+ 500 => {
+ Playback(demo-abouttotry);
+ Dial(IAX2/guest at misery.digium.com);
+ Playback(demo-nogo);
+ goto s|instructions;
+ };
+ 600 => {
+ Playback(demo-echotest);
+ Echo();
+ Playback(demo-echodone);
+ goto s|instructions;
+ };
+ # => {
+hangup:
+ Playback(demo-thanks);
+ Hangup();
+ };
+ t => goto #|hangup;
+ i => Playback(invalid);
+};
+
+
+Syntax Note
+------------------------
+Please note that all opening {'s are on the same line as the keyword. For
+the time being, that syntax is mandatory. We are looking at ways to allow
+other syntax in the future for flexibility, but for now, that is the way
+you must write AEL clauses.
+
Propchange: team/group/autoconf_and_menuselect/doc/ael.txt
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Added: team/group/autoconf_and_menuselect/doc/app-sms.txt
URL: http://svn.digium.com/view/asterisk/team/group/autoconf_and_menuselect/doc/app-sms.txt?rev=18300&view=auto
==============================================================================
--- team/group/autoconf_and_menuselect/doc/app-sms.txt (added)
+++ team/group/autoconf_and_menuselect/doc/app-sms.txt Fri Apr 7 10:40:08 2006
@@ -1,0 +1,470 @@
+
+ * Application SMS
+
+ The SMS module for Asterisk was developed by Adrian Kennard, and is an
+ implementation of the ETSI specification for landline SMS, ETSI ES 201
+ 912, which is available from www.etsi.org. Landline SMS is starting to
+ be available in various parts of Europe, and is available from BT in
+ the UK. However, Asterisk would allow gateways to be created in other
+ locations such as the US, and use of SMS capable phones such as the
+ Magic Messenger. SMS works using analogue or ISDN lines.
+
+Background
+
+ Short Message Service (SMS), or texting is very popular between mobile
+ phones. A message can be sent between two phones, and normally
+ contains 160 characters. There are ways in which various types of data
+ can be encoded in a text message such as ring tones, and small
+ graphic, etc. Text messaging is being used for voting and
+ competitions, and also SPAM...
+ Sending a message involves the mobile phone contacting a message
+ centre (SMSC) and passing the message to it. The message centre then
+ contacts the destination mobile to deliver the message. The SMSC is
+ responsible for storing the message and trying to send it until the
+ destination mobile is available, or a timeout.
+ Landline SMS works in basically the same way. You would normally have
+ a suitable text capable landline phone, or a separate texting box such
+ as a Magic Messenger on your phone line. This sends a message to a
+ message centre your telco provides by making a normal call and sending
+ the data using 1200 Baud FSK signaling according to the ETSI spec. To
+ receive a message the message centre calls the line with a specific
+ calling number, and the text capable phone answers the call and
+ receives the data using 1200 Baud FSK signaling. This works
+ particularly well in the UK as the calling line identity is sent
+ before the first ring, so no phones in the house would ring when a
+ message arrives.
+
+Typical use with Asterisk
+
+ Sending messages from an Asterisk box can be used for a variety of
+ reasons, including notification from any monitoring systems, email
+ subject lines, etc.
+ Receiving messages to an Asterisk box is typically used just to email
+ the messages to someone appropriate - we email and texts that are
+ received to our direct numbers to the appropriate person. Received
+ messages could also be used to control applications, manage
+ competitions, votes, post items to IRC, anything.
+ Using a terminal such as a magic messenger, an Asterisk box could ask
+ as a message centre sending messages to the terminal, which will beep
+ and pop up the message (and remember 100 or so messages in its
+ memory).
+
+Terminology
+
+ SMS
+ Short Message Service
+ i.e. text messages
+ SMSC
+ Short Message Service Centre
+ The system responsible for storing and forwarding messages
+ MO
+ Mobile Originated
+ A message on its way from a mobile or landline device to the SMSC
+ MT
+ Mobile Terminated
+ A message on its way from the SMSC to the mobile or landline device
+ RX
+ Receive
+ A message coming in to the Asterisk box
+ TX
+ Transmit
+ A message going out of the Asterisk box
+
+Sub address
+
+ When sending a message to a landline, you simply send to the landline
+ number. In the UK, all of the mobile operators (bar one) understand
+ sending messages to landlines and pass the messages to the BTText
+ system for delivery to the landline.
+ The specification for landline SMS allows for the possibility of more
+ than one device on a single landline. These can be configured with Sub
+ addresses which are a single digit. To send a message to a specific
+ device the message is sent to the landline number with an extra digit
+ appended to the end. The telco can define a default sub address (9 in
+ the UK) which is used when the extra digit is not appended to the end.
+ When the call comes in, part of the calling line ID is the sub
+ address, so that only one device on the line answers the call and
+ receives the message.
+ Sub addresses also work for outgoing messages. Part of the number
+ called by the device to send a message is its sub address. Sending
+ from the default sub address (9 in the UK) means the message is
+ delivered with the sender being the normal landline number. Sending
+ from any other sub address makes the sender the landline number with
+ an extra digit on the end.
+ Using Asterisk, you can make use of the sub addresses for sending and
+ receiving messages. Using DDI (DID, i.e. multiple numbers on the line
+ on ISDN) you can also make use of many different numbers for SMS.
+
+Build / installation
+
+ app_sms.c is included in the Asterisk source apps directory and is
+ included in the object list (app_sms.so) in apps/Makefile.
+ smsq.c is a stand alone helper application which is used to send SMSs
+ from the command line. It uses the popt library. A line for your make
+ file is:-
+smsq: smsq.c
+ cc -O -o smsq smsq.c -lpopt
+
+extensions.conf
+
+ The following contexts are recommended.
+; Mobile Terminated, RX. This is used when an incoming call from the SMS arrive
+s, with the queue (called number and sub address) in ${EXTEN}
+; Running an app after receipt of the text allows the app to find all messages
+in the queue and handle them, e.g. email them.
+; The app may be something like smsq --process=somecommand --queue=${EXTEN}
+to run a command for each received message
+; See below for usage
+[smsmtrx]
+exten = _X.,1, SMS(${EXTEN}|a)
+exten = _X.,2,System("someapptohandleincomingsms ${EXTEN}")
+exten = _X.,3,Hangup
+; Mobile originated, RX. This is receiving a message from a device, e.g. a Magi
+c Messenger on a sip extension
+; Running an app after receipt of the text allows the app to find all messages
+in the queue and handle then, e.g. sending them to the public SMSC
+; The app may be something like smsq --process=somecommand --queue=${EXTEN}
+to run a command for each received message
+; See below for example usage
+[smsmorx]
+exten = _X.,1, SMS(${EXTEN}|sa)
+exten = _X.,2,System("someapptohandlelocalsms ${EXTEN}")
+exten = _X.,3,Hangup
+
+ smsmtrx is normally accessed by an incoming call from the SMSC. In the
+ UK this call is from a CLI of 080058752X0 where X is the sub address.
+ As such a typical usage in the extensions.conf at the point of
+ handling an incoming call is:-
+exten = _X./8005875290,1,Goto(smsmtrx,${EXTEN},1)
+exten = _X./_80058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERIDNUM:8:1},1)
+
+ Alternatively, if you have the correct national prefix on incoming
+ CLI, e.g. using zaphfc, you might use:-
+exten = _X./08005875290,1,Goto(smsmtrx,${EXTEN},1)
+exten = _X./_080058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERIDNUM:9:1},1)
+
+ smsmorx is normally accessed by a call from a local sip device
+ connected to a Magic Messenger. It could however by that you are
+ operating Asterisk as a message centre for calls from outside. Either
+ way, you look at the called number and goto smsmorx. In the UK, the
+ SMSC number that would be dialed is 1709400X where X is the caller sub
+ address. As such typical usage in extension.config at the point of
+ handling a call from a sip phone is:-
+exten = 17094009,1,Goto(smsmorx,${CALLERIDNUM},1)
+exten = _1709400[0-8],1,Goto(smsmorx,${CALLERIDNUM}-{EXTEN:7:1},1)
+
+Using smsq
+
+ smsq is a simple helper application designed to make it easy to send
+ messages from a command line. it is intended to run on the Asterisk
+ box and have direct access to the queue directories for SMS and for
+ Asterisk.
+ In its simplest form you can send an SMS by a command such as
+ smsq 0123456789 This is a test to 0123456789
+ This would create a queue file for a mobile originated TX message in
+ queue 0 to send the text "This is a test to 0123456789" to 0123456789.
+ It would then place a file in the /var/spool/asterisk/outgoing
+ directory to initiate a call to 17094009 (the default message centre
+ in smsq) attached to application SMS with argument of the queue name
+ (0).
+ Normally smsq will queue a message ready to send, and will then create
+ a file in the Asterisk outgoing directory causing Asterisk to actually
+ connect to the message centre or device and actually send the pending
+ message(s).
+ Using --process, smsq can however be used on received queues to run a
+ command for each file (matching the queue if specified) with various
+ environment variables set based on the message (see below);
+ smsq options:-
+
+ --help
+ Show help text
+ --usage
+ Show usage
+ --queue
+ -q
+ Specify a specific queue
+ In no specified, messages are queued under queue "0"
+ --da
+ -d
+ Specify destination address
+ --oa
+ -o
+ Specify originating address
+ This also implies that we are generating a mobile terminated message
+ --ud
+ -m
+ Specify the actual message
+ --ud-file
+ -f
+ Specify a file to be read for the context of the message
+ A blank filename (e.g. --ud-file= on its own) means read stdin. Very
+ useful when using via ssh where command line parsing could mess up the
+ message.
+ --mt
+ -t
+ Mobile terminated message to be generated
+ --mo
+ Mobile originated message to be generated
+ Default
+ --tx
+ Transmit message
+ Default
+ --rx
+ -r
+ Generate a message in the receive queue
+ --UTF-8
+ Treat the file as UTF-8 encoded (default)
+ --UCS-1
+ Treat the file as raw 8 bit UCS-1 data, not UTF-8 encoded
+ --UCS-2
+ Treat the file as raw 16 bit bigendian USC-2 data
+ --process
+ Specific a command to process for each file in the queue
+ Implies --rx and --mt if not otherwise specified.
+ Sets environment variables for every possible variable, and also ud,
+ ud8 (USC-1 hex), and ud16 (USC-2 hex) for each call. Removes files.
+ --motx-channel
+ Specify the channel for motx calls
+ May contain X to use sub address based on queue name or may be full
+ number
+ Default is Local/1709400X
+ --motx-callerid
+ Specify the caller ID for motx calls
+ The default is the queue name without -X suffix
+ --motx-wait
+ Wait time for motx call
+ Default 10
+ --motx-delay
+ Retry time for motx call
+ Default 1
+ --motx-retries
+ Retries for motx call
+ Default 10
+ --mttx-channel
+ Specify the channel for mttx calls
+ Default is Local/ and the queue name without -X suffix
+ --mtttx-callerid
+ Specify the callerid for mttx calls
+ May include X to use sub address based on queue name or may be full
+ number
+ Default is 080058752X0
+ --mttx-wait
+ Wait time for mttx call
+ Default 10
+ --mttx-delay
+ Retry time for mttx call
+ Default 30
+ --mttx-retries
+ Retries for mttx call
+ Default 100
+ --default-sub-address
+ The default sub address assumed (e.g. for X in CLI and dialled numbers
+ as above) when none added (-X) to queue
+ Default 9
+ --no-dial
+ -x
+ Create queue, but do not dial to send message
+ --no-wait
+ Do not wait if a call appears to be in progress
+ This could have a small window where a mesdsage is queued but not
+ sent, so regular calls to smsq should be done to pick up any missed
+ messages
+ --concurrent
+ How many concurrent calls to allow (per queue), default 1
+ --mr
+ -n
+ Message reference
+ --pid
+ -p
+ Protocol ID
+ --dcs
+ Data coding scheme
+ --udh
+ Specific hex string of user data header specified (not including the
+ initial length byte)
+ May be a blank string to indicate header is included in the user data
+ already but user data header indication to be set.
+ --srr
+ Status report requested
+ --rp
+ Return path requested
+ --vp
+ Specify validity period (seconds)
+ --scts
+ Specify timestamp (YYYY-MM-DDTHH:MM:SS)
+ --spool-dir
+ Spool dir (in which sms and outgoing are found)
+ Default /var/spool/asterisk
+
+ Other arguments starting '-' or '--' are invalid and will cause an
+ error. Any trailing arguments are processed as follows:-
+ * If the message is mobile originating and no destination address
+ has been specified, then the first argument is assumed to be a
+ destination address
+ * If the message is mobile terminating and no destination address
+ has been specified, then the first argument is assumed to be the
+ queue name
+ * If there is no user data, or user data file specified, then any
+ following arguments are assumed to be the message, which are
+ concatenated.
+ * If no user data is specified, then no message is sent. However,
+ unless --no-dial is specified, smsq checks for pending messages
+ and generates an outgoing anyway
+
+ Note that when smsq attempts to make a file in
+ /var/spool/asterisk/outgoing, it checks if there is already a call
+ queued for that queue. It will try several filenames, up to the
+ --concorrent setting. If these files exists, then this means Asterisk
+ is already queued to send all messages for that queue, and so Asterisk
+ should pick up the message just queued. However, this alone could
+ create a race condition, so if the files exist then smsq will wait up
+ to 3 seconds to confirm it still exists or if the queued messages have
+ been sent already. The --no-wait turns off this behaviour. Basically,
+ this means that if you have a lot of messages to send all at once,
+ Asterisk will not make unlimited concurrent calls to the same message
+ centre or device for the same queue. This is because it is generally
+ more efficient to make one call and send all of the messages one after
+ the other.
+ smsq can be used with no arguments, or with a queue name only, and it
+ will check for any pending messages and cause an outgoing if there are
+ any. It only sets up one outgoing call at a time based on the first
+ queued message it finds. A outgoing call will normally send all queued
+ messages for that queue. One way to use smsq would be to run with no
+ queue name (so any queue) every minute or every few seconds to send
+ pending message. This is not normally necessary unless --no-dial is
+ selected. Note that smsq does only check motx or mttx depending on the
+ options selected, so it would need to be called twice as a general
+ check.
+ UTF-8 is used to parse command line arguments for user data, and is
+ the default when reading a file. If an invalid UTF-8 sequence is
+ found, it is treated as UCS-1 data (i.e, as is).
+ The --process option causes smsq to scan the specified queue (default
+ is mtrx) for messages (matching the queue specified, or any if queue
+ not specified) and run a command and delete the file. The command is
+ run with a number of environment variables set as follows. Note that
+ these are unset if not needed and not just taken from the calling
+ environment. This allows simple processing of incoming messages
+
+ $queue
+ Set if a queue specified
+ $?srr
+ srr is set (to blank) if srr defined and has value 1.
+ $?rp
+ rp is set (to blank) if rp defined and has value 1.
+ $ud
+ User data, UTF-8 encoding, including any control characters, but with
+ nulls stripped out
+ Useful for the content of emails, for example, as it includes any
+ newlines, etc.
+ $ude
+ User data, escaped UTF-8, including all characters, but control
+ characters \n, \r, \t, \f, \xxx and \ is escaped as \\
+ Useful fGuaranteed one line printable text, so useful in Subject lines
+ of emails, etc
+ $ud8
+ Hex UCS-1 coding of user data (2 hex digits per character)
+ Present only if all user data is in range U+0000 to U+00FF
+ $ud16
+ Hex UCS-2 coding of user data (4 hex digits per chartacter)
+ other
+ Other fields set using their field name, e.g. mr, pid, dcs, etc. udh
+ is a hex byte string
+
+File formats
+
+ By default all queues are held in a director /var/spool/asterisk/sms.
+ Within this directory are sub directories mtrx, mttx, morx, motx which
+ hold the received messages and the messages ready to send. Also,
+ /var/log/asterisk/sms is a log file of all messages handled.
+ The file name in each queue directory starts with the queue parameter
+ to SMS which is normally the CLI used for an outgoing message or the
+ called number on an incoming message, and may have -X (X being sub
+ address) appended. If no queue ID is known, then 0 is used by smsq by
+ default. After this is a dot, and then any text. Files are scanned for
+ matching queue ID and a dot at the start. This means temporary files
+ being created can be given a different name not starting with a queue
+ (we recommend a . on the start of the file name for temp files).
+ Files in these queues are in the form of a simple text file where each
+ line starts with a keyword and an = and then data. udh and ud have
+ options for hex encoding, see below.
+ UTF-8. The user data (ud) field is treated as being UTF-8 encoded
+ unless the DCS is specified indicating 8 bit formart. If 8 bit format
+ is specified then the user data is sent as is.
+ The keywords are as follows:-
+
+ oa Originating address
+ The phone number from which the message came
+ Present on mobile terminated messages and is the CLI for morx messages
+ da
+ Destination Address
+ The phone number to which the message is sent
+ Present on mobile originated messages
+ scts
+ The service centre time stamp
+ Format YYYY-MM-DDTHH:MM:SS
+ Present on mobile terminated messages
+ pid
+ One byte decimal protocol ID
+ See GSM specs for more details
+ Normally 0 or absent
+ dcs
+ One byte decimal data coding scheme
+ If omitted, a sensible default is used (see below)
+ See GSM specs for more details
+ mr
+ One byte decimal message reference
+ Present on mobile originated messages, added by default if absent
+ srr
+ 0 or 1 for status report request
+ Does not work in UK yet, not implemented in app_sms yet
+ rp
+ 0 or 1 return path
+ See GSM specs for details
+ vp
+ Validity period in seconds
+ Does not work in UK yet
+ udh
+ Hex string of user data header prepended to the SMS contents,
+ excluding initial length byte.
+ Consistent with ud, this is specified as udh# rather than udh=
+ If blank, this means that the udhi flag will be set but any user data
+ header must be in the ud field
+ ud
+ User data, may be text, or hex, see below
+
+ udh is specified as as udh# followed by hex (2 hex digits per byte).
+ If present, then the user data header indicator bit is set, and the
+ length plus the user data header is added to the start of the user
+ data, with padding if necessary (to septet boundary in 7 bit format).
+ User data can hold an USC character codes U+0000 to U+FFFF. Any other
+ characters are coded as U+FEFF
+ ud can be specified as ud= followed by UTF-8 encoded text if it
+ contains no control characters, i.e. only (U+0020 to U+FFFF). Any
+ invalid UTF-8 sequences are treated as is (U+0080-U+00FF).
+ ud can also be specified as ud# followed by hex (2 hex digits per
+ byte) containing characters U+0000 to U+00FF only.
+ ud can also be specified as ud## followed by hex (4 hex digits per
+ byte) containing UCS-2 characters.
+ When written by app_sms (e.g. incoming messages), the file is written
+ with ud= if it can be (no control characters). If it cannot, the a
+ comment line ;ud= is used to show the user data for human readability
+ and ud# or ud## is used.
+
+Delivery reports
+
+ The SMS specification allows for delivery reports. These are requested
+ using the srr bit. However, as these do not work in the UK yet they
+ are not fully implemented in this application. If anyone has a telco
+ that does implement these, please let me know. BT in the UK have a non
+ standard way to do this by starting the message with *0#, and so this
+ application may have a UK specific bodge in the near future to handle
+ these.
+ The main changes that are proposed for delivery report handling are :-
+ * New queues for sent messages, one file for each destination
+ address and message reference.
+ * New field in message format, user reference, allowing applications
+ to tie up their original message with a report.
+ * Handling of the delivery confirmation/rejection and connecting to
+ the outgoing message - the received message file would then have
+ fields for the original outgoing message and user reference
+ allowing applications to handle confirmations better.
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@@ -1,0 +1,78 @@
+Asterisk Main Configuration File
+-----------------------------------------------------
+Below is a sample of the main Asterisk configuration file,
+asterisk.conf. Note that this file is _not_ provided in
+sample form, because the Makefile creates it when needed
+and does not touch it when it already exists.
+
+---------------
+
+[directories]
+; Make sure these directoriess have the right permissions if not
+; running Asterisk as root
+
+; Where the configuration files (except for this one) are located
+astetcdir => /etc/asterisk
+
+; Where the Asterisk loadable modules are located
+astmoddir => /usr/lib/asterisk/modules
+
+; Where additional 'library' elements (scripts, etc.) are located
+astvarlibdir => /var/lib/asterisk
+
+; Where AGI scripts/programs are located
+astagidir => /var/lib/asterisk/agi-bin
+
+; Where spool directories are located
+; Voicemail, monitor, dictation and other apps will create files here
+; and outgoing call files (used with pbx_spool) must be placed here
+astspooldir => /var/spool/asterisk
+
+; Where the Asterisk process ID (pid) file should be created
+astrundir => /var/run/asterisk
+
+; Where the Asterisk log files should be created
+astlogdir => /var/log/asterisk
+
+
+[options]
+;Under "options" you can enter configuration options
+;that you also can set with command line options
+
+verbose = 0 ; Verbosity level for logging (-v)
+debug = 3 ; Debug: "No" or value (1-4)
+nofork=yes | no ; Background execution disabled (-f)
+console= yes | no ; Console mode (-c)
+highpriority = yes | no ; Execute with high priority (-p)
+initcrypto = yes | no ; Initialize crypto at startup (-i)
+nocolor = yes | no ; Disable ANSI colors (-n)
+dumpcore = yes | no ; Dump core on failure (-g)
+quiet = yes | no ; Run quietly (-q)
+timestamp = yes | no ; Force timestamping on log entries to console (-T)
+runuser = asterisk ; User to run asterisk as (-U) NOTE: will require changes to
+ ; directory and device permisions
+rungroup = asterisk ; Group to run asterisk as (-G)
+internal_timing = yes | no ; Enable internal timing support (-I)
+
+;These options have no command line equivalent
+cache_record_files = yes | no ; Cache record() files in another directory until completion
+record_cache_dir = <dir>
+transcode_via_sln = yes | no ; Build transcode paths via SLINEAR
+transmit_silence_during_record = yes | no ; send SLINEAR silence while channel is being recorded
+maxload = 1.0 ; The maximum load average we accept calls for
+maxcalls = 255 ; The maximum number of concurrent calls you want to allow
+execincludes = yes | no ; Allow #exec entries in configuration files
+dontwarn = yes | no ; Don't over-inform the Asterisk sysadm, he's a guru
+systemname = <a_string> ; System name. Used to prefix CDR uniqueid and to fill ${SYSTEMNAME}
+
+[files]
+; Changing the following lines may compromise your security
+; Asterisk.ctl is the pipe that is used to connect the remote CLI
+; (asterisk -r) to Asterisk. Changing these settings change the
+; permissions and ownership of this file.
+; The file is created when Asterisk starts, in the "astrundir" above.
+
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = asterisk
+;astctl = asterisk.ctl
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--- team/group/autoconf_and_menuselect/doc/backtrace.txt (added)
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+This document is to provide information on how to obtain the
+backtraces required on the asterisk bug tracker, available at
+http://bugs.digium.com. The information is required by developers to
+help fix problem with bugs of any kind. Backtraces provide information
+about what was wrong when a program crashed; in our case,
+Asterisk. There are two kind of backtraces (aka 'bt'), which are
+useful: bt and bt full.
+
+First of all, when you start Asterisk, you MUST start it with option
+-g (this tells Asterisk to produce a core file if it crashes).
+
+If you start Asterisk with the safe_asterisk script, it automatically
+starts using the option -g.
+
+If you're not sure if Asterisk is running with the -g option, type the
+following command in your shell:
+
+debian:/tmp# ps aux | grep asterisk
+root 17832 0.0 1.2 2348 788 pts/1 S Aug12 0:00 /bin/sh /usr/sbin/safe_asterisk
+root 26686 0.0 2.8 15544 1744 pts/1 S Aug13 0:02 asterisk -vvvg -c
+[...]
+
+The interesting information is located in the last column.
+
+Second, your copy of Asterisk must have been built without
+optimization or the backtrace will be (nearly) unusable. This can be
+done by using 'make dont-optimize' intead of 'make install' to build
+and install the Asterisk binary and modules.
+
+After Asterisk crashes, a core file will be "dumped" in your /tmp/
+directory. To make sure it's really there, you can just type the
+following command in your shell:
+
+debian:/tmp# ls -l /tmp/core.*
+-rw------- 1 root root 10592256 Aug 12 19:40 /tmp/core.26252
+-rw------- 1 root root 9924608 Aug 12 20:12 /tmp/core.26340
+-rw------- 1 root root 10862592 Aug 12 20:14 /tmp/core.26374
+-rw------- 1 root root 9105408 Aug 12 20:19 /tmp/core.26426
+-rw------- 1 root root 9441280 Aug 12 20:20 /tmp/core.26462
+-rw------- 1 root root 8331264 Aug 13 00:32 /tmp/core.26647
+debian:/tmp#
+
+Now that we've verified the core file has been written to disk, the
+final part is to extract 'bt' from the core file. Core files are
+pretty big, don't be scared, it's normal.
+
+*** NOTE: Don't attach core files on the bug tracker, we only need the bt and bt full. ***
+
+For extraction, we use a really nice tool, called gdb. To verify that
+you have gdb installed on your system:
+
+debian:/tmp# gdb -v
+GNU gdb 6.3-debian
+Copyright 2004 Free Software Foundation, Inc.
+GDB is free software, covered by the GNU General Public License, and you are
+welcome to change it and/or distribute copies of it under certain conditions.
+Type "show copying" to see the conditions.
+There is absolutely no warranty for GDB. Type "show warranty" for details.
+This GDB was configured as "i386-linux".
+debian:/tmp#
+
+Which is great, we can continue. If you don't have gdb installed, go install gdb.
+
+Now load the core file in gdb, as follows:
+
+debian:/tmp# gdb -se "asterisk" -c /tmp/core.26252
+[...]
+(You would see a lot of output here.)
+[...]
+Reading symbols from /usr/lib/asterisk/modules/app_externalivr.so...done.
+Loaded symbols for /usr/lib/asterisk/modules/app_externalivr.so
+#0 0x29b45d7e in ?? ()
+(gdb)
+
+Now at the gdb prompt, type: bt
+You would see output similar to:
+(gdb) bt
+#0 0x29b45d7e in ?? ()
+#1 0x08180bf8 in ?? ()
+#2 0xbcdffa58 in ?? ()
+#3 0x08180bf8 in ?? ()
+#4 0xbcdffa60 in ?? ()
+#5 0x08180bf8 in ?? ()
+#6 0x180bf894 in ?? ()
+#7 0x0bf80008 in ?? ()
+#8 0x180b0818 in ?? ()
+#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
+#10 0x000000a0 in ?? ()
+#11 0x000000a0 in ?? ()
+#12 0x00000000 in ?? ()
+#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262
+#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
+#15 0xbcdffbe0 in ?? ()
+#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
+#17 0x401ec92a in clone () from /lib/libc.so.6
+(gdb)
+
+
+The bt's output is the information that we need on the bug tracker.
+
+Now do a bt full as follows:
[... 4438 lines stripped ...]
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