Index: configs/alarmreceiver.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/alarmreceiver.conf.sample,v retrieving revision 1.1 diff -u -r1.1 alarmreceiver.conf.sample --- configs/alarmreceiver.conf.sample 21 Jun 2004 19:28:34 -0000 1.1 +++ configs/alarmreceiver.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -22,8 +22,9 @@ ;eventcmd = yourprogram -yourargs ... ; -; Specify a spool directory for the event files. This setting is required if you want the app to be useful. -; Event files written to the spool directory will be of the template event-XXXXXX, where XXXXXX is a random +; Specify a spool directory for the event files. This setting is required +; if you want the app to be useful. Event files written to the spool +; directory will be of the template event-XXXXXX, where XXXXXX is a random ; and unique alphanumeric string. ; ; Default is none, and the events will be dropped on the floor. @@ -32,8 +33,9 @@ eventspooldir = /tmp ; -; The alarmreceiver app can either log the events one-at-a-time to individual files in the spool -; directory, or it can store them until the caller disconnects and write them all to one file. +; The alarmreceiver app can either log the events one-at-a-time to individual +; files in the spool directory, or it can store them until the caller +; disconnects and write them all to one file. ; ; The default setting for logindividualevents is no. ; @@ -41,32 +43,34 @@ logindividualevents = no ; -; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. to 10000 msec. The -; default is 2000 msec. Note: if you wish to test the receiver by entering digits manually, set this -; to a reasonable time out like 10000 milliseconds. +; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. +; to 10000 msec. The default is 2000 msec. Note: if you wish to test the +; receiver by entering digits manually, set this to a reasonable time out +; like 10000 milliseconds. fdtimeout = 2000 ; -; The timeout for receiving subsequent DTMF digits is adjustable from 110 msec. to 4000 msec. The -; default is 200 msec. Note: if you wish to test the receiver by entering digits manually, set this -; to a reasonable time out like 4000 milliseconds. +; The timeout for receiving subsequent DTMF digits is adjustable from +; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test +; the receiver by entering digits manually, set this to a reasonable time out +; like 4000 milliseconds. ; sdtimeout = 200 ; -; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. The default is 8192 -; This shouldn't need to be messed with, but is included just in case there are problems with -; signal levels. +; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. +; The default is 8192. This shouldn't need to be messed with, but is included +; just in case there are problems with signal levels. ; loudness = 8192 ; -; The db-family setting allows the user to capture statistics on the number of calls, and the errors -; the alarm receiver sees. The default is for no db-family name to be defined and the database logging -; to be turned off. +; The db-family setting allows the user to capture statistics on the number of +; calls, and the errors the alarm receiver sees. The default is for no +; db-family name to be defined and the database logging to be turned off. ; ;db-family = yourfamily: Index: configs/codecs.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/codecs.conf.sample,v retrieving revision 1.4 diff -u -r1.4 codecs.conf.sample --- configs/codecs.conf.sample 26 Aug 2005 20:14:06 -0000 1.4 +++ configs/codecs.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -12,7 +12,8 @@ enhancement => true ; voice activity detection [true / false] -; reduces bitrate when no voice detected, used only for CBR (implicit in VBR/ABR) +; reduces bitrate when no voice detected, used only for CBR +; (implicit in VBR/ABR) vad => true ; variable bit rate [true / false] Index: configs/extensions.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/extensions.conf.sample,v retrieving revision 1.46 diff -u -r1.46 extensions.conf.sample --- configs/extensions.conf.sample 27 Jul 2005 05:45:52 -0000 1.46 +++ configs/extensions.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -52,13 +52,15 @@ ; priorityjumping=no ; -; You can include other config files, use the #include command (without the ';') -; Note that this is different from the "include" command that includes contexts within -; other contexts. The #include command works in all asterisk configuration files. +; You can include other config files, use the #include command +; (without the ';'). Note that this is different from the "include" command +; that includes contexts within other contexts. The #include command works +; in all asterisk configuration files. ;#include "filename.conf" ; The "Globals" category contains global variables that can be referenced -; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable +; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental +; variables, ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals] @@ -73,10 +75,14 @@ ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in ; the specified group. The four possible options are: ; -; g: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group). -; G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group). -; r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group). -; R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group). +; g: select the lowest-numbered non-busy Zap channel +; (aka. ascending sequential hunt group). +; G: select the highest-numbered non-busy Zap channel +; (aka. descending sequential hunt group). +; r: use a round-robin search, starting at the next highest channel than last +; time (aka. ascending rotary hunt group). +; R: use a round-robin search, starting at the next lowest channel than last +; time (aka. descending rotary hunt group). ; TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:pass@provider @@ -443,11 +449,11 @@ ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) -; Real extensions would go here. Generally you want real extensions to be 4 or 5 -; digits long (although there is no such requirement) and start with a single -; digit that is fairly large (like 6 or 7) so that you have plenty of room to -; overlap extensions and menu options without conflict. You can alias them with -; names, too and use global variables +; Real extensions would go here. Generally you want real extensions to be +; 4 or 5 digits long (although there is no such requirement) and start with a +; single digit that is fairly large (like 6 or 7) so that you have plenty of +; room to overlap extensions and menu options without conflict. You can alias +; them with names, too, and use global variables ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer Index: configs/iax.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/iax.conf.sample,v retrieving revision 1.55 diff -u -r1.55 iax.conf.sample --- configs/iax.conf.sample 15 Sep 2005 02:25:06 -0000 1.55 +++ configs/iax.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -74,11 +74,11 @@ ; The jitter buffer's function is to compensate for varying ; network delay. ; -; There are presently two jitterbuffer implementations available for * and chan_iax2; -; the classic and the new, channel/application independent implementation. These -; are controlled at compile-time. The new jitterbuffer additionally has support for PLC -; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost -; packets. +; There are presently two jitterbuffer implementations available for Asterisk +; and chan_iax2; the classic and the new, channel/application independent +; implementation. These are controlled at compile-time. The new jitterbuffer +; additionally has support for PLC which greatly improves quality as the +; jitterbuffer adapts size, and in compensating for lost packets. ; ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio @@ -90,7 +90,8 @@ ; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels ; we don't want to do jitterbuffering on the switch, since the endpoints ; can each handle this. However, some endpoints may have poor jitterbuffers -; themselves, so this option will force * to always jitterbuffer, even in this case. +; themselves, so this option will force * to always jitterbuffer, even in this +; case. ; [This option presently applies only to the new jitterbuffer implementation] ; ; dropcount: the jitter buffer is sized such that no more than "dropcount" @@ -105,15 +106,17 @@ ; ; resyncthreshold: when the jitterbuffer notices a significant change in delay ; that continues over a few frames, it will resync, assuming that the change in -; delay was caused by a timestamping mix-up. The threshold for noticing a change -; in delay is measured as twice the measured jitter plus this resync threshold. -; Resycning can be disabled by setting this parameter to -1. +; delay was caused by a timestamping mix-up. The threshold for noticing a +; change in delay is measured as twice the measured jitter plus this resync +; threshold. +; Resyncing can be disabled by setting this parameter to -1. ; [This option presently applies only to the new jitterbuffer implementation] ; -; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should -; return in a row. Since some clients do not send CNG/DTX frames to indicate -; silence, the jitterbuffer will assume silence has begun after returning this -; many interpolations. This prevents interpolating throughout a long silence. +; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer +; should return in a row. Since some clients do not send CNG/DTX frames to +; indicate silence, the jitterbuffer will assume silence has begun after +; returning this many interpolations. This prevents interpolating throughout +; a long silence. ; [This option presently applies only to the new jitterbuffer implementation] ; ; maxexcessbuffer: If conditions improve after a period of high jitter, @@ -147,11 +150,11 @@ ;trunkfreq=20 ; How frequently to send trunk msgs (in ms) ; Should we send timestamps for the individual sub-frames within trunk frames? -; There is a small bandwidth use for these (less than 1kbps/call), but they ensure -; that frame timestamps get sent end-to-end properly. If both ends of all your trunks -; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you -; can probably suppress these. The receiver must also support this feature, although -; they do not also need to have it enabled. +; There is a small bandwidth use for these (less than 1kbps/call), but they +; ensure that frame timestamps get sent end-to-end properly. If both ends of +; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame +; length for your codecs, you can probably suppress these. The receiver must +; also support this feature, although they do not also need to have it enabled. ; ; trunktimestamps=yes ; @@ -217,22 +220,21 @@ ; ;mailboxdetail=yes ; -; If regcontext is specified, Asterisk will dynamically -; create and destroy a NoOp priority 1 extension for a given -; peer who registers or unregisters with us. The actual extension -; is the 'regexten' parameter of the registering peer or its -; name if 'regexten' is not provided. More than one regexten may be supplied -; if they are separated by '&'. Patterns may be used in regexten. +; If regcontext is specified, Asterisk will dynamically create and destroy +; a NoOp priority 1 extension for a given peer who registers or unregisters +; with us. The actual extension is the 'regexten' parameter of the registering +; peer or its name if 'regexten' is not provided. More than one regexten +; may be supplied if they are separated by '&'. Patterns may be used in +; regexten. ; ;regcontext=iaxregistrations ; -; If we don't get ACK to our NEW within 2000ms, and autokill is set -; to yes, then we cancel the whole thing (that's enough time for one -; retransmission only). This is used to keep things from stalling for a long -; time for a host that is not available, but would be ill advised for bad -; connections. In addition to 'yes' or 'no' you can also specify a number -; of milliseconds. See 'qualify' for individual peers to turn on for just -; a specific peer. +; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes, +; then we cancel the whole thing (that's enough time for one retransmission +; only). This is used to keep things from stalling for a long time for a host +; that is not available, but would be ill advised for bad connections. In +; addition to 'yes' or 'no' you can also specify a number of milliseconds. +; See 'qualify' for individual peers to turn on for just a specific peer. ; autokill=yes ; @@ -274,8 +276,8 @@ ; has expired based on its registration interval, used the stored ; address information regardless. (yes|no) -; Guest sections for unauthenticated connection attempts. Just -; specify an empty secret, or provide no secret section. +; Guest sections for unauthenticated connection attempts. Just specify an +; empty secret, or provide no secret section. ; [guest] type=user @@ -310,14 +312,13 @@ ;context=dundi-e164-local ; -; Further user sections may be added, specifying a context and a -; secret used for connections with that given authentication name. -; Limited IP based access control is allowed by use of "allow" and -; "deny" keywords. Multiple rules are permitted. Multiple permitted -; contexts may be specified, in which case the first will be the default. -; You can also override caller*ID so that when you receive a call you -; set the Caller*ID to be what you want instead of trusting what -; the remote user provides +; Further user sections may be added, specifying a context and a secret used +; for connections with that given authentication name. Limited IP based +; access control is allowed by use of "allow" and "deny" keywords. Multiple +; rules are permitted. Multiple permitted contexts may be specified, in +; which case the first will be the default. You can also override caller*ID +; so that when you receive a call you set the Caller*ID to be what you want +; instead of trusting what the remote user provides ; ; There are three authentication methods that are supported: md5, plaintext, ; and rsa. The least secure is "plaintext", which sends passwords cleartext @@ -372,11 +373,10 @@ ;jitterbuffer=no ; Turn off jitter buffer for this peer ; -; Peers can remotely register as well, so that they can be -; mobile. Default IP's can also optionally be given but -; are not required. Caller*ID can be suggested to the other -; side as well if it is for example a phone instead of another -; PBX. +; Peers can remotely register as well, so that they can be mobile. Default +; IP's can also optionally be given but are not required. Caller*ID can be +; suggested to the other side as well if it is for example a phone instead of +; another PBX. ; ;[dynamichost] @@ -410,3 +410,4 @@ ;secret=moofoo ;context=default ;permit=0.0.0.0/0.0.0.0 + Index: configs/iaxprov.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/iaxprov.conf.sample,v retrieving revision 1.1 diff -u -r1.1 iaxprov.conf.sample --- configs/iaxprov.conf.sample 7 Jul 2004 09:34:01 -0000 1.1 +++ configs/iaxprov.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -1,25 +1,22 @@ ; ; IAX2 Provisioning Information ; -; Contains provisioning information for templates -; and for specific service entries. +; Contains provisioning information for templates and for specific service +; entries. ; -; Templates provide a group of settings from which provisioning takes -; place. A template may be based upon any template that has been -; specified before it. If the template that an entry is based on is not -; specified then it is presumed to be 'default' (unless it is the first -; of course). -; -; Templates which begin with 'si-' are used for provisioning -; units with specific service identifiers. For example the -; entry "si-000364000126" would be used when the device with the -; corresponding service identifier of "000364000126" attempts -; to register or make a call. +; Templates provide a group of settings from which provisioning takes place. +; A template may be based upon any template that has been specified before +; it. If the template that an entry is based on is not specified then it is +; presumed to be 'default' (unless it is the first of course). +; +; Templates which begin with 'si-' are used for provisioning units with +; specific service identifiers. For example the entry "si-000364000126" +; would be used when the device with the corresponding service identifier of +; "000364000126" attempts to register or make a call. ; [default] ; -; The port number the device should use to bind to. The default -; is 4569 +; The port number the device should use to bind to. The default is 4569. ; ;port=4569 ; @@ -27,14 +24,13 @@ ; ;server=192.168.69.3 ; -; altserver is the BACKUP server for registration and placing calls -; in the event the primary server is unavailable. +; altserver is the BACKUP server for registration and placing calls in the +; event the primary server is unavailable. ; ;altserver=192.168.69.4 ; -; port is the port number to use for IAX2 outbound. The -; connections to the server and altserver -- default is of course -; 4569. +; port is the port number to use for IAX2 outbound. The connections to the +; server and altserver -- default is of course 4569. ;serverport=4569 ; ; language is the preferred language for the device @@ -78,9 +74,10 @@ ; ;[*] ; -; If specified, the '*' provisioning is used for all devices which do -; not have another provisioning entry within the file. If unspecified, no +; If specified, the '*' provisioning is used for all devices which do not +; have another provisioning entry within the file. If unspecified, no ; provisioning will take place for devices which have no entry. DO NOT ; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING. ; ;template=default + Index: configs/indications.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/indications.conf.sample,v retrieving revision 1.29 diff -u -r1.29 indications.conf.sample --- configs/indications.conf.sample 23 Aug 2005 12:38:48 -0000 1.29 +++ configs/indications.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -16,7 +16,7 @@ ; [example] ; description = string -; The full name of your country, in English +; The full name of your country, in English. ; alias = iso[,iso]* ; List of other countries 2-letter iso codes, which have the same ; tone indications. @@ -31,14 +31,16 @@ ; callwaiting = tonelist ; Set of tones played when there is a call waiting in the background. ; dialrecall = tonelist -; Not well defined, many phone systems play a recall dial tone after hook flash +; Not well defined; many phone systems play a recall dial tone after hook +; flash. ; record = tonelist -; Set of tones played when call recording is in progress +; Set of tones played when call recording is in progress. ; info = tonelist -; Set of tones played with special information messages (e.g., "number is out of service") +; Set of tones played with special information messages (e.g., "number is +; out of service") ; 'name' = tonelist -; Every other variable will be available as a shortcut for the "PlayList" command -; but will not automaticly be used by Asterisk. +; Every other variable will be available as a shortcut for the "PlayList" command +; but will not be used automatically by Asterisk. ; ; ; The tonelist itself is defined by a comma-separated sequence of elements. @@ -587,8 +589,8 @@ description = South Africa ; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm ; (definitions for other countries can also be found there) -; Note, though, that South Africa uses two switch types in their network - Alcatel -; switches - mainly in the Western Cape, and Siemens elsewhere. +; Note, though, that South Africa uses two switch types in their network -- +; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere. ; The former use 383+417 in dial, ringback etc. The latter use 400*33 ; I've provided both, uncomment the ones you prefer ringcadance = 400,200,400,2000 Index: configs/logger.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/logger.conf.sample,v retrieving revision 1.14 diff -u -r1.14 logger.conf.sample --- configs/logger.conf.sample 22 Aug 2005 21:19:59 -0000 1.14 +++ configs/logger.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -16,10 +16,12 @@ ; This appends the hostname to the name of the log files. ;appendhostname = yes ; -; This determines whether or not we log queue events to a file (defaults to yes). +; This determines whether or not we log queue events to a file +; (defaults to yes). ;queue_log = no ; -; This determines whether or not we log generic events to a file (defaults to yes). +; This determines whether or not we log generic events to a file +; (defaults to yes). ;event_log = no ; ; @@ -44,17 +46,16 @@ ; ; Special filename "console" represents the system console ; -; We highly recommend that you DO NOT turn on debug mode if you -; are simply running a production system. Debug mode turns on a -; LOT of extra messages, most of which you are unlikely to understand -; without an understanding of the underlying code. Do NOT report -; debug messages as code issues, unless you have a specific issue that -; you are attempting to debug. They are messages for just that -- -; debugging -- and do not rise to the level of something that merit -; your attention as an Asterisk administrator. Debug messages are also -; very verbose and can and do fill up logfiles quickly; this is another -; reason not to have debug mode on a production system unless you are -; in the process of debugging a specific issue. +; We highly recommend that you DO NOT turn on debug mode if you are simply +; running a production system. Debug mode turns on a LOT of extra messages, +; most of which you are unlikely to understand without an understanding of +; the underlying code. Do NOT report debug messages as code issues, unless +; you have a specific issue that you are attempting to debug. They are +; messages for just that -- debugging -- and do not rise to the level of +; something that merit your attention as an Asterisk administrator. Debug +; messages are also very verbose and can and do fill up logfiles quickly; +; this is another reason not to have debug mode on a production system unless +; you are in the process of debugging a specific issue. ; ;debug => debug console => notice,warning,error Index: configs/manager.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/manager.conf.sample,v retrieving revision 1.5 diff -u -r1.5 manager.conf.sample --- configs/manager.conf.sample 17 Mar 2005 15:56:55 -0000 1.5 +++ configs/manager.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -1,23 +1,19 @@ ; ; AMI - The Asterisk Manager Interface ; -; Third party application call management support -; and PBX event supervision +; Third party application call management support and PBX event supervision ; -; This configuration file is read every time someone -; logs in +; This configuration file is read every time someone logs in ; -; Use the "show manager commands" at the CLI to list -; availabale manager commands and their authorization -; levels. +; Use the "show manager commands" at the CLI to list available manager commands +; and their authorization levels. ; ; "show manager command " will show a help text. ; -; ------------------- SECURITY NOTE ----------------- -; Note that you should not enable the AMI on a public -; IP address. If needed, block this TCP port with -; iptables (or another FW software) and reach it -; with IPsec, SSH or SSL vpn tunnel +; ---------------------------- SECURITY NOTE ------------------------------- +; Note that you should not enable the AMI on a public IP address. If needed, +; block this TCP port with iptables (or another FW software) and reach it +; with IPsec, SSH, or SSL vpn tunnel ; [general] enabled = no Index: configs/meetme.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/meetme.conf.sample,v retrieving revision 1.4 diff -u -r1.4 meetme.conf.sample --- configs/meetme.conf.sample 17 Mar 2005 15:56:55 -0000 1.4 +++ configs/meetme.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -1,6 +1,5 @@ ; -; Configuration file for MeetMe simple conference rooms -; for Asterisk of course. +; Configuration file for MeetMe simple conference rooms for Asterisk of course. ; ; This configuration file is read every time you call app meetme() ; @@ -10,3 +9,4 @@ ; ;conf => 1234 ;conf => 2345,9938 + Index: configs/mgcp.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/mgcp.conf.sample,v retrieving revision 1.11 diff -u -r1.11 mgcp.conf.sample --- configs/mgcp.conf.sample 5 Apr 2005 21:40:37 -0000 1.11 +++ configs/mgcp.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -45,7 +45,8 @@ ; ;context=local ;host=dynamic -;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or 'hybrid' which starts in none and moves to inband. Default is none. +;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or + ; 'hybrid' which starts in none and moves to inband. Default is none. ;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing ;line => aaln/1 Index: configs/modules.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/modules.conf.sample,v retrieving revision 1.6 diff -u -r1.6 modules.conf.sample --- configs/modules.conf.sample 5 Jul 2005 22:11:42 -0000 1.6 +++ configs/modules.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -7,11 +7,12 @@ [modules] autoload=yes ; -; Any modules that need to be loaded before the Asterisk core has been initialized -; (just after the logger has been initialized) can be loaded using 'preload'. This -; will frequently be needed if you wish to map all module configuration files into -; Realtime storage, since the Realtime driver will need to be loaded before the -; modules using those configuration files are initialized. +; Any modules that need to be loaded before the Asterisk core has been +; initialized (just after the logger has been initialized) can be loaded +; using 'preload'. This will frequently be needed if you wish to map all +; module configuration files into Realtime storage, since the Realtime +; driver will need to be loaded before the modules using those configuration +; files are initialized. ; ; An example of loading ODBC support would be: ;preload => res_odbc.so Index: configs/musiconhold.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/musiconhold.conf.sample,v retrieving revision 1.9 diff -u -r1.9 musiconhold.conf.sample --- configs/musiconhold.conf.sample 25 Aug 2005 16:11:46 -0000 1.9 +++ configs/musiconhold.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -26,7 +26,8 @@ ;application=/usr/bin/streamplayer 192.168.100.52 888 ;format=ulaw -; mpg123 on Solaris does not always exit properly; madplay may be a better choice +; mpg123 on Solaris does not always exit properly; madplay may be a better +; choice ;[solaris] ;mode=custom ;directory=/var/lib/asterisk/mohmp3 Index: configs/queues.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/queues.conf.sample,v retrieving revision 1.31 diff -u -r1.31 queues.conf.sample --- configs/queues.conf.sample 2 Sep 2005 19:27:01 -0000 1.31 +++ configs/queues.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -9,8 +9,8 @@ ; persistentmembers = yes ; -; Note that a timeout to fail out of a queue may be passed as part of application call -; from extensions.conf: +; Note that a timeout to fail out of a queue may be passed as part of +; an application call from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) @@ -43,7 +43,8 @@ ;strategy = ringall ; ; Second settings for service level (default 0) -; Used for service level statistics (calls answered within service level time frame) +; Used for service level statistics (calls answered within service level time +; frame) ;servicelevel = 60 ; ; A context may be specified, in which if the user types a SINGLE @@ -94,7 +95,8 @@ ; ; What's the rounding time for the seconds? -; If this is non zero then we announce the seconds as well as the minutes rounded to this value +; If this is non-zero, then we announce the seconds as well as the minutes +; rounded to this value. ; ; announce-round-seconds = 10 ; @@ -119,26 +121,29 @@ ; To enable monitoring, simply specify "monitor-format"; it will be disabled ; otherwise. ; -; You can specify the monitor filename with by calling Set(MONITOR_FILENAME=foo) -; Otherwise it will use ${UNIQUEID} +; You can specify the monitor filename with by calling +; Set(MONITOR_FILENAME=foo) +; Otherwise it will use MONITOR_FILENAME=${UNIQUEID} ; ; monitor-format = gsm|wav|wav49 ; -; If you wish to have the two files joined together when the call ends set this to yes +; If you wish to have the two files joined together when the call ends, set this +; to yes. ; ; monitor-join = yes ; -; This setting controls whether callers can join a queue with no members. There are three -; choices: +; This setting controls whether callers can join a queue with no members. There +; are three choices: ; -; yes - callers can join a queue with no members or only unavailable members -; no - callers cannot join a queue with no members -; strict - callers cannot join a queue with no members or only unavailable members +; yes - callers can join a queue with no members or only unavailable members +; no - callers cannot join a queue with no members +; strict - callers cannot join a queue with no members or only unavailable +; members ; ; joinempty = yes ; -; If you wish to remove callers from the queue when new callers cannot join, set this setting -; to one of the same choices for 'joinempty' +; If you wish to remove callers from the queue when new callers cannot join, +; set this setting to one of the same choices for 'joinempty' ; ; leavewhenempty = yes ; @@ -155,14 +160,15 @@ ; ; eventmemberstatusoff = no ; -; If you wish to report the caller's hold time to the member before they are connected -; to the caller, set this to yes. +; If you wish to report the caller's hold time to the member before they are +; connected to the caller, set this to yes. ; ; reportholdtime = no ; ; -; If you wish to have a delay before the member is connected to the caller (or before the member -; hears any announcement messages), set this to the number of seconds to delay. +; If you wish to have a delay before the member is connected to the caller (or +; before the member hears any announcement messages), set this to the number of +; seconds to delay. ; ; memberdelay = 0 ; Index: configs/sip.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v retrieving revision 1.71 diff -u -r1.71 sip.conf.sample --- configs/sip.conf.sample 27 Sep 2005 01:54:17 -0000 1.71 +++ configs/sip.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -91,12 +91,11 @@ ; the moment the channel loads this configuration ; -; If regcontext is specified, Asterisk will dynamically -; create and destroy a NoOp priority 1 extension for a given -; peer who registers or unregisters with us. The actual extension -; is the 'regexten' parameter of the registering peer or its -; name if 'regexten' is not provided. More than one regexten may be supplied -; if they are separated by '&'. Patterns may be used in regexten. +; If regcontext is specified, Asterisk will dynamically create and destroy a +; NoOp priority 1 extension for a given peer who registers or unregisters with +; us. The actual extension is the 'regexten' parameter of the registering +; peer or its name if 'regexten' is not provided. More than one regexten may +; be supplied if they are separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations ; @@ -104,12 +103,12 @@ ; Format for the register statement is: ; register => user[:secret[:authuser]]@host[:port][/extension] ; -; If no extension is given, the 's' extension is used. The extension -; needs to be defined in extensions.conf to be able to accept calls -; from this SIP proxy (provider) +; If no extension is given, the 's' extension is used. The extension needs to +; be defined in extensions.conf to be able to accept calls from this SIP proxy +; (provider). ; -; host is either a host name defined in DNS or the name of a -; section defined below. +; host is either a host name defined in DNS or the name of a section defined +; below. ; ; Examples: ; @@ -120,12 +119,13 @@ ; ;register => 2345:password@sip_proxy/1234 ; -; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local -; extension 1234 in extensions.conf default context, unless you define -; unless you configure a [sip_proxy] section below, and configure a context. -; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] -; Tip 2: Use separate type=peer and type=user sections for SIP providers -; (instead of type=friend) if you have calls in both directions +; Register 2345 at sip provider 'sip_proxy'. Calls from this provider +; connect to local extension 1234 in extensions.conf, default context, +; unless you configure a [sip_proxy] section below, and configure a +; context. +; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] +; Tip 2: Use separate type=peer and type=user sections for SIP providers +; (instead of type=friend) if you have calls in both directions ;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registerattempts=10 ; Number of registration attempts before we give up @@ -134,9 +134,9 @@ ; Default is 10 tries ;callevents=no ; generate manager events when sip ua performs events (e.g. hold) -;---------------------------------------------- NAT SUPPORT ------------------------ -; The externip, externhost and localnet settings are used if you use Asterisk behind -; a NAT device to communicate with services on the outside. +;----------------------------------------- NAT SUPPORT ------------------------ +; The externip, externhost and localnet settings are used if you use Asterisk +; behind a NAT device to communicate with services on the outside. ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT @@ -159,10 +159,10 @@ ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network ; The nat= setting is used when Asterisk is on a public IP, communicating with -; devices hidden behind a NAT device (broadband router). -; If you have one-way audio problems, you usually have problems with your NAT -; configuration or your firewalls support of SIP+RTP ports. -; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf +; devices hidden behind a NAT device (broadband router). If you have one-way +; audio problems, you usually have problems with your NAT configuration or your +; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP +; ports for incoming audio in rtp.conf ; ;nat=no ; Global NAT settings (Affects all peers and users) ; yes = Always ignore info and assume NAT @@ -225,7 +225,7 @@ ; You may also add auth= statements to [peer] definitions ; Peer auth= override all other authentication settings if we match on realm -;----------------------------------------------------------------------------------- +;------------------------------------------------------------------------------ ; Users and peers have different settings available. Friends have all settings, ; since a friend is both a peer and a user ; @@ -251,10 +251,10 @@ ; useclientcode useclientcode ; accountcode accountcode ; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit -; restrictcid restrictcid +; callerid callerid +; amaflags amaflags +; call-limit call-limit +; restrictcid restrictcid ; mailbox ; username ; template @@ -323,6 +323,7 @@ ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729 ; Pass-thru only unless g729 license obtained +;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists ;[xlite1] Index: configs/voicemail.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/voicemail.conf.sample,v retrieving revision 1.53 diff -u -r1.53 voicemail.conf.sample --- configs/voicemail.conf.sample 1 Oct 2005 01:24:15 -0000 1.53 +++ configs/voicemail.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -10,8 +10,8 @@ ;serveremail=asterisk@linux-support.net ; Should the email contain the voicemail as an attachment attach=yes -; Maximum number of messages per folder. If not specified a default value (100) is used. -; Maximum value for this option is 9999. +; Maximum number of messages per folder. If not specified, a default value +; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 @@ -28,13 +28,12 @@ silencethreshold=128 ; Max number of failed login attempts maxlogins=3 -; If you need to have an external program, i.e. /usr/bin/myapp -; called when a voicemail is left, delivered, or your voicemailbox -; is checked, uncomment this: +; If you need to have an external program, i.e. /usr/bin/myapp called when a +; voicemail is left, delivered, or your voicemailbox is checked, uncomment +; this: ;externnotify=/usr/bin/myapp -; If you need to have an external program, i.e. /usr/bin/myapp -; called when a voicemail password is changed, -; uncomment this: +; If you need to have an external program, i.e. /usr/bin/myapp called when a +; voicemail password is changed, uncomment this: ;externpass=/usr/bin/myapp ; For the directory, you can override the intro file if you want ;directoryintro=dir-intro @@ -54,13 +53,15 @@ ;usedirectory=yes ; ; Change the from, body and/or subject, variables: -; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE +; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, +; VM_CIDNAME, VM_DATE ; -; Note: The emailbody config row can be up to 512 characters due to a limitation in -; asterisk config files. +; Note: The emailbody config row can only be up to 512 characters due to a +; limitation in the Asterisk configuration subsystem. ;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} -; The following definition is very close to the default, but the default shows just -; the CIDNAME, if it is not null, else just the CIDNUM, or "an unknown caller" if they are both null. +; The following definition is very close to the default, but the default shows +; just the CIDNAME, if it is not null, otherise just the CIDNUM, or "an unknown +; caller", if they are both null. ;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n ; ; You can also change the Pager From: string, the pager body and/or subject. @@ -69,7 +70,8 @@ ;pagersubject=New VM ;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE} ; -; Set the date format on outgoing mails. Valid arguments can be found on the strftime(3) man page +; Set the date format on outgoing mails. Valid arguments can be found on the +; strftime(3) man page ; ; Default emaildateformat=%A, %B %d, %Y at %r @@ -93,7 +95,8 @@ ; variable substitution is done on the values below. ; ; Supported values: -; 'filename' filename of a soundfile (single ticks around the filename required) +; 'filename' filename of a soundfile (single ticks around the filename +; required) ; ${VAR} variable substitution ; A or a Day of week (Saturday, Sunday, ...) ; B or b or h Month name (January, February, ...) @@ -105,8 +108,10 @@ ; M Minute, with 00 pronounced as "o'clock" ; N Minute, with 00 pronounced as "hundred" (US military time) ; P or p AM or PM -; Q "today", "yesterday" or ABdY (*note: not standard strftime value) -; q "" (for today), "yesterday", weekday, or ABdY (*note: not standard strftime value) +; Q "today", "yesterday" or ABdY +; (*note: not standard strftime value) +; q "" (for today), "yesterday", weekday, or ABdY +; (*note: not standard strftime value) ; R 24 hour time, including minute ; ; @@ -114,11 +119,13 @@ ; ; Each mailbox is listed in the form =,,,, ; if the e-mail is specified, a message will be sent when a message is -; received, to the given mailbox. If pager is specified, a message will be sent there as well. If the password is prefixed by '-' then it is considered to be unchangable +; received, to the given mailbox. If pager is specified, a message will be +; sent there as well. If the password is prefixed by '-', then it is +; considered to be unchangable. ; ; Advanced options example is extension 4069 -; NOTE: All options can be expressed globally in the general section, and overriden in the per-mailbox -; settings, unless listed otherwise. +; NOTE: All options can be expressed globally in the general section, and +; overriden in the per-mailbox settings, unless listed otherwise. ; ; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no. ; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email Index: configs/vpb.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/vpb.conf.sample,v retrieving revision 1.11 diff -u -r1.11 vpb.conf.sample --- configs/vpb.conf.sample 22 Jun 2005 23:54:47 -0000 1.11 +++ configs/vpb.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -1,17 +1,28 @@ +; ; V6PCI/V12PCI config file for VoiceTronix Hardware -; Options -; For [general] section +; +; Options for [general] section +; ; type = v12pci|v6pci|v4pci ; cards = number of cards -; indication = 1 ( To use Asterisk indication tones) -; ecsuppthres = 0|2048|4096 (none,-24db,-18db only for use with OpenLine4) -; dtmfidd = 3000 (Inter Digit Delay timeout for when collecting DTMF tones for dialling from a Station port, in ms) -; ast-dtmf-det=1 ( To use Asterisk DTMF detection ) -; relaxdtmf=1 ( Used with ast-dtmf-det ) -; break-for-dtmf=no (When a native bridge occurs between 2 vpb channels, it will only break the connection for '#' and '*') -; timer_period_ring=4000 (Set the maximum period between received rings, default 4000ms) +; To use Asterisk indication tones +; indication = 1 +; none,-24db,-18db only for use with OpenLine4 +; ecsuppthres = 0|2048|4096 +; Inter Digit Delay timeout for when collecting DTMF tones for dialling +; from a Station port, in ms +; dtmfidd = 3000 +; To use Asterisk DTMF detection +; ast-dtmf-det=1 +; Used with ast-dtmf-det +; relaxdtmf=1 +; When a native bridge occurs between 2 vpb channels, it will only break +; the connection for '#' and '*' +; break-for-dtmf=no +; Set the maximum period between received rings, default 4000ms +; timer_period_ring=4000 ; -; For [interface] section +; Options for [interface] section ; board = board_number (1, 2, 3, ...) ; channel = channel_number (1,2,3...) ; mode = fxo|immediate|dialtone -- for type of line and line handling Index: configs/zapata.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/zapata.conf.sample,v retrieving revision 1.53 diff -u -r1.53 zapata.conf.sample --- configs/zapata.conf.sample 1 Sep 2005 19:02:37 -0000 1.53 +++ configs/zapata.conf.sample 4 Oct 2005 18:54:51 -0000 @@ -103,9 +103,9 @@ ;privateprefix = +497115678 ;unknownprefix = ; -; PRI resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 -; minimum 60 seconds -; some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 100000000 +; PRI resetinterval: sets the time in seconds between restart of unused +; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like +; channel restarts. so set the interval to a very long interval e.g. 100000000 ; or 'never' to disable *entirely*. ; ;resetinterval = 3600 @@ -129,58 +129,66 @@ ; priexclusive = yes ; ; ISDN Timers -; All of the ISDN timers and counters that are used are configurable. Specify -; the timer name, and its value (in ms for timers) +; All of the ISDN timers and counters that are used are configurable. Specify +; the timer name, and its value (in ms for timers). ; ; pritimer => t200,1000 ; pritimer => t313,4000 ; ; To enable transmission of facility-based ISDN supplementary services (such -; as caller name from CPE over facility) enable this option. +; as caller name from CPE over facility), enable this option. ; facilityenable = yes ; ; ; Signalling method (default is fxs). Valid values: -; em: E & M -; em_w: E & M Wink -; featd: Feature Group D (The fake, Adtran style, DTMF) -; featdmf: Feature Group D (The real thing, MF (domestic, US)) -; featdmf_ta : Feature Group D (The real thing, MF (domestic, US)) through a Tandem Access point -; featb: Feature Group B (MF (domestic, US)) -; fxs_ls: FXS (Loop Start) -; fxs_gs: FXS (Ground Start) -; fxs_ks: FXS (Kewl Start) -; fxo_ls: FXO (Loop Start) -; fxo_gs: FXO (Ground Start) -; fxo_ks: FXO (Kewl Start) -; pri_cpe: PRI signalling, CPE side -; pri_net: PRI signalling, Network side +; em: E & M +; em_w: E & M Wink +; featd: Feature Group D (The fake, Adtran style, DTMF) +; featdmf: Feature Group D (The real thing, MF (domestic, US)) +; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through +; a Tandem Access point +; featb: Feature Group B (MF (domestic, US)) +; fxs_ls: FXS (Loop Start) +; fxs_gs: FXS (Ground Start) +; fxs_ks: FXS (Kewl Start) +; fxo_ls: FXO (Loop Start) +; fxo_gs: FXO (Ground Start) +; fxo_ks: FXO (Kewl Start) +; pri_cpe: PRI signalling, CPE side +; pri_net: PRI signalling, Network side ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side -; sf: SF (Inband Tone) Signalling -; sf_w: SF Wink -; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) -; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) -; sf_featb: SF Feature Group B (MF (domestic, US)) -; e911: E911 (MF) style signalling +; sf: SF (Inband Tone) Signalling +; sf_w: SF Wink +; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) +; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) +; sf_featb: SF Feature Group B (MF (domestic, US)) +; e911: E911 (MF) style signalling +; ; The following are used for Radio interfaces: -; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank) -; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank) -; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank) -; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank) -; em_rx: Receive audio/COR on an E&M interface (1-way) -; em_tx: Transmit audio/PTT on an E&M interface (1-way) -; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way) -; em_rxtx: same as em_txrx (for our dyslexic friends) -; sf_rx: Receive audio/COR on an SF interface (1-way) -; sf_tx: Transmit audio/PTT on an SF interface (1-way) -; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way) -; sf_rxtx: same as sf_txrx (for our dyslexic friends) +; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the +; channel bank) +; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the +; channel bank) +; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the +; channel bank) +; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at +; the channel bank) +; em_rx: Receive audio/COR on an E&M interface (1-way) +; em_tx: Transmit audio/PTT on an E&M interface (1-way) +; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface +; (2-way) +; em_rxtx: Same as em_txrx (for our dyslexic friends) +; sf_rx: Receive audio/COR on an SF interface (1-way) +; sf_tx: Transmit audio/PTT on an SF interface (1-way) +; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface +; (2-way) +; sf_rxtx: Same as sf_txrx (for our dyslexic friends) ; signalling=fxo_ls ; -; For Feature Group D Tandem access, to set the default CIC and OZZ use -; these parameters: +; For Feature Group D Tandem access, to set the default CIC and OZZ use these +; parameters: ;defaultozz=0000 ;defaultcic=303 ; @@ -197,7 +205,8 @@ ; rxwink=300 ; Atlas seems to use long (250ms) winks ; -; How long generated tones (DTMF and MF) will be played on the channel (in miliseconds) +; How long generated tones (DTMF and MF) will be played on the channel +; (in miliseconds) ;toneduration=100 ; ; Whether or not to do distinctive ring detection on FXO lines @@ -210,12 +219,15 @@ usecallerid=yes ; ; Type of caller ID signalling in use -; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands +; bell = bell202 as used in US +; v23 = v23 as used in the UK +; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; ;cidsignalling=bell ; ; What signals the start of caller ID -; ring = a ring signals the start, polarity = polarity reversal signals the start +; ring = a ring signals the start +; polarity = polarity reversal signals the start ; ;cidstart=ring ; @@ -227,12 +239,14 @@ ; callwaiting=yes ; -; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user) +; Whether or not restrict outgoing caller ID (will be sent as ANI only, not +; available for the user) ; Mostly use with FXS ports ; ;restrictcid=no ; -; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending +; Whether or not use the caller ID presentation for the outgoing call that the +; calling switch is sending. ; usecallingpres=yes ; @@ -271,31 +285,29 @@ ; ; Stutter dialtone support: If a mailbox is specified without a voicemail ; context, then when voicemail is received in a mailbox in the default -; voicemail context in voicemail.conf, taking the phone off hook will -; cause a stutter dialtone instead of a normal one. +; voicemail context in voicemail.conf, taking the phone off hook will cause a +; stutter dialtone instead of a normal one. ; -; If a mailbox is specified *with* a voicemail context, the same will -; result if voicemail recieved in mailbox in the specified voicemail -; context +; If a mailbox is specified *with* a voicemail context, the same will result +; if voicemail recieved in mailbox in the specified voicemail context. ; ; for default voicemail context, the example below is fine: ; ;mailbox=1234 ; -; for any other voicemail context, the following will produce the -; stutter tone: +; for any other voicemail context, the following will produce the stutter tone: ; ;mailbox=1234@context ; ; Enable echo cancellation -; Use either "yes", "no", or a power of two from 32 to 256 if you wish -; to actually set the number of taps of cancellation. +; Use either "yes", "no", or a power of two from 32 to 256 if you wish to +; actually set the number of taps of cancellation. ; echocancel=yes ; -; Generally, it is not necessary (and in fact undesirable) to echo cancel -; when the circuit path is entirely TDM. You may, however, reverse this -; behavior by enabling the echo cancel during pure TDM bridging below. +; Generally, it is not necessary (and in fact undesirable) to echo cancel when +; the circuit path is entirely TDM. You may, however, reverse this behavior +; by enabling the echo cancel during pure TDM bridging below. ; echocancelwhenbridged=yes ; @@ -309,10 +321,9 @@ ;echotraining=yes ;echotraining=800 ; -; If you are having trouble with DTMF detection, you can relax the -; DTMF detection parameters. Relaxing them may make the DTMF detector -; more likely to have "talkoff" where DTMF is detected when it -; shouldn't be. +; If you are having trouble with DTMF detection, you can relax the DTMF +; detection parameters. Relaxing them may make the DTMF detector more likely +; to have "talkoff" where DTMF is detected when it shouldn't be. ; ;relaxdtmf=yes ; @@ -321,8 +332,8 @@ rxgain=0.0 txgain=0.0 ; -; Logical groups can be assigned to allow outgoing rollover. Groups -; range from 0 to 63, and multiple groups can be specified. +; Logical groups can be assigned to allow outgoing rollover. Groups range +; from 0 to 63, and multiple groups can be specified. ; group=1 ; @@ -335,19 +346,18 @@ pickupgroup=1 ; -; Specify whether the channel should be answered immediately or -; if the simple switch should provide dialtone, read digits, etc. +; Specify whether the channel should be answered immediately or if the simple +; switch should provide dialtone, read digits, etc. ; immediate=no ; -; Specify whether flash-hook transfers to 'busy' channels should complete -; or return to the caller performing the transfer (default is yes). +; Specify whether flash-hook transfers to 'busy' channels should complete or +; return to the caller performing the transfer (default is yes). ; ;transfertobusy=no ; -; CallerID can be set to "asreceived" or a specific number -; if you want to override it. Note that "asreceived" only -; applies to trunk interfaces. +; CallerID can be set to "asreceived" or a specific number if you want to +; override it. Note that "asreceived" only applies to trunk interfaces. ; ;callerid=2564286000 ; @@ -373,39 +383,36 @@ ; ;busydetect=yes ; -; If busydetect is enabled, is also possible to specify how many -; busy tones to wait for before hanging up. The default is 4, but -; better results can be achieved if set to 6 or even 8. Mind that -; higher the number, more time is needed to hangup a channel, but -; lower is probability to get random hangups +; If busydetect is enabled, it is also possible to specify how many busy tones +; to wait for before hanging up. The default is 4, but better results can be +; achieved if set to 6 or even 8. Mind that the higher the number, the more +; time that will be needed to hangup a channel, but lowers the probability +; that you will get random hangups. ; ;busycount=4 ; -; If busydetect is enabled, is also possible to specify the -; cadence of your busy signal. In many countries it is 500mec -; on, 500msec off. -; Without busypattern specified, we'll accept any regular -; sound-silence pattern than repeats busycount times as a busy -; signal. -; If you specify busypattern then we'll further check the length -; of the sound (tone) and silence, which will further reduce the -; chance of a false positive. +; If busydetect is enabled, it is also possible to specify the cadence of your +; busy signal. In many countries, it is 500msec on, 500msec off. Without +; busypattern specified, we'll accept any regular sound-silence pattern that +; repeats times as a busy signal. If you specify busypattern, +; then we'll further check the length of the sound (tone) and silence, which +; will further reduce the chance of a false positive. ; ;busypattern=500,500 ; -; NOTE: In the Asterisk Makefile you'll find further options to tweak -; the busy detector. If your country has a busy tone with the same -; lengh tone and silence (as many countries do), consider defining -; the -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option. +; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy +; detector. If your country has a busy tone with the same length tone and +; silence (as many countries do), consider defining the +; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option. ; ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes ; -; In some countries, a polarity reversal is used to signal the disconnect -; of a phone line. If the hanguponpolarityswitch option is selected, the -; call will be considered "hung up" on a polarity reversal +; In some countries, a polarity reversal is used to signal the disconnect of a +; phone line. If the hanguponpolarityswitch option is selected, the call will +; be considered "hung up" on a polarity reversal. ; ;hanguponpolarityswitch=yes ; @@ -413,13 +420,13 @@ ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, -; so don't count on it being very accurate. +; so don't count on it being very accurate. ; -; Few zones are supported at the time of this writing, but may -; be selected with "progzone" +; Few zones are supported at the time of this writing, but may be selected +; with "progzone" ; -; This feature can also easily detect false hangups. The symptoms of this -; is being disconnected in the middle of a call for no reason. +; This feature can also easily detect false hangups. The symptoms of this is +; being disconnected in the middle of a call for no reason. ; ;callprogress=yes ;progzone=us @@ -446,15 +453,15 @@ ; ;musiconhold=default ; -; PRI channels can have an idle extension and a minunused number. So long -; as at least "minunused" channels are idle, chan_zap will try to call -; "idledial" on them, and then dump them into the PBX in the "idleext" -; extension (which is of the form exten@context). When channels are needed -; the "idle" calls are disconnected (so long as there are at least "minidle" -; calls still running, of course) to make more channels available. The -; primary use of this is to create a dynamic service, where idle channels -; are bundled through multilink PPP, thus more efficiently utilizing -; combined voice/data services than conventional fixed mappings/muxings. +; PRI channels can have an idle extension and a minunused number. So long as +; at least "minunused" channels are idle, chan_zap will try to call "idledial" +; on them, and then dump them into the PBX in the "idleext" extension (which +; is of the form exten@context). When channels are needed the "idle" calls +; are disconnected (so long as there are at least "minidle" calls still +; running, of course) to make more channels available. The primary use of +; this is to create a dynamic service, where idle channels are bundled through +; multilink PPP, thus more efficiently utilizing combined voice/data services +; than conventional fixed mappings/muxings. ; ;idledial=6999 ;idleext=6999@dialout @@ -465,10 +472,10 @@ ; ;jitterbuffers=4 ; -; You can define your own custom ring cadences here. You can define up to -; 8 pairs. If the silence is negative, it indicates where the callerid -; spill is to be placed. Also, if you define any custom cadences, the -; default cadences will be turned off. +; You can define your own custom ring cadences here. You can define up to 8 +; pairs. If the silence is negative, it indicates where the callerid spill is +; to be placed. Also, if you define any custom cadences, the default cadences +; will be turned off. ; ; Syntax is: cadence=ring,silence[,ring,silence[...]] ; @@ -479,11 +486,11 @@ ;cadence=125,125,125,125,125,-4000 ;cadence=1000,500,2500,-5000 ; -; Each channel consists of the channel number or range. It -; inherits the parameters that were specified above its declaration +; Each channel consists of the channel number or range. It inherits the +; parameters that were specified above its declaration. ; -; For GR-303, CRV's are created like channels except they must start -; with the trunk group followed by a colon, e.g.: +; For GR-303, CRV's are created like channels except they must start with the +; trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 @@ -506,9 +513,8 @@ ;callerid="Main TA 750" <(256) 428-6127> ;channel => 44 ; -; For example, maybe we have some other channels -; which start out in a different context and use -; E & M signalling instead. +; For example, maybe we have some other channels which start out in a +; different context and use E & M signalling instead. ; ;context=remote ;sigalling=em @@ -538,9 +544,9 @@ ;callerid="Larry Moe" <(256) 428-6234> ;channel => 28 ; -; Sample PRI (CPE) config: Specify the switchtype, the signalling as -; either pri_cpe or pri_net for CPE or Network termination, and generally -; you will want to create a single "group" for all channels of the PRI. +; Sample PRI (CPE) config: Specify the switchtype, the signalling as either +; pri_cpe or pri_net for CPE or Network termination, and generally you will +; want to create a single "group" for all channels of the PRI. ; ; switchtype = national ; signalling = pri_cpe