[Asterisk-cvs] asterisk-addons/asterisk-ooh323c COPYING, NONE,
1.1 ChangeLog, NONE, 1.1 INSTALL, NONE, 1.1 Makefile.am, NONE,
1.1 Makefile.in, NONE, 1.1 NEWS, NONE, 1.1 README, NONE,
1.1 config.h.in, NONE, 1.1 configure, NONE, 1.1 configure.in,
NONE, 1.1 extensions.conf.sample, NONE, 1.1 h323.conf.sample,
NONE, 1.1
vphirke at lists.digium.com
vphirke at lists.digium.com
Mon May 23 16:47:56 CDT 2005
Update of /usr/cvsroot/asterisk-addons/asterisk-ooh323c
In directory mongoose.digium.com:/tmp/cvs-serv20168
Added Files:
COPYING ChangeLog INSTALL Makefile.am Makefile.in NEWS README
config.h.in configure configure.in extensions.conf.sample
h323.conf.sample
Log Message:
Added
--- NEW FILE: COPYING ---
GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
Copyright (C) 1989, 1991 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
Preamble
The licenses for most software are designed to take away your
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When we speak of free software, we are referring to freedom, not
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TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains
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How to Apply These Terms to Your New Programs
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<one line to give the program's name and a brief idea of what it does.>
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Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this
when it starts in an interactive mode:
Gnomovision version 69, Copyright (C) year name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, the commands you use may
be called something other than `show w' and `show c'; they could even be
mouse-clicks or menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Library General
Public License instead of this License.
--- NEW FILE: ChangeLog ---
Added support for gsm, g729a, g723.1
Added call end cause conversion logic from H323 cause to asterisk and viceversa
First Release 0.1
Can do basic call setup stuff.
--- NEW FILE: INSTALL ---
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opt:
$(MAKE) "CFLAGS = -DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -O2" all
debug:
$(MAKE) "CFLAGS = -g -DGNU -D_GNU_SOURCE -D_REENTRANT" all
sample:
cp h323.conf.sample /etc/asterisk/h323.conf
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
--- NEW FILE: NEWS ---
--- NEW FILE: README ---
Copyright (C) 2005, Objective Systems, Inc.
Objective Systems asterisk-ooh323c driver for supporting H323 on asterisk
Thank you for downloading Objective Systems H323 driver for asterisk. This
package contains a driver which will enable ooh323c, a C based open source
H.323 stack from Objective Systems(www.obj-sys.com/open), to be used with
asterisk, an open source IP-PBX solution, from Digium, Inc.(www.asterisk.org)
Capabilities supported: ulaw, gsm, g729a, g723.1, rfc2833
Package Contents:
asterisk-ooh323c-0.1
|
|-src [H323 channel driver code]
|
|_ooh323c [ooh323c stack code]
How to build:
1. To build this package successfully, you have to have latest cvs version of
asterisk installed on your system.
To get asterisk sources from cvs:
cvs -d:pserver:anoncvs at cvs.digium.com:/usr/cvsroot login
CVS Password:anoncvs
cvs -d:pserver:anoncvs at cvs.digium.com:/usr/cvsroot co asterisk
To install asterisk.(You will have to be in super user mode for this)
>cd asterisk
>make install
>make samples
2. To build asterisk-ooh323c-0.1
>cd asterisk-oh323c-0.1
>./configure
>make
3. To install chan_h323.so in /usr/libs/modules/ change to super user mode
and then run:
>make install
4. Open h323.conf.sample in asterisk-ooh323c-0.1 directory. Under [general]
you will see global configuration setting. Modify ip addresses of asterisk
server "bindaddr" to match your configuration.
5. To install sample h323.conf in /etc/asterisk
>make sample
6. Now run asterisk as
/usr/sbin/asterisk -vvvc
7. From asterisk console
To check whether H323 channel is registered properly
CLI>show channeltypes
To see all the defined H323 users.
CLI>h323 show users
To see details of a specific user
CLI>h323 show user myuser1
To see all the defined peers
CLI>h323 show peers
To see details of a specific peer
CLI>h323 show peer mypeer1
Getting Started with some simple examples:
-----------------------------------------
First stop asterisk as follows:
CLI>stop now
Now to run tests, you will have to copy extensions.conf.sample in
asterisk-ooh323c-0.1 directory to /etc/asterisk as extensions.conf. Make
sure to save a backup copy of your existing extensions.conf. Now restart
the pbx as follows:
>/usr/sbin/asterisk -vvvc
Now you have asterisk with our sample dial plan running.
Test1:
--------
Someone from outside calls into asterisk server and a recorded message is
playedback to him.
1. Now you need an h323 phone(phone1) to test. You can use any H323 phones
including NetMeeting or OpenH323 based ohphone.
You can download ohphone executable
from http://www.openh323.org/code.html#linux or
http://www.openh323.org/code.html#windows
2. Now run ohphone as follows:
>ohphone -n -ttttt -o trace.log <ip address of asterisk server>
Test2:
--------
A registered users calls into asterisk server at a particular extension and
a different messages is played back to him.
1. Use ohphone as a registered user myuser1 to call extension 100 in
asterisk as follows:
>./ohphone -n -u myuser1 -tttt -o trace.log 100@<ip of asterisk server>
The ohphone will send call to asterisk, which will identify the user as
myuser1, and use the users context to handle extension 100, which
playbacks a pre-recorded message.
Test3:
--------
A registered user calls an extension which belongs to a peer and the call is
routed to the peer's phone
1. Use one more h323 phone (phone2)as a peer.
2. Stop asterisk as:
CLI>stop now
3. Open /etc/asterisk/h323.conf. Modify ip/port of mypeer1 to match those of
the h323 phone you are going to use as peer.
4. Run your peer phone phone2 as follows:
>./ohphone -n -u mypeer1 -tttttt -o trace.log -l
This keeps phone2 in listen mode.
5. Now run asterisk as follows:
>/usr/sbin/asterisk -vvvc
6. Check details of peer for ip/port
CLI>h323 show peer mypeer1
7. Now use phone1 to make a call to asterisk at extension 101 as follows.
>./ohphone -n -u myuser1 -ttttt -o trace.log 101@<ip of asterisk>
Reporting Problems:
Report problems you encounter by sending E-mail to support at obj-sys.com.
If you have any further questions or comments on what you would like to
see in the product or what is difficult to use or understand, please
communicate them to us. Your feedback is important to us. Please let us
know how it works out for you - either good or bad.
--- NEW FILE: config.h.in ---
/* config.h.in. Generated from configure.in by autoheader. */
/* Define to 1 if you have the <arpa/inet.h> header file. */
#undef HAVE_ARPA_INET_H
/* Define to 1 if you have the <dlfcn.h> header file. */
#undef HAVE_DLFCN_H
/* Define to 1 if you don't have `vprintf' but do have `_doprnt.' */
#undef HAVE_DOPRNT
/* Define to 1 if you have the <fcntl.h> header file. */
#undef HAVE_FCNTL_H
/* Define to 1 if you have the `gethostbyname' function. */
#undef HAVE_GETHOSTBYNAME
/* Define to 1 if you have the `gethostname' function. */
#undef HAVE_GETHOSTNAME
/* Define to 1 if you have the `gettimeofday' function. */
#undef HAVE_GETTIMEOFDAY
/* Define to 1 if you have the `inet_ntoa' function. */
#undef HAVE_INET_NTOA
/* Define to 1 if you have the <inttypes.h> header file. */
#undef HAVE_INTTYPES_H
/* Define to 1 if you have the <limits.h> header file. */
#undef HAVE_LIMITS_H
/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
to 0 otherwise. */
#undef HAVE_MALLOC
/* Define to 1 if you have the `memchr' function. */
#undef HAVE_MEMCHR
/* Define to 1 if you have the <memory.h> header file. */
#undef HAVE_MEMORY_H
/* Define to 1 if you have the `memset' function. */
#undef HAVE_MEMSET
/* Define to 1 if you have the <netdb.h> header file. */
#undef HAVE_NETDB_H
/* Define to 1 if you have the <netinet/in.h> header file. */
#undef HAVE_NETINET_IN_H
/* Define to 1 if you have the `pipe' function. */
#undef HAVE_PIPE
/* Define to 1 if you have the `select' function. */
#undef HAVE_SELECT
/* Define to 1 if you have the `socket' function. */
#undef HAVE_SOCKET
/* Define to 1 if you have the <stddef.h> header file. */
#undef HAVE_STDDEF_H
/* Define to 1 if you have the <stdint.h> header file. */
#undef HAVE_STDINT_H
/* Define to 1 if you have the <stdlib.h> header file. */
#undef HAVE_STDLIB_H
/* Define to 1 if you have the `strftime' function. */
#undef HAVE_STRFTIME
/* Define to 1 if you have the <strings.h> header file. */
#undef HAVE_STRINGS_H
/* Define to 1 if you have the <string.h> header file. */
#undef HAVE_STRING_H
/* Define to 1 if you have the <sys/ioctl.h> header file. */
#undef HAVE_SYS_IOCTL_H
/* Define to 1 if you have the <sys/select.h> header file. */
#undef HAVE_SYS_SELECT_H
/* Define to 1 if you have the <sys/socket.h> header file. */
#undef HAVE_SYS_SOCKET_H
/* Define to 1 if you have the <sys/stat.h> header file. */
#undef HAVE_SYS_STAT_H
/* Define to 1 if you have the <sys/types.h> header file. */
#undef HAVE_SYS_TYPES_H
/* Define to 1 if you have the <unistd.h> header file. */
#undef HAVE_UNISTD_H
/* Define to 1 if you have the `vprintf' function. */
#undef HAVE_VPRINTF
/* Name of package */
#undef PACKAGE
/* Define to the address where bug reports for this package should be sent. */
#undef PACKAGE_BUGREPORT
/* Define to the full name of this package. */
#undef PACKAGE_NAME
/* Define to the full name and version of this package. */
#undef PACKAGE_STRING
/* Define to the one symbol short name of this package. */
#undef PACKAGE_TARNAME
/* Define to the version of this package. */
#undef PACKAGE_VERSION
/* Define to the type of arg 1 for `select'. */
#undef SELECT_TYPE_ARG1
/* Define to the type of args 2, 3 and 4 for `select'. */
#undef SELECT_TYPE_ARG234
/* Define to the type of arg 5 for `select'. */
#undef SELECT_TYPE_ARG5
/* Define to 1 if you have the ANSI C header files. */
#undef STDC_HEADERS
/* Define to 1 if you can safely include both <sys/time.h> and <time.h>. */
#undef TIME_WITH_SYS_TIME
/* Define to 1 if your <sys/time.h> declares `struct tm'. */
#undef TM_IN_SYS_TIME
/* Version number of package */
#undef VERSION
/* Define to empty if `const' does not conform to ANSI C. */
#undef const
/* Define to rpl_malloc if the replacement function should be used. */
#undef malloc
/* Define to `unsigned' if <sys/types.h> does not define. */
#undef size_t
--- NEW FILE: configure ---
#! /bin/sh
# Guess values for system-dependent variables and create Makefiles.
# Generated by GNU Autoconf 2.57 for ASTERISK-OOH323C 0.1.
#
# Report bugs to <Objective Systems <support at obj-sys.com>>.
#
# Copyright 1992, 1993, 1994, 1995, 1996, 1998, 1999, 2000, 2001, 2002
# Free Software Foundation, Inc.
# This configure script is free software; the Free Software Foundation
# gives unlimited permission to copy, distribute and modify it.
## --------------------- ##
## M4sh Initialization. ##
## --------------------- ##
# Be Bourne compatible
if test -n "${ZSH_VERSION+set}" && (emulate sh) >/dev/null 2>&1; then
emulate sh
NULLCMD=:
# Zsh 3.x and 4.x performs word splitting on ${1+"$@"}, which
[...10505 lines suppressed...]
# Unfortunately, on DOS this fails, as config.log is still kept open
# by configure, so config.status won't be able to write to it; its
# output is simply discarded. So we exec the FD to /dev/null,
# effectively closing config.log, so it can be properly (re)opened and
# appended to by config.status. When coming back to configure, we
# need to make the FD available again.
if test "$no_create" != yes; then
ac_cs_success=:
ac_config_status_args=
test "$silent" = yes &&
ac_config_status_args="$ac_config_status_args --quiet"
exec 5>/dev/null
$SHELL $CONFIG_STATUS $ac_config_status_args || ac_cs_success=false
exec 5>>config.log
# Use ||, not &&, to avoid exiting from the if with $? = 1, which
# would make configure fail if this is the last instruction.
$ac_cs_success || { (exit 1); exit 1; }
fi
--- NEW FILE: configure.in ---
AC_PREREQ(2.57)
AC_INIT([ASTERISK-OOH323C],[0.1],[Objective Systems <support at obj-sys.com>],[asterisk-ooh323c])
PACKAGE=asterisk-ooh323c
VERSION=0.1
AC_CONFIG_AUX_DIR(config)
AM_INIT_AUTOMAKE($PACKAGE, $VERSION)
AC_CONFIG_SRCDIR([src/chan_h323.c])
AM_CONFIG_HEADER(config.h)
dnl should include it in src files before using here
OOH323CDRIVERDIR=`pwd`
AC_SUBST(OOH323CDRIVERDIR)
if test "x$prefix" = "xNONE"; then
INSTALLPREFIX=.
else
INSTALLPREFIX="${prefix}"
fi
AC_SUBST(INSTALLPREFIX)
OOH323CFLAGS="-DGNU -D_GNU_SOURCE -D_OOINFO -D_REENTRANT -D_COMPACT"
AC_SUBST(OOH323CFLAGS)
AC_PROG_CC
AC_PROG_LIBTOOL
AC_LANG_C
dnl not required as using libtool AC_PROG_RANLIB
AC_HEADER_STDC
AC_CHECK_FILE(/usr/include/asterisk/channel.h,, AC_MSG_ERROR([No asterisk installation found]))
AC_CHECK_HEADERS([arpa/inet.h fcntl.h limits.h netdb.h netinet/in.h stddef.h stdlib.h string.h sys/socket.h unistd.h sys/ioctl.h])
dnl Checks for typedefs, structures, and compiler characteristics.
AC_C_CONST
AC_TYPE_SIZE_T
AC_HEADER_TIME
AC_STRUCT_TM
dnl Checks for library functions.
AC_FUNC_MALLOC
AC_FUNC_SELECT_ARGTYPES
AC_FUNC_STRFTIME
AC_FUNC_VPRINTF
AC_CHECK_FUNCS([gettimeofday inet_ntoa memchr memset select socket gethostbyname gethostname pipe])
dnl AC_OUTPUT
AC_OUTPUT(Makefile src/Makefile ooh323c/Makefile ooh323c/src/Makefile ooh323c/tests/Makefile ooh323c/tests/chansetup/Makefile)
--- NEW FILE: extensions.conf.sample ---
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
; 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
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass at provider
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; "next" or "n" means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').
; Priorities may then also have an alias, or label, in
; parenthesis after their name which can be used in goto situations
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority[+offset][(alias)],application(arg1,arg2,...)
;exten => someexten,priority[+offset][(alias)],application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;
;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
exten => s, 1, Playback(demo-congrats)
exten => s, n, Wait, 2
;exten => s, n, Hangup
exten => mypeer1, 1, Dial(H323/mypeer1)
exten => vishal, 2, Goto(s, 1)
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password at bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[context1]
exten => 100, 1, Answer
exten => 100, 2, Wait, 2
exten => 100, 3, Playback(demo-congrats)
exten => 100, 4, Wait, 2
exten => 100, 5, Hangup
exten => 101, 1, Dial(H323/mypeer1)
exten => 101, 2, Hangup
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
exten => s, 1, Answer
exten => s, 2, Wait, 2
exten => s, 3, Playback(demo-thanks)
exten => s, 4, Hangup
;include => demo
;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
; Real extensions would go here. Generally you want real extensions to be 4 or 5
; digits long (although there is no such requirement) and start with a single
; digit that is fairly large (like 6 or 7) so that you have plenty of room to
; overlap extensions and menu options without conflict. You can alias them with
; names, too and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,Voicemail(u6245) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(b6245) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at your
; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan.
;
--- NEW FILE: h323.conf.sample ---
; Objective System's H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as "dynamic" is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; H323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; H323/ip[:port] OR if gk is used H323/alias where alias can be any H323
; alias
;
; For dialing into another asterisk peer at a specific exten
; H323/exten/peer OR H323/exten at ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.
[general]
;Define the asetrisk server h323 endpoint
;The port asterisk should listen for incoming H323 connections.
;Default - 1720
;port=1720
;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it's own
bindaddr=0.0.0.0 ;UPDATE this to proper ip address of your asterisk box
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=no
;h245tunneling=no
;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
e164=100
;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk
;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE
;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;logfile=/var/log/asterisk/h323_log
;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition
;Sets default context all clients will be placed in.
;Default - default
context=default
;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay
;amaflags = default
;The account code used by default for all clients.
;accountcode=h3230101
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw and gsm supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=gsm
allow=ulaw
; dtmf mode to be used by default for all clients. Only rfc2833 supported as
; of now.
;Default - rfc 2833
dtmfmode=rfc2833
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; incominglimit outgoinglimit
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout
;
;Define users here
;Section header is extension
[myuser1]
type=user
context=context1
disallow=all
allow=gsm
allow=ulaw ; only ulaw and gsm supported as of now
incominglimit=4
[mypeer1]
type=peer
context=context2
ip=a.b.c.d ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
outgoinglimit=4
e164=101
email=mypeer1 at xyz.com
[myfriend1]
type=friend
context=context3
ip=a.b.c.d ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
disallow=all
allow=ulaw
outgoinglimit=4
incominglimit=4
email=myfriend1 at obj-sys.com
e164=12345
rtptimeout=60
dtmfmode=rfc2833
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