<p>Corey Farrell has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/7344">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">README: Convert to README.md.<br><br>Convert the README file to markdown format, remove the old README. This<br>causes websites like github to display the README in a much nicer<br>format with live links. The raw file is still very readable from<br>plain text editors and terminals.<br><br>Change-Id: I7d13131764a9a9026e5f8a6ddb245a01bbd788e7<br>---<br>D README<br>A README.md<br>2 files changed, 272 insertions(+), 296 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/44/7344/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">diff --git a/README b/README<br>deleted file mode 100644<br>index a1ccf2c..0000000<br>--- a/README<br>+++ /dev/null<br>@@ -1,296 +0,0 @@<br>-===============================================================================<br>-=== The Asterisk(R) Open Source PBX<br>-===<br>-=== by Mark Spencer <markster@digium.com><br>-=== and the Asterisk.org developer community<br>-===<br>-=== Copyright (C) 2001-2016 Digium, Inc.<br>-=== and other copyright holders.<br>-===============================================================================<br>-<br>--------------------------------------------------------------------------------<br>---- SECURITY ------------------------------------------------------------------<br>-<br>- It is imperative that you read and fully understand the contents of<br>-the security information document before you attempt to configure and run<br>-an Asterisk server.<br>-<br>- If you downloaded Asterisk as a tarball, see the security section in the PDF<br>-version of the documentation in doc/tex/asterisk.pdf. Alternatively, pull up<br>-the HTML version of the documentation in doc/tex/asterisk/index.html. The<br>-source for the security document is available in doc/tex/security.tex.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- WHAT IS ASTERISK ? --------------------------------------------------------<br>-<br>- Asterisk is an Open Source PBX and telephony toolkit. It is, in a<br>-sense, middleware between Internet and telephony channels on the bottom,<br>-and Internet and telephony applications at the top. However, Asterisk supports<br>-more telephony interfaces than just Internet telephony. Asterisk also has a<br>-vast amount of support for traditional PSTN telephony, as well. For more<br>-information on the project itself, please visit the Asterisk home page at:<br>-<br>- https://www.asterisk.org<br>-<br>- The official Asterisk wiki can be found at:<br>-<br>- https://wiki.asterisk.org<br>-<br>- In addition you'll find lots of information compiled by the Asterisk<br>-community on this Wiki:<br>-<br>- https://www.voip-info.org/wiki-Asterisk<br>-<br>- There is a book on Asterisk published by O'Reilly under the Creative Commons<br>-License. It is available in book stores as well as in a downloadable version on<br>-the http://www.asteriskdocs.org web site.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- SUPPORTED OPERATING SYSTEMS -----------------------------------------------<br>-<br>---- Linux<br>- The Asterisk Open Source PBX is developed and tested primarily on the<br>-GNU/Linux operating system, and is supported on every major GNU/Linux<br>-distribution.<br>-<br>---- Others<br>- Asterisk has also been 'ported' and reportedly runs properly on other<br>-operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,<br>-and the BSD variants.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- GETTING STARTED -----------------------------------------------------------<br>-<br>- First, be sure you've got supported hardware (but note that you don't need<br>-ANY special hardware, not even a sound card) to install and run Asterisk.<br>-<br>- Supported telephony hardware includes:<br>-<br>- * All Analog and Digital Interface cards from Digium (www.digium.com)<br>- * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)<br>- * any full duplex sound card supported by ALSA, OSS, or PortAudio<br>- * any ISDN card supported by mISDN on Linux<br>- * The Xorcom Astribank channel bank<br>- * VoiceTronix OpenLine products<br>-<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- UPGRADING FROM AN EARLIER VERSION -----------------------------------------<br>-<br>- If you are updating from a previous version of Asterisk, make sure you<br>-read the UPGRADE.txt file in the source directory. There are some files<br>-and configuration options that you will have to change, even though we<br>-made every effort possible to maintain backwards compatibility.<br>-<br>- In order to discover new features to use, please check the configuration<br>-examples in the /configs directory of the source code distribution. For a<br>-list of new features in this version of Asterisk, see the CHANGES file.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- NEW INSTALLATIONS ---------------------------------------------------------<br>-<br>- Ensure that your system contains a compatible compiler and development<br>-libraries. Asterisk requires either the GNU Compiler Collection (GCC) version<br>-3.0 or higher, or a compiler that supports the C99 specification and some of<br>-the gcc language extensions. In addition, your system needs to have the C<br>-library headers available, and the headers and libraries for ncurses.<br>-<br>- There are many modules that have additional dependencies. To see what<br>-libraries are being looked for, see ./configure --help, or run<br>-"make menuselect" to view the dependencies for specific modules.<br>-<br>- On many distributions, these dependencies are installed by packages with names<br>-like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'<br>-or similar.<br>-<br>- So, let's proceed:<br>-<br>-1) Read this README file.<br>-<br>- There are more documents than this one in the doc/ directory. You may also<br>-want to check the configuration files that contain examples and reference<br>-guides. They are all in the configs/ directory.<br>-<br>-2) Run "./configure"<br>-<br>- Execute the configure script to guess values for system-dependent<br>-variables used during compilation.<br>-<br>-3) Run "make menuselect" [optional]<br>-<br>- This is needed if you want to select the modules that will be compiled and to<br>-check dependencies for various optional modules.<br>-<br>-4) Run "make"<br>-<br>- Assuming the build completes successfully:<br>-<br>-5) Run "make install"<br>-<br>- If this is your first time working with Asterisk, you may wish to install<br>-the sample PBX, with demonstration extensions, etc. If so, run:<br>-<br>-6) "make samples"<br>-<br>- Doing so will overwrite any existing configuration files you have installed.<br>-<br>- Finally, you can launch Asterisk in the foreground mode (not a daemon) with:<br>-<br>-# asterisk -vvvc<br>-<br>- You'll see a bunch of verbose messages fly by your screen as Asterisk<br>-initializes (that's the "very very verbose" mode). When it's ready, if<br>-you specified the "c" then you'll get a command line console, that looks<br>-like this:<br>-<br>-*CLI><br>-<br>- You can type "core show help" at any time to get help with the system. For help<br>-with a specific command, type "core show help <command>". To start the PBX using<br>-your sound card, you can type "console dial" to dial the PBX. Then you can use<br>-"console answer", "console hangup", and "console dial" to simulate the actions<br>-of a telephone. Remember that if you don't have a full duplex sound card<br>-(and Asterisk will tell you somewhere in its verbose messages if you do/don't)<br>-then it won't work right (not yet).<br>-<br>- "man asterisk" at the Unix/Linux command prompt will give you detailed<br>-information on how to start and stop Asterisk, as well as all the command<br>-line options for starting Asterisk.<br>-<br>- Feel free to look over the configuration files in /etc/asterisk, where you<br>-will find a lot of information about what you can do with Asterisk.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- ABOUT CONFIGURATION FILES -------------------------------------------------<br>-<br>- All Asterisk configuration files share a common format. Comments are<br>-delimited by ';' (since '#' of course, being a DTMF digit, may occur in<br>-many places). A configuration file is divided into sections whose names<br>-appear in []'s. Each section typically contains two types of statements,<br>-those of the form 'variable = value', and those of the form 'object =><br>-parameters'. Internally the use of '=' and '=>' is exactly the same, so<br>-they're used only to help make the configuration file easier to<br>-understand, and do not affect how it is actually parsed.<br>-<br>- Entries of the form 'variable=value' set the value of some parameter in<br>-asterisk. For example, in dahdi.conf, one might specify:<br>-<br>- switchtype=national<br>-<br>- In order to indicate to Asterisk that the switch they are connecting to is<br>-of the type "national". In general, the parameter will apply to<br>-instantiations which occur below its specification. For example, if the<br>-configuration file read:<br>-<br>- switchtype = national<br>- channel => 1-4<br>- channel => 10-12<br>- switchtype = dms100<br>- channel => 25-47<br>-<br>- The "national" switchtype would be applied to channels one through<br>-four and channels 10 through 12, whereas the "dms100" switchtype would<br>-apply to channels 25 through 47.<br>-<br>- The "object => parameters" instantiates an object with the given<br>-parameters. For example, the line "channel => 25-47" creates objects for<br>-the channels 25 through 47 of the card, obtaining the settings<br>-from the variables specified above.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- SPECIAL NOTE ON TIME ------------------------------------------------------<br>-<br>- Those using SIP phones should be aware that Asterisk is sensitive to<br>-large jumps in time. Manually changing the system time using date(1)<br>-(or other similar commands) may cause SIP registrations and other<br>-internal processes to fail. If your system cannot keep accurate time<br>-by itself use NTP (http://www.ntp.org/) to keep the system clock<br>-synchronized to "real time". NTP is designed to keep the system clock<br>-synchronized by speeding up or slowing down the system clock until it<br>-is synchronized to "real time" rather than by jumping the time and<br>-causing discontinuities. Most Linux distributions include precompiled<br>-versions of NTP. Beware of some time synchronization methods that get<br>-the correct real time periodically and then manually set the system<br>-clock.<br>-<br>- Apparent time changes due to daylight savings time are just that,<br>-apparent. The use of daylight savings time in a Linux system is<br>-purely a user interface issue and does not affect the operation of the<br>-Linux kernel or Asterisk. The system clock on Linux kernels operates<br>-on UTC. UTC does not use daylight savings time.<br>-<br>- Also note that this issue is separate from the clocking of TDM<br>-channels, and is known to at least affect SIP registrations.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- FILE DESCRIPTORS ----------------------------------------------------------<br>-<br>- Depending on the size of your system and your configuration,<br>-Asterisk can consume a large number of file descriptors. In UNIX,<br>-file descriptors are used for more than just files on disk. File<br>-descriptors are also used for handling network communication<br>-(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and<br>-digital trunk hardware). Asterisk accesses many on-disk files for<br>-everything from configuration information to voicemail storage.<br>-<br>- Most systems limit the number of file descriptors that Asterisk can<br>-have open at one time. This can limit the number of simultaneous<br>-calls that your system can handle. For example, if the limit is set<br>-at 1024 (a common default value) Asterisk can handle approximately 150<br>-SIP calls simultaneously. To change the number of file descriptors<br>-follow the instructions for your system below:<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- PAM-based Linux System ----------------------------------------------------<br>-<br>- If your system uses PAM (Pluggable Authentication Modules) edit<br>-/etc/security/limits.conf. Add these lines to the bottom of the file:<br>-<br>-root soft nofile 4096<br>-root hard nofile 8196<br>-asterisk soft nofile 4096<br>-asterisk hard nofile 8196<br>-<br>-(adjust the numbers to taste). You may need to reboot the system for<br>-these changes to take effect.<br>-<br>-== Generic UNIX System ==<br>-<br>- If there are no instructions specifically adapted to your system<br>-above you can try adding the command "ulimit -n 8192" to the script<br>-that starts Asterisk.<br>--------------------------------------------------------------------------------<br>-<br>--------------------------------------------------------------------------------<br>---- MORE INFORMATION ----------------------------------------------------------<br>-<br>- See the doc directory for more documentation on various features. Again,<br>-please read all the configuration samples that include documentation on<br>-the configuration options.<br>-<br>- If this release of Asterisk was downloaded from a tarball, then some<br>-additional documentation should have been included.<br>- * doc/tex/asterisk.pdf --- PDF version of the documentation<br>- * doc/tex/asterisk/index.html --- HTML version of the documentation<br>-<br>- Finally, you may wish to visit the web site and join the mailing list if<br>-you're interested in getting more information.<br>-<br>- https://www.asterisk.org/support<br>-<br>- Welcome to the growing worldwide community of Asterisk users!<br>--------------------------------------------------------------------------------<br>-<br>---- Mark Spencer, and the Asterisk.org development community<br>-<br>--------------------------------------------------------------------------------<br>-Asterisk is a trademark of Digium, Inc.<br>diff --git a/README.md b/README.md<br>new file mode 100644<br>index 0000000..4ed9b3e<br>--- /dev/null<br>+++ b/README.md<br>@@ -0,0 +1,272 @@<br>+# The Asterisk(R) Open Source PBX<br>+```text<br>+ By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.<br>+ Copyright (C) 2001-2016 Digium, Inc. and other copyright holders.<br>+```<br>+## SECURITY<br>+<br>+ It is imperative that you read and fully understand the contents of<br>+the security information document before you attempt to configure and run<br>+an Asterisk server.<br>+<br>+ If you downloaded Asterisk as a tarball, see the security section in the PDF<br>+version of the documentation in doc/tex/asterisk.pdf. Alternatively, pull up<br>+the HTML version of the documentation in doc/tex/asterisk/index.html. The<br>+source for the security document is available in doc/tex/security.tex.<br>+<br>+## WHAT IS ASTERISK ?<br>+<br>+ Asterisk is an Open Source PBX and telephony toolkit. It is, in a<br>+sense, middleware between Internet and telephony channels on the bottom,<br>+and Internet and telephony applications at the top. However, Asterisk supports<br>+more telephony interfaces than just Internet telephony. Asterisk also has a<br>+vast amount of support for traditional PSTN telephony, as well.<br>+<br>+ For more information on the project itself, please visit the Asterisk<br>+[home page] and the official [wiki]. In addition you'll find lots<br>+of information compiled by the Asterisk community at [voip-info.org].<br>+<br>+ There is a book on Asterisk published by O'Reilly under the Creative Commons<br>+License. It is available in book stores as well as in a downloadable version on<br>+the [asteriskdocs.org] web site.<br>+<br>+## SUPPORTED OPERATING SYSTEMS<br>+<br>+### Linux<br>+<br>+ The Asterisk Open Source PBX is developed and tested primarily on the<br>+GNU/Linux operating system, and is supported on every major GNU/Linux<br>+distribution.<br>+<br>+### Others<br>+<br>+ Asterisk has also been 'ported' and reportedly runs properly on other<br>+operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,<br>+and the BSD variants.<br>+<br>+## GETTING STARTED<br>+<br>+ First, be sure you've got supported hardware (but note that you don't need<br>+ANY special hardware, not even a sound card) to install and run Asterisk.<br>+<br>+Supported telephony hardware includes:<br>+* All Analog and Digital Interface cards from [Digium]<br>+* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)<br>+* any full duplex sound card supported by ALSA, OSS, or PortAudio<br>+* any ISDN card supported by mISDN on Linux<br>+* The Xorcom Astribank channel bank<br>+* VoiceTronix OpenLine products<br>+<br>+### UPGRADING FROM AN EARLIER VERSION<br>+<br>+ If you are updating from a previous version of Asterisk, make sure you<br>+read the [UPGRADE.txt] file in the source directory. There are some files<br>+and configuration options that you will have to change, even though we<br>+made every effort possible to maintain backwards compatibility.<br>+<br>+ In order to discover new features to use, please check the configuration<br>+examples in the [configs] directory of the source code distribution. For a<br>+list of new features in this version of Asterisk, see the [CHANGES] file.<br>+<br>+### NEW INSTALLATIONS<br>+<br>+ Ensure that your system contains a compatible compiler and development<br>+libraries. Asterisk requires either the GNU Compiler Collection (GCC) version<br>+3.0 or higher, or a compiler that supports the C99 specification and some of<br>+the gcc language extensions. In addition, your system needs to have the C<br>+library headers available, and the headers and libraries for ncurses.<br>+<br>+ There are many modules that have additional dependencies. To see what<br>+libraries are being looked for, see `./configure --help`, or run<br>+`make menuselect` to view the dependencies for specific modules.<br>+<br>+ On many distributions, these dependencies are installed by packages with names<br>+like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'<br>+or similar.<br>+<br>+So, let's proceed:<br>+1. Read this file.<br>+<br>+ There are more documents than this one in the [doc] directory. You may also<br>+want to check the configuration files that contain examples and reference<br>+guides in the [configs] directory.<br>+<br>+2. Run `./configure`<br>+<br>+ Execute the configure script to guess values for system-dependent<br>+variables used during compilation.<br>+<br>+3. Run `make menuselect` _\[optional]_<br>+<br>+ This is needed if you want to select the modules that will be compiled and to<br>+check dependencies for various optional modules.<br>+<br>+4. Run `make`<br>+<br>+Assuming the build completes successfully:<br>+<br>+5. Run `make install`<br>+<br>+ If this is your first time working with Asterisk, you may wish to install<br>+the sample PBX, with demonstration extensions, etc. If so, run:<br>+<br>+6. Run `make samples`<br>+<br>+ Doing so will overwrite any existing configuration files you have installed.<br>+<br>+7. Finally, you can launch Asterisk in the foreground mode (not a daemon) with:<br>+```<br>+ # asterisk -vvvc<br>+```<br>+ You'll see a bunch of verbose messages fly by your screen as Asterisk<br>+initializes (that's the "very very verbose" mode). When it's ready, if<br>+you specified the "c" then you'll get a command line console, that looks<br>+like this:<br>+```<br>+ *CLI><br>+```<br>+ You can type "core show help" at any time to get help with the system. For help<br>+with a specific command, type "core show help <command>". To start the PBX using<br>+your sound card, you can type "console dial" to dial the PBX. Then you can use<br>+"console answer", "console hangup", and "console dial" to simulate the actions<br>+of a telephone. Remember that if you don't have a full duplex sound card<br>+(and Asterisk will tell you somewhere in its verbose messages if you do/don't)<br>+then it won't work right (not yet).<br>+<br>+ "man asterisk" at the Unix/Linux command prompt will give you detailed<br>+information on how to start and stop Asterisk, as well as all the command<br>+line options for starting Asterisk.<br>+<br>+ Feel free to look over the configuration files in `/etc/asterisk`, where you<br>+will find a lot of information about what you can do with Asterisk.<br>+<br>+### ABOUT CONFIGURATION FILES<br>+<br>+ All Asterisk configuration files share a common format. Comments are<br>+delimited by ';' (since '#' of course, being a DTMF digit, may occur in<br>+many places). A configuration file is divided into sections whose names<br>+appear in []'s. Each section typically contains two types of statements,<br>+those of the form 'variable = value', and those of the form 'object =><br>+parameters'. Internally the use of '=' and '=>' is exactly the same, so<br>+they're used only to help make the configuration file easier to<br>+understand, and do not affect how it is actually parsed.<br>+<br>+ Entries of the form 'variable=value' set the value of some parameter in<br>+asterisk. For example, in [chan_dahdi.conf], one might specify:<br>+```<br>+ switchtype=national<br>+```<br>+ In order to indicate to Asterisk that the switch they are connecting to is<br>+of the type "national". In general, the parameter will apply to<br>+instantiations which occur below its specification. For example, if the<br>+configuration file read:<br>+```<br>+ switchtype = national<br>+ channel => 1-4<br>+ channel => 10-12<br>+ switchtype = dms100<br>+ channel => 25-47<br>+```<br>+<br>+ The "national" switchtype would be applied to channels one through<br>+four and channels 10 through 12, whereas the "dms100" switchtype would<br>+apply to channels 25 through 47.<br>+<br>+ The "object => parameters" instantiates an object with the given<br>+parameters. For example, the line "channel => 25-47" creates objects for<br>+the channels 25 through 47 of the card, obtaining the settings<br>+from the variables specified above.<br>+<br>+### SPECIAL NOTE ON TIME<br>+<br>+ Those using SIP phones should be aware that Asterisk is sensitive to<br>+large jumps in time. Manually changing the system time using date(1)<br>+(or other similar commands) may cause SIP registrations and other<br>+internal processes to fail. If your system cannot keep accurate time<br>+by itself use [NTP] to keep the system clock<br>+synchronized to "real time". NTP is designed to keep the system clock<br>+synchronized by speeding up or slowing down the system clock until it<br>+is synchronized to "real time" rather than by jumping the time and<br>+causing discontinuities. Most Linux distributions include precompiled<br>+versions of NTP. Beware of some time synchronization methods that get<br>+the correct real time periodically and then manually set the system<br>+clock.<br>+<br>+ Apparent time changes due to daylight savings time are just that,<br>+apparent. The use of daylight savings time in a Linux system is<br>+purely a user interface issue and does not affect the operation of the<br>+Linux kernel or Asterisk. The system clock on Linux kernels operates<br>+on UTC. UTC does not use daylight savings time.<br>+<br>+ Also note that this issue is separate from the clocking of TDM<br>+channels, and is known to at least affect SIP registrations.<br>+<br>+### FILE DESCRIPTORS<br>+<br>+ Depending on the size of your system and your configuration,<br>+Asterisk can consume a large number of file descriptors. In UNIX,<br>+file descriptors are used for more than just files on disk. File<br>+descriptors are also used for handling network communication<br>+(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and<br>+digital trunk hardware). Asterisk accesses many on-disk files for<br>+everything from configuration information to voicemail storage.<br>+<br>+ Most systems limit the number of file descriptors that Asterisk can<br>+have open at one time. This can limit the number of simultaneous<br>+calls that your system can handle. For example, if the limit is set<br>+at 1024 (a common default value) Asterisk can handle approximately 150<br>+SIP calls simultaneously. To change the number of file descriptors<br>+follow the instructions for your system below:<br>+<br>+#### PAM-BASED LINUX SYSTEM<br>+<br>+ If your system uses PAM (Pluggable Authentication Modules) edit<br>+`/etc/security/limits.conf`. Add these lines to the bottom of the file:<br>+```text<br>+root soft nofile 4096<br>+root hard nofile 8196<br>+asterisk soft nofile 4096<br>+asterisk hard nofile 8196<br>+```<br>+<br>+(adjust the numbers to taste). You may need to reboot the system for<br>+these changes to take effect.<br>+<br>+#### GENERIC UNIX SYSTEM<br>+<br>+ If there are no instructions specifically adapted to your system<br>+above you can try adding the command `ulimit -n 8192` to the script<br>+that starts Asterisk.<br>+<br>+## MORE INFORMATION<br>+<br>+ See the [doc] directory for more documentation on various features.<br>+Again, please read all the configuration samples that include documentation<br>+on the configuration options.<br>+<br>+ Finally, you may wish to visit the [support] site and join the [mailing<br>+list] if you're interested in getting more information.<br>+<br>+Welcome to the growing worldwide community of Asterisk users!<br>+```<br>+ Mark Spencer, and the Asterisk.org development community<br>+```<br>+<br>+---<br>+<br>+Asterisk is a trademark of Digium, Inc.<br>+<br>+[home page]: https://www.asterisk.org<br>+[support]: https://www.asterisk.org/support<br>+[wiki]: https://wiki.asterisk.org/<br>+[mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users<br>+[chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample<br>+[voip-info.org]: http://www.voip-info.org/wiki-Asterisk<br>+[asteriskdocs.org]: http://www.asteriskdocs.org<br>+[NTP]: http://www.ntp.org/<br>+[Digium]: https://www.digium.com/<br>+[UPGRADE.txt]: UPGRADE.txt<br>+[CHANGES]: CHANGES<br>+[configs]: configs<br>+[doc]: doc<br>+<br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/7344">change 7344</a>. To unsubscribe, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/7344"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: I7d13131764a9a9026e5f8a6ddb245a01bbd788e7 </div>
<div style="display:none"> Gerrit-Change-Number: 7344 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Corey Farrell <git@cfware.com> </div>