[asterisk-commits] branch oej/test-this-branch r14185 - /team/oej/test-this-branch/

Wed Mar 22 13:27:29 MST 2006

Author: oej
Date: Wed Mar 22 14:27:26 2006
New Revision: 14185

URL: http://svn.digium.com/view/asterisk?rev=14185&view=rev
Log:
Add html README with links to bug reports (thanks Mike Taht)

Added:
    team/oej/test-this-branch/README.test-this-branch.html   (with props)

Added: team/oej/test-this-branch/README.test-this-branch.html
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/README.test-this-branch.html?rev=14185&view=auto
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--- team/oej/test-this-branch/README.test-this-branch.html (added)
+++ team/oej/test-this-branch/README.test-this-branch.html Wed Mar 22 14:27:26 2006
@@ -1,0 +1,245 @@
+<html>
+<head>
+<title>Test This Branch</title>
+<body>
+
+<h2>TESTING BRANCH - WELCOME!!</h2>
+
+<p>Asterisk is developed by the Asterisk.org user community. The
+development team does not only consist of coders, but also
+testers and people that write documentation and check for 
+security problems.</p>
+
+<p>This is a combined branch of many patches and branches from the
+bug tracker that needs your testing.  Please test and report
+your results in the bug tracker reports for each patch.</p>
+
+<a href=http://www.voip-forum.com/?p=189&more=1>How do I work with subversion branches?</a>
+
+<h2>What's in this branch?</h2><p><i>This branch includes the following branches</i>:<br />
+- sipdiversion: Additional support for the Diversion: header<br />
+- jitterbuffer: Jitterbuffer for RTP in chan_sip (<a href="http://bugs.digium.com/view.php?id=3854">#3854</a>, Securax/ZOA)<br />
+- videosupport: Improved support for video (<a href="http://bugs.digium.com/view.php?id=5427">#5427</a>, John Martin)<br />
+- peermatch: New peer matching algorithm (<a href="http://bugs.digium.com/view.php?id=6612">#6612</a>, oej)<br />
+- rtcp: Improved support for RTCP (<a href="http://bugs.digium.com/view.php?id=2863">#2863</a>, folsson/John Martin)<br />
+- dialplan-ami-events: Report dialplan reload in manager (<a href="http://bugs.digium.com/view.php?id=5741">#5741</a>, oej)<br />
+- sipregister: A new registration architecture (rizzo, oej <a href="http://bugs.digium.com/view.php?id=5834">#5834</a>)<br />
+- subscribemwi: Support for SIP subscription of MWI notification (oej <a href="http://bugs.digium.com/view.php?id=6390">#6390</a>)<br />
+- iptos: New IPtos support, separate audio and signalling (<a href="http://bugs.digium.com/view.php?id=6355">#6355</a>)<br />
+- multiparking: Multiple parking lots (<a href="http://bugs.digium.com/view.php?id=6113">#6113</a>)<br />
+- cdr_radius: CDR support for Radius (<a href="http://bugs.digium.com/view.php?id=6639">#6639</a>, phsultan). See doc/cdrdriver.txt <br />
+- amieventhook-561 rev 11712: Support for manager event hooks (<a href="http://bugs.digium.com/view.php?id=5161">#5161</a>, anthm)<br />
+- func_realtime: Realtime dialplan function (bweschke, <a href="http://bugs.digium.com/view.php?id=5695">#5695</a>)<br />
+- <a href=#metermaids>metermaids</a>: Subscription support for parking lots (<a href="http://bugs.digium.com/view.php?id=5779">#5779</a>)<br />
+- filenamelen: Some code changes for file name lengths (oej)<br />
+- disable-ol-and-sub: Settings for disabling sip subscriptions and overlap 
+  dialing (<a href="http://bugs.digium.com/view.php?id=6705">#6705</a>, oej) - See configs/sip.conf.sample<br />
+- t38passthrough: Support for Fax passthrough in SIP (<a href="http://bugs.digium.com/view.php?id=5090">#5090</a>, Steve Underwood)<br />
+</p>
+<i>And the following stand-alone patches:</i><p>
+- Additional options for the CHANNEL dialplan function (oej, <a href="http://bugs.digium.com/view.php?id=6670">#6670</a>)<br />
+- Manager sendtext event (ZX81, <a href="http://bugs.digium.com/view.php?id=6131">#6131</a>)<br />
+- Carrier ENUM support (otmar, <a href="http://bugs.digium.com/view.php?id=5526">#5526</a>)<br />
+- HDLC mode support for ZAP channels (crich, <a href="http://bugs.digium.com/view.php?id=6251">#6251</a>)<br />
+- Show manager CLI commands (junky, <a href="http://bugs.digium.com/view.php?id=5240">#5240</a>)<br />
+- Support SIP_CODEC for early media (oej, <a href="http://bugs.digium.com/view.php?id=6576">#6576</a>)<br />
+- Show threads CLI command (rizzo, <a href="http://bugs.digium.com/view.php?id=6053">#6053</a>)<br />
+- IFMODULE dialplan function (oej, <a href="http://bugs.digium.com/view.php?id=6671">#6671</a>)<br />
+- LDAP realtime driver (mguesdon, <a href="http://bugs.digium.com/view.php?id=5738">#5768</a>)
+  See doc/rt_ldap.txt!<br />
+- PostgreSQL realtime driver (mguesdoon, <a href="http://bugs.digium.com/view.php?id=5637">#5637</a>)<br />
+- Mute logging in remote console (mavetju, <a href="http://bugs.digium.com/view.php?id=6524">#6524</a>)<br />
+- Manager playDTMF command (squinky, <a href="http://bugs.digium.com/view.php?id=6682">#6682</a>)<br /> 
+  (Note: I changed the name in this version...)<br />
+- G.722 support in Asterisk (passthrough, formats) (andrew, <a href="http://bugs.digium.com/view.php?id=5084">#5084</a>)<br />
+- Fix race condition in voicemail (corydon76, <a href="http://bugs.digium.com/view.php?id=6714">#6714</a>)<br />
+- TOUPPER and TOLOWER ASCII functions (powerkill, <a href="http://bugs.digium.com/view.php?id=6668">#6668</a>)
+  (With some changes)<br />
+
+<h3>Things that has been commited to svn trunk:</h3>
+- Abandon Queue manager event (tim_ringenbach, <a href="http://bugs.digium.com/view.php?id=6489">#6459</a>)<br />
+- End CDR before 'h' extension (russellb, <a href="http://bugs.digium.com/view.php?id=6193">#6193</a>)<br />
+
+Coming here soon:<br />
+- siptransfer: Improved SIP transfer support (branch)<br />
+
+<p>All of these exist in the <a http://bugs.digium.com>bug tracker</a>. Please report your findings
+in each open issue report, so that we get the feedback for each
+patch. Your opinion, good or bad, is important for us.
+http://bugs.digium.com></p>
+
+<p>The time you use to test this branch is a very important donation
+to the Asterisk Open Source project. Thank you!</p>
+
+<h2>Open BUGS (fixes welcome!)</h3><p>
+- ???<br /></p>
+<h2>Patch Descriptions<h3><p>
+<h4><a name=metermaids>Metermaids</a></h4>
+
+<p>Metermaids is a way to watch the status of a parking lot through a SIP
+subscription. Add a hint pointing to Local/exten at parkinglot and subscribe
+to that extension. Whenever someone parkings in that parking lot, you
+will see it in the subscribing phone.</p>
+
+<h4>The Generic Jitterbuffer</h4>
+<p>Enable the jitter buffer in the Makefile for testing it.</p>
+
+<h4>CLI commands for globals variables</h4>
+<p>With this patch, you get a CLI command to read global channel
+variables and one to set a global dialplan variable. By using
+this in comibination with "asterisk -rx" from the CLI you can
+set and read counters of various kinds that you implement in the
+dialplan from an external application.
+	<i>- This function is now integrated in svn trunk.</i></p>
+
+<h4>SIPregister</h4>
+
+<p>This patch tries to address several problems. It adds a register=yes/no
+clause to the peer section as well as making sure that the incoming
+call from the service you register with matches the peer directly,
+avoiding the clumsy old IP-based matching paradigm.</p>
+
+<h4>Videosupport</h4>
+
+Enables turning on video support per device in sip.conf instead of
+doing it for all of the devices in the SIP channel. Also implements
+a bandwidth maximum rate used for video phones.
+
+<h4>IPtos</h4><p>
+The IPtos patch adds the ability to define separate TOS values for
+SIP signalling and multimedia streams.
+Please read doc/ip-tos.txt for information about this new feature</p>
+
+<h4>Carrier ENUM support</h4><p>--------
+Documented in doc/enum.txt as well as configs/enum.conf.sample
+
+<h4>PEERMATCH: New object match for incoming calls. Skip the "user" :-)</h4><p>
+In this code, we will match incoming calls like this:<br />
+
+- First user on From: user name <br />
+- Then peer on From: user name   *** NEW **** <br />
+- Then peer on IP and port <br />
+
+<p>This means that in most configurations, you can configure a phone entry as
+"type=peer" instead of "type=friend". Subscriptions will work much better
+with just one object to match.</p>
+
+<p>We have also removed the sip_user structure and only uses one device
+data structure in the code. For type=friend, we use *one* object instead of
+two, still linking it to the user and the peer list.
+This means that call_limits now are implemented for the device only
+once. Incoming and outgoing channels are counted on the same object,
+not separate objects as before. This also improves the SIP SUBSCRIBE
+support (blinking lamps) as outbound and incoming calls are handled
+by the same in-memory object. In most cases, type=peer will work as
+well. </p>
+
+<h4>Multiparking - multiple parking lots in Asterisk</h4><p>
+Multiple parking support gives you the ability to define multiple
+parking lots within your Asterisk pbx. You can have one per company
+in a virtual PBX solution, or one per department or one per board
+member... Whatever you wants. Read the sample configuration files
+for more informaiton, features.conf for configuration of additional
+parking lots and the channel configurations for setting default
+parking lot for a device/channel. </p>
+
+<h4>End CDR before 'h' extension</h4><p>
+In Asterisk 1.2 you will not get values from ${CDR(end)} or ${CDR(billsec)}
+from the 'h' extension. The reason is that the CDR is not ended before
+executing the 'h' extension.</p>
+
+<p>This means that a customer would be billed for all of the time that
+the 'h' extension is being executed.</p>
+
+<p>This patch simply ends the CDR before starting to execute the 'h' extension.
+If the 'h' extension does not exist, the cdr will be ended shortly after
+in ast_hangup().</p>
+
+<p>The CDR is still only posted in ast_hangup, because we want the CDR to
+be ended, but still present in the 'h' extension so that all values can
+still be read from it.</p>
+
+<h4>6251: Support for HDLC mode in ZAP channels</h4><p>
+The patch adds 2 things to chan_zap:</p>
+<p>
+1. an application: ZapSetHDLC <br />
+2. an option to the dialstring h ( dial(zap/g0h/123) ) <br />
+</p>
+
+<p>This patch adds an application ZapSetHDLC and an option to the zap channel dialstring, 
+which allow incoming and outgoing zap channels to be setted into HDLC Mode. </p>
+
+<p>The patch adds an element hldc to the zt_pvt struct. When either for incoming calls 
+ZapSetHDLC is called or for outgoing calls the h option is given, the corresponding 
+zap channel is set into HDLC mode via ioctl(ZT_HDLCFCSMODE) in PRI_HANGUP the patch 
+sets back the channel into ZT_AUDIOMODE. </p>
+
+<p>Additionally in my_zt_write and zt_read it is checked at particular places if hdlc mode 
+is set, and then the sample rate is set directly for example, rather then using 
+READ_SIZE and so forth. </p>
+
+<p>The feature plans are to add a frame subclass AST_MODEM_HDLC. Applications or channels
+that need to transfer data in hdlc mode could just send with that type and the other
+channel driver could activate its hdlc controller. </p>
+
+<p>With this construction you can make a successful data call between chan_misdn and 
+chan_zap T1/E1 channel.</p>
+
+<p>The Patch does NOT change anything when hdlc mode is not set, which means it will not 
+touch current installations.</p>
+
+<pre>
+
+DIALPLAN EXAMPLE:
+
+misdn.conf:
+
+ports=1
+context=bri2Zap
+
+
+extensions.conf:
+
+[bri2Zap]
+exten => _X.,1,misdn_set_opt(h1)
+exten => _X.,1,Dial(Zap/g0h/1234)
+
+
+
+the other way would be:
+
+zapata.conf:
+context=zap2bri
+channels=>1-15,17-31
+
+
+extensions.conf:
+
+[zap2bri]
+exten => _X.,1,ZapSetHDLC()
+exten => _X.,2,Dial(mISDN/g:Computer1/123/h1)
+</pre>
+</p>
+
+<h3>CREDITS</h3><p>
+Thanks to the following companies for sponsoring this work:<p>
+
+* VOOP A/S, Norway (http://www.voop.no)<br />
+
+<h4>For the SIP transfer code:</h4><p>
+
+* Nuvio, inc, USA <br />
+* Foniris Telecom, Denmark<br />
+* VOOP A/S, Norway<br />
+* Inotel, Poznan, Poland<br />
+
+<h4>The SIP MWI subscription patch was funded by<h4>
+
+* Dus.net GMBH, Germany</br />
+
+<h4>MAINTAINER</h4><p>
+
+Olle E. Johansson, Asterisk developer, trainer, bug marshal
+<a href=http://edvina.net>http://edvina.net</a></p>
+</body>
+</html>

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