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2008-04-17Merged revisions 114202 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) | 2 lines fileio.h does not exist; io.h does, though. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@114203 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28Merged revisions 111908-111909 via svnmerge from russell1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r111908 | russell | 2008-03-28 17:45:43 -0500 (Fri, 28 Mar 2008) | 3 lines Note a minor race condition that I noticed while reviewing Jeff's changes to this code. ........ r111909 | russell | 2008-03-28 17:50:46 -0500 (Fri, 28 Mar 2008) | 3 lines Make some notes about common usage of pbx_builtin_getvar_helper() that is not thread-safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@111910 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28Merged revisions 111606 via svnmerge from tilghman1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) | 11 lines Merged revisions 111605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines Update debugging text, since Valgrind eliminated the --log-file-exactly option. (Closes issue #12320) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@111607 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Merged revisions 110911 via svnmerge from juggie1-4/+18
https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 | juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines update documentation to reflect the changes in the way configure detects net-snmp. (closes issue #12067) Reported by: juggie Patches: 12067_snmp_doc.patch uploaded by juggie (license 24) Tested by: juggie ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@110959 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107827 via svnmerge from qwell1-24/+50
https://origsvn.digium.com/svn/asterisk/trunk ................ r107827 | qwell | 2008-03-11 18:38:00 -0500 (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines Update documentation for pgsql ODBC voicemail. (closes issue #12186) Reported by: jsmith Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license 15) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@107828 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106684 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 | russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines fix example usage ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106686 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106518 via svnmerge from russell1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 | russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line minor text changes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106520 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106507 via svnmerge from russell1-18/+118
https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 | russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines Add updated SMDI documentation that I had only sitting in my email ... oops ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106508 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-06Merged revisions 106186 via svnmerge from russell1-0/+14
https://origsvn.digium.com/svn/asterisk/trunk ........ r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008) | 7 lines document var_metric usage to prevent bugreports that are actually configuration issues (closes issue #12151) Reported by: caio1982 Patches: DB_metric3.diff uploaded by caio1982 (license 22) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106315 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27note that the chan_sip conversion is already in progressrussell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104473 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27add another janitor projectrussell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104444 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27Add the stuff from the janitor projects page that is still relevant. I figurerussell1-0/+15
that if we keep this in the tree, it will be much easier to keep up to date. The page on asterisk.org just links to this on svn.digium.com/view git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104419 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27Create placeholder file...for now.qwell1-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104418 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-261) Make braces mandatory for if/for/while, even around single statements.tilghman1-10/+14
2) Revise the argument parsing section, showing use of the standard macros. 3) Fix a typo. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104176 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Adding more tls configuration details to sip.conf sample, with a list of ↵bbryant1-0/+2
valid ciphers provided in both files. .. First commit since July, woot git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-22Move Originate to a separate privilege and require the additional System ↵tilghman1-0/+5
privilege to call out to a subshell. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104039 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15Merged revisions 103722 via svnmerge from mmichelson1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines Final round of changes for configure script logic for IMAP Now if a directory is specified, then we will search that directory for a source installation of the IMAP toolkit. If none is found, then we will use that directory as the basis for detecting a package installation of the IMAP c-client. If that check fails, then configure will fail. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103725 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15Same changes as made to 1.4 in revision 103710mmichelson1-4/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103711 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14Trunk version of 1.4's imap documentation updatesmmichelson1-8/+21
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103705 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-28Reintroduce more chan_vpb stuff that was removed in r100421 and r100422qwell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100679 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-25Get rid of that last little bit.qwell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100422 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Add res_config_ldap for realtime LDAP engine.tilghman1-0/+65
(closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99696 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Documentation updates for BRIDGEPVTCALLIDoej1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Small fixesoej1-8/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99482 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18Merge changes from team/group/sip-tcptlsrussell1-0/+94
This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Update res_phoneprov to default to setting the SERVER variable to the IPtwilson1-9/+15
the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98988 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Update documentation.file2-2/+5
(closes issue #11763) Reported by: IgorG Patches: docupd.v1.diff uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98695 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add some extra checking to help out with a potential error when trying torussell1-0/+9
run "make asterisk.pdf" when not all of the right packages are installed. (closes issue #10763) Reported by: Corydon76 Patches: 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98454 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Documentation updatestilghman1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98152 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Added a new module, res_phoneprov, which allows auto-provisioning of phonestwilson2-1/+305
based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Adding user-configurable TCP timeout settings to IMAP voicemail. This couldmmichelson1-0/+4
go a long way towards preventing unexplainable hangs experienced by people. In the case of MWI hangs, this also will mean that the SIP port isn't blocked anymore. (closes issue #11665, reported by yehavi) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96934 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Fix -s socket option, and document it as well.qwell1-1/+5
Closes issue #11645, patch by Laureano. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95070 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27Adding documentation for new manager actions and events in app_queuemmichelson1-0/+43
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94903 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Merging the queue-penalty branch. In short, this allows one to dynamically ↵mmichelson1-0/+1
adjust the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending on the amount of time passed. The purpose is to allow the call to open up to more (or maybe just different) members without the caller's losing his place in the queue. See configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample for how to associate a rule with a queue. Along with the functional changes, new CLI and manager commands exist to show the rules defined and there is an additional CLI command to reload the queue rules. Future enhancements that may be made: support for realtime queue rules and support for dynamically adding a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write this myself very soon). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94370 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Add option for starting remote Asterisk by naming the actual runtime socket ↵oej1-5/+14
instead of pointing to configuration file with -C Reported by: sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax (license 359) doc changes by committer (closes issue #11598) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93854 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18A minor update, caused by a recent bug report ;-)oej1-6/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18small documentation update (nothing important).rizzo1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16HUGE improvements to QoS/CoS handling by IgorGoej1-66/+85
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10Add a few extra headers in the voicemail users listing inoej1-0/+4
manager 1.1. Update documentation too. (closes issue #11495) Reported by: caio1982 Patches: extra_vm_manager_info1.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10add a bit of info on the build infrastructurerizzo1-0/+41
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92084 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Add count of total number of calls processed by asterisk during it's lifetime.qwell1-2/+22
Add number of total calls and current calls to SNMP. Closes issue #10057, patch by jcmoore. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91779 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06Adding documentation for the massive manager changes to manageroej1-0/+236
version 1.1 - hopefully a more consistent manager interface. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91438 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Change all instances of "CALLERID(number)" to "CALLERID(num)" formmichelson1-7/+7
consistency's sake (closes issue #11381, reported and patched by jon) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89621 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22new info on the management of headersrizzo1-2/+79
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89526 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Merged revisions 89450 via svnmerge from murf1-10/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 line closes issue #11324; break statements missing in switch cases. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89451 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Merged revisions 89416 via svnmerge from file1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov 2007) | 4 lines Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen. (closes issue #11246) Reported by: jon ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89417 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Merged revisions 89103 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) | 2 lines Typo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89104 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Adding documentation regarding imapfolder, imapgreetings, and ↵mmichelson1-0/+14
greetingsfolder options in voicemail.conf (closes issue #11133, reported by selsky, patched by blitzrage) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89075 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Added the S() and L() options to the MeetMe application. These are prettyrussell1-6/+9
much identical to the S() and L() options to Dial(). They let you set timeouts for the conference, as well as have warning sounds played to let the caller know how much time is left, and when it is running out. (closes issue #8030) Reported by: areski Patches: meetme_timeout_timelimit_v2.patch uploaded by areski (license 29) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89069 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Merge the code from asterisk/team/group/chan_unistim:russell1-0/+127
This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88368 f38db490-d61c-443f-a65b-d21fe96a405b
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