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2017-12-21Fix some invalid Unicode charactersSean Bright
configs/samples/minivm.conf.sample contains invalid UTF-8, but that appears to be intentional. Change-Id: I7b1e0d332f3380fd0425962a3c9c55f9b200c8cc
2015-04-24CREDITS: Update credits for Olle JohanssonOlle E. Johansson
Change-Id: I8f3d0a6c3f1075a1f7d8308593394611a96749de
2013-12-20res_pjsip: Add PJSIP CLI commandsMatthew Jordan
Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-05Add RFC 3327 Path header support to chan_sipMatthew Jordan
This patch adds support for RFC 3327 "Path" headers. This can be enabled in sip.conf using the 'supportpath' setting, either on a global basis or on a peer basis. This setting enables Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. This patch also adds Realtime support for dynamically updating the Path information for a peer. A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts in writing this patch. Review: https://reviewboard.asterisk.org/r/2235/ Review: https://reviewboard.asterisk.org/r/991/ (closes issue ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej, mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000 (License 5054) oolong-path-support-trunk in team branch by oej (License 5267) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13Update CREDITSAndrew Latham
Update Jean-Denis and add myself (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11CREDITS clean upAndrew Latham
As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning. This is 95% whitespace with a few additions found in file headers. Further additions should be added here instead of in the file being updated. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-15Merged revisions 336042 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-12New sip.conf option for setting default tonezone for channel or individual ↵Olle Johansson
devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-14Add Device State Information CCSS for Generic Devices.Richard Mudgett
Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add Despegar.com (my main sponsor) to the CREDITS file.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Adding a few more to the list of CREDITSOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Adding a few more creditsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-10Add support for devices with less than 3 lines on the LCD.Russell Bryant
(closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24Calendaring support for Exchange Server 2007+ via EWSTerry Wilson
This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-18Convert this branch to Opsound music-on-hold.Kevin P. Fleming
For more details: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16Update my e-mail address (thanks for the props, russell :))Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16Add Sean Bright to CREDITS - Thanks, Sean!Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29Apply anti-spam obfuscation to an email address.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15add elielMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-16Add MFC/R2 support for chan_dahdi.Russell Bryant
This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Related to issue #14246Olle Johansson
Update changes for SIPRemoveHeader() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss ↵Olle Johansson
in text stream Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-23Add Sergey Tamkovich to CREDITS. Thank you for your contributions!Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-21UpdateOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-18Merge changes from team/group/sip-tcptlsRussell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-16Merge the changes from issue #10665 from the team/group/sip_session_timers ↵Russell Bryant
branch. This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-18Name the people responsible for some recent contributions to the tree.Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merge the code from asterisk/team/group/chan_unistim:Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-31Formatting cleanups, remove obsolete contributions (modules no longer inTilghman Lesher
Asterisk), and obfuscate email addresses enough to stop most spam harvesters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-06Philippe was listed twiceRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-07Adding Philippe to CREDITS for hard work on detecting bugs in our ↵Olle Johansson
jabber/jingle integration git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02Updating CREDITSOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Adding Realtime Text support (T.140) to AsteriskOlle Johansson
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25Ok, second attempt...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25On the other hand, don't use 1.4 patches for trunk... Sorry.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audioOlle Johansson
when the number of channels fill the MTU on a given link. In the future, this needs to be configurable per peer with trunking enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-16UpdateOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-19Work!!!Matthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-20Merged revisions 40692 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r40692 | tilghman | 2006-08-20 17:09:57 -0500 (Sun, 20 Aug 2006) | 2 lines Reformat to match the contribution style of other contributors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08support for imap in app_voicemail as well as some Matt O'Gorman
credits fixed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08Merged revisions 39379 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r39379 | kpfleming | 2006-08-08 13:39:16 -0500 (Tue, 08 Aug 2006) | 2 lines add explicit listing of anthm's contributions (issue #7683) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-14add Grandstream to credits tooKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-13I am the king of typos....Matt O'Gorman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-13added thanks to voipsupply and steve underwoodMatt O'Gorman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-03Adding John Martin to CREDITS for his video workOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-02add credits for cdr_radiusRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-02Adding credits for SIP transfer workOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01- add slav, zoa, and royk to the CREDITS for the generic jitterbufferRussell Bryant
- change references to the "scx" jitterbuffer to be called "fixed" and change references to the "stevek" jitterbuffer to be called "adaptive", instead git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23allows for configurable answer timeout on attended transferMatt O'Gorman
patch 0006763 with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-20Add support for logging CDR recrods to a radius server (issue #6639, phsultan)Russell Bryant
- with contributions from miconda, jcollie, and sb - branch maintained by oej Thanks everyone! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-21 18:41:44 +0000