path: root/CREDITS
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2011-04-14Add Device State Information CCSS for Generic Devices.rmudgett1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@313744 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add Despegar.com (my main sponsor) to the CREDITS file.eliel1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277103 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Adding a few more to the list of CREDITSoej1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277027 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Adding a few more creditsoej1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276952 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10Add support for devices with less than 3 lines on the LCD.russell1-0/+2
(closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275466 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Calendaring support for Exchange Server 2007+ via EWStwilson1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@265317 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Convert this branch to Opsound music-on-hold.kpfleming1-1/+1
For more details: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212922 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Update my e-mail address (thanks for the props, russell :))seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201190 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Add Sean Bright to CREDITS - Thanks, Sean!russell1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200942 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-29Apply anti-spam obfuscation to an email address.eliel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198083 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-15add elielmvanbaak1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194649 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Add MFC/R2 support for chan_dahdi.russell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Related to issue #14246oej1-0/+2
Update changes for SIPRemoveHeader() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168639 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss ↵oej1-0/+3
in text stream Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-23Add Sergey Tamkovich to CREDITS. Thank you for your contributions!russell1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100021 f38db490-d61c-443f-a65b-d21fe96a405b
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99280 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18Merge changes from team/group/sip-tcptlsrussell1-1/+6
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
2008-01-16Merge the changes from issue #10665 from the team/group/sip_session_timers ↵russell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@98978 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Name the people responsible for some recent contributions to the tree.rizzo1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93559 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Merge the code from asterisk/team/group/chan_unistim:russell1-0/+4
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
2007-10-31Formatting cleanups, remove obsolete contributions (modules no longer intilghman1-54/+60
Asterisk), and obfuscate email addresses enough to stop most spam harvesters. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87817 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06Philippe was listed twicerussell1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73631 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-07Adding Philippe to CREDITS for hard work on detecting bugs in our ↵oej1-0/+2
jabber/jingle integration git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68201 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Updating CREDITSoej1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62609 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Adding Realtime Text support (T.140) to Asteriskoej1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@54838 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25Ok, second attempt...oej1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46199 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25On the other hand, don't use 1.4 patches for trunk... Sorry.oej1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46197 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audiooej1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@46195 f38db490-d61c-443f-a65b-d21fe96a405b
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45224 f38db490-d61c-443f-a65b-d21fe96a405b
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43287 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-20Merged revisions 40692 via svnmerge from tilghman1-33/+8
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: http://svn.digium.com/svn/asterisk/trunk@40693 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-08support for imap in app_voicemail as well as some mogorman1-0/+10
credits fixed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39404 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-08Merged revisions 39379 via svnmerge from kpfleming1-0/+33
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: http://svn.digium.com/svn/asterisk/trunk@39380 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-14add Grandstream to credits tookpfleming1-2/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34043 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-13I am the king of typos....mogorman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33913 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-13added thanks to voipsupply and steve underwoodmogorman1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33911 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-03Adding John Martin to CREDITS for his video workoej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31715 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-02add credits for cdr_radiusrussell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31664 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-02Adding credits for SIP transfer workoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31636 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-01- add slav, zoa, and royk to the CREDITS for the generic jitterbufferrussell1-0/+36
- 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: http://svn.digium.com/svn/asterisk/trunk@31356 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-23allows for configurable answer timeout on attended transfermogorman1-0/+2
patch 0006763 with minor changes. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29766 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-20Add support for logging CDR recrods to a radius server (issue #6639, phsultan)russell1-2/+3
- with contributions from miconda, jcollie, and sb - branch maintained by oej Thanks everyone! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29094 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-24Merge Steve Murphy's (murf) complete re-implementation of AEL, which is now ↵kpfleming1-1/+1
no longer considered experimental :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22273 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-05Issue #6654: Enum crash on ADDRESS record, possibly bad record, but still a ↵oej1-0/+2
crash (imported from 1.2) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17490 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-30Add NetBSD for credits for editlinemarkster1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16602 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-06Add credit for the poll.c emulation layer for BSDoej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12107 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-03Adding res_snmp to Tholo's listoej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11689 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-29git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7221 ↵kpfleming1-0/+0
2005-10-26Add Claude Patry to the Credits. Thank you Junk-Y!!!russell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6863 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-18Fix various documentation issues (bugs #5464-5467)markster1-19/+29
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6826 f38db490-d61c-443f-a65b-d21fe96a405b