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755 Commits (897e4b5eaaa4dbbae307be9652ebf8815544677d)

Author SHA1 Message Date
Joshua Colp 34eb4f54ba Use lower case 'x' instead of a UTF-8 character (issue #7888 reported by flefoll)
19 years ago
Mark Spencer 47d8e14871 Comment out default from extensions.ael
19 years ago
Jason Parker 4ba458e352 Merged revisions 42014 via svnmerge from
19 years ago
Olle Johansson 3f888b84f3 Use GLOBAL() in dialplan examples
19 years ago
Joshua Colp c6977b9983 Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
19 years ago
Kevin P. Fleming b281acf0f8 change default setting for autofallthrough
19 years ago
Jason Parker 0850bb72ed Kevins last commit made me spot a typo.
19 years ago
Kevin P. Fleming ece7018515 add one remaining bit of functionality to the features.conf applicationmap (from Matt Nicholson in Digium Express Services)
19 years ago
Kevin P. Fleming 7b65cdc0c7 remove documentation of 'global' section in modules.conf, since it is no longer needed or supported
19 years ago
Joshua Colp 236c23269d Merged revisions 40979 via svnmerge from
19 years ago
Joshua Colp e5c0665caf Merged revisions 40971 via svnmerge from
19 years ago
Kevin P. Fleming 749cd217c3 Merged revisions 40392 via svnmerge from
19 years ago
Russell Bryant 87ac16847e - unregister SLA apps on module unload and add sample config (issue #7701, junky)
19 years ago
Joshua Colp a0bd41f79b Add support for Sigma Designs cards. These basically allow you to offload dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
20 years ago
Joshua Colp 0f0323dbab Clarify volgain option a bit, it needs sox to work.
20 years ago
Tilghman Lesher f37a4e3e12 Bug 6237 - add volgain parameter, such that voicemail messages may be amplified after recording
20 years ago
Kevin P. Fleming 9d26f46fc7 remove some extraneous 'followme' in prompt names
20 years ago
Olle Johansson 2f69bec40e Add placeholder for sla.conf sample in configs/. Please update with
20 years ago
Russell Bryant 4d7c67fc72 Merge my applicationmap_fixup branch to address the issues described in this
20 years ago
Tilghman Lesher 0d902b3033 Update documentation on realtime; add a workaround for lack of realtime hints by using func_odbc
20 years ago
Kevin P. Fleming 6d0742fc16 merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
20 years ago
Mark Spencer 837910062b First pass at in-place file manipulation via manager
20 years ago
North Antara e69e056012 config sample for the previous, regarding ADSI
20 years ago
Kevin P. Fleming 4376af0080 actually make the non-standard G726-32 behavior available for SIP clients
20 years ago
Christian Richter 54ce0f0a22 added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
20 years ago
Olle Johansson 3b5a2aafa4 - Change filename to current file name
20 years ago
Joshua Colp ba092c1244 And now the trunk version! Add an option for IAX2 users that allows you to set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
20 years ago
Olle Johansson 0e0059c0f3 Remove configuration option "restrictcid" that is nowhere to
20 years ago
Matthew Fredrickson de03118578 Asterisk portion of the T309 patch. (#7271)
20 years ago
Christian Richter bd0b801a0d * removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
20 years ago
Olle Johansson ec9d4711d7 - Add notes about voicemail depending on res_adsi
20 years ago
Olle Johansson b971f65978 - Make use of system name in realtime SIP peers optional
20 years ago
Olle Johansson f3594bd1a0 Removing configuration options that does not do anything yet. No need to
20 years ago
Christian Richter f5c0cd2ddc added better L2 handling for ptp, if it's down we don't try to call on that port in groupdial anymore, also we try to get it up then. Additionally added the configoptions ntdebugflags and ntdebugfile to debug the mISDNuser NT Stack (should have done that ages before..). isdn_lib.c compiles again.
20 years ago
Kevin P. Fleming dec3d7d4c0 Merged revisions 36253-36254 via svnmerge from
20 years ago
Olle Johansson 4177596e8d reformatting sip.conf.sample a bit, adding dumphistory that was not documented
20 years ago
Olle Johansson cc43f0bdc7 Speling error. Avoid swenglish :-) (thanks, jtodd!)
20 years ago
Olle Johansson 6399ac438d Add explanation and warning about the "s" extension. (Hi Mike :-)
20 years ago
Olle Johansson f8311adcda METERMAIDS:
20 years ago
Olle Johansson e2b0c5b558 Add example of permit/deny to sip.conf.sample
20 years ago
Tilghman Lesher 89f6ffe1e5 Bug 6589 - option to display channel variables in queue events
20 years ago
Joshua Colp 4c066de7cf Merged revisions 35334 via svnmerge from
20 years ago
North Antara a5d6979fac Finally merge chan_skinny fixes into trunk.
20 years ago
Russell Bryant 035a8b4278 Merged revisions 34627 via svnmerge from
20 years ago
Joshua Colp 0e5e744fb2 Add bulgarian indications (issue #7314 reported by KNK)
20 years ago
Joshua Colp 5456f425c6 Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support
20 years ago
Matt O'Gorman 1e530787f3 solves some bugs with memory allocation, and adds
20 years ago
Kevin P. Fleming bc49d5bfb3 moh files will now be distributed in native format, not mp3, so...
20 years ago
BJ Weschke 3d973a0686 Introducing app_followme into /trunk!
20 years ago
Kevin P. Fleming dd6de5ee4e it's time... only enable global priority jumping if the config file says to do so
20 years ago
Olle Johansson 4e74b8fb75 Issue #7231 - Missing indications from libtonezone (tzafrir)
20 years ago
Olle Johansson 2dc6947144 Issue #2863 - Improved RTCP support (John Martin, Fredrik Olsson)
20 years ago
Matt O'Gorman 18b135f215 oops my config file was out of date not the sample
20 years ago
Matt O'Gorman 58cd5f2440 oops fixing example config
20 years ago
Russell Bryant 4c76028de9 - add the ability to configure forced jitterbuffers on h323, jingle,
20 years ago
Christian Richter 4be235a974 added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
20 years ago
Kevin P. Fleming 6bce269454 Merged revisions 31321 via svnmerge from
20 years ago
Matt O'Gorman 8cfb992c1e adds statusmessage customization from Julian Lyndon-Smith
20 years ago
North Antara e25c4621b4 Nobody saw this coming, I bet.
20 years ago
Russell Bryant bb7dd96cfe Add support for using a jitterbuffer for RTP on bridged calls. This includes
20 years ago
Kevin P. Fleming 18606233da fix various typos and other bits (from Ian Kinner)
20 years ago
Joshua Colp 18248f092f Merged revisions 30239 via svnmerge from
20 years ago
Christian Richter 8122c35675 fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
20 years ago
Kevin P. Fleming 3e99be68d1 add a new option for 'obscuring' SIP user/peer names from fishers
20 years ago
Matt O'Gorman 45107ed763 allows for configurable answer timeout on attended transfer
20 years ago
Matt O'Gorman 7aa1a77e75 asterisk-xmpp merge in
20 years ago
BJ Weschke 5235890be4 This is part 2/2 of the patches for #7090. Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.
20 years ago
Christian Richter 19d46333bf added callcounters for incoming and outgoing calls
20 years ago
Tilghman Lesher 9e81cc3e0c Escaping commas within fields isn't always desireable.
20 years ago
Russell Bryant 1fcc86d905 Add support for logging CDR recrods to a radius server (issue #6639, phsultan)
20 years ago
Kevin P. Fleming 42cf0b0a8f add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
20 years ago
Joshua Colp 6d603ec09c Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)
20 years ago
Joshua Colp 15358932ec Add distinctive ring detection with Caller ID for Australia, New Zealand, and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)
20 years ago
Olle Johansson 5237a0e06d - Use systemname for realm in sip, if we have no configuration for realm
20 years ago
Kevin P. Fleming 5fb4e7019f and chan_iax2 gets smaller... remove the old jitterbuffer
20 years ago
Luigi Rizzo e0f0f4b4a4 german syntax for numbers from christian richter
20 years ago
Mark Spencer 66ed134473 Allow media to go directly between IAX endpoints while signalling still
20 years ago
Olle Johansson ca6cf552f9 Add documentation on "allowtransfer"
20 years ago
BJ Weschke 3e2079e46c Fix output delimiters and add prefix parameter to func_odbc #7025 (Corydon76)
20 years ago
BJ Weschke d83bd4d136 Integrate the MixMonitor functionality (introduced in 1.2) as an option for recording queue member conversations with callers. #7084
20 years ago
Russell Bryant 0794168428 add support for having the user reminded that their temporary greeting
20 years ago
BJ Weschke 85e0c889e4 Allow for the execution of an AGI to the caller's channel right before they get bridged with the queue member that is going to take their call. Add the option to set a MEMBERINTERFACE variable on the caller's channel that will contain the interface of the queue member that is going to/did take the call. #6843
20 years ago
Christian Richter efccf89eae Added option far_alerting. This option makes it possible to generate a Ringing on other channels if they feel that they should have inband ringing, but there is non in reality. I need this due to the fact that asterisk has not the possibility to transmit progress indicators thus chan_sip and others do not know wether they should generate a Rining tone themselves if they receive AST_CONTROL_RINGING..
20 years ago
BJ Weschke 7b3f3db65d Fix autofill behavior in app_queue and document it's functionality in queues.conf.sample and UPGRADE.txt
20 years ago
Olle Johansson 7bbb6bd3aa - fix typo in rtp.c, devicestate.h
20 years ago
Russell Bryant c38fbd246e note that group assignments must be from 0 to 63 (issue #7048)
20 years ago
Christian Richter 0b6bd0073b put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
20 years ago
Kevin P. Fleming 5f58cc8770 Merge Steve Murphy's (murf) complete re-implementation of AEL, which is now no longer considered experimental :-)
20 years ago
Tilghman Lesher e3f569532f Deprecate prefixed options in voicemail
20 years ago
Olle Johansson 5873462c2e - Add doxygen documentation for sipsock_read locking
20 years ago
Luigi Rizzo 68730ba487 update configuration, generalize date format and
20 years ago
Luigi Rizzo 64fbe4cbc5 add example syntax for new-style number and date spelling
20 years ago
Joshua Colp e8a94a71e2 Allow the attachment format to be specified differently for different mailboxes (issue #6961 reported by the ever fabulous Corydon76)
20 years ago
Russell Bryant 8a5436c72f add indications for Thailand (issue #6971)
20 years ago
Russell Bryant 717445c1d8 add the ability to turn off the feature that allows agents to end calls
20 years ago
Josh Roberson b04c61eeb3 Note that the res_speech module will need to be loaded first, and add a conveient line to uncomment to do so for the time being.
20 years ago
Joshua Colp afcefc4a68 Convert chan_iax2 to use linked lists for multithreading, and add dynamic threads. These are used when all pool threads are in use, and will stick around until load dies down. The theory is that during high load you'll have more threads available, and during low load you'll only have the normal pool threads sticking around.
20 years ago
Olle Johansson 7089dc1341 Issue #6899 - remove OSP support code from chan_sip.c and app_dial.c
20 years ago
Olle Johansson 9d8260c68e Formatting fixes
20 years ago
Olle Johansson 8e22245b09 Formatting fixes
20 years ago