Commit Graph

785 Commits (7ba16c98781d2ff10b01d4c706b45e7d5b4ef6bd)

Author SHA1 Message Date
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
Olle Johansson 023e27f695 Formatting fixes
20 years ago
Olle Johansson 95de51526a Added information on call-limit and realtime
20 years ago
Olle Johansson bf4b484e62 Clarify the need for numeric parking positions (imported from 1.2)
20 years ago
Mark Spencer bfba044b5f Flesh out the remainder of the manager + http changes and create a sample application to partially
20 years ago
BJ Weschke cc0b49d927 Provide warning about current behavior of autofill = yes
20 years ago
Olle Johansson eb94c40702 Typo
20 years ago
North Antara 139b07c76c whitespace "fixes", and general cleanup
20 years ago
Luigi Rizzo c01fc0ee03 the comment character is ';' not '#' ...
20 years ago
North Antara 150e0b72cc Added more "valid" phone types to skinny sample config.
20 years ago
Luigi Rizzo 08df3610a6 update example file
20 years ago
Russell Bryant 41f8e3728e disable the http server by default at the request of people on IRC
20 years ago
Kevin P. Fleming 8410e0d681 support subscription-based MWI, and use proper Call-ID on NOTIFY messages (issue #6390)
20 years ago
Kevin P. Fleming 278b8e8fc7 improve IP TOS support for SIP and IAX2 (issue #6355, code from jcollie plus modifications)
20 years ago
Olle Johansson 83d9331261 Issue #5427
20 years ago
Olle Johansson 18de2b7787 Issue #6705 (oej)
20 years ago
Mark Spencer 9164eac21a Add micro-http server and abstract manager interface, make snmp not die
20 years ago
Matthew Fredrickson ba8f7b8819 Allow channels to be moved if channel change is requested in SETUP_ACK, also add a WAY cool new field to the nsf option
20 years ago
Jim Dixon 7cfb9b3515 Added separate outsignalling specification, and fixed FEATDMF to allow for
20 years ago
Russell Bryant 1abe304427 add a CLI command that allows converting files to other formats using
20 years ago
Russell Bryant f882197157 Merged revisions 13964 via svnmerge from
20 years ago
Russell Bryant c114587344 add indications for Malaysia (issue #6758)
20 years ago
Christian Richter 52eb1ad9d1 removed dynamic switching from transparent to hdlc mode. Instead we've got a config option hdlc=yes now which enables the hdlc controller for a data call
20 years ago
Christian Richter a0800bd179 these traceing option do not exist anymore
20 years ago
Olle Johansson d7b5a18f4c Fix reference to README files
20 years ago
Olle Johansson 0efbe1aa5d Add reference to examples for files and custom, too make it more obious
20 years ago
Olle Johansson 1a206c1bf8 Clarify documentation for "progressinband" - imported from 1.2
20 years ago
Russell Bryant 9df72acbe9 deprecate the mailboxdetail option and always use its behavior, instead (issue #6665)
20 years ago
Russell Bryant 4e6af293f9 add an option to cdr.conf that enables ending CDRs before executing
20 years ago
Christian Richter 8e7dd52695 added option to change the connected party number dialplan (ton)
20 years ago
Matt O'Gorman dad9d7709b allows the table field to be configurable for
20 years ago
Christian Richter 21735de56d added a bit more detailed description for the echotraining parameter, also changed the default from 1 to 2000. The default for the upper_threshold is now 0
20 years ago
Russell Bryant 99206286fb Merged revisions 11946 via svnmerge from
20 years ago
Matt O'Gorman 7377ebbd2e cdr_csv logging parameters in cdr.conf
20 years ago
Olle Johansson 6b8701cffa Whitespace changes
20 years ago
Christian Richter bd9c89a710 better default values for jitterbuffer in code and config
20 years ago
Mark Spencer 6a86c7c5c9 Add SNMP support (bug #6439)
20 years ago
Mark Spencer 16109b9d2c Make IAX2 multithreaded
20 years ago
Kevin P. Fleming 7092b4475c Merged revisions 10511,10535,10736 via svnmerge from
20 years ago
Matt O'Gorman fecae4f64e Changing syntax once again slightly and standardizing
20 years ago
Christian Richter afaf8e4c04 adde incoming_early_audio option, to avoid sending tone indications to the remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
20 years ago
Christian Richter f6bd1b8559 added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
20 years ago
Kevin P. Fleming f0495e8944 add option to avoid calling members whose channels are 'in use' (issue #6315, plus documentation)
20 years ago
Matt O'Gorman 49a04b91a2 changed naming scheme for variables so it matches
20 years ago
Kevin P. Fleming b9918fb16b set properties for new files (i need to get this documented)
20 years ago
Matt O'Gorman dacbca4699 Commiting 5959 with minor formatting and typo
20 years ago
Tilghman Lesher 973c12effd Bug 6477 - minor syntax error, plus a few other syntax fixes
20 years ago
Matthew Fredrickson af07dc8883 Add smdi support for asterisk (see doc/smdi.txt for config info) (#5945)
20 years ago
Kevin P. Fleming b40bd71a9a restore 'rfc2833' naming for DTMF mode in chan_sip
20 years ago
Olle Johansson 4d07b89fdd - Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.
20 years ago
Olle Johansson 24ceb84434 - Adding example on using european time zones in voicemail.conf
20 years ago
Christian Richter b42dd639ee default values of jitterbuffer and jitterbuffer_upper_threshold should be > 0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
20 years ago
Christian Richter 7133d1b006 * removed unnecessary struct elements and functions
20 years ago
Olle Johansson 8204f581ab - Adding a doc/00README.1st with an INDEX over README files
20 years ago
Olle Johansson 3f6cc5c544 - Clarify default setting of canreinvite (thanks royk)
20 years ago
Olle Johansson 125fd8446c Issue 5892: Set a minimum T1 timer for calls. Reporter: twisted
20 years ago
Olle Johansson b64404e039 From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel
20 years ago
Olle Johansson 0ba27e0a6b Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183)
20 years ago
Matt O'Gorman 83ab52cdc4 changed some settings to app_args and some code
20 years ago
Matthew Fredrickson 4dc76fbcc1 Fix comments in sip.conf (#6134)
20 years ago
Matthew Fredrickson bab4d7a6b9 Update config. Apprently default doesn't always work now.
20 years ago
Olle Johansson 125db028c3 - Add DOC file about caller ID presentation values
20 years ago
Olle Johansson 3d456ce4f0 Add support for "musicclass" instead of "musiconhold" to make chan_zap compatible
20 years ago
Tilghman Lesher 2a9a03e6f0 Bug 5090 - sample configuration for udptl packets
20 years ago
Matthew Fredrickson b5ed1a1a59 Add mission options to agents.conf sample file (#6234)
20 years ago
Matt O'Gorman 06f2040e6f added feature for pausing and unpausing the
20 years ago
BJ Weschke 1874f21ff8 Implement the autologoffunavail option in chan_agent (#6038 with some minor mods)
20 years ago
Matt O'Gorman 130ce286ba added two new features to meetme, autofill and
20 years ago
Matt O'Gorman 8d4ee3882a forgot to take out createlink sample along with
20 years ago
Matt O'Gorman 1dc0312d01 Added option for limiting a user from logging in
20 years ago
Matt O'Gorman 64f01df137 added page macro examples from 6077 with minor typo
20 years ago