Commit Graph

903 Commits (41d513f443becff2378370ffae4163f2551a883c)

Author SHA1 Message Date
Russell Bryant 4a523b1b2d Add the ability to customize some of the prompts used within the voicemail
19 years ago
Christian Richter f19300635f Merged revisions 46351-46353 via svnmerge from
19 years ago
Luigi Rizzo e85d8e98d1 document the match_auth_username option
19 years ago
Matthew Fredrickson 67926b9ac4 Update changes to do US style point code parsing/formatting (xxx.xxx.xxx)
19 years ago
Olle Johansson c30f1d12c5 Ok, second attempt...
19 years ago
Olle Johansson 25b8f577b8 On the other hand, don't use 1.4 patches for trunk... Sorry.
19 years ago
Olle Johansson 13ea5fc0d0 Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audio
19 years ago
Luigi Rizzo c15f7953c8 Fix a few issues in the previous (disabled) HTTPS code,
19 years ago
Luigi Rizzo d171c3d864 remove unused fields and unimplemented options.
19 years ago
Russell Bryant d8e688ece9 Merged revisions 45439 via svnmerge from
19 years ago
Olle Johansson a8a26ad389 Update of docs
19 years ago
Joshua Colp c62784c10d In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
19 years ago
Joshua Colp da330feb60 Merged revisions 45280 via svnmerge from
19 years ago
Joshua Colp b58cc9e1bd Merged revisions 45262 via svnmerge from
19 years ago
Christian Richter e09ad744af Merged revisions 44561 via svnmerge from
19 years ago
Olle Johansson 77c69dc4ef Recommend using "sip reload" since it's much easier to learn and
19 years ago
Luigi Rizzo b19b4b9764 document a bit the use of templates.
19 years ago
Luigi Rizzo f94849ca2a document the "contact" option a bit better.
19 years ago
Luigi Rizzo ccca5843fd Two things:
19 years ago
Luigi Rizzo 2a7ac3f735 update example commands to match current syntax
19 years ago
Steve Murphy 8135d5016a I've been meaning to add some explanation about muted... here it is
19 years ago
Steve Murphy 6e42aa676c CLI reverbification update to this config file
19 years ago
Joshua Colp 8ff3dd273a Expand setinterfacevar option to also set a variable, MEMBERNAME, which contains the member's name. (issue #8046 reported by jmls)
19 years ago
Paul Cadach b4ef9599de Merged revisions 44186 via svnmerge from
19 years ago
Joshua Colp e5203bb283 Add option to logger to rename log files with timestamp (issue #8020 reported by jmls)
19 years ago
Joshua Colp cc1945ce1b Add option 'keepstats' which will keep queue statistics during a reload. (issue #7908 reported by jmls)
19 years ago
Russell Bryant 8a5cf10121 Merged revisions 44111 via svnmerge from
19 years ago
Paul Cadach 9cf1f14ed5 Handle HOLD/RETRIEVE notifications
19 years ago
Steve Murphy 2b7debf368 Merged revisions 43739 via svnmerge from
19 years ago
Jason Parker fe1a3b8877 Add optional queue_log_name config option for logger.conf, to change the
19 years ago
Paul Cadach c479c25182 Support for negotiation and receiption of Cisco's RTP DTMF
19 years ago
Paul Cadach 1af96a0b21 Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated)
19 years ago
Tilghman Lesher 859bc68383 Merged revisions 43464 via svnmerge from
19 years ago
Matthew Fredrickson bd76cda68d Merge in SS7 changes.... need to still cleanup zapata.conf
19 years ago
Jason Parker 8bd82ebc0d Add documentation on rtp packetization.
19 years ago
Jason Parker 3c224654c2 Document member name logging functionality.
19 years ago
Matthew Fredrickson 2e5118cc49 Add the h323 config file. Arrr!!! for international talk like a pirate's day.
19 years ago
Matt O'Gorman ec4bf7a849 seperate jingle and gtalk so it will be easier to track
19 years ago
Steve Murphy 4ed1578104 Clarified the meaning of the callwaiting variable in the zapata.conf file.
19 years ago
Joshua Colp 956b837a41 Merged revisions 43159 via svnmerge from
19 years ago
Tilghman Lesher ee27f9efee Remove the suggestion of realtime hints, since that functionality will not be available until post-1.4
19 years ago
Mark Spencer c2d959c2f9 Improve documentation of users.conf items.
19 years ago
Jason Parker 6d5809297b Skinny hold support.
19 years ago
Kevin P. Fleming c2c4f86c72 merge markster's usersconf branch with some slight changes
19 years ago
Tilghman Lesher 091e1aed8d Merged revisions 42716 via svnmerge from
19 years ago
Tilghman Lesher b0666488f3 Merged revisions 42697 via svnmerge from
19 years ago
Voicetronix Support e02897acd4 Board numbers and channel numbers are now 0 based, since vpb driver
19 years ago
Matthew Fredrickson 75822388a4 Make sure we give a little warning about the echotraining option
19 years ago
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
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