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2010-01-05Add a missing part of the connected line work into trunk.Mark Michelson
Part of the work done for connected line was to add an optional argument to the 'f' option to allow for the connected party information of the outgoing channel to be set to the argument provided. This was overlooked during the merge of the work to trunk and is being added back now. The CHANGES file has also been updated to note this change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05Spell "aficionado" like someone who isn't stupid.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23Update CHANGES to reflect new QUEUE_MEMBER option, "ready"David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23update CHANGES to reflect new 'R' app_queue option plus a minor optimization ↵David Vossel
to the feature patch (issue #16384) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-22update CHANGES to reflect the addition of the test frameworkDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-19app_dial optional parameter to option 'r' to allow play indication from ↵Alec L Davis
indications.conf (closes issue #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-16Add auth_policy option to jabber.conf for auto user registration.Jeff Peeler
The option is global and currently the acceptable values as noted in the sample config are accept or deny. (closes issue #15228) Reported by: lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-16Enhance AMI redirect to allow channels to be redirected to different places.Jeff Peeler
New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added to redirect the second channel to a different location. Previously, it was only possible to redirect both channels to the same place. (closes issue #15853) Reported by: haakon Patches: trunk-manager.c.patch uploaded by haakon (license 880) Tested by: jpeeler git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14Allow greetings-only mailboxes for Voicemail.Tilghman Lesher
(closes issue #15132) Reported by: floletarmo Patches: voicemail_changes.patch uploaded by floletarmo (license 784) (with some additional changes by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-10Add audio announcement option to app_pageJeff Peeler
As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Move an entry from CHANGES to UPGRADE.txt.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Move an entry from CHANGES that should be in UPGRADE.txt.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Provide a real description of LOCAL_PEEK().Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Remove a feature from CHANGES that was listed twice for 1.6.2.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Fix up the faxdetect entry in CHANGES.Russell Bryant
This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X versions. The description of the feature was also no longer accurate. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-09Remove an entry from CHANGES that is already in UPGRADE.txt (where it should ↵Russell Bryant
be). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-07Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchatJeff Peeler
(closes issue #14352) Reported by: fiddur Patches: trunk-14352-2.diff uploaded by phsultan (license 73) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04update CHANGES file for .m3u support in Mp3Player applicationDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04update CHANGES for new queue option, penaltymemberslimit.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Add pagerdateformat, to allow shorter dates for SMS messages.Tilghman Lesher
(closes issue #16263) Reported by: andrew Patches: pagerdate.patch uploaded by andrew (license 240) (with a slight modification by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Extend voicemail to allow IMAP folders to be specified per mailbox.Jeff Peeler
Previously only possible per context, new option called imapfolder. (closes issue #14298) Reported by: jablko Patches: patch-200906202 uploaded by jablko (license 675) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 ↵David Vossel
and 1.6.2 (closes issue #16212) Reported by: miki git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Add an 'X' option to the asterisk application which enables #exec for ↵Joshua Colp
configuration files. This option can be used to enable #exec support in the asterisk.conf configuration file. (closes issue #16260) Reported by: atis Patches: exec_includes.patch uploaded by atis (license 242) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Add an option to Record which enables a mode where any DTMF digit will ↵Joshua Colp
terminate recording. (closes issue #15436) Reported by: Vince Patches: app_record.diff uploaded by Vince (license 823) Tested by: dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-24Updated CHANGES file to describe the new 'd' option to app_followme added in ↵Matthew Nicholson
r230964 (related to issue #14155) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-24Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for ↵Tilghman Lesher
the ast_get_encoded_* functions. * Add REPLACE function, which searches a given variable for a set of characters and replaces each with a given character. * Add PASSTHRU function, which passes a literal string back, like a NoOp for functions. Intent is to be able to specify a literal string to another function that takes a variable name as an argument. * Let the array manipulation functions work with dialplan functions, in addition to variables. This allows the array manipulation functions to modify ASTDB and ODBC backends, assuming the func_odbc configuration has both read and write functions. (closes issue #15223) Reported by: ajohnson Patches: 20091112__issue15223.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Display a list of channel variables in each channel-oriented event.Tilghman Lesher
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Store the cause code that is returned when trying to create a channel in ↵Joshua Colp
ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of overwriting the device state in AVAILSTATUS. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Add the capability to require a module to be loaded, or else Asterisk exits.Olle Johansson
Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-11Update CHANGES file.Leif Madsen
Updating the CHANGES file after noticing an email on the asterisk-dev mailing list from Russell. (issue #15874) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-09Add the 'relative-periodic-announce' option to app_queue to allow for ↵Matthew Nicholson
calculating the time of announcments from the end of the previous announcment rather than from the beginning. (closes issue #15260) Reported by: tonils git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-06Created standard location to add options to chan_dahdi for ISDN dialing.Richard Mudgett
Dial(DAHDI/g1[/extension[/options]]) Current options: K(<keypad_digits>) R Reverse charging indication (Collect calls) The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was variable and did not allow for the easy addition of more options. The earlier 'C' prefix character for reverse charge indiation would conflict with the a-d DTMF digits if ISDN uses them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Added the 'a' option to app dial and modified app_dial to set the answertime ↵Matthew Nicholson
when the called channel answers. This change causes answertime to be correct even if the called channel hangs up during an announcement triggered by the A() option. (closes issue #15936) Reported by: falves11 Patches: dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96) dial-caller-answer1.diff uploaded by mnicholson (license 96) Tested by: falves11, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03This patch adds a sequence field to CDRs that can be combined with the ↵Matthew Nicholson
linkedid or uniqueid field to uniquely identify a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networksTilghman Lesher
(closes issue #12950) Reported by: alea-soluciones Patches: ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514) Tested by: alea-soluciones, adomjan, urtho, nahuelgreco git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02This patch adds support for a draft proposal for adding Q.850 reason headers ↵Matthew Nicholson
to sip messages. (closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-27Add support for receiving unsolicited MWI NOTIFY messages.Joshua Colp
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.Richard Mudgett
* Added handling of received HOLD/RETRIEVE messages and the optional ability to transfer a held call on disconnect similar to an analog phone. * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. Will reroute/deflect an outgoing call when receive the message. Can use the DAHDISendCallreroutingFacility to send the message for the supported switches. * Added ability to send/receive keypad digits in the SETUP message. Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * Added support for BRI PTMP NT mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22Permit storage of voicemail secrets in a separate file, located within the ↵Tilghman Lesher
spool directory. (closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22Add support for calling and called subaddress. Partial support for COLP ↵Richard Mudgett
subaddress. The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Add support for specifying the IP address to use for media streams in sip.confJoshua Colp
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Turn on DENOISE filter for all conference participants.Tilghman Lesher
(Fixes SWP-238) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Revert media_address commit, I'm going to roll a fix to the SDP generation ↵Joshua Colp
in the next version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Add support for specifying the IP address to use for media streams in sip.confJoshua Colp
(closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-20Added information to CHANGES about the dynamic range compression feature ↵Matthew Nicholson
added to dahdi. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-14Allow for adding message body to the SIP NOTIFY messageJeff Peeler
Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-06Updates CHANGES to reflect the new externtcpport and externtlsport sip optionsDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-01Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-01Prevent deadlock if chan_dahdi attempts to change PRI channel names.Richard Mudgett
The PRI channels can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ISDN-<span>-<sequence-number> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channelsPhilippe Sultan
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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