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${ noResults }
501 Commits (d4894b3d25de78ca3b2e289261a73e3aec07209e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
89a1af1d38 |
Allow greetings-only mailboxes for Voicemail.
(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 |
16 years ago |
|
|
2414bc8005 |
Add audio announcement option to app_page
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 |
16 years ago |
|
|
c207825dc7 |
Move an entry from CHANGES to UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2a1dce85b4 |
Move an entry from CHANGES that should be in UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0aa5aae587 |
Provide a real description of LOCAL_PEEK().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d50779cfa3 |
Remove a feature from CHANGES that was listed twice for 1.6.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e14393f97c |
Fix up the faxdetect entry in CHANGES.
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 |
16 years ago |
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cdd1a8616a |
Remove an entry from CHANGES that is already in UPGRADE.txt (where it should be).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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26daf50863 |
Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchat
(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 |
16 years ago |
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52570bfa84 |
update CHANGES file for .m3u support in Mp3Player application
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5cbe65e96b |
update CHANGES for new queue option, penaltymemberslimit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233198 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8599628e0b |
Add pagerdateformat, to allow shorter dates for SMS messages.
(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 |
16 years ago |
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e5aa8cad9b |
Extend voicemail to allow IMAP folders to be specified per mailbox.
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 |
16 years ago |
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2081809b07 |
update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 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 |
16 years ago |
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d0e431ce3d |
Add an 'X' option to the asterisk application which enables #exec for 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 |
16 years ago |
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a0b1c785c6 |
Add an option to Record which enables a mode where any DTMF digit will 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 |
16 years ago |
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1c7dd506d6 |
Updated CHANGES file to describe the new 'd' option to app_followme added in r230964
(related to issue #14155) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0bccc4fbe6 |
Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for 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 |
16 years ago |
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5e2aa190fe |
Display a list of channel variables in each channel-oriented event.
(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 |
16 years ago |
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f62d03a8f3 |
Store the cause code that is returned when trying to create a channel in 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 |
16 years ago |
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75c015bfff |
Add the capability to require a module to be loaded, or else Asterisk exits.
Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0ce378d3da |
Update CHANGES file.
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 |
16 years ago |
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aabff54c4b |
Add the 'relative-periodic-announce' option to app_queue to allow for 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 |
16 years ago |
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20e56c9d36 |
Created standard location to add options to chan_dahdi for ISDN dialing.
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 |
16 years ago |
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317435a932 |
Added the 'a' option to app dial and modified app_dial to set the answertime 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 |
16 years ago |
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7ed425ec80 |
This patch adds a sequence field to CDRs that can be combined with the 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 |
16 years ago |
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66579d9d49 |
Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
(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 |
16 years ago |
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93e43578ec |
This patch adds support for a draft proposal for adding Q.850 reason headers 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 |
16 years ago |
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5825f68e8b |
Add support for receiving unsolicited MWI NOTIFY messages.
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 |
16 years ago |
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cff6d02b53 |
Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.
* 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
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16 years ago |
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d9f72c1893 |
Permit storage of voicemail secrets in a separate file, located within the 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 |
16 years ago |
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1174a61612 |
Add support for calling and called subaddress. Partial support for COLP 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 |
16 years ago |
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01ab66275a |
Add support for specifying the IP address to use for media streams in sip.conf
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 |
16 years ago |
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a2f809c127 |
Turn on DENOISE filter for all conference participants.
(Fixes SWP-238) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a31eb5bb35 |
Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28d0ec5421 |
Add support for specifying the IP address to use for media streams in sip.conf
(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 |
16 years ago |
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e90ff8de4c |
Added information to CHANGES about the dynamic range compression feature added to dahdi.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e3f473f4f3 |
Allow for adding message body to the SIP NOTIFY message
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 |
16 years ago |
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ccdfe1c2d7 |
Updates CHANGES to reflect the new externtcpport and externtlsport sip options
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222399 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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dd0c76a9d3 |
Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9c05faf76d |
Prevent deadlock if chan_dahdi attempts to change PRI channel names.
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 |
16 years ago |
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b11b94a083 |
Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels
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 |
16 years ago |
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c68a2d9d30 |
Add support for 'setvar=' for MGCP device lines, like other channel drivers provide.
(closes issue #14818) Reported by: alea-soluciones Patches: chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cd88adfc6a |
Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which I discovered that the last feature added to this application duplicated an option (oops!) and changed that option so that it now works. (closes issue #14909) Reported by: junky Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10) Tested by: amilcar, junky, flujan, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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75d8960740 |
Allow multiple rows to be fetched within the normal mode of operation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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595ab444af |
Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ISDN PTMP CPE spans.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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eca8f9082c |
Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach)
Review: https://reviewboard.asterisk.org/r/345/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2cfddf8cb6 |
Add MASTER_CHANNEL() dialplan function, as well as a useful usage.
(closes issue #13140) Reported by: cpina Patches: 20090807__issue13140.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen Change-type: feature git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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29e1e05e13 |
Add two new dialplan variables when using features
Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the unique channel name on the other side and is set when a dynamic feature is triggered. (closes issue #14663) Reported by: tamiel Patches: 20090313_features.diff uploaded by tamiel (license 712) Tested by: tamiel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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39c9838d77 |
Add COLP support to chan_dahdi/sig_pri.
Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP support is now available for any switch for which libpri supports COLP (currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/340/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |