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${ noResults }
1020 Commits (01f3a08f8a45b7e5bdcf185207a16335f02fd19a)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
1e066813ac |
Add support for authenticating on a NOTIFY request. This is useful for phones that require it when sending them a special packet to get them to do something (such as reload their configuration).
(closes issue #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
|
41fd7a6a21 |
(closes issue #6113)
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Steve Murphy
|
5fb4b1bbe5 |
This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
|
0dd46a6bf0 |
Make the sample config match the contributed LDAP schema
(Closes issue #12421) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
ded5ec5b5d |
Merged revisions 113874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines If the [csv] section does not exist in cdr.conf, then an unload/load sequence is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
137c02a020 |
Permit message wrap-around during message retrieval.
(closes issue #12254) Reported by: andrew Patches: bug-12253.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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36cd3d0107 |
Additional note
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jason Parker
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763da3332a |
Document 'originate' permission in manager sample config.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
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63f574ceb4 |
Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
c6453ded22 |
Update sample configurations to make virtual hosting more obvious.
(closes issue #11969) Reported by: pprindeville Patches: acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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7741ed8bcc |
Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293) Reported by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
|
a567b41083 |
Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
|
58fa8e6e9e |
Change back to using ldap_initialize() and let the user specify a URL directly,
instead of trying to piece it together, badly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Mark Michelson
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cd7efcf4e7 |
Add option 'randomperiodicannounce' to queues.conf. Setting this will
allow the list of periodic announcments specified to be played in a random order instead of being played sequentially. (closes issue #6681) Reported by: alt_phil Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Olle Johansson
|
0de4eba640 |
Add manager peerstatus events when peer can't authenticate.
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
93b0f037b4 |
Add sample events for aastra phones.
aastra-check-cfg is the same as the other check-cfg entries, and aastra-xml is to load a pre-configured xml script. (closes issue #12229) Reported by: gowen72 Patches: aastra.patch uploaded by gowen72 (license 432) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Kevin P. Fleming
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a3a8aa6547 |
add support for named sections in zapata.conf, and fix a few bugs in config file parsing
(closes issue #9503) Reported by: tzafrir Patches: fix_cleanups uploaded by tzafrir (license 46) zapata_sections uploaded by tzafrir (license 46) skipchannel_options uploaded by tzafrir (license 46) conf_sample uploaded by tzafrir (license 46) patches updated by me to better conform to coding guidelines and fix some problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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0b97554307 |
Add contributed script for separation of database access from Asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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8a411ccf83 |
Create a centralized configuration option for silencethreshold
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
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7422f0ee37 |
Add documentation for setting username/password in SIP dial string.
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
4aff24881b |
Bring Voicetronix driver up to date with current drivers
(closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
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3a8756c9b4 |
Merged revisions 104119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Brett Bryant
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55aaa80d15 |
Adding more tls configuration details to sip.conf sample, with a list of valid ciphers provided in both files. .. First commit since July, woot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
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44810652d6 |
Change the queue holdtime announcement to happen at any interval (not just greater than two minutes). Remove
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using '1' as a queue-round-seconds value is no longer valid. (closes issue #9736) Reported by: caio1982 Patches: queue_announce5.diff uploaded by caio1982 (license 22) Tested by: caio1982, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
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a33932047d |
Merged revisions 103315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
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cdff02c08f |
Merged revisions 102807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb 2008) | 2 lines document usage of 'transfer' configuration option for ISDN PRI switch-side transfers ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
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31d411d393 |
Merged revisions 102651 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels. (due to a discussion between me and a user via email) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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f910cb5cb9 |
Change examples to use G here also.
Closes issue #11875 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
de0d0ad137 |
Clarify the pooling functionality by changing the config file keyword
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Olle Johansson
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9d07e7e9ee |
Clarify configuration file that can be misunderstood
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
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a1bf177286 |
Removing applications that wasn't ready for svn trunk, as trunk now has
pre-release status. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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0065508b25 |
Merged revisions 101219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37 -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use descending channel order of groups, rather than ascending. Fixes a potential source of confusion in glare-type situations. Issue 11875, reported by JimVanM. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
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11455c0898 |
Add rtppage() application to do multicast or unicast RTP paging to SIP phones.
(closes issue #11797) Reported by: macbrody Patches: app_rtppage-20080130.c uploaded by macbrody (license 352) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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7928888ecd |
Reintroduce more chan_vpb stuff that was removed in r100421 and r100422
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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838310187b |
Remove more remnants of chan_vpb
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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3bf7daa0c0 |
Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which drops packets that do not come from the remote party.
(closes issue #8952) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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cfa0ec1f97 |
Add res_config_ldap for realtime LDAP engine.
(closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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d1ba37f1c9 |
Change the Asterisk CLI startup commands feature to read commands to run from cli.conf
after a discussion on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99642 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Olle Johansson
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c85b71bf72 |
Documentation updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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6d57a8c873 |
Adding the QUEUENAME variable to the variables set using the setqueuevar option
in queues.conf. Suggestion comes from Shaun2222 on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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6181e386b5 |
Merged revisions 99341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines Permit the user to specify number of seconds that a connection may remain idle, which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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12a6e88d8c |
correct the name of a CLI command for getting available device names
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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f20450ea03 |
Merge changes from team/russell/console_devices
- Add support for multiple devices. All devices are configured in console.conf. - Add "console list devices" CLI command to show configured devices. Also, changed the old "list devices" to be "list available", which queries PortAudio for all audio devices that are available for use. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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b995c78c31 |
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
|
8dc5e09ccb |
Add several busy detection related defines to menuselect.
Allow better busy detect debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: busydetect_enhancement.patch uploaded by agx (license 298) busydetect-r94975.diff uploaded by sergee (license 138) Additional changes/cleanup by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
4346a37106 |
Merged revisions 98991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600 (Thu, 17 Jan 2008) | 4 lines Add a clarification about the immediate= option of zapata.conf Issue 11784, patch by klaus3000. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98992 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
cd4cc27c93 |
major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Terry Wilson
|
417c6dcb1d |
Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
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6aaa992301 |
Merge the changes from issue #10665 from the team/group/sip_session_timers branch.
This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |