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2008-04-11Merged revisions 114088 via svnmerge from tilghman1-91/+91
https://origsvn.digium.com/svn/asterisk/trunk ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines Make the sample config match the contributed LDAP schema (Closes issue #12421) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@114089 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-09Merged revisions 113875 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@113876 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Merged revisions 113245 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines Additional note ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@113246 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Merged revisions 113243 via svnmerge from qwell1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line Document 'originate' permission in manager sample config. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@113244 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Merged revisions 113119 via svnmerge from qwell1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@113174 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-25Merged revisions 110691 via svnmerge from tilghman2-0/+60
https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110692 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-25Merged revisions 110689 via svnmerge from tilghman1-26/+68
https://origsvn.digium.com/svn/asterisk/trunk ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110690 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-21Merged revisions 110499 via svnmerge from russell1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110501 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Merged revisions 109775 via svnmerge from tilghman1-7/+24
https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@109779 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Merged revisions 109111 via svnmerge from russell1-0/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@109454 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29Add documentation for setting username/password in SIP dial string.file1-1/+10
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105378 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27Bring Voicetronix driver up to date with current driverstilghman1-98/+238
(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: http://svn.digium.com/svn/asterisk/trunk@104502 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-26Merged revisions 104119 via svnmerge from russell1-0/+32
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: http://svn.digium.com/svn/asterisk/trunk@104120 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Adding more tls configuration details to sip.conf sample, with a list of ↵bbryant1-0/+21
valid ciphers provided in both files. .. First commit since July, woot git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14Change the queue holdtime announcement to happen at any interval (not just ↵mmichelson1-5/+2
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: http://svn.digium.com/svn/asterisk/trunk@103687 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-11Merged revisions 103315 via svnmerge from kpfleming1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@103316 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-07Merged revisions 102807 via svnmerge from kpfleming1-2/+6
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: http://svn.digium.com/svn/asterisk/trunk@102808 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-06Merged revisions 102651 via svnmerge from russell1-1/+4
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: http://svn.digium.com/svn/asterisk/trunk@102652 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-04Change examples to use G here also.qwell2-5/+5
Closes issue #11875 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@102262 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-01Clarify the pooling functionality by changing the config file keywordtilghman1-5/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101824 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Clarify configuration file that can be misunderstoodoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101322 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Removing applications that wasn't ready for svn trunk, as trunk now hasoej1-20/+0
pre-release status. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101271 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Merged revisions 101219 via svnmerge from qwell1-4/+4
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: http://svn.digium.com/svn/asterisk/trunk@101220 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Add rtppage() application to do multicast or unicast RTP paging to SIP phones.oej1-0/+20
(closes issue #11797) Reported by: macbrody Patches: app_rtppage-20080130.c uploaded by macbrody (license 352) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-28Reintroduce more chan_vpb stuff that was removed in r100421 and r100422qwell1-0/+108
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100679 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-25Remove more remnants of chan_vpbqwell1-108/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100421 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-24Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which ↵file1-0/+4
drops packets that do not come from the remote party. (closes issue #8952) Reported by: amorsen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100206 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Add res_config_ldap for realtime LDAP engine.tilghman1-0/+141
(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: http://svn.digium.com/svn/asterisk/trunk@99696 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Change the Asterisk CLI startup commands feature to read commands to run ↵russell1-0/+12
from cli.conf after a discussion on the -dev list. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99642 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Documentation updatesoej1-8/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99483 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-21Adding the QUEUENAME variable to the variables set using the setqueuevar option mmichelson1-0/+1
in queues.conf. Suggestion comes from Shaun2222 on IRC. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99406 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-21Merged revisions 99341 via svnmerge from tilghman1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@99350 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-20correct the name of a CLI command for getting available device namesrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99232 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-20Merge changes from team/russell/console_devicesrussell1-1/+23
- 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: http://svn.digium.com/svn/asterisk/trunk@99227 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18Merge changes from team/group/sip-tcptlsrussell1-2/+14
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: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Add several busy detection related defines to menuselect.qwell1-3/+7
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: http://svn.digium.com/svn/asterisk/trunk@98998 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Merged revisions 98991 via svnmerge from qwell1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@98992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17major reliability and performance improvement in VWMI monitoring for FXO ↵kpfleming1-1/+3
ports (code by markster, me and dbailey) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98990 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Update res_phoneprov to default to setting the SERVER variable to the IPtwilson1-5/+10
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: http://svn.digium.com/svn/asterisk/trunk@98988 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Merge the changes from issue #10665 from the team/group/sip_session_timers ↵russell1-7/+35
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: http://svn.digium.com/svn/asterisk/trunk@98978 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-15Add the "filter" keywordtilghman1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98947 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add backupdeleted option to app_voicemailqwell1-0/+7
(closes issue #10740) Reported by: ruffle Patches: app_voicemail.diff uploaded by ruffle (license 201) 10740-voicemail.diff uploaded by qwell (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak (license 7) Tested by: blitzrage, mvanbaak, qwell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98889 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add 'auto' signalling mode for Zaptel channels.kpfleming1-5/+6
(closes issue #11690) Reported by: tzafrir Patches: signaling_to_signalling.diff uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir (license 46) zap_no_default_sig.diff uploaded by tzafrir (license 46) zap_signal_auto.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98436 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add a new global and per-peer option to chan_sip, qualifyfreq, which allows yourussell1-0/+9
to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98027 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Merged revisions 97753 via svnmerge from russell1-3/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines Remove other remnants of pbx_kdeconsole ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97758 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Several manager changes:tilghman1-2/+24
1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Added a new module, res_phoneprov, which allows auto-provisioning of phonestwilson2-0/+58
based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Adding the option of specifying a second interface in a member definition ↵mmichelson1-1/+5
for a queue. app_queue will monitor this second device's state for the member, even though it actually calls the first interface. This ability has been added for statically defined queue members, realtime queue members, and dynamic queue members added through the CLI, dialplan, or manager. (closes issue #11603, reported by acidv) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97203 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Merged revisions 96932 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r96932 | russell | 2008-01-07 14:47:52 -0600 (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96933 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Add a note about viewing the default set of documentation using the built-in ↵russell1-0/+9
http server git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96888 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-01 11:12:36 +0000