aboutsummaryrefslogtreecommitdiffstats
path: root/configs
AgeCommit message (Collapse)AuthorFilesLines
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
2008-01-02another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl ↵kpfleming1-2/+13
if it is present, but doesn't parse any supplied parameters yet (this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96019 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge the main set of changes from team/russell/chan_console.russell2-2/+71
Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Adding support for storing the queue log entries in a realtime backend.mmichelson1-0/+1
(closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94782 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-25Change the abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMFtilghman1-9/+9
character. Also, fix the documentation to match the code. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94772 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-22Change the name of config file entries for keypad regionsrizzo1-18/+18
from 'keypad_entry' to 'region'. Fix the example file accordingly. Also make some fixes in the code do reset entries on reload of the keypad. The recently committed kpad2.jpg has the correct names. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94638 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Merging the queue-penalty branch. In short, this allows one to dynamically ↵mmichelson2-0/+27
adjust the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending on the amount of time passed. The purpose is to allow the call to open up to more (or maybe just different) members without the caller's losing his place in the queue. See configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample for how to associate a rule with a queue. Along with the functional changes, new CLI and manager commands exist to show the rules defined and there is an additional CLI command to reload the queue rules. Future enhancements that may be made: support for realtime queue rules and support for dynamically adding a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write this myself very soon). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94370 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-20Add a bit more to the description of the "mwimonitor" option.russell1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94320 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Adding the ability to specify the To: header in an outbound INVITEoej1-11/+31
by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16HUGE improvements to QoS/CoS handling by IgorGoej8-11/+32
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Update documentationoej1-7/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93160 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Make more timers settable in SIP so that we can force timeout earlier on ↵oej1-0/+7
non-responsive SIP servers. Thanks, jcmoore, for the patch! Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9) (closes issue #9771) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93159 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-15configuration options related to video support.rizzo1-0/+63
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93145 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Remove use of privacy.conf by the Privacy app.tilghman1-3/+0
Reported by: eliel Patch by: eliel (Closes issue #11344) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93066 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Update documentation for pbx_lua.qwell1-3/+3
Closes issue #11492, patch by mnicholson. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91832 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Change cdr_manager to use a "CDR" level, rather than the (overcrowded) ↵tilghman1-2/+2
"call" level. (Closes issue #11015) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Remove second prefix line. Only need it documented once in the same file.file1-1/+0
(closes issue #11472) Reported by: eserra Patches: http.conf.sample.diff uploaded by eserra (license 45) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Rename "username" to "defaultuser" to match with "defaultip".oej1-5/+6
"Username" still works, but is deprecated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Add support for monitoring MWI on FXO lines.russell1-0/+13
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90949 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Updating sample queues.conf file to show how multiple periodic announcementsmmichelson1-0/+7
may be specified since this was not documented previously (closes issue #11432, reported and patched by Laureano) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90528 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Adding support for the "automixmonitor" dial and queue options.mmichelson1-0/+1
This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Merged revisions 90098 via svnmerge from kpfleming1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90100 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Adding support for realtime music on hold. The following are the main points:mmichelson2-0/+6
1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89946 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89634 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89635 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89624 via svnmerge from oej1-2/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89625 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89622 via svnmerge from murf1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.oej1-1/+1
Both still works in this version. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89613 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Thanks to pnlarsson for noting the spelling error in the cli commands. Also, ↵murf1-5/+18
added some verbage about the new algorithm to CHANGES. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89583 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Merged revisions 89559 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines We previously attempted to use the ESCAPE clause to set the escape delimiter to a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89561 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵oej1-15/+21
call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24closes issue #11363; where the pattern _20x. buried in an included context, ↵murf1-0/+20
didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89547 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23Merged revisions 89527 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) | 3 lines mvanbaak pointed out a spelling error in this sample configuration file. While I was at it, I went ahead and tweaked it a little bit more. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89529 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Changed occurrences of "busy-level" to "busylevel" in sip.conf.samplemmichelson1-4/+4
in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89453 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Add support for application/dtmf SIP INFO dtmf handling. Yep, anotheroej1-1/+2
way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89278 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89173 via svnmerge from crichter1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89179 f38db490-d61c-443f-a65b-d21fe96a405b