path: root/configs
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2011-07-22Merge branch 'master' of McHardy5-3/+62
2011-07-21Merged revisions 329204 via svnmerge from rmudgett1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@329205 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-13Merged revisions 328014 via svnmerge from rmudgett1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line Add ATXFER_NULL_TECH note in features.conf.sample. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328016 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-10Full T.38 handshaking and fax detectionmay1-0/+12
Add full t.38 handshaking for OOH323 that are required for newest T.38 gateway codes. Add fax detection (cng tone, t38) and dialplan redirection to fax ext on fax event detected. Add OOH323() function to set/get t38support and faxdetect parameters. (closes issue ASTERISK-17754) Reported by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot (license 52) issue19183-final.patch uploaded by may213 (license 454) Tested by: may213, irroot Review: https://reviewboard.asterisk.org/r/1174/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327359 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Adds pass-through support for codec CELT.dvossel1-1/+18
This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326855 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Updates confbridge.conf video documentation and adds dtmf action for ↵dvossel1-2/+21
releasing video src. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326782 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Merge branch 'master' of McHardy4-2/+35
2011-06-30Merged revisions 325935 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325936 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Video support for ConfBridge.dvossel1-0/+22
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Commit "distrotech" app_queue changes to Trunkirroot1-0/+11
* Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325483 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-20Merged revisions 324241 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324242 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-17Merge McHardy1-0/+5
2011-06-13Addition of "outofcall_message_context" sip.conf option.dvossel1-0/+5
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merge McHardy15-11/+449
2011-06-07Merged revisions 322189 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322190 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321685 via svnmerge from lmadsen1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321689 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-01Support routing text messages outside of a call.russell2-0/+13
Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321546 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 319938 via svnmerge from jrose1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319939 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.rmudgett1-0/+13
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319427 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-09Merged revisions 318148 via svnmerge from jrose1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318162 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Updated the sample pbx_lua config file to reflect autoservice changes.mnicholson1-6/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317818 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Add CEL extra field to cel_pgsql.russell1-1/+1
(closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317482 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317058 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317059 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-21Merged revisions 314628 via svnmerge from mnicholson3-0/+24
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314666 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-21New HD ConfBridge conferencing application.dvossel1-0/+302
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-18Problems with ISDN MWI to phones.rmudgett1-1/+16
The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314116 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-14Add Device State Information CCSS for Generic Devices.rmudgett1-1/+43
Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313744 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-13Add 'description' field for CLI and Manager outputlmadsen4-0/+14
(closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313528 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-10Merge branch 'master' of McHardy2-15/+27
2011-04-05Merged revisions 312766 via svnmerge from mnicholson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@312767 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-31Merged revisions 311930 via svnmerge from tilghman1-15/+16
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines Incorrect default example; the field is actually internally named "clid", not "callerid". (closes issue #19040) Reported by: wcselby Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@311931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-18Merge branch 'master' of McHardy4-3/+21
2011-03-17Merged revisions 311050 via svnmerge from alecdavis1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@311051 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-07Merged revisions 309765 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309766 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-04Add setvar option to calendaringtwilson1-0/+7
Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309640 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-04Add support for defining hints from pbx_luamnicholson1-0/+11
(closes issue #16024) Reported by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309493 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-26chan_dect: support configured timeouts for location registrationPatrick McHardy1-1/+4
Signed-off-by: Patrick McHardy <kaber@trash.net>
2011-02-26chan_dect: use IPEI as primary key for PPs in databasePatrick McHardy1-25/+0
Signed-off-by: Patrick McHardy <kaber@trash.net>
2011-02-26chan_dect: fix up for latest libdect changesPatrick McHardy1-0/+2
Specify the cluster to bind to. Signed-off-by: Patrick McHardy <kaber@trash.net>
2011-02-26chan_dect: add authentication, ciphering and key allocationPatrick McHardy1-0/+6
Signed-off-by: Patrick McHardy <kaber@trash.net>
2011-02-26Import chan_dectPatrick McHardy1-0/+50
Re-import chan_dect due to a switch to the trunk branch. Signed-off-by: Patrick McHardy <kaber@trash.net>
2011-02-24Merged revisions 308679 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308680 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵dvossel1-0/+71
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Merged revisions 307467 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample configuration. Discovered by Philippe Lindheimer and pointed out on #asterisk-dev ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307468 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Documentation Updateslathama2-1/+5
Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Define the MCID acronym in chan_dahdi.conf.sample.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306793 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Pass a MCID request to the bridged channel.rmudgett1-0/+5
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306755 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.rmudgett1-1/+26
The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306396 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 supportlathama1-0/+4
(issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305988 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-02Replacing doc/* and asterisk.pdf with wiki linkslathama9-12/+12
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305843 f38db490-d61c-443f-a65b-d21fe96a405b