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2009-10-21Merged revisions 225169 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines Isolate the frame returned from ast_translate(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Blocked revisions 225103 via svnmergeTilghman Lesher
........ r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines Suffix is not needed for a match ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Apparently, I don't need to specify the ".so" suffix to get a matchTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Add support for specifying the IP address to use for media streams in sip.confJoshua Colp
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Turn on DENOISE filter for all conference participants.Tilghman Lesher
(Fixes SWP-238) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Revert media_address commit, I'm going to roll a fix to the SDP generation ↵Joshua Colp
in the next version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Merged revisions 225032 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Add support for specifying the IP address to use for media streams in sip.confJoshua Colp
(closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Merged revisions 224931 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Make PRI_SUBCMD_xxx handling subaddress friendly.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-20Merged revisions 224855 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-20Merged revisions 224773 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-20Added information to CHANGES about the dynamic range compression feature ↵Matthew Nicholson
added to dahdi. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Merged revisions 224670 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Add dynamic range compression support for analog channels.Matthew Nicholson
(closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Merged revisions 224565 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Remove useless debugging message.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Remove a completed project and add anotherTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Add a callback to sig_pri which is called when sig_pri is going to queue a ↵Joshua Colp
control frame on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19Allow ODBC storage to be queried with multiple mailboxes, and remove ↵Tilghman Lesher
multiple goto's. This corrects an issue reported on the -users list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-18Clarify that "forcecommit" is NOT an alias for "autocommit", but instead ↵Tilghman Lesher
controls the default disposition of uncommitted transactions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-17Remove unnecessary typedefTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-17fix typo, sorryJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-17Merged revisions 224330 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-16Merged revisions 224260 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (issue #14292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15Create an API for adding an optional time unit onto the ends of time periods.Tilghman Lesher
Two examples of its use are included, and the usage could be expanded in some cases into certain configuration options where time periods are specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15Readd removed ability to allow listening to one side of the call in app_chanspyJeff Peeler
(Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15chan_dahdi.conf.sample changes for DTMF CID detect Doug Bailey
Explains new options for detecting DTMF CID on fxo lines (issue #9096) Reported by: fleed Patches: chan_dahid_sample_config.patch uploaded by sum (license 766) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15Properly handle PUT requests for CALENDAR_WRITE()Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-14Add missing 'getnum' fieldTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-14Allow for adding message body to the SIP NOTIFY messageJeff Peeler
Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13use Calendar: instead of Calendar/ for devstateTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix some doxygen format problems and trim trailing whitespace.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix compiler warning.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Revert inadvertant code commit to app_originateTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-13Fix handling of notification calls w/ the dialing apiTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Merged revisions 223804 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 optionsDavid Vossel
SWP-151 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Recorded merge of revisions 223692 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover. (issue #16025) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12Remove automatic switching from T.38 to voice mode in chan_sip.Kevin P. Fleming
chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-11Check the proper page for the SENDRPID flag.Mark Michelson
If a pending reinvite were sent, we might not properly send connected party info since we were checking the wrong flag. This was a rare occurrence, but could still happen nevertheless. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-11Merged revisions 223550 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009) | 2 lines Remove a duplicate ao2_iterator_destroy(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-11Merged revisions 223485-223486 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Fix handling of floating times and datesTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Adding note about TLS usageOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Add an additional note on TLS supportOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Adding some information on TLS supportOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09Properly return "free" on confirmed events that are freeTerry Wilson
CONFIRMED status doesn't imply busy or free, that is handled with the TRANSP field. Luckily, libical already sets the is_busy status on the span for us. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09Initiate T.38 switchover when acting as called party, regardless of FAX ↵Kevin P. Fleming
direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09Merged revisions 223225 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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