mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
1210 Commits (29bbb76c71dc5cafa4c0943e238960ff123f4eaa)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Leif Madsen
|
fc5a0587c1 |
Merged revisions 239834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r239834 | lmadsen | 2010-01-13 13:31:13 -0600 (Wed, 13 Jan 2010) | 8 lines Add more examples to extensions.conf showing how to use various functionality and provide commonly useful features. (closes issue #16090) Reported by: pprindeville Patches: extensions.conf-bugid16090.patch#3 uploaded by pprindeville (license 347) Tested by: tzafrir, pprindeville, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@239837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
a50f9f456f |
Merged revisions 239520 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r239520 | lmadsen | 2010-01-12 12:22:45 -0600 (Tue, 12 Jan 2010) | 6 lines Note that direct T.38 is not supported. (closes issue #16411) Reported by: stanusr Patches: __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen (license 10) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@239523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tzafrir Cohen
|
c213189326 |
Merged revisions 238313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | 2 lines Document the usefulness of explicit udp:// in the register string ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@238349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jared Smith
|
f27a27f0c8 |
Merged revisions 235298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r235298 | jsmith | 2009-12-15 23:24:58 -0600 (Tue, 15 Dec 2009) | 11 lines Merged revisions 235181 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines Add a line showing that we can use CIDR notation. patch by jsmith, after discussion with jtodd ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@235334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
faaf1ffc47 |
Merged revisions 233280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@233281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
fbbd80432b |
Merged revisions 230881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@230884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
487e679dc8 |
Merged revisions 229966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | 13 lines Merged revisions 229965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 lines Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@229969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
6a76e138e9 |
Merged revisions 227361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@227364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
0b59c7a796 |
Merged revisions 227162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@227165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
8a19549935 |
Merged revisions 226384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@226385 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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25a24086a6 |
Merged revisions 225033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from 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/branches/1.6.2@225035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
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4f02b5922d |
Merged revisions 224446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18 Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@224447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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26470fe031 |
Merged revisions 223756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
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86435bd7d2 |
Merged revisions 222548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct 2009) | 5 lines Remove 'keepstats' queue option from sample config, as it's no longer used. https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
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ba54bb0506 |
Merged revisions 222110 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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f820a30e22 |
Merged revisions 221592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
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5d6d3d7676 |
Merged revisions 221432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthias Nick
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e28d4da865 |
Merged revisions 221368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221371 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
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9a61ba3e28 |
Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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eb13d6330b |
Merged revisions 219061 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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b292004c84 |
Merged revisions 218361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
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9e57fd58e2 |
Merged revisions 216438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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574c8c2144 |
Merged revisions 215955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jason Parker
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bb0909ed90 |
Merged revisions 213494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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8afb651407 |
Merged revisions 213098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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1f254de8a8 |
Merged revisions 212857 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18 Aug 2009) | 4 lines Make the default extconfig.conf match entries with the sample res_mysql.conf. This eliminates a future source of possible confusion with the configuration of 1.6.1 and higher. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@212866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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bbf4a08fe7 |
Merged revisions 210190 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
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12a938f833 |
Merged revisions 209132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209135 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
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27447a42bd |
Merged revisions 208813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 | mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 lines add default alias reload to run module reload. Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
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53533451ae |
Merged revisions 207095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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8ff8696c38 |
Merged revisions 206873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
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8300f67982 |
Merged revisions 204440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 | russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing .sample). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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45d9ad47ca |
Merged revisions 203699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Moises Silva
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285034ebdd |
Merged revisions 200799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 | moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@200807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Moises Silva
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b9f6ee94dd |
Merged revisions 200477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@200512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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2292d0e19b |
Merged revisions 198791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@198794 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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de6cc4ba3d |
Merged revisions 198186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines Suggesting that only a single timing module be loaded is no longer necessary. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@198188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Gavin Henry
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e1587340e2 |
issue #15155 and issue #15156 from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@197441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Sean Bright
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b99254b007 |
Merged revisions 197089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@197092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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49cda05d53 |
Merged revisions 196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@196452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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c74e17490c |
Merged revisions 194765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@194768 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Kevin P. Fleming
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11d58b0448 |
Merged revisions 193194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@193197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Kevin P. Fleming
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60afdc8089 |
Merged revisions 191955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@191958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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0b21b28a55 |
Merged revisions 186444,186447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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0dcd1d7601 |
Merged revisions 186175 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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f381fb5421 |
Merged revisions 186060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Richard Mudgett
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9b6189b180 |
Merged revisions 185123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@185129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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f50cb85123 |
Merged revisions 183914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@183917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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77a6840fd3 |
Add MFC/R2 support for chan_dahdi.
This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Michiel van Baak
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f1ae8e9f3b |
Provide correct hint to debug SIP trouble in the default config
(closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |