mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
1166 Commits (bb0b64d8420e15910ee7e8a637595dd67893e8a4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
2ea15c9ac6 |
Merged revisions 257768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 Apr 2010) | 2 lines Removing unused configuration parameters ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@257770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
ab4cb6e425 |
Merged revisions 257262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 lines Yet another issue where the conversion of the application delimiter to comma caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@257264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
|
a58fd30392 |
Merged revisions 257146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr 2010) | 16 lines Merged revisions 257070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@257214 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Leif Madsen
|
8585bbf16e |
Merged revisions 255504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@255506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Leif Madsen
|
cf7512bf0d |
Finish syncing documentation for tlsbindaddr.
(issue #17054) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@255068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
ef9708211b |
Merged revisions 255021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@255026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
dbdb786ef4 |
Merged revisions 253712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 | tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@253713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
5e188c61da |
Merged revisions 253028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@253030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
a9eaa67660 |
Merged revisions 252762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@252764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
47fa7c3768 |
Merged revisions 252534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@252536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
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77bd4bac6a |
Merged revisions 252089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@252135 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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4e48910948 |
Merged revisions 250045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@250048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
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8da2b4c813 |
Merged revisions 249893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@249896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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31bafc03f3 |
Merged revisions 245945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) | 9 lines Merged revisions 245944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@245947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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27f5939d6d |
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/1.6.1@239836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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2bf66584ca |
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/1.6.1@239522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tzafrir Cohen
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ec6d1db5e9 |
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/1.6.1@238351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jared Smith
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b5aa2a5e49 |
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/1.6.1@235333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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7d19415db3 |
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/1.6.1@233283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
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d4fecf8429 |
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/1.6.1@230883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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7b8f311cd2 |
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/1.6.1@229968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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daf31332fd |
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/1.6.1@227363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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e999ce942d |
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/1.6.1@227164 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
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8ed14498d8 |
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/1.6.1@226386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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219e2238a9 |
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/1.6.1@225062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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2c5df9ffd4 |
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/1.6.1@223758 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
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d605a00c13 |
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/1.6.1@222112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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4df41b1609 |
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/1.6.1@221602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Nicholson
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f52743ced9 |
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/1.6.1@221478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthias Nick
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7c411eeb39 |
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/1.6.1@221370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Terry Wilson
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1a56b67549 |
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/1.6.1@221302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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23faa9744a |
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/1.6.1@219062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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7a56903be3 |
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/1.6.1@218363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
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57accf67ed |
Make code and documentation agree with each other
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@216657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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6108a2a894 |
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/1.6.1@216646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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38bbe9653f |
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/1.6.1@216004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
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1bd918db02 |
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/1.6.1@213496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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0e582be3ec |
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/1.6.1@212862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
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15c964f355 |
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/1.6.1@209134 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
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6063455e79 |
Merged revisions 145293,158010 from
https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@207287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
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621747431c |
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/1.6.1@207101 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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003c34549b |
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/1.6.1@206875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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642b571683 |
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/1.6.1@203703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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7375533824 |
Merged revisions 165180,200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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fcdc8c20f4 |
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/1.6.1@198793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
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75b4fa611d |
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/1.6.1@198187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Gavin Henry
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89a59c618e |
issue #15155 and issue #15156 from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
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7cf0275af0 |
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/1.6.1@197091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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28a71581e0 |
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/1.6.1@196453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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31de47c770 |
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/1.6.1@194767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |