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${ noResults }
18763 Commits (1c9fe00920c4d887328400bf43c3e16fc9d24c37)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kevin P. Fleming
|
1c9fe00920 |
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
20743ec07d |
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/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
|
10e8ee1746 |
Add a few missing events to analog_handle_event.
The reported bug was actually only for pulsedigit, dtmfup, and dtmfdown handling. Also added recognition for fax events (just some verbose output) and fixed handling for the ec_disabled_event. In order to make comparing the analog version of events to the DAHDI events easier, the ordering has been changed to follow that of the DAHDI events. (closes issue #15924) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
3cce68d329 |
Merged revisions 222026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
c1c25181af |
Merged revisions 221970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
ba10edfcac |
Initialize a variable that we check immediately upon startup.
(closes issue #15973) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
|
80f0a242a7 |
Whitespace change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
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1a02b4c659 |
Whitespace change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
|
3b83d2b414 |
Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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bd7ca4b764 |
One more off-by-one in trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221781 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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8c7b3cf738 |
Merged revisions 221776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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dd0c76a9d3 |
Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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c0a884ba29 |
Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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9c05faf76d |
Prevent deadlock if chan_dahdi attempts to change PRI channel names.
The PRI channels can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ISDN-<span>-<sequence-number> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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aaa7284c00 |
outbound tls connections were not defaulting to port 5061
(closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/357/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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ff555b5e12 |
Sync up UPGRADE.txt with the 1.6.2 version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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19ba91cd22 |
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/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
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da169b2db4 |
Merged revisions 221588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Olle Johansson
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73697dc2c7 |
Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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d043f52a2d |
Cleaned up merge from r221432
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthias Nick
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00bb578898 |
Prevents from division by zero
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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a5eee590f4 |
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/trunk@221432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthias Nick
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63984d5c21 |
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/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Terry Wilson
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717d2ec3c9 |
Remove spurious debug
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
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10ce6cd757 |
Use rtp properties instead of adding a callback
Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
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865daf4858 |
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/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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a4ece92018 |
Merged revisions 221200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
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0e805a51ec |
Modify VoiceMailMain()'s a() argument to allow mailboxes to be specified by name.
(closes issue #14740) Reported by: pj Patches: issue14740_09022009.diff uploaded by seanbright (license 71) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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3ac28f1e0f |
Clarify documentation for VoiceMailMain()'s a() option.
We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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751f191dfa |
Allow locks to be inherited through a masquerade without causing starvation.
(closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
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01181a27a0 |
Fix channel reference leak.
ast_cel_report_event would geet a reference to the bridged channel. However, certain return paths, such as if CEL was not enabled, would result in a reference leak. All return paths now properly unref the channel. (closes issue #15991) Reported by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220995 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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75725abeb4 |
Blocked revisions 220907 via svnmerge
........ r220907 | mnicholson | 2009-09-29 15:14:29 -0500 (Tue, 29 Sep 2009) | 10 lines Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220928 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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cee8c6cd47 |
Get rid of annoying and cryptic debug messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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6f5e763fe5 |
Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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4b4432b67e |
Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel.
(closes issue #15964) Reported by: shrift git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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a9154a905a |
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
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f3f456f8b6 |
Miscellaneous minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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c8a141905e |
Merged revisions 220717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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05f94a05c2 |
Fix building of registration entry in build_peer when using callbackextension
Check for remotesecret option was unintentionally always true, which therefore caused the secret option to never be used. Thanks to dvossel for pointing out the exact fix. (closes issue #15943) Reported by: tpsast git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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307bf124d2 |
Locking issues dealing with service_lock.
* Removed unneeded and uninitialized service_lock. * Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Michiel van Baak
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7eac18b09c |
add name argument for the CALLERID dialplan function to the xml documentation.
Pointed out to me on IRC by snuff-home. Thanks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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cd7287dbfb |
Allow AES to compile, when OpenSSL is not present.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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146c352144 |
Reduce indentation in sig_pri_available().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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d04158f5b1 |
Eliminate unnecessary include of version.h in manager.c.
Including version.h here causes this file to get recompiled after every commit or update, which is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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8c30540269 |
Correct sense of logic test committed in revision 220494.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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aabdc575a5 |
Don't use hash-based lookups for ast_channel_get_by_name_prefix().
ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based channel lookups, but this will not work properly when the channel's full name was not supplied; for name-prefix searches, there is no value in doing a hash-based lookup, and in fact doing so could result in many channels being skipped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220494 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Philippe Sultan
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b11b94a083 |
Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
17180120bf |
Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
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90746d26f3 |
fixes tcptls_session memory leak caused by ref count error
(closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
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f150b48bc0 |
Add bridge related dial flags to the bridge app
Most of the functionality here is gained simply by setting the feature flag on the bridge config. However, the dial limit functionality has been moved from app_dial to the features code and has been made public so both app_dial and the bridge app can use it. (closes issue #13165) Reported by: tim_ringenbach Patches: app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |