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${ noResults }
2009 Commits (dd0c76a9d3248f670c62100e4f3423a762dc72f5)
Author | SHA1 | Message | Date |
---|---|---|---|
|
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 |
|
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 |
|
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 |
|
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 |
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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 |
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a5612f6e6f |
Add Mantis work flow documention.
This commit adds the doxygen changes that I've made to describe the Mantis work flow documentation for the open source issue tracker. This should make it easier to determine the flow of issues through the issue tracker, and what those statuses mean. (closes issue #15902) Reported by: lmadsen Patches: mantisworkflow.h uploaded by lmadsen (license 10) Review: https://reviewboard.asterisk.org/r/367/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b27a54b8de |
Merged revisions 219136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219139 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d8457eb18c |
Detect whether we actually have the long double type, before looking for those functions.
(closes issue #15017) Reported by: tzafrir Patches: 20090916__issue15017.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c9dd40c1f6 |
Verify support for wide ODBC character types before using them.
(closes issue #15870) Reported by: nic_bellamy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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29b72bc343 |
Add doxygen to ast_event_subscribe for the description.
Most importantly, note that a NULL description will cause a crash, as I just experienced that firsthand. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5d0790027a |
Ensure that the default autoconf CFLAGS are not used.
A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5dfaf5c8b7 |
Fix trunk breakage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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43f36d9582 |
make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8430c87faa |
Added detection DTMF CID without polarity change alert.
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi monitoring loop so DTMF CID can be detected without the need of a polarity change precursor. (closes issue #9096) Reported by: fleed Patches: 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819) Tested by: cyberplant, sum, maturs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d09f9fd00a |
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/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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80973cb97f |
Revert attempt to standardize with _POSIX_C_SOURCE.
This did not function in the way that was intended, causing more compatibility issues than it solved. It is best, therefore, that it be simply removed. (Discussed with kpfleming; agreement to remove was reached.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6227229264 |
Let's compile again on OpenBSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9f7a3466ef |
Various patches, to enable Asterisk to once again compile on Mac OS X.
One note on defining _POSIX_C_SOURCE: while this feature test macro works to require certain behaviors on Linux, it works differently on *BSD platforms to REMOVE certain API calls that are not in the POSIX specification, such as vasprintf(3). Thus, defining it while depending upon vasprintf (and other extensions to the POSIX standard) to be defined is a recipe to ensure that Asterisk is only buildable on Linux. Hence, this define which was meant to INCREASE portability, effectively ensures the opposite. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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01fd08d5b9 |
If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly.
Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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802b79e3ca |
Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved.
Cross-compilation environments want to provide 'defaults' for compiler and linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the environment or as command-line arguments to the configure script. This patch modifies the configure script and Makefile to preserve these settings and ensure they are used in the build process. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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27fb59b001 |
Fix some incorrect documentation of sched_thread functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214650 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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faa0b8efae |
Merged revisions 214517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before. (closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214518 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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74d7f7f788 |
Merged revisions 214436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines One more build system change, to make the descriptions look better, if we have better information. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f2e9a73c81 |
Merged revisions 214357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines Make autoheader descriptions render correctly in our autoconfig.h file. (Figured out while working with issue #14906) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ddf5a08d83 |
Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all.
(closes issue #15169) Reported by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8fb871b1ad |
Merged revisions 213559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. (closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0f0ad824f0 |
Relax check for XOPEN_VERSION.
It's not clear that we actually require XOPEN_VERSION to be 600 or greater at this time, so skip the check for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4222e1c367 |
Define our desires for POSIX and X/OPEN API features properly.
Based on a post on the gcc-help mailing list and some subsequent reading, we can increase our portability to various platforms by directly defining the POSIX and X/OPEN API feature sets we wish to have available. This patch does that, and also includes a double-check to ensure that the system we are compiling on can actually provide the requested feature sets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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606112e234 |
Add two more API calls for getting the current glue and channel in bridging code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1effb11ef5 |
Add an API call for retrieving the engine in use by an RTP instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a737df8603 |
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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eb449d514e |
Minor improvements to app_fax.
This patch makes some small changes to handle watchdog timeouts in a better way, and also uses a 'cleaner' method of including the spandsp header files. (closes issue #14769) Reported by: andrew Patches: app_fax-20090406.diff uploaded by andrew (license 240) v1-14769.patch uploaded by dimas (license 88) Tested by: freh, deti, caspy, dimas, sgimeno, Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28ad5ced1a |
Initial minimum ast_party_caller support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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48363c16e1 |
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ba020fc390 |
Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d81d6d3415 |
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
(closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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17e2d9fdbc |
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0a6e06c7ff |
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d223e3636f |
Merged revisions 208083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) | 4 lines Export symbols for functions included in our compatibility headers. (closes issue #15556) Reported by: smw1218 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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299a9ff3fa |
Remove trailing whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208018 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4cf8a968fd |
Add an API for reporting security events, and a security event logging module.
This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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67d1957e60 |
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f3560397be |
Merged revisions 205599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ba2a8457b8 |
Merged revisions 205471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e39a252b1e |
Merged revisions 205409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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15b94d1182 |
Merged revisions 205215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e75ae63ac2 |
Fix a few compilation problems found when building Asterisk against uClibc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205214 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0e8c630224 |
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ee0cd5a32c |
Add a configure check for Reverse Charging Indication support in LibPRI.
Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c381cf82e7 |
Wrap rtp_engine.h header comments to 80 characters.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |