mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
25390 Commits (e81b873fa2f23a277194722a00ec445127a68e57)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Jonathan Rose
|
e81b873fa2 |
chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'
ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ ........ Merged revisions 413876 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Walter Doekes
|
0eda637fc4 |
h264: Fix H264 SDP payload format.
https://tools.ietf.org/html/rfc3984#section-8.1 says profile-level-id takes 3 bytes in base16 (6 hex digits). This fixes video setup in certain cases. ASTERISK-23664 #close ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ ........ Merged revisions 413791 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413792 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Walter Doekes
|
f66e9d6c9e |
rtp: Fix case typo in H263+ mime.
http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical mime subtype is "H263-1998", not "h263-1998". Original code was added in r183101 on 2009-03-19 02:26:50 +0100. This fixes issues with Polycom phones. ASTERISK-23665 #close ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me. Review: https://reviewboard.asterisk.org/r/3529/ ........ Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413788 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413789 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Richard Mudgett
|
8b6ab4782a |
chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled.
When overlap dialing is enabled, the lack of inband audio available information in the SETUP_ACKNOWLEDGE events causes an interoperability problem with SIP. sig_pri doesn't know if there is dialtone present when a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183 Session Progress and blocks the desired 180 Ringing message when the ALERTING message comes in. * Made the configure script detect if the installed version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * Using the new API, made generate an AST_CONTROL_PROGRESS frame on an incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio is present instead of assuming that dialtone is present. * Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio available indication only if dialtone is expected. The change also makes the fallback behaviour of sending the PROGRESS message better by sending it only if dialtone is expected. * Changed receiving a PROCEEDING message to not generate an AST_CONTROL_PROGRESS frame if the progress indication ie indicates non-end-to-end-ISDN. This helps interoperability with SIP. * Changed sending a PROCEEDING message in response to an AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It was silly to do so anyway because the channel driver doesn't know if inband audio is even available. This helps interoperability with SIP. This patch and a corresponding change in libpri work together to allow Asterisk to control the inband audio available progress indication ie on the SETUP_ACKNOWLEDGE message when dialtone is present. AST-1338 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ ........ Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
|
552dbe531e |
Fix compiler warning from GCC 4.10 fixup.
........ Merged revisions 413766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Jonathan Rose
|
643a7f02d6 |
app_chanspy: Fix a test that was failing on account of r413551
ASTERISK-23381 #close ASTERISK-23381 #comment Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413710 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413712 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Joshua Colp
|
d134150be2 |
framehooks: Add callback for determining if a hook is consuming frames of a specific type.
In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Joshua Colp
|
e2ed86e4ca |
Undoing framehook support. Issues were uncovered by Bamboo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Joshua Colp
|
3b3e4b9b95 |
framehooks: Add callback for determining if a hook is consuming frames of a specific type.
In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Kinsey Moore
|
21949def8e |
Fix 32bit build for func_env
........ Merged revisions 413592 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413597 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Kinsey Moore
|
abd3e4040b |
Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
|
f3b55da1b8 |
http.c: Remove dead code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413572 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Jonathan Rose
|
5770483217 |
app_chanspy: Fix a bug where Barge mode could fail
If the barge audiohook was attached prior to the spyee and its peer actually being bridged, the audiohook would not be applied and the connected peer would not be able to hear audio from the spy when the spy is in barge mode. (closes issue ASTERISK-23381) Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413551 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413556 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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f2ca3438e7 |
app_queue: Extend documentation for various Manager actions and events.
........ Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413486 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413487 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Mark Michelson
|
68066907e1 |
Ensure that presence state is decoded properly on Asterisk startup.
The CustomPresence provider callback will automatically base64 decode stored data if the 'e' option was present when the state was set. However, since the provider callback was bypassed on Asterisk startup, encoded presence subtypes and messages were being sent instead. This fix makes it so the provider callback is always used when providing presence state updates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
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a92f0a9e83 |
app_confbridge: Fixed "CBAnn" channels not going away.
Fixed a ref leak in conf_handle_talker_cb() everytime the conference bridge was found to report a channel's talker status change. The resulting leak caused the "CBAnn" channels and the conference bridge to never be destroyed. Thanks to Richard Kenner on the asterisk-user's list for locating the problem. Reported by: Richard Kenner ........ Merged revisions 413454 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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90b9413a0d |
app_confbridge: Fix ref leak in CLI "confbridge kick" command.
Fixed ref leak in the CLI "confbridge kick" command when the channel to be kicked was not in the conference. ........ Merged revisions 413451 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413452 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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2d572eafb9 |
Fix encoding of custom prepare extra data.
Patches: res_config_odbc-take2.patch by John Hardin (License #6512) ........ Merged revisions 413396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413397 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413399 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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065bd7d703 |
Improve XML sanitization in NOTIFYs, especially for presence subtypes and messages.
Embedded carriage return line feed combinations may appear in presence subtypes and messages since they may be derived from user input in an instant messenger client. As such, they need to be properly escaped so that XML parsers do not vomit when the messages are received. ........ Merged revisions 413372 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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9eae6c3f5b |
Check for an act on failures to update contacts during registration.
There was an underlying issue in a realtime backend where database updates would fail. Since we were not checking for failure, we would end up in a strange state where the old database entry was still present but Asterisk thought that it had been updated. Now when an entry fails to update, we print a warning and delete the old contact from sorcery so there is no mismatch between foreground and backend state. Patches: res_pjsip_registrar.patch by John Hardin (License #6512) ........ Merged revisions 413358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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3f5d4516bd |
Ensure that all parts of SQL UPDATEs and DELETEs are encoded.
Patches: res_config_odbc.patch by John Hardin (License #6512) ........ Merged revisions 413304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413305 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413306 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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795ed566d9 |
Blocked revisions 413282
........ Correct variable traversal logic in res_config_odbc's update_odbc function. Closes issue ASTERISK-23675 Reported by Leando Dardini Patches: asterisk-23675-odbc-linkedlist-traversal_12.diff uploaded by Michael L. Young (license #5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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ff1841fcfb |
Prevent crashes in res_config_odbc due to uninitialized string fields.
Patches:
odbc-crash.patch by John Hardin (License #6512)
........
Merged revisions 413241 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 413251 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 413258 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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11 years ago |
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Mark Michelson
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ff1658ed3b |
Return the number of rows affected by a SQL insert, rather than an object ID.
The realtime API specifies that the store callback is supposed to return the number of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were inserted causes problems to the function's callers. To give an idea of how strange code can be, this is the necessary code change to fix a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that the registrar would attempt to insert contacts into the database. Because of the 0 return from res_config_pgsql, the registrar would think that the contact was not successfully inserted, even though it actually was. As such, even though the contact was query-able and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered, meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE. The necessary fix applies to all versions of Asterisk, so even though the bug reported only applies to Asterisk 12+, the code correction is being inserted into 1.8+. Closes issue ASTERISK-23707 Reported by Mark Michelson ........ Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
|
119599407b |
res_pjsip_refer: Add Referred-By header on INVITE for blind transfers.
Per rfc3892, the Referred-By header in a REFER must be copied into the referenced request (IE. The outgoing INVITE to the transfer target). * Automatically put the Referred-By header in the outgoing INVITE message if the SIPREFERREDBYHDR channel variable is defined with a value. * Made chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance so chan_pjsip has a better chance to interoperate. * Fixed refer_blind_callback() and refer_incoming_refer_request() to not modify the data in the pointer returned by pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data since the calling routine doesn't own the buffer. ASTERISK-23501 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3514/ ........ Merged revisions 413210 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Jonathan Rose
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57372e61d2 |
Parking: Add 'AnnounceChannel' argument to manager action 'Park'
(closes ASTERISK-23397) Reported by: Denis Review: https://reviewboard.asterisk.org/r/3446/ ........ Merged revisions 413196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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120ac66df9 |
Make behavior of the PRESENCE_STATE 'e' option more consistent.
When writing presence state, if 'e' is specified, then the presence state will be stored in the astdb encoded. However, consumers of presence state events or those that query for the presence state will be given decoded information. If base64 encoding is desired for consumers, then the information can be base64-encoded manually and the 'e' option can be omitted. closes issue ASTERISK-23671 Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3482 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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fc4c5ca3de |
Remove unnecessary repetition checks from res_pjsip_exten_state
The PBX core already takes care of ensuring that repeated state changes are not communicated to exten state consumers. Because the check in res_pjsip_exten_state was incomplete, it was causing valid presence state changes not to be sent out. For instance, if the presence state did not change but the message or subtype did, then no presence-related NOTIFY request would be sent out. closes issue ASTERISK-23672 Reported by Mark Michelson ........ Merged revisions 413173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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45a7132480 |
res_pjsip: Add the ability to configure ciphers based on name.
Previously this code would only accept the OpenSSL identifier instead of the documented name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by: Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/ ........ Merged revisions 413159 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413160 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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20750e261b |
chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues.
* Fixed early exit in sip_msg_send() not destroying the message iterator. * Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() tolerant of a NULL iter parameter in case ast_msg_var_iterator_init() fails. * Made ast_msg_var_iterator_destroy() clean up any current message data ref. * Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() use iter instead of i. * Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers(). ........ Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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f2a060f502 |
chan_pjsip: Fix deadlock when retrieving call-id of channel.
If a task was in-flight which required the channel or bridge lock it was possible for the synchronous task retrieving the call-id to deadlock as it holds those locks. After discussing with Mark Michelson the synchronous task was removed and the call-id accessed directly. This should be safe as each object involved is guaranteed to exist and the call-id will never change. ........ Merged revisions 413140 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Kinsey Moore
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a7fc217837 |
Websocket: Add session locking and delay close
This resolves a race condition where data could be written to a NULL FILE pointer causing a crash as a websocket connection was in the process of shutting down by adding locking to websocket session writes and by deferring session teardown until session destruction. (closes issue ASTERISK-23605) Review: https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan ........ Merged revisions 413123 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413124 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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10f4d0f65d |
res_stasis: Add progress indications to operations which perform media.
This change fixes operations which did not account for the fact that they may be executed on channels which have not been answered. These operations will now indicate progress when invoked. ASTERISK-23560 #close ASTERISk-23560 #comment Reported by: Jan Svoboda Review: https://reviewboard.asterisk.org/r/3495/ ........ Merged revisions 413121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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7378d3e054 |
res_pjsip_sdp_rtp: Fix issue where sending a hold SDP twice could cause an unhold.
This change fixes a bug where if an SDP with media address and sendonly was received twice the underlying call would go off hold, instead of remaining on hold. This occured because the code did not properly take into account that the SDP may contain both a valid media address and the sendonly attribute. The code now examines the sendonly attribute and media address first, so if the SDP is received again no change will occur. ASTERISK-23558 #comment Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3472/ ........ Merged revisions 413119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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56ca10c7f1 |
chan_pjsip: Add support for picking up calls in the configured pickup group.
AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ ........ Merged revisions 413117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
George Joseph
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c6ed85748c |
Add "destroy" implementation for spinlock.
The original commit for spinlock was missing "destroy" implementations. Most of them are no-ops but phtread_spin and pthread_mutex do need their locks destroyed. ........ Merged revisions 413102 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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2e39da35d0 |
chan_pjsip: Implement core ability to get Call-ID of a channel.
This changes implement the "get_pvt_uniqueid" which is used to return the technology specific unique identifier. In the case of SIP this is the Call-ID of the dialog. Review: https://reviewboard.asterisk.org/r/3480/ ........ Merged revisions 413088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Kinsey Moore
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f7caf4e249 |
Bridging: Don't lock NULL bridges
When bridge locking was added for bridge snapshot creation, some locations where bridge locking was added were not guaranteed to actually have a bridge and locking NULL AO2 objects tends to cause segfaults. This ensures that NULL bridges aren't locked. ........ Merged revisions 413073 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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7dd64ff993 |
Add DeviceStateChanged and PresenceStateChanged AMI events.
These events are controlled by two new modules, res_manager_devicestate and res_manager_presencestate. Review: https://reviewboard.asterisk.org/r/3417 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Igor Goncharovskiy
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d3433771c9 |
Introducing changes proposed to chan_unistim driver:
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11 years ago |
Olle Johansson
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7c276f9fef |
tcptls.c : Log errors as ERROR, not warning or something else.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Matthew Jordan
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bf81470083 |
res_rtp_asterisk: Add support for DTLS handshake retransmissions
On congested networks, it is possible for the DTLS handshake messages to get lost. This patch adds a timer to res_rtp_asterisk that will periodically check to see if the handshake has succeeded. If not, it will retransmit the DTLS handshake. Review: https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close Reported by: Nitesh Bansal patches: dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418) ........ Merged revisions 413008 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413009 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Kevin Harwell
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798b21a914 |
pjsip realtime: increase the size of some columns
The string lengths on certain columns created through alembic for PJSIP were too short. For instance, columns containing URIs are currently set to 40 characters, but this can be too small and result in truncated values. Added an alembic migration script that increases the size of these columns and a few others to 255. ASTERISK-23639 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3475/ ........ Merged revisions 412992 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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George Joseph
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64045f0b57 |
This patch adds support for spinlocks in Asterisk.
There are cases in Asterisk where it might be desirable to lock a short critical code section but not incur the context switch and yield penalty of a mutex or rwlock. The primary spinlock implementations execute exclusively in userspace and therefore don't incur those penalties. Spinlocks are NOT meant to be a general replacement for mutexes. They should be used only for protecting short blocks of critical code such as simple compares and assignments. Operations that may block, hold a lock, or cause the thread to give up it's timeslice should NEVER be attempted in a spinlock. The first use case for spinlocks is in astobj2 - internal_ao2_ref. Currently the manipulation of the reference counter is done with an ast_atomic_fetchadd_int which works fine. When weak reference containers are introduced however, there's an additional comparison and assignment that'll need to be done while the lock is held. A mutex would be way too expensive here, hence the spinlock. Given that lock contention in this situation would be infrequent, the overhead of the spinlock is only a few more machine instructions than the current ast_atomic_fetchadd_int call. ASTERISK-23553 #close Review: https://reviewboard.asterisk.org/r/3405/ ........ Merged revisions 412976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412977 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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e6c4b97521 |
http: Fix spurious ERROR message in responses with no content.
Backport -r411687 and fix the fix because content_length is the length of out plus the length of the file controlled by fd. When a response has an out content length of 0, fwrite would be called to write a buffer with no data in it. This resulted in the following classic error message: [Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success This patch makes it so that we only attempt to write the content of out if the out string is non-zero. ........ Merged revisions 412922 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412923 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412924 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Russell Bryant
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4b9b4790d9 |
Fix error loading res_monitor.
For some odd reason, loading app_mixmonitor was fine, but res_monitor was not. This patch fixes a set of issues related to func_periodic_hook exporting the beep functions that gets res_monitor working again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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9b71a87108 |
res_stasis: Fix crash when handling a failed blind transfer message.
This changes fixes a crash that occurs when stasis determines if it should send a message out to an application or not. The code incorrectly assumed that a bridge snapshot would always be present when in reality for failure cases it may not be. ASTERISK-23573 #close ........ Merged revisions 412882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Jonathan Rose
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86c68bc437 |
chan_sip: trust_id_outbound CHANGES message improvement
(closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski ........ Merged revisions 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412822 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412823 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Jonathan Rose
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ae21162a69 |
chan_sip: Add sendrpid trust options
In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ ........ Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Kinsey Moore
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dcb2ea657c |
HTTP: Add TCP_NODELAY to accepted connections
This adds the TCP_NODELAY option to accepted connections on the HTTP server built into Asterisk. This option disables the Nagle algorithm which controls queueing of outbound data and in some cases can cause delays on receipt of response by the client due to how the Nagle algorithm interacts with TCP delayed ACK. This option is already set on all non-HTTP AMI connections and this change would cover standard HTTP requests, manager HTTP connections, and ARI HTTP requests and websockets in Asterisk 12+ along with any future use of the HTTP server. Review: https://reviewboard.asterisk.org/r/3466/ ........ Merged revisions 412745 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412748 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |