mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
25433 Commits (4e292ea3af5484b10b6df9dddf6b718f3efc3d20)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
c704795dcb |
res_parking: Minor tweaks.
* Use ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables. * Use ast_copy_string() instead of inlining it. * Remove an already done TODO comment. * Some whitespace tweaks. ........ Merged revisions 411638 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
|
7542361f4a |
stasis_channels.c: Eliminate another overuse of RAII_VAR().
........ Merged revisions 411636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Joshua Colp
|
c7b8633c26 |
app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.
This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ ........ Merged revisions 411584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411585 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411586 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411587 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Matthew Jordan
|
ef0c9fe4d8 |
res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP
This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a *lot* of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Alexandr Anikin
|
a35ce3924b |
process stack command even if gatekeeper client isn't register
don't destroy gatekeeper client if it is not started don't destroy gatekeeper client in some sort of gatekeeper errors signal rtp create condition when call cleared before rtp structure created (closes issue ASTERISK-23460) Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry Melekhov ........ Merged revisions 411531 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411532 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
597f25db69 |
Update API versions and UPGRADE/CHANGES for 12.2.0
This patch does the following: * It updates the AMI version to 2.2.0 to indicate backwards compatible changes have been made since the last release * It updates the ARI version to 1.2.0 to indicate backwards compatible changes have been made since the last release * It updates the UPGRADE/CHANGES files with changes that were not mentioned ........ Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
a438a0e65f |
res_config_odbc: Fix for nullable integer columns and keyfield existence check in update_odbc.
This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. Also, the check for existence of a mandatory column checked for the first column in the list instead of the key field lookup column. This patch fixes that issue as well. Finally, the compatibility option allow_empty_string_in_nontext, which was added to previous revisions to allow for some database backends with certain schemas to function, has been removed. Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459 #close ASTERISK-23351 #close (closes issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
24452b852d |
Blocked revisions 411512
........ res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values. This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. This patch also adds a compatibility setting in res_odbc.conf, allow_empty_string_in_nontext. It is enabled by default. It should be disabled for database backends (such as PostgreSQL) that require NULL instead of an empty string for Integer columns. Review: https://reviewboard.asterisk.org/r/3375 (issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) ........ Merged revisions 411399 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411408 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Scott Griepentrog
|
0d057e6791 |
http: response body often missing after specific request
This patch works around a problem with the HTTP body being dropped from the response to a specific client and under specific circumstances: a) Client request comes from node.js user agent "Shred" via use of swagger-client library. b) Asterisk and Client are *not* on the same host or TCP/IP stack In testing this problem, it has been determined that the write of the HTTP body is lost, even if the data is written using low level write function. The only solution found is to instruct the TCP stack with the shutdown function to flush the last write and finish the transmission. See review for more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) Reported by: Sam Galarneau Review: https://reviewboard.asterisk.org/r/3402/ ........ Merged revisions 411462 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411463 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
2cfa9b4b77 |
UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems.
........ Merged revisions 411457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411458 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411459 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
2dae3d6ea3 |
contrib/realtime: Remove empty SQL script files
Since the relatime scripts are now managed by Alembic, the previous realtime scripts were previously removed. However, the removal process messed up, as the files were still in the repository. The contents were just empty. This removes the files from the tree. ........ Merged revisions 411442 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411443 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Matthew Jordan
|
562182f5b0 |
chan_sip: Add MESSAGE request to allowed methods
The allowed methods advertised by chan_sip did not previously note the MESSAGE request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we should advertise that we support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 #comment Reported by: Martin Kontsek ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged revisions 411372 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411373 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411374 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Corey Farrell
|
fbe0dfaf44 |
Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Corey Farrell
|
44409401ec |
main/formats: Fix crash in ast_format_cmp during non-clean shutdown.
* Update asterisk.h to reflect availability of ast_register_cleanup in 11.9. * Use ast_register_cleanup for format_attr_shutdown. (closes issue ASTERISK-23103) Reported by: JoshE ........ Merged revisions 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Mark Michelson
|
a8629e53c1 |
Give sorcery instances a reference to their wizards.
On graceful shutdown, sorcery wizards are all killed off, but it is possible for sorcery instances to still have dangling pointers after this, possibly causing a crash. Giving the sorcery instances a reference to their wizards ensures that the wizard reference will remain valid for the lifetime of the sorcery instance. Review: https://reviewboard.asterisk.org/r/3401 ........ Merged revisions 411295 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Joshua Colp
|
7dddd694cb |
say: Fix a bug where SayNumber in Polish tries to play incorrect sound.
This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ Merged revisions 411243 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Jonathan Rose
|
fa3a2f8eca |
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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4b18b3bb4d |
Fix 'alembic branches' merge conflict as described by the web page.
........ Merged revisions 411191 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Sean Bright
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c32fe8b8e5 |
ARI: Don't complain about missing ARI users when we aren't enabled
Currently, if ARI is not enabled it will still complain that there are no configured users. This patch checks to see if ARI is enabled before logging and error or iterating the container to validate the users. Review: https://reviewboard.asterisk.org/r/3391/ ........ Merged revisions 411173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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2bf37a417d |
Add a "message_context" option for PJSIP endpoints.
........ Merged revisions 411157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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c1c8300e27 |
res_pjsip: Fix contact authenticate_qualify endpoint lookup when qualifing a contact.
* Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of find_endpoints() with find_an_endpoint() since only the first found endpoint is ever needed. * Fixed qualify_contact_cb() to update the contact with the aor authenticate_qualify setting. Otherwise, permanent contacts in the aor type sections would have a config line order dependancy. * Fixed off nominal path contact ref leak in qualify_contact(). The comment saying the unref is not needed was wrong. * Fixed off nominal path use of the endpoint parameter if it is NULL in send_out_of_dialog_request(). * Added missing off nominal path unref of pjsip tdata in send_out_of_dialog_request(). * Fixed off nominal path failing to call the callback in send_request_cb() when the request is challenged for authentication. * Eliminated silly RAII_VAR() use in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen to better reflect reality. (closes issue ASTERISK-23254) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/ ........ Merged revisions 411141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Kinsey Moore
|
a4890eddfb |
chan_sip: Fix incorrect use of timers
If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) ........ Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411089 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411091 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Jonathan Rose
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eb0a982f8c |
ARI: Resolve a subscription leak against implicit bridge subscriptions
When a channel in a stasis application is joined to a bridge, a subscription for that bridge is created implicitly for the stasis application serving the channel. Prior to this patch, subsequent removals of the channel from the bridge would leave the subscription open. Review: https://reviewboard.asterisk.org/r/3380/ ........ Merged revisions 411086 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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c8ebf3e3c7 |
Revert -r411073. It didn't help and blew up the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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89e12de79d |
locking: Add temporary sanity checks.
Add some temporary sanity checks to hunt for locking problems with the masquerade supertest. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411073 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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6d81951f0d |
chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted.
(closes issue ASTERISK-20841) Reported by: Kelly Goedert ........ Merged revisions 411021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411022 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411023 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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236d17362d |
res_pjsip_registrar.c: Miscellaneous cleanup in rx_task().
* Fix variable shadowing of 'updated' by renaming it to 'contact_update'. * Checked 'contact_update' for ast_sorcery_copy() failure. * Removed silly use of RAII_VAR() for 'contact_update'. ........ Merged revisions 410995 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Sean Bright
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b44d324891 |
Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug. This time without breaking the build. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Sean Bright
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14942ecb17 |
Revert r410981. aelparse blew up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Sean Bright
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df2d959d7d |
Remove a LOG_NOTICE from ast_config_engine_register.
There is enough indication from the CLI that we are loading a realtime engine as it is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Sean Bright
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922d0b7565 |
Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Jonathan Rose
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3c16865fc2 |
app_confbridge: Fix bug - users with startmuted set don't start muted
(closes issue ASTERISK-23461) Reported by: Chico Manobela Review: https://reviewboard.asterisk.org/r/3373/ ........ Merged revisions 410965 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410966 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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1ba13718fc |
assigned-uniqueids: Miscellaneous cleanup and fixes.
* Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2 uniqueid on a stack variable instead of mallocing it. * Made send error response to ARI and AMI requests instead of just logging excessive uniqueid length and allowing truncation. action_originate() and ari_channels_handle_originate_with_id(). * Fixed minor truncating uniqueid hole when generating the ;2 uniqueid string length. Created public and internal lengths of uniqueid. The internal length can handle a max public uniqueid plus an appended ;2. * free() and ast_free() are NULL tolerant so they don't need a NULL test before calling. * Made use better struct initialization format instead of the position dependent initialization format. Also anything not explicitly initialized in the struct is initialized to zero by the compiler. * Made ast_channel_internal_set_fake_ids() use the safer ast_copy_string() instead of strncpy(). Review: https://reviewboard.asterisk.org/r/3371/ ........ Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
|
57239bfe37 |
PJSIP: Allow for identify sections to be specified in sorcery.conf.
"identify" is a special type of configuration object in PJSIP because unlike the other objects, it is not provided by the base res_pjsip module. Instead, it is provided by the res_pjsip_endpoint_identifier_ip module. If using the default sorcery wizard (config,criteria=type=identify) then things work because the module that applies the default wizard is the correct module. However, if attempting to use sorcery.conf to apply an alternate wizard, it was not possible. If you attempted to specify the identify object type in the res_pjsip section, then the object could not be registered since the object was undocumented for the res_pjsip module. There was no alternate configuration section defined for it, so you were out of luck if you wanted to override the default wizard. With this change, the identify section will properly have a sorcery.conf-based wizard applied when the identify definition is within the res_pjsip_endpoint_identifier_ip section. ........ Merged revisions 410933 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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326153d949 |
res_stasis: Fix a bug where the default bridge type was not set.
........ Merged revisions 410918 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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1cf74b8776 |
res_stasis: Extend bridge type to be a comma separated list of bridge attributes.
This change turns the bridge type field into a comma separated list of attributes. These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting the various attributes a user can control the type of bridge created with the behavior they need for their application. (closes issue ASTERISK-23437) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........ Merged revisions 410904 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Matthew Jordan
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e33e003f78 |
res_ari: Fix documentation schema error
........ Merged revisions 410890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Rusty Newton
|
35fb3a564b |
res_ari: Add notes about Asterisk HTTP server to the "enabled" config option for the res_ari general section
Added note and see-also reminding user to enable the HTTP server. (closes issue ASTERISK-22499) Reported by: Rusty Newton ........ Merged revisions 410876 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Scott Griepentrog
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eecb74a9a7 |
ARI: allow json content type with zero length body
When a request was received with a Content-type of json, the body was sent for json parsing - even if it was zero length. This resulted in ARI requests failing that were valid, such as a channel DELETE with no parameters. The code has now been changed to skip json parsing with zero content length. (closes issue SWP-6748) Reported by: Samuel Galarneau Review: https://reviewboard.asterisk.org/r/3360/ ........ Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Matthew Jordan
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77db94a25a |
cdr: Add asserts for when we don't know about a CDR for a channel
In the CDR core, every channel should either be filtered out (due to being an 'internal' channel used as an implementation detail, such as playing media back into a bridge) or it should get a CDR. Even if that CDR ends up being discarded, we still give the channel a CDR in case we end up needing it. If we hit a situation where a channel does not have a CDR, we should blow up in -dev-mode. Asserts are appropriate for that. This patch adds those asserts, as they would have quickly caught the error fixed by r410814. ........ Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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216b04e6f4 |
res_pjsip: Fix memory leak of nameservers in off-nominal resolver creation failure.
Thanks Walter Doekes! ........ Merged revisions 410844 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Sean Bright
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8357027080 |
res_fax_spandsp: Use g711_free() when available.
Per Johann Steinwendtner on the asterisk-dev mailing list: http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a noop. I opted not to remove the call to g711_release() since it is harmless and to call g711_free() if we have a sufficiently recent version of spandsp. (issue ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged revisions 410829 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410830 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Richard Mudgett
|
614b6abc38 |
stasis_cache: Use the right variable in the cache entry ao2 cmp function.
........ Merged revisions 410813 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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cc40bf5317 |
res_pjsip: Enable PJSIP DNS client support.
This change enables DNS client support within PJSIP. System nameservers are automatically discovered using res_init or res_ninit. If this fails then PJSIP will resort to using gethostbyname for resolution. By enabling this support we gain SRV support, failover, and weight support. (closes issue ASTERISK-23435) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3343/ ........ Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Joshua Colp
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932fb5a6e2 |
res_pjsip_multihomed: Make address replacement less aggressive.
This change makes the res_pjsip_multihomed module less aggressive when changing the address in messages. It will now only occur if the transport in use is bound to the any address OR if the system determined source address matches the bound address of the transport in use. Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged revisions 410793 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410794 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
Russ Meyerriecks
|
ed50ef4dc8 |
callerid: Logic error in checksum processing
Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) ........ This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a broken patch to be comitted to trunk so I pre-merge merged them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
Mark Michelson
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eba91d2a98 |
Revert changes to sorcery that accidentally got committed.
These changes were still up for review and have not been approved yet. I must have had the changes in my working copy when making a different change. ........ Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Mark Michelson
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d44aefeef4 |
Fix stuck channel in ARI through the introduction of synchronous bridge actions.
Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Richard Mudgett
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1900bae7b6 |
app_confbridge: Add missing destructor call to announcer channel destructor.
........ Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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Matthew Jordan
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2c5484c869 |
stasis/app.c: Add some extra debugging for subscription counts
Events are sent to a connected ARI application based on the things that ARI application cares about. These subscriptions can be set up implicitly - such as when that ARI application creates a new object - or explicitly, via the application resource's subscription operations. Debugging *why* something was being sent to an application - or why something was not being sent to an application - was a bit tricky, as there was no debug information for the subscriptions. This patch adds some debug level 3 statements that show the subscription counts for applications. (Level 3 was chosen as it matches the verbose level 3 statements elsewhere) ........ Merged revisions 410650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |