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10.12.0
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10.3.0
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10.3.1
10.4.0
10.4.0-digiumphones-rc1
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${ noResults }
26990 Commits (6fcd361540a2b38724d5f49bd193d97f5522616c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
118d628e08 |
Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs
This patch adds a rule for installing the Super Awesome Company based 'Basic PBX' configuration files. As part of adding this rule, a bit of the content that makes up installing the configuration files under the 'samples' target was refactored into a make subroutine for usage by additional later config make targets. Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404 |
10 years ago |
|
9a021a42ad |
res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.
When compiled with assertions enabled one will occur when destroying the subscription tree when UAS dialog creation fails. This is because the code assumes that a dialog will always exist on a subscription tree when in reality during this specific scenario it won't. This change makes it so a dialog is not removed from the subscription tree if it is not present. ASTERISK-25505 #close Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee |
10 years ago |
|
b640858e9b |
Merge "chan_sip: Do not send all codecs on INVITE." into 13
|
10 years ago |
|
ba566b0f0f |
Merge "res_pjsip: Add "like" processing to pjsip list and show commands" into 13
|
10 years ago |
|
1256aedf66 |
chan_sip: Do not send all codecs on INVITE.
Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 |
10 years ago |
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31f13a1e93 |
Merge "build: GCC 5.1.x catches some new const, array bounds and missing paren issues" into 13
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10 years ago |
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7d1a2b0839 |
Merge "format: Update the maximum packetization time for iLBC 30." into 13
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10 years ago |
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6875944771 |
Merge "res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog." into 13
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10 years ago |
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131c3c181d |
Merge "res_pjsip_pubsub: Ensure dialog lock balance." into 13
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10 years ago |
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c7651da5d1 |
Merge "res_pjsip_pubsub: Prevent crashes on final NOTIFY." into 13
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10 years ago |
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90f0ed5475 |
Merge "res_pjsip_pubsub: Remove serializer when sending final NOTIFY." into 13
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10 years ago |
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d9be087da4 |
Merge "res_pjsip_pubsub: Fix crash on destruction of empty subscription tree." into 13
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10 years ago |
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4d576fe6d9 |
Merge "res_pjsip_pubsub: Solidify lifetime and ownership of objects." into 13
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10 years ago |
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5f593e7c38 |
build: GCC 5.1.x catches some new const, array bounds and missing paren issues
Fixed 1 issue in each of the affected files. ASTERISK-25494 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77 |
10 years ago |
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162acd45f7 |
res_pjsip: Add "like" processing to pjsip list and show commands
Add the ability to filter output from pjsip list and show commands using the "like" predicate like chan_sip. For endpoints, aors, auths, registrations, identifyies and transports, the modification was a simple change of an ast_sorcery_retrieve_by_fields call to ast_sorcery_retrieve_by_regex. For channels and contacts a little more work had to be done because neither of those objects are true sorcery objects. That was just removing the non-matching object from the final container. Of course, a little extra plumbing in the common pjsip_cli code was needed to parse the "like" and pass the regex to the get_container callbacks. Some of the get_container code in res_pjsip_endpoint_identifier was also refactored for simplicity. ASTERISK-25477 #close Reported by: Bryant Zimmerman Tested by: George Joseph Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1 |
10 years ago |
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d818e6edce |
Merge "res_pjsip_outbound_registration: registration stops due to fatal 4xx response" into 13
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10 years ago |
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c58091737d |
res_pjsip_outbound_registration: registration stops due to fatal 4xx response
During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2 |
10 years ago |
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8315479929 |
Merge "chan_sip: Fix autoframing=yes." into 13
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10 years ago |
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84f1068bab |
Merge topic 'fix_oom_crash' into 13
* changes: strings.c: Fix __ast_str_helper() to always return a terminated string. Add missing failure checks to ast_str_set_va() callers. |
10 years ago |
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53ed5a0675 |
Merge "res_pjsip: Move URI validation to use time." into 13
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10 years ago |
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ebe69dee0d |
format_cap: Detect vector allocation failures.
A crash was seen on a system that ran out of memory due to Asterisk not checking for vector allocation failures in format_cap.c. With this change, if either of the AST_VECTOR_INIT calls fail, we will return a value indicating failure. Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 |
10 years ago |
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3b19efefef |
res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.
A certain situation can result in our attempting to send a NOTIFY on a destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but that subscriber has dropped off the network. We end up retransmitting that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY transaction. When the pjsip evsub code is told that the transaction has been terminated, it responds in kind by alerting us that the subscription has been terminated, destroying the subscription, and then removing its reference to the dialog, thus destroying the dialog. The problem is that when we get told that the subscription is being terminated, we detect that we have not sent a terminating NOTIFY request, so we queue up such a NOTIFY to be sent out. By the time that queued NOTIFY gets sent, the dialog has been destroyed, so attempting to send that NOTIFY can result in a crash. The fix being introduced here is actually a reintroduction of something the pubsub code used to employ. We hold a reference to the dialog and wait to decrement our reference to the dialog until our subscription tree object is destroyed. This way, we can send messages on the dialog even if the PJSIP evsub code wants to terminate earlier than we would like. In doing this, some NULL checks for subscription tree dialogs have been removed since NULL dialogs are no longer actually possible. Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5 |
10 years ago |
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0a346f095f |
res_pjsip_pubsub: Ensure dialog lock balance.
When sending a NOTIFY, we lock the dialog and then unlock the dialog when finished. A recent change made it so that the subscription tree's dialog pointer will be set NULL when sending the final NOTIFY request out. This means that when we attempt to unlock the dialog, we pass a NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog remains locked after we think we have unlocked it. When a response to the NOTIFY arrives, the monitor thread attempts to lock the dialog, but it cannot because we never released the dialog lock. This results in Asterisk being unable to process incoming SIP traffic any longer. The fix in this patch is to use a local pointer to save off the pointer value of the subscription tree's dialog when locking and unlocking the dialog. This way, if the subscription tree's dialog pointer is NULLed out, the local pointer will still have point to the proper place and the dialog lock will be unlocked as we expect. Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a |
10 years ago |
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ad39508095 |
res_pjsip_pubsub: Prevent crashes on final NOTIFY.
The SIP dialog is removed from the subscription tree when the final NOTIFY is sent. However, after the final NOTIFY is sent, the persistence update function still attempts to access the cseq from the dialog, resulting in a crash. This fix removes the subscription persistence at the same time that the dialog is removed from the subscription tree. This way, there is no attempt to update persistence when the subscription is being destroyed. Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb |
10 years ago |
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067f408760 |
res_pjsip_pubsub: Remove serializer when sending final NOTIFY.
There have been crashes seen where a taskprocessor's listener is NULL unexpectedly. Looking at backtraces, the problem was specifically seen in PJSIP serializers. Subscriptions make the mistake of removing a serializer from a dialog during subscription tree destruction. Since subscription trees are reference-counted, guaranteeing the circumstances behind the destruction are not possible. This makes it so that the dialog serializer can be removed while not holding the dialog lock. This makes it possible for the distributor to get a pointer to the dialog serializer and have that serializer get freed out from under it. The fix for this is to remove the serializer from a subscription dialog when sending the final NOTIFY. This guarantees that the serializer is removed with the dialog lock held. By doing this, we guarantee that if the distributor gains access to the dialog's serializer, it will not be possible for the serializer to get freed by another thread. Change-Id: I21f5dac33529f65cec45679bdace60670800ff66 |
10 years ago |
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1bcc592765 |
res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.
If an old persistent subscription is recreated but then immediately destroyed because it is out of date, the subscription tree will have no leaf subscriptions on it. This was resulting in a crash when attempting to destroy the subscription tree. A simple NULL check fixes this problem. Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac |
10 years ago |
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b3cc2bd7df |
res_pjsip_pubsub: Solidify lifetime and ownership of objects.
There have been crashes and general instability seen in the pubsub code, so this patch introduces three changes to increase the stability. First, the ownership model for subscriptions has been modified. Due to RLS, subscriptions are stored in memory as a tree structure. Prior to my patch, the PJSIP subscription was the owner of the subscription tree. When the PJSIP subscription told us that it was terminating, we started destroying the subscription tree along with all of the individual leaf subscriptions that belong to the tree. The problem with this model is that the two actors in play here, the PJSIP subscription and the individual leaf subscriptions, need to have joint ownership of the subscription tree. So now, the PJSIP subscription and the individual leaf subscriptions each have a reference to the subscription tree. This way, we will not actually free memory until no players are left that care. The PJSIP subscription is a bigger stakeholder, in that if the PJSIP subscription's reference to the subscription tree is removed, the subscription tree instructs the leaf subscriptions to shut down and drop their references to the subscription tree when possible. The individual leaf subscriptions, upon being told to shut down, can drop their stasis subscriptions or whatever they use to learn of new state, and then drop their reference to the subscription tree once they are ready to die. Second, the lifetime of a PJSIP subscription's reference to our subscription tree has been altered. As I learned from doing a deep dive, the PJSIP evsub code can tell Asterisk multiple times that the subscription has been terminated, and not all of these times are especially helpful. I have altered the message flow that we use for SIP subscriptions such that we will always drop the PJSIP subscription's reference to the subscription tree when we send the NOTIFY that terminates a SIP subscription. This also means that we will now queue NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so that we can have predictable state changes from the PJSIP evsub code. Third, the synchronization of operations has been improved. PJSIP can call into our code from a serializer thread (e.g. upon receiving an incoming request) or from the monitor thread (e.g. when a subscription times out). Because of this, there is the possibility of competing threads stepping on each other. PJSIP attempts to do some synchronization on its own by always keeping the dialog lock held when it calls into us. However, since we end up pushing tasks into the serializer, the result was that serialized operations were not grabbing the dialog lock and could, as a result, step on something that was being attempted by a different thread. Now we ensure that serialized operations grab the dialog lock, then check for extenuating circumstances, then proceed with their operation if they can. Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5 |
10 years ago |
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c8c65dfa41 |
strings.c: Fix __ast_str_helper() to always return a terminated string.
Users of functions which call __ast_str_helper() such as the ones listed below are likely to not check the return value for failure so ensuring that the string is always nil terminated is a good safety measure. ast_str_set_va() ast_str_append_va() ast_str_set() ast_str_append() Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 |
10 years ago |
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b271d4a28a |
Add missing failure checks to ast_str_set_va() callers.
Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f |
10 years ago |
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04030b0dd0 |
Merge "funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function" into 13
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10 years ago |
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f2725c8b77 |
res_pjsip: Move URI validation to use time.
In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a |
10 years ago |
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84ff075d41 |
format: Update the maximum packetization time for iLBC 30.
In September 2006, the maximum packetization time (ptime) were set to such a low value, packetization was disabled for many codecs actually. This was fixed for many codecs but not for iLBC 30. This enables packetization for iLBC which can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. ASTERISK-7803 Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 |
10 years ago |
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869ef2a8ee |
chan_sip: Fix autoframing=yes.
With Asterisk 13, the structures ast_format and ast_codec changed. Because of that, the paketization timing (framing) of the RTP channel moved away from the formats/codecs. In the course of that change, the ptime of the callee was not honored anymore, when the optional autoframing was enabled. ASTERISK-25484 #close Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4 |
10 years ago |
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9fd2adc204 |
rest-api-templates: Wikify error code response reasons
Error response code descriptions may contain wiki markup that need to be escaped. Without this patch, Confluence will reject the document being sent and the responsible script will raise an exception. Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5 |
10 years ago |
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72cbb6df55 |
funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function
When
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10 years ago |
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7c560a9a31 |
Merge "contrib/scripts/autosupport: Update for Asterisk 13" into 13
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10 years ago |
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9fc9777fa3 |
contrib/scripts/autosupport: Update for Asterisk 13
This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 |
10 years ago |
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25ec63ab0c |
Merge topic 'ASTERISK-25461' into 13
* changes: config.c: Fix off-nominal memory leak. config.c: Fix potential memory corruption after [section](+). config.c: Fix #include after [section](+). |
10 years ago |
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5db999d813 |
Merge "res_config_pgsql.c: Fix deadlock loading realtime configuration." into 13
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10 years ago |
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ebdfd38511 |
Merge "Build: Add menuselect options for using compiler sanitizers" into 13
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10 years ago |
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dc6ec661b3 |
res_config_pgsql.c: Fix deadlock loading realtime configuration.
On v13, loading several thousand PJSIP endpoints on Asterisk start causes a deadlock most of the time. Thanks to mdu113 for discovering that there was a call to pgsql_exec() not protected by the pgsql_lock reentrancy lock. {quote} I believe a code path exists that attempts to use pgsql connection without locking pgsql_lock. I believe what happens during that deadlock that I see is two concurrent threads are both attempting to send query to pgsql, one of the thread is using a code path without locking pgsql_lock. If they managed to send queries at the same time, it seems postgres ignores one of the queries and replies only to the one of them. If it happens so that the thread holding the lock didn't receive the reply it will wait for it (and hold the lock) forever (or at least for very long time), thus completely blocking all access to db. {quote} * Added missing reentrancy locking around pgsql_exec() in find_table(). * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion between the psql_tables list lock and the pgsql_lock. ASTERISK-25455 #close Reported by: mdu113 Patches: res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113 Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2 |
10 years ago |
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f8707ae9a5 |
channels/chan_sip: Set cause code to 44 on RTP timeout
To quote Olle: "When issuing a hangup due to RTP timeouts the cause code is not set. I have selected 44 based on Cisco's implementation..." ASTERISK-25135 #close Reported by: Olle Johansson patches: rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc |
10 years ago |
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486b172b50 |
Build: Add menuselect options for using compiler sanitizers
This patch adds menuselect options for building Asterisk with various sanitizers provided by gcc and clang. When one of *SANITIZER flags is set in menuselect, the appropriate option is added to CFLAGS ad LDFLAGS for the build. Information on sanitizers in the project wiki: https://github.com/google/sanitizers/wiki GCC Manual: https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html Clang Compiler User's Manual: http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation ASTERISK-24718 #close Reported by: Badalian Vyacheslav Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0 |
10 years ago |
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e14023ca35 |
config.c: Fix off-nominal memory leak.
Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 |
10 years ago |
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a99e821520 |
config.c: Fix potential memory corruption after [section](+).
The memory corruption could happen if the [section](+) is the last section in the file with trailing comments. In this case process_text_line() has left *last_cat is set to newcat and newcat is destroyed. Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 |
10 years ago |
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8d31d2526b |
config.c: Fix #include after [section](+).
An #include right after a [section](+) would associate any variable assignments before a new section in the #include with the wrong section. * Fix section association by setting the current section to the appended section. * Fix '+' and '!' section flag interaction corner case depending upon which flag came first. If the '!' came first then it would be ignored. If the '!' came after then it would affect the appended section. The '!' will now no longer be ignored. ASTERISK-25461 #close Reported by: Sean Pimental Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 |
10 years ago |
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1c799b1f1e |
Merge "chan_pjsip: Fix crash on reINVITE before initial INVITE completes." into 13
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10 years ago |
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3329c714f7 |
res_pjsip: Fix deadlock when sending out-of-dialog requests.
The struct send_request_wrapper has a pjsip lock associated with it that is created non-recursive. There is a code path for the struct send_request_wrapper lock that will attempt to lock it recursively. The reporter's deadlock showed that the thread calling endpt_send_request() deadlocked itself right after the wrapper object got created. Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited MWI NOTIFY messages can hit this deadlock. * Replaced the struct send_request_wrapper pjsip lock with the mutex lock that can come with an ao2 object since all of Asterisk's mutexes are recursive. Benefits include removal of code maintaining the pjsip non-recursive lock since ao2 objects already know how to maintain their own lock and the lock will show up in the CLI "core show locks" output. ASTERISK-25435 #close Reported by: Dmitriy Serov Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d |
10 years ago |
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a1435aa3fa |
res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending
In ast_rtp_read, the value of the variable 'mark' which we try to assign to a frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate it to 0 or 1. ASTERISK-25451 #close Change-Id: I53bdf5c026041730184a6a809009c028549ce626 |
10 years ago |
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3357678b94 |
func_presencestate: Return "not_set" when no data is set in AstDB
Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not exist, i.e. when a new CustomPresence is added in the dialplan. ASTERISK-25400 #close Reported by: Andrew Nagy Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a |
10 years ago |