- Aug 15, 2016
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Matt Jordan authored
* Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b
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- Aug 12, 2016
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Joshua Colp authored
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zuul authored
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zuul authored
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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zuul authored
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zuul authored
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zuul authored
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Corey Farrell authored
Errors during startup result in an exit. These error branches should be calling ast_run_atexit(0) to ensure mandatory cleanup is run. ASTERISK-26267 #close Change-Id: If226f2326ae2df7add20040696132214cf2bb680
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zuul authored
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- Aug 11, 2016
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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George Joseph authored
When compact_headers was set, we were sending a zero-length header name for PAI and RPID because we always forced the short header name length to 0. We did this because we cloned the header from "From" and wanted to clear "f" from the sname. By cloning however, we bypass pjproject's automatic logic that sets sname to name if there's no compact form of the header, which there isn't for PAI and RPID. So now we force sname to be the same as name right after we set name. res_pjsip_diversion needed the same treatment for the Diversion header. ASTERISK-26241 #close Change-Id: I633ec139630cd83809aae00336cee4a10077e467
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Matt Jordan authored
When a call forward attempt is made from a Queue member, the current code will hang up the forwarding channel in an off-nominal condition prior to raising the Stasis events informing the rest of Asterisk that the call was forwarded. This will result in a slew of dreaded FRACKs, most likely leading to a crash. This patch modifies the code such that we don't hang up the forwarding channel even in an off-nominal condition until we've safely raised the Stasis messages. ASTERISK-25797 #close Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38
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zuul authored
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George Joseph authored
If debug was specified in the global configuration but left blank, the logger would treat it as a wildcard and log all hosts. If default_from_user was empty, a crash would result. The global apply handler now checks for empty strings. ASTERISK-26239 #close ASTERISK-26238 #close Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336
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Richard Mudgett authored
* Eliminated RAII_VAR() usage in ast_sip_persistent_endpoint_update_state(). * Added a missing allocation failure check to persistent_endpoint_find_or_create(). * Made persistent_endpoint_find_or_create() create the new object without a lock as it isn't needed. * Cleaned up some ao2 container allocation idioms. * Reordered res_pjsip_mwi.c load_module() and unload_module() Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8
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Richard Mudgett authored
* Eliminated most RAII_VAR() usage. * Added several missing allocation failure checks. * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without a lock as it is not needed. Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc
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George Joseph authored
Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433
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Richard Mudgett authored
* The high water check in ast_taskprocessor_alert_set_levels() would trigger immediately if the new high water level is zero and the queue was empty. * The high water check in taskprocessor_push() was off by one. Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d
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Richard Mudgett authored
The named aor lock was always being locked for writes so a rwlock adds no benefit and may be slower because rwlocks are biased toward read locking. Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
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Richard Mudgett authored
Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28
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zuul authored
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David M. Lee authored
The non-module libs libasteriskssl.dylib and libasteriskpj.dylib have long been missing the AST_NOT_MODULE compile flag. This was mostly okay, until a recent fix to improve compiler warnings when the AST_MODULE_SELF_SYM is missing broke the build on OS X/macOS/whatever they are calling it these days. Change-Id: I2cb51c890824f001280a5114f2e775f97c163516
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Kevin Harwell authored
A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8
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zuul authored
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zuul authored
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- Aug 10, 2016
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Richard Mudgett authored
A patch made to the master branch (Now the 14 branch) inadvertently made libsrtp a required dependency in order to compile Asterisk. Rather than create dummy defines to substitute for the defines supplied by libsrtp when libsrtp is not available, most of the code in sdp_srtp.c is moved into res_srtp.c. This gets more code out of Asterisk's core that isn't used when SRTP is not available. This also makes another inadvertent required dependency on libsrtp by Asterisk's core unlikely. ASTERISK-26253 #close Reported by: Ben Merrills Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
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Alexei Gradinari authored
If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
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Kevin Harwell authored
The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b
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George Joseph authored
libunbound at version 1.4.20 (which CentOS still uses) declared all of their string function parameters as as 'char *'. 1.4.21 changed them all to 'const char *'. Thankfully 1.4.21 also introduced the UNBOUND_VERSION_MAJOR define so configure now checks for that and sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks that and casts away the 'const' if it's not set. Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and Fedora24 (1.5.4). There are a few failing tests to be addressed though. ASTERISK-26283 #close Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
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Matt Jordan authored
This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b (cherry picked from commit eec60dd7)
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zuul authored
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zuul authored
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- Aug 09, 2016
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zuul authored
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Mark Michelson authored
When an RTCP packet is sent or received, res_rtp_asterisk generates a Stasis event that contains the RTCP report as well as the local and remote addresses that the report pertains to. The addresses are determined using ast_find_ourip(). For the local address, this will typically result in a lookup of the hostname of the server, and then a DNS lookup of that hostname. If you do not have the host in /etc/hosts, then this results in a full DNS lookup, which can potentially block for some time. This is especially problematic when performing RTCP reads, since those are done on the same thread responsible for reading and writing media. This patch addresses the issue by performing a lookup of the local address when RTCP is allocated. We then use this cached local address for the Stasis events when necessary. ASTERISK-26280 #close Reported by Mark Michelson Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556
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