asterisk

Commit Graph

Author	SHA1	Message	Date
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
zuul	0cc14597b2	Merge "rtp_engine: Allow more than 32 dynamic payload types."	9 years ago
Kevin Harwell	bf01ff53f8	Revert "chan_sip: Fix lastrtprx always updated" This reverts commit `93332cb1d0`. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d	9 years ago
Joshua Colp	a1bbdabb8e	Merge "chan_sip: add missing account code"	9 years ago
zuul	180e550520	Merge "app_dial: Fix incorrect device state when channel is picked up."	9 years ago
zuul	673964d330	Merge "chan_dahdi: remove by_name support"	9 years ago
Sebastian Gutierrez	0904c1f4cc	chan_sip: add missing account code Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482	9 years ago
Joshua Colp	4de5454ef1	app_dial: Fix incorrect device state when channel is picked up. Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f	9 years ago
Alexander Traud	9ac53877f6	rtp_engine: Allow more than 32 dynamic payload types. Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, this change utilizes payload types in the range between 35 and 63 giving room for another 29 payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	9 years ago
Joshua Colp	a380bba1e3	Merge "res_pjsip_sdp_rtp: Limit number of formats to defined maximum."	9 years ago
Joshua Colp	dd84684272	Merge "define PATH_MAX for HURD"	9 years ago
Joshua Colp	12b81851f4	Merge "netsock.c: fix includes for HURD"	9 years ago
zuul	bff5c27426	Merge "bundled pjproject: Fix DNS write to freed memory."	9 years ago
zuul	de2c964e86	Merge "pjproject_bundled: Fix compile of pjsua so it handles audio"	9 years ago
Joshua Colp	f0236b2d87	Merge "codecs.conf.sample: Add sample and option descriptions for codec_opus"	9 years ago
Joshua Colp	d94e2250c2	Merge "res_pjsip_outbound_publish: Fix crash when publishing device state."	9 years ago
Joshua Colp	a2dde4a543	Merge "res/stasis: Add CLI commands for displaying/debugging ARI apps"	9 years ago
Richard Mudgett	f29b8d62bb	bundled pjproject: Fix DNS write to freed memory. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patch below fixes a write to freed memory under cartain DNS lookup conditions. 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch ASTERISK-26516 Reported by: Richard Mudgett Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5	9 years ago
zuul	4361a24b63	Merge "chan_sip: Incorrect display option Outbound reg. retry 403"	9 years ago
Joshua Colp	6233e146c6	res_pjsip_sdp_rtp: Limit number of formats to defined maximum. The res_pjsip_sdp_rtp module did not restrict the number of formats added to a media stream in the SDP to the defined limit. If allow=all was used with additional loaded codecs this could result in the next media stream being overwritten some. This change restricts the module to limit it to the defined maximum and also increases the maximum in our bundled pjproject. ASTERISK-26541 #close Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8	9 years ago
Kevin Harwell	8060cd1ec1	codecs.conf.sample: Add sample and option descriptions for codec_opus codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b	9 years ago
Matt Jordan	c30d677333	res/stasis: Add CLI commands for displaying/debugging ARI apps This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5	9 years ago
Grachev Sergey	2526dff94d	chan_sip: Incorrect display option Outbound reg. retry 403 If in sip.conf (general section) set option register_retry_403=no, the command "sip show settings" return value: Outbound reg. retry 403:0 If in sip.conf (general section) set option register_retry_403=yes, the command "sip show settings" return value: Outbound reg. retry 403:-1 * In static char "sip show settings" for "Outbound.reg. retry 403" option use AST_CLI_YESNO ASTERISK-26476 #close Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9	9 years ago
Tzafrir Cohen	ed08811e64	netsock.c: fix includes for HURD ASTERISK-25070 Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814	9 years ago
Tzafrir Cohen	69fed26deb	define PATH_MAX for HURD PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD define it to a constant. It is indeed not safe to assume there won't be longer paths and Asterisk generally does err safely on such cases. So even for HURD we'll just pretend PATH_MAX is 4096. ASTERISK-25070 #close Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3	9 years ago
George Joseph	f27f837a9f	pjproject_bundled: Fix compile of pjsua so it handles audio In order for pjsua and its python binding to actually negotiate audio for the testsuite tests, it needs g711 and resample. The pj* libraries themselves do not. Unfortunately, pjproject relies on a brand new libresample that most distros don't ship so we need to use the libresample already bundled with pjproject. Only the pjsua executable and the _pjsua.so python library are linked with it so it shouldn't interfere with asterisk itself. Also it was pointed out that apply_patches couldn't handle multiple patches that depended on each other during the dry-run, so the dry-run was removed. Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098	9 years ago
Etienne Lessard	1648ca06c3	manager: Add documentation for NewConnectedLine event. The NewConnectedLine event has been added by commit `fe7671f`, but the documentation was missing. ASTERISK-26537 #close Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6	9 years ago
zuul	3ad4719917	Merge "bundled pjproject: Crashes while resolving DNS names."	9 years ago
zuul	d9cf38dc51	Merge "astobj2: Declare private variable data_size for AO2_DEBUG only."	9 years ago
Corey Farrell	273debd261	vector: Prevent NULL argument to memcpy. Headers declare that memcpy does not accept NULL argument for the first two parameters. Add a conditional block to prevent memcpy and ast_free from running on vectors with NULL element array. ASTERISK-26526 #close Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71	9 years ago
Corey Farrell	ad60927a40	astobj2: Declare private variable data_size for AO2_DEBUG only. Every ao2 object contains storage for a private variable data_size, though the value is never read if AO2_DEBUG is disabled. This change makes the variable conditional, reducing memory usage. ASTERISK-26524 #close Change-Id: If859929e507676ebc58b0f84247a4231e11da07f	9 years ago
Richard Mudgett	6feee22e09	bundled pjproject: Crashes while resolving DNS names. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patches below fix the DNS lookup race condition crash caused by attempting to send the same message twice for the single DNS lookup. 0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch 0006-r5473-svn-backport-Fix-pending-query.patch The patch below removes a cached DNS response from the hash table when another thread is referencing the old entry. The table still contained the entry when it was destroyed which can result in inexplicable crashes. 0006-r5475-svn-backport-Remove-DNS-cache-entry.patch ASTERISK-26344 #close Reported by: Ian Gilmour ASTERISK-26387 #close Reported by: Harley Peters Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4	9 years ago
George Joseph	12bdde6a6c	pjproject_bundled: Fix issue where "/version.mak" wasn't found main/Makefile includes third-party/pjproject/build.mak but doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak" evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile before the include. Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604	9 years ago
zuul	2995b31d18	Merge "Fix shutdown crash caused by modules being left open."	9 years ago
mkrokosz	9d8b9b6ca5	res_pjsip_outbound_publish: Fix crash when publishing device state. While publishing device state between multiple instances of Asterisk, a crash will sporadically occur under high CPS which looks to be a race condition operating on the publisher queue. ASTERISK-26506 Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6	9 years ago
Corey Farrell	d6ad867897	Fix shutdown crash caused by modules being left open. It is only safe to run ast_register_cleanup callbacks when all modules have been unloaded. Previously these callbacks were run during graceful shutdown, making it possible to crash during shutdown. ASTERISK-26513 #close Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21	9 years ago
Rusty Newton	badd38f031	SAC documentation: don't specify transports for endpoints and registrations Removing explicit transport definition for endpoints and registrations. It isn't necessary and isn't generally advised. ASTERISK-26514 #close Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb	9 years ago
zuul	0ec5abe592	Merge "Remove ASTERISK_REGISTER_FILE."	9 years ago
zuul	0fc942e7cd	Merge "res_pjsip_sdp_rtp: Fix address family of explicit media_address."	9 years ago
Joshua Colp	24d0907849	Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads."	9 years ago
zuul	6f73da9c11	Merge "app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS."	9 years ago
Tzafrir Cohen	0646b48ece	chan_dahdi: remove by_name support Support for referring to DAHDI channels by logical names was added in (FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support of refering to channels by name. While technically usable, it has never been properly supported in dahdi-tools, as using it would require many changes at the Asterisk level. Instead logical mapping was added at the kernel level. Thus it seems that refering to DAHDI channels by name is not really used by anyone, and therefore should probably be removed. Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485	9 years ago
zuul	90d1b3262b	Merge "res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls."	9 years ago
George Joseph	4f45d62653	pjproject_bundled: Remove usage of tar's --strip-components option Older versions of tar don't support the --strip-components option so instead of doing 'tar --strip-components=1 -C source', we now just untar to the tarball's root directory (pjproject-<version>) and rename that directory to 'source'. Also fixed an issue where the pjproject source directory is a hard coded absolute pathname. ASTERISK-26510 #close ASTERISK-22480 #close Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Joshua Colp	6993f3c9c3	res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. The res_pjsip_caller_id module wrongly assumed that a saved From header would always exist on sessions. This is true until an inbound call is received and a session timer causes an UPDATE to be sent. In this case there will be no saved From header and a crash will occur. This change makes it fall back to the From header of the outgoing request if no saved From header is present. ASTERISK-26307 #close Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa	9 years ago
zuul	211a1003e4	Merge "test_astobj2_thrash: Fix multithreaded issues"	9 years ago
zuul	761b32633c	Merge "cdr_radius,cel_radius: Fix old memleak in unload"	9 years ago
Joshua Colp	95062fe220	app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice mailbox was not cleared, causing it to try to free garbage. ASTERISK-26503 #close Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3	9 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago

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