asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	4d5e96ab53	res_pjsip_caller_id: Copy header name to short header name 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	9 years ago
zuul	dbc78c9fab	Merge "pjsip: Fix deadlock with suspend taskprocessor on masquerade" into 13	9 years ago
Alexei Gradinari	1589452fdc	pjsip: Fix deadlock with suspend taskprocessor on masquerade 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	9 years ago
Joshua Colp	c864ebab52	Merge "res_rtp_asterisk: Cache local RTCP address." into 13	9 years ago
Mark Michelson	a119bab6a6	res_rtp_asterisk: Cache local RTCP address. 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	9 years ago
zuul	5a5b949333	Merge "res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack" into 13	9 years ago
Joshua Colp	926c1c72bd	Merge "res_pjsip_outbound_publish: Use a serializer shutdown group for unload." into 13	9 years ago
Alexei Gradinari	a06a1af0eb	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	9 years ago
Joshua Colp	485fd27f7c	res_pjsip_outbound_publish: Use a serializer shutdown group for unload. This change replaces the custom unload process for the outbound publish module with the common serializer shutdown group. ASTERISK-25217 #close Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6	9 years ago
Joshua Colp	2a0f42c494	Merge "res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports." into 13	9 years ago
Joshua Colp	f1b0286aa4	Merge "rest-api: Code out of sync with the model" into 13	9 years ago
Kevin Harwell	efc4034d72	rest-api: Code out of sync with the model Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006	9 years ago
Joshua Colp	102d28c11a	sorcery: Use more compatible regex for local expressions. This changes the use of an empty regex for both res_sorcery_config and res_sorcery_memory to "." instead. This is a more compatible regular expression which also works on FreeBSD. ASTERISK-26206 #close Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388	9 years ago
Alexander Traud	b78d10a2df	res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058	9 years ago
zuul	8d6a7b89bd	Merge "res_pjsip: Whitespace and comment cleanup." into 13	9 years ago
zuul	e3fbb4e099	Merge "res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice." into 13	9 years ago
Richard Mudgett	33716106e0	res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	9 years ago
zuul	ffbaefa48f	Merge "res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook." into 13	9 years ago
Joshua Colp	0b8448a74b	Merge changes from topic 'ASTERISK-26214' into 13 * changes: res_fax: Fix FAXOPT(faxdetect) timeout option. chan_dahdi: Add faxdetect_timeout option.	9 years ago
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." into 13	9 years ago
Alexei Gradinari	5997ec7c9e	res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a	9 years ago
zuul	7dacb14c03	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only." into 13	9 years ago
zuul	290269bb23	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets." into 13	9 years ago
zuul	7ce180a754	Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts." into 13	9 years ago
Richard Mudgett	628e8c91d5	res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. The fax_detect_framehook() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. * Made only detach the framehook if we detected a fax and not on other possible frames. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d	9 years ago
Richard Mudgett	676aeede36	res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976	9 years ago
Richard Mudgett	851b1c3a17	res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	9 years ago
Corey Farrell	c8e41d14a1	Unit tests: Use AST_TEST_DEFINE in conditional code only. If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686	9 years ago
Alexander Traud	e404f51b42	res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. With this change, the initial RTP sequence number is randomly chosen not between 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over counter (ROC) synchronization is not lost for sRTP, when the very first RTP packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 ASTERISK-26207 #close Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464	9 years ago
zuul	4b2031226d	Merge "Update support for SILK format." into 13	9 years ago
Alexei Gradinari	cb58f853e1	res_pjsip_mwi: remove unneeded check on endpoint's contacts. The function create_mwi_subscriptions_for_endpoint checks if there is active contacts by retrieving aors and contacts. This function is used to create all unsolicited mwi subscriptions on startup and is used when contact added. In both cases it's not necessary to check if there are contacts. The contacts are needed when asterisk sends mwi. ASTERISK-26200 #close Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa	9 years ago
Mark Michelson	28501051b4	Update support for SILK format. This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e	9 years ago
zuul	b12aee68be	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS." into 13	9 years ago
zuul	56668e3e9c	Merge "pjsip_options.c: Fix container operation." into 13	9 years ago
zuul	91148fdd4f	Merge "pjsip_configuration.c: Misc cleanups." into 13	9 years ago
zuul	8c3d301dc6	Merge "res/res_pjsip_session: Check for presence of an active negotiator" into 13	9 years ago
Joshua Colp	ca98b6cea2	Merge "res/res_pjsip_pubsub: Add additional debug statements" into 13	9 years ago
Joshua Colp	355fc081e6	Merge "res/res_corosync: Raise a Stasis message on node join/leave events" into 13	9 years ago
Alexander Traud	332beb27d8	res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c	9 years ago
Richard Mudgett	fea201f7e6	pjsip_options.c: Fix container operation. aor_observer_deleted() needs to operate on all contacts found for the deleted AOR instead of only the first one found. This is really only a problem if there is more than one contact for the AOR. Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1	9 years ago
Richard Mudgett	02877b4b4f	pjsip_configuration.c: Misc cleanups. * Fix some whitespace in various routines. * Rename i to iter in persistent_endpoint_update_state(). * Fix off-nominal copy/paste message wording in persistent_endpoint_contact_deleted_observer() Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8	9 years ago
zuul	8cea01ab1b	Merge "res_pjsip: Fix statsd regression." into 13	9 years ago
zuul	afd7aba585	Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed" into 13	9 years ago
zuul	daec52a8e6	Merge "res_pjsip: Added "subscribe_context" to endpoint" into 13	9 years ago
Richard Mudgett	97b4c7a5b4	res_pjsip: Fix statsd regression. The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1	9 years ago
Joshua Colp	17efed6cf7	func_odbc: Fix connection deadlock. The func_odbc module was modified to ensure that the previous behavior of using a single database connection was maintained. This was done by getting a single database connection and holding on to it. With the new multiple connection support in res_odbc this will actually starve every other thread from getting access to the database as it also maintains the previous behavior of having only a single database connection. This change disables the func_odbc specific behavior if the res_odbc module is running with only a single database connection active. The connection is only kept for the duration of the request. ASTERISK-26177 #close Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f	9 years ago
zuul	8019f32129	Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled." into 13	9 years ago
Scott Griepentrog	e26bd15e7a	PJSIP: provide valid tcp nodelay option for reuse When using TCP transport with chan_pjsip, the TCP_NODELAY option value was allocated on the stack, then passed as a pointer to the tcp transport configuration structure, and later re-used on subsequently created sockets when it was no longer valid. This patch changes the allocation to be a static. ASTERISK-26180 #close Reported by: Scott Griepentrog Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0	9 years ago
Joshua Colp	77b0145a25	chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c	9 years ago
Alexei Gradinari	b4a9fa2c9e	res_sorcery_realtime: fix bug when successful UPDATE is treated as failed If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. This value should be treated as success. But the function sorcery_realtime_update treats it as failed. This bug was found using stress tests on PJSIP. If there are 2 consecutive SIP REGISTER requests with the same contact data during 1 second then res_pjsip_registrar adds contact location on 1st request and tries to update contact location on 2nd. The update fails and res_pjsip_registrar even removes correct contact location. The test "object_update_uncreated" was removed from test_sorcery_realtime.c because it's now a valid situation. This patch also adds missing debug of extra SQL parameter. ASTERISK-26172 #close Change-Id: I05a7f3051455336c9dda29efc229decf86071303	9 years ago

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3576 Commits (4d5e96ab53f7076496e62b251dd349f320120d88)