asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	852e763571	rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c	9 years ago
Richard Mudgett	159e437e5a	dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc	9 years ago
zuul	7ec9819403	Merge "Fix sqlalchemy error regarding identifier length." into 13	9 years ago
zuul	17e0e058ca	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)." into 13	9 years ago
Mark Michelson	eda95236d1	Fix sqlalchemy error regarding identifier length. sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9	9 years ago
zuul	8d6a7b89bd	Merge "res_pjsip: Whitespace and comment cleanup." into 13	9 years ago
Alexander Traud	66c9dfb272	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957	9 years ago
Joshua Colp	0de05c2938	Merge "chan_sip: Prevent deadlock when issuing "sip show channels"" 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	80a9899100	Merge "chan_dahdi.c: Fix deadlock potential in fax redirection." into 13	9 years ago
zuul	2b001fd6aa	Merge "chan_sip.c: Fix deadlock potential in fax redirection." into 13	9 years ago
zuul	00ed6b74ea	Merge "chan_pjsip.c: Fix deadlock potential in fax redirection." into 13	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
George Joseph	52ab0bf258	chan_sip: Prevent deadlock when issuing "sip show channels" sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990	9 years ago
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." into 13	9 years ago
zuul	dec1e31f45	Merge "Add conditional support for noreturn functions." 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	d4242d6250	Merge "Makefile: Retain XML Declaration and DTD in docs." into 13	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
Corey Farrell	7fdf7c3d4c	Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753	9 years ago
Richard Mudgett	dcb8aa8c1c	chan_dahdi.c: Fix deadlock potential in fax redirection. The dahdi_handle_dtmf() and my_handle_dtmf() have 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. ASTERISK-26216 #close Reported by: Richard Mudgett Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa	9 years ago
Richard Mudgett	fa91cf3eec	chan_sip.c: Fix deadlock potential in fax redirection. The sip_read() 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. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e	9 years ago
Richard Mudgett	2e1bdc3775	chan_pjsip.c: Fix deadlock potential in fax redirection. The chan_pjsip_cng_tone_detected() 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. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5	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	652130feb2	chan_dahdi: Add faxdetect_timeout option. The new option allows the channel driver's faxdetect 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. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810	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
Alexander Traud	021d4892cd	Makefile: Retain XML Declaration and DTD in docs. Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, the XML Declaration and DTD were overwritten by this. ASTERISK-26212 #close Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd	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
Alexander Traud	5f24874ebb	Makefile: Suppress echoing of target 'config' again. ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f	9 years ago
zuul	962c7ef5d9	Merge "app_queue: Only remove queue member from pending when state changes." into 13	9 years ago
Corey Farrell	76d4983c15	features.c: Remove unneeded adsi.h include. adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59	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	714ceb88df	Merge "translate: explicit format destination not properly set" into 13	9 years ago
Joshua Colp	8892029a03	Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path." into 13	9 years ago
zuul	b4976b9038	Merge "pbx: Fix leak of timezone for time based includes." into 13	9 years ago
zuul	64a09963be	Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf." into 13	9 years ago
zuul	b12aee68be	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS." into 13	9 years ago
zuul	3c1a40eb8d	Merge "stasis_endpoint.c: Fix contactstatus_to_json()." into 13	9 years ago
Joshua Colp	43b5f8d57b	app_queue: Only remove queue member from pending when state changes. It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0	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
Corey Farrell	a17b071e36	pbx: Fix leak of timezone for time based includes. Create include_free to run ast_destroy_timing and ast_free, use that in all places that freed an ast_include structure. This fixes a couple of paths that previously did not run ast_destroy_timing. ASTERISK-26196 #close Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838	9 years ago

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