asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	9de5cd209e	res_ari: Correct Location headers returned by some ARI resources The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	10 years ago
zuul	e6a946400f	Merge "res_hep: Provide an option to pick the UUID type" into 13	10 years ago
zuul	c735ce1a05	Merge "config_transport: Tell pjproject to allow all SSL/TLS protocols" into 13	10 years ago
Matt Jordan	89ae4466ea	res_hep: Provide an option to pick the UUID type At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago
zuul	1705c5d2ba	Merge "pjsip_distributor: Add missing newline to NOTICE" into 13	10 years ago
Joshua Colp	dd9900849b	Merge "basic-cfg: asterisk.conf: don't set languages" into 13	10 years ago
zuul	8143a00c69	Merge "basic-cfg: asterisk.conf: defaults of options" into 13	10 years ago
zuul	515f49747c	Merge "basic-cfg: asterisk.conf: remove [directories]" into 13	10 years ago
zuul	bf3c7a891a	Merge "basic-cfg: asterisk.conf: debug level 5 spams" into 13	10 years ago
zuul	25db2a8bf5	Merge "followme: delete the right recorded name file" into 13	10 years ago
Joshua Colp	a83bf5dabd	Merge "Use doubles instead of floats for conversions when comparing strings." into 13	10 years ago
Tzafrir Cohen	a73d79c22f	basic-cfg: asterisk.conf: remove [directories] A minimal configuration does not need to explicitly spell out the directories. The built-in defaults will do just fine. In many cases they are wrong. Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	1c56de9453	basic-cfg: asterisk.conf: defaults of options Note the default of remmed-out options. To clarify that those values are not the defaults. Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	d7af591c59	basic-cfg: asterisk.conf: debug level 5 spams Don't suggest users to use debug level 5, which spews (usually non-useful) debug information. Reduce the suggestion to (an arbitrarily-selected) level 2. Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	9b7db18fc1	basic-cfg: asterisk.conf: don't set languages * No need to set language in a miniml configuration. 'en' will do just fine. * It would be useful to have an example of setting it to a different language. * Setting the documentation language explicitly is likewise not required. Setting it to a different value is not common. At least until there is a set of translated documentation. Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	eec539a46e	followme: delete the right recorded name file FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
zuul	95285c6d23	Merge "res_pjsip_outbound_registration: generate correct Contact URI for TLS" into 13	10 years ago
Mark Michelson	02d30e171e	Use doubles instead of floats for conversions when comparing strings. In 13.9.0, there was an issue where PJSIP contacts added to an AOR would be deleted at seemingly random times. One reason this was happening was because of an operation to retrieve the contacts whose expiration time was less than or equal to the current time. When retrieving existing contacts, the contact's expiration time and the current time were converted from a string to a float, and those two floats were compared. On some systems, including mine, this conversion was horribly off. For instance, I could regularly see the string "1463079214" get converted into 1463079168.000000. When switching from using a float to using a double, the conversion was as expected. Why was the conversion to float off? My best guess is that the conversion to float was attempting to store the entire value in the 23 bit significand of the IEEE-754 floating point number. In particular, if you take only the 23 most significant bits of 1463079214, you get the messed up 1463079168 that we were seeing in the conversion. It likely was possible to get a more precise value by composing the number using an exponent, but the conversion did not work that way. With a double, you have a 52 bit significand, allowing the entire value to fit there, and thereby allowing an accurate conversion. ASTERISK-26007 #close Reported by Greg Siemon Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070	10 years ago
George Joseph	e2df15bae9	pjsip_distributor: Add missing newline to NOTICE There was a newline missing from the end of the "no matching endpoint" notice. Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181	10 years ago
Sebastian Damm	a94a12bbf7	res_pjsip_outbound_registration: generate correct Contact URI for TLS There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2	10 years ago
Matt Jordan	37214b0bdf	configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER When running on a system that does not support or use AST_UNDEFINED_SANITIZER or AST_LEAK_SANITIZER, the configure script would incorrectly set those constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would cause menuselect to error out, complaining that a blank value is not a valid option. This patch corrects the issue by setting the value to 0 if the options that those constants enable/disable is not found. Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba	10 years ago
zuul	a01ce2b889	Merge "res_pjsip: improve realtime performance" into 13	10 years ago
zuul	81773ceb9c	Merge "res_fax/t38_gateway: Peer V.21 session is created on wrong channel" into 13	10 years ago
Joshua Colp	2d4c818ac1	Merge "app_confbridge: Add a regcontext option for confbridge bridge profiles." into 13	10 years ago
zuul	577d65b701	Merge "res_pjsip_authenticator_digest: Don't use source port in nonce verification" into 13	10 years ago
Joshua Colp	7561ffa64a	Merge "pjproject_bundled: Check for python-dev and TEST_FRAMEWORK" into 13	10 years ago
Kevin Harwell	4d063814ba	res_pjsip_authenticator_digest: Don't use source port in nonce verification From the issue reporter: "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of the timestamp, the source address, the source port, a server UUID that is calculated at startup, and the authentication realm. Rather than caching nonces that we create, we instead attempt to re-calculate the nonce when receiving an incoming request with authentication. We then compare the re-calculated nonce to the incoming nonce, and if they don't match, then authentication has failed early. The problem is that it is possible, especially when using TCP, to receive two requests from the same endpoint but have differing source ports for those requests. Asterisk itself commonly will use different source ports for outbound TCP requests." This patch removes the source port dependency when building the nonce. ASTERISK-25978 #close Change-Id: I871b5f4adce102df1c4988066283095ec509dffe	10 years ago
George Joseph	fb6227a372	config_transport: Tell pjproject to allow all SSL/TLS protocols The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. SSL is not allowed. So, even if you specify "sslv3" for a transport method, it's silently ignored and one of the TLS protocols is used. This was a new behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that we never caught. Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). This tells pjproject to set the socket protocol to match the method. ASTERISK-26004 #close Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078	10 years ago
zuul	0f54a5ee98	Merge "res_pjsip: module load priority" into 13	10 years ago
zuul	5261151286	Merge "file: Ensure nativeformats remains valid for lifetime of use." into 13	10 years ago
Jaco Kroon	2db17a793c	app_confbridge: Add a regcontext option for confbridge bridge profiles. This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f	10 years ago
zuul	60cc53ecb3	Merge "stasis_endpoints: Add new Status and Headers to ContactStatus" into 13	10 years ago
George Joseph	2a7130b8b0	pjproject_bundled: Check for python-dev and TEST_FRAMEWORK The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. The python bindings are now built only if TEST_FRAMEWORK is set and a python development package is installed. libresample was also disabled. ASTERISK-25993 #close Reported-by: Joshua Colp Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03	10 years ago
Alexei Gradinari	72eb7c8301	res_pjsip: module load priority The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* and res_pjsip_registrar modules should load ASAP to avoid "No matching endpoint found" for legitimate endpoint. ASTERISK-25994 Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b	10 years ago
Chris Trobridge	dd00c71aae	config_options.c: Expand #ifdef to contain whole if statement. ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38	10 years ago
Alexei Gradinari	e6eb17efd9	stasis_endpoints: Add new Status and Headers to ContactStatus ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. These additions should be also in stasis_endpoints to include in command "manager show event ContactStatus" Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a	10 years ago
zuul	3fd8113386	Merge "pjsip: Added "reg_server" to contacts (fixed alembic)" into 13	10 years ago
Joshua Colp	fa11f4c920	file: Ensure nativeformats remains valid for lifetime of use. It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915	10 years ago
Alexei Gradinari	9c2032240e	res_pjsip: improve realtime performance This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05	10 years ago
zuul	168a7b3dd8	Merge "res_fax: add FAXMODE variable" into 13	10 years ago
Alexei Gradinari	fe38d21c2a	pjsip: Added "reg_server" to contacts (fixed alembic) ASTERISK-25931 Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549	10 years ago
zuul	8aef2c4156	Merge "res_pjsip/AMI: add contact.updated event" into 13	10 years ago
Joshua Colp	3fa70be0f5	Merge "app_voicemail: always copy dynamic struct to avoid race condition" into 13	10 years ago
zuul	2970f72dff	Merge "pjproject_bundled: Various fixes discovered during testing of OSes" into 13	10 years ago
Alexei Gradinari	7a14e669f0	res_pjsip/AMI: add contact.updated event With the old SIP module AMI sends PeerStatus event on every successfully REGISTER requests, ie, on start registration, update registration and stop registration. With PJSIP AMI sends ContactStatus only when status is changed. Regarding registration: on start registration - Created on stop registration - Removed but on update registration nothing This patch added contact.updated event. ASTERISK-25904 Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f	10 years ago
Alexei Gradinari	06d4ac0355	res_fax: add FAXMODE variable The app_fax set FAXMODE variable, but res_fax missing this feature. This patch add FAXMODE variable which is set to either "audio" or "T38". ASTERISK-25980 Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b	10 years ago
Alexei Gradinari	2d17fe06c5	res_fax/t38_gateway: Peer V.21 session is created on wrong channel The channel and peer V.21 sessions are created on the same channel now. The peer V.21 session should be created only on peer channel when one of channel can handle T.38. Also this patch enable debug for T.38 gateway session if global fax debug enabled. ASTERISK-25982 Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e	10 years ago
Joshua Colp	3677409116	Merge "pjsip: Added "reg_server" to contacts." into 13	10 years ago
Diederik de Groot	a2f19d82a8	configs/basic-pbx/asterisk.conf: contains incorrect path separator Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004	10 years ago
George Joseph	f39089f17c	pjproject_bundled: Various fixes discovered during testing of OSes For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c	10 years ago

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