asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	3f6ef63099	res_pjsip_outbound_registration: Clean up state when registration is deleted Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23	9 years ago
zuul	bcb133ce93	Merge "configs/samples/pjsip.conf.sample: Fix typo" into 13	9 years ago
George Joseph	b6f9392a12	res_pjsip: Set TCP_NODELAY on TCP transports Although it's perfectly legal to place multiple SIP messages in the same packet, it can cause problems because the Linux default is to enable Path MTU Discovery which sets the Don't Fragment bit on the packets. If adding a second message to the packet causes the MTU to be exceeded, and the destination isn't equipped to send a FRAGMENTATION NEEDED response to a large packet, the packet will just be dropped. We can't specifically tell the stack to send only 1 message per packet, but we can turn on TCP_NODELAY when we create the transport. This will at least tell the stack to send packets as soon as possible. ASTERISK-26005 #close Reported-by: Ross Beer Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd	9 years ago
Matt Jordan	361a16f316	configs/samples/pjsip.conf.sample: Fix typo A ':' is not a valid token for starting a comment. Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad	9 years ago
Matt Jordan	f91a7dc993	res/res_hep_pjsip: Fix reported local IP address when bound to 'any' When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3	9 years ago
zuul	12db14b9b1	Merge "logger: Add PID to syslog messages." into 13	9 years ago
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	9 years ago
zuul	e6a946400f	Merge "res_hep: Provide an option to pick the UUID type" into 13	9 years ago
zuul	c735ce1a05	Merge "config_transport: Tell pjproject to allow all SSL/TLS protocols" into 13	9 years ago
Alexei Gradinari	524a302974	res_pjsip: Endpoint IP Access Controls With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02	9 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	9 years ago
zuul	1705c5d2ba	Merge "pjsip_distributor: Add missing newline to NOTICE" into 13	9 years ago
Joshua Colp	dd9900849b	Merge "basic-cfg: asterisk.conf: don't set languages" into 13	9 years ago
zuul	8143a00c69	Merge "basic-cfg: asterisk.conf: defaults of options" into 13	9 years ago
zuul	515f49747c	Merge "basic-cfg: asterisk.conf: remove [directories]" into 13	9 years ago
zuul	bf3c7a891a	Merge "basic-cfg: asterisk.conf: debug level 5 spams" into 13	9 years ago
zuul	25db2a8bf5	Merge "followme: delete the right recorded name file" into 13	9 years ago
Joshua Colp	a83bf5dabd	Merge "Use doubles instead of floats for conversions when comparing strings." into 13	9 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>	9 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>	9 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>	9 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>	9 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>	9 years ago
zuul	95285c6d23	Merge "res_pjsip_outbound_registration: generate correct Contact URI for TLS" into 13	9 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	9 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	9 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	9 years ago
Alexei Gradinari	36d66a23e0	logger: Add PID to syslog messages. During refactoring of this support the addition of the PID to messages was removed. This change adds it back in. ASTERISK-25538 #close Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36	9 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	9 years ago
zuul	a01ce2b889	Merge "res_pjsip: improve realtime performance" into 13	9 years ago
Kevin Harwell	49b25a0956	res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches When reloading, or fetching realtime data, if the "apply" failed for any numerous reasons the current state object would not be maintained. This potentially resulted in publishes being stopped for some states/clients when they should not have been. This patch makes it so the current state object is kept upon any type of reload/ fetch failures. Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30	9 years ago
Kevin Harwell	1b5c91b7be	res_pjsip_outbound_publish: Potential crash due to off nominal path It was possible for the explicit publish destroy function to be called without the pjsip client ever being initialized. This fix checks to make sure there is a client to destroy before attempting. Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c	9 years ago
Kevin Harwell	10de553c9d	res_pjsip_outbound_publishing: After unloading the library won't load again The same thing was happening in res_pjsip_publish_asterisk. When the library was unloaded it did not unregister the object type from sorcery. Subsequent loads resulted in a failed load due to the sorcery type already existing. Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9	9 years ago
Kevin Harwell	1a833b9739	res_pjsip_outbound_publish: Ref leak in off nominal callback paths There were a few spots where the client object's reference was being leaked in sip_outbound_publish_callback. This patch cleans up those leaks. Change-Id: I485d0bc9335090f373026f77c548042e258461df	9 years ago
Kevin Harwell	4752ef02e0	res_pjsip_outbound_publish: Won't unload if condition wait times out When res_pjsip_outbound_publish unloads it has to wait for all current publishing objects to get done. However if the wait condition times out then it does not fail the unload. This sometimes results in an infinite loop check while unloading. This patch now fails the unload operation if the condition times out. Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec	9 years ago
zuul	81773ceb9c	Merge "res_fax/t38_gateway: Peer V.21 session is created on wrong channel" into 13	9 years ago
Joshua Colp	2d4c818ac1	Merge "app_confbridge: Add a regcontext option for confbridge bridge profiles." into 13	9 years ago
zuul	577d65b701	Merge "res_pjsip_authenticator_digest: Don't use source port in nonce verification" into 13	9 years ago
Joshua Colp	7561ffa64a	Merge "pjproject_bundled: Check for python-dev and TEST_FRAMEWORK" into 13	9 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	9 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	9 years ago
zuul	0f54a5ee98	Merge "res_pjsip: module load priority" into 13	9 years ago
zuul	5261151286	Merge "file: Ensure nativeformats remains valid for lifetime of use." into 13	9 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	9 years ago
zuul	60cc53ecb3	Merge "stasis_endpoints: Add new Status and Headers to ContactStatus" into 13	9 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	9 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	9 years ago
Chris Trobridge	dd00c71aae	config_options.c: Expand #ifdef to contain whole if statement. ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38	9 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	9 years ago
zuul	3fd8113386	Merge "pjsip: Added "reg_server" to contacts (fixed alembic)" into 13	9 years ago

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