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Author	SHA1	Message	Date
George Joseph	b6bf6d091e	res_pjsip_messaging: Overwrite user in existing contact URI ... When the MessageSend destination is in the form PJSIP/<number>@<endpoint> and the endpoint's contact URI already has a user component, that user component will now be replaced with <number> when creating the request URI. ASTERISK_29404 Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5	5 years ago
Bernd Zobl	fe40c3cc74	res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter ... Set preferred transport when querying the local address to use in filter_on_tx_messages(). This prevents the module to erroneously select the wrong transport if more than one transports of the same type (TCP or TLS) are configured. ASTERISK-29241 Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6	5 years ago
Naveen Albert	80be0edae7	pbx_builtins: Corrects SayNumber warning ... Previously, SayNumber always emitted a warning if the caller hung up during execution. Usually this isn't correct, so check if the channel hung up and, if so, don't emit a warning. ASTERISK-29475 Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594	5 years ago
Jaco Kroon	2dbf8cd135	func_lock: Add "dialplan locks show" cli command. ... For example: arthur*CLI> dialplan locks show func_lock locks: Name Requesters Owner uls-autoref 0 (unlocked) 1 total locks listed. Obviously other potentially useful stats could be added (eg, how many times there was contention, how many times it failed etc ... but that would require keeping the stats and I'm not convinced that's worth the effort. This was useful to troubleshoot some other issues so submitting it. Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Jaco Kroon	5d7d174e1f	func_lock: Prevent module unloading in-use module. ... The scenario where a channel still has an associated datastore we cannot unload since there is a function pointer to the destroy and fixup functions in play. Thus increase the module ref count whenever we allocate a datastore, and decrease it during destroy. In order to tighten the race that still exists in spite of this (below) add some extra failure cases to prevent allocations in these cases. Race: If module ref is zero, an LOCK or TRYLOCK is invoked (near) simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if in such a case the datastore is created *prior* to unloading being set to true (first step in module unload) then it's possible that the module will unload with the destructor being called (and segfault) post the module being unloaded. The module will however wait for such locks to release prior to unloading. If post that we can recheck the module ref before returning the we can (in theory, I think) eliminate the last of the race. This race is mostly theoretical in nature. Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Jaco Kroon	4040095ea4	func_lock: Fix memory corruption during unload. ... AST_TRAVERSE accessess current as current = current->(field).next ... and since we free current (and ast_free poisons the memory) we either end up on a ast_mutex_lock to a non-existing lock that can never be obtained, or a segfault. Incidentally add logging in the "we have to wait for a lock to release" case, and remove an ineffective statement that sets memory that was just cleared by ast_calloc to zero. Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Jaco Kroon	5008d9fee1	func_lock: Fix requesters counter in error paths. ... In two places we bail out with failure after we've already incremented the requesters counter, if this occured then it would effectively result in unload to wait indefinitely, thus preventing clean shutdown. Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Naveen Albert	b4e77b7f10	app_originate: Allow setting Caller ID and variables ... Caller ID can now be set on the called channel and Variables can now be set on the destination using the Originate application, just as they can be currently using call files or the Manager Action. ASTERISK-29450 Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66	5 years ago
Sean Bright	e824138a26	menuselect: Fix description of several modules. ... The text description needs to be the last thing on the AST_MODULE_INFO line to be pulled in properly by menuselect. Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832	5 years ago
Naveen Albert	2c2dc7d97d	app_confbridge: New ConfKick() application ... Adds a new ConfKick() application, which may be used to kick a specific channel, all channels, or all non-admin channels from a specified conference bridge, similar to existing CLI and AMI commands. ASTERISK-29446 Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b	5 years ago
Naveen Albert	17b9c5c5cf	res_pjsip_dtmf_info: Hook flash ... Adds hook flash recognition support for application/hook-flash. ASTERISK-29460 Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea	5 years ago
Naveen Albert	2376425785	sip_to_pjsip: Fix missing cases ... Adds the "auto" case which is valid with both chan_sip dtmfmode and chan_pjsip's dtmf_mode, adds subscribecontext to subscribe_context conversion, and accounts for cipher = ALL being invalid. ASTERISK-29459 Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2	5 years ago
Naveen Albert	95f588496d	app_confbridge: New option to prevent answer supervision ... A new user option, answer_channel, adds the capability to prevent answering the channel if it hasn't already been answered yet. ASTERISK-29440 Change-Id: I26642729d0345f178c7b8045506605c8402de54b	5 years ago
George Joseph	6cd89c4f0a	res_pjsip_messaging: Refactor outgoing URI processing ... * Implemented the new "to" parameter of the MessageSend() dialplan application. This allows a user to specify a complete SIP "To" header separate from the Request URI. * Completely refactored the get_outbound_endpoint() function to actually handle all the destination combinations that we advertized as supporting. * We now also accept a destination in the same format as Dial()... PJSIP/number@endpoint * Added lots of debugging. ASTERISK-29404 Reported by Brian J. Murrell Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce	5 years ago
Naveen Albert	8236f2f155	func_math: Three new dialplan functions ... Introduces three new dialplan functions, MIN and MAX, which can be used to calculate the minimum or maximum of up to two numbers, and ABS, an absolute value function. ASTERISK-29431 Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d	5 years ago
Ben Ford	cee88c9826	STIR/SHAKEN: Add Date header, dest->tn, and URL checking. ... STIR/SHAKEN requires a Date header alongside the Identity header, so that has been added. Still on the outgoing side, we were missing the dest->tn section of the JSON payload, so that has been added as well. Moving to the incoming side, URL checking has been added to the public cert URL to ensure that it starts with http. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c	5 years ago
Joshua C. Colp	e07fd35238	res_pjsip: On partial transport reload also move factories. ... For connection oriented transports PJSIP uses factories to produce transports. When doing a partial transport reload we need to also move the factory of the transport over so that anything referencing the transport (such as an endpoint) has the factory available. ASTERISK-29441 Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161	5 years ago
Naveen Albert	d1305af137	func_volume: Add read capability to function. ... Up until now, the VOLUME function has been write only, so that TX/RX values can be set but not read afterwards. Now, previously set TX/RX values can be read later. ASTERISK-29439 Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f	5 years ago
Joshua C. Colp	bb8e868066	res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates. ... RTCP ICE candidates use a base address derived from the RTP candidate. The port on the base address was not being updated to the RTCP port. This change sets the base port to the RTCP port and all is well. ASTERISK-29433 Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040	5 years ago
Evgenios_Greek	6d5f087138	stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing ... When unsubscribing from an endpoint technology a FRACK would occur due to incorrect reference counting. This fixes that issue, along with some other issues. Fixed a typo in get_subscription when calling ao2_find as it needed to pass the endpoint ID and not the entire object. Fixed scenario where a subscription would get returned when it shouldn't have been when searching based on endpoint technology. A doulbe unreference has also been resolved by only explicitly releasing the reference held by tech_subscriptions. ASTERISK-28237 #close Reported by: Lucas Tardioli Silveira Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729	5 years ago
Joseph Nadiv	a123e8cb0e	res_pjsip.c: Support endpoints with domain info in username ... In multidomain environments, it is desirable to create PJSIP endpoints with the domain info in the endpoint name in pjsip_endpoint.conf. This resulted in an error with registrations, NOTIFY, and OPTIONS packet generation. This commit will detect if there is an @ in the endpoint identifier and generate the URI accordingly so NOTIFY and OPTIONS From headers will generate correctly. ASTERISK-28393 Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619	5 years ago
Joshua C. Colp	0ee56cc2bd	asterisk: We've moved to Libera Chat! ... Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575	5 years ago
Jeremy Lainé	0cd5d5150a	res_rtp_asterisk: make it possible to remove SOFTWARE attribute ... By default Asterisk reports the PJSIP version in a SOFTWARE attribute of every STUN packet it sends. This may not be desired in a production environment, and RFC5389 recommends making the use of the SOFTWARE attribute a configurable option: https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2 This patch adds a `stun_software_attribute` yes/no option to make it possible to omit the SOFTWARE attribute from STUN packets. ASTERISK-29434 Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b	5 years ago
George Joseph	7c0b8f0701	res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs ... RFC7616 and RFC8760 allow more than one WWW-Authenticate or Proxy-Authenticate header per realm, each with different digest algorithms (including new ones like SHA-256 and SHA-512-256). Thankfully however a UAS can NOT send back multiple Authenticate headers for the same realm with the same digest algorithm. The UAS is also supposed to send the headers in order of preference with the first one being the most preferred. We're supposed to send an Authorization header for the first one we encounter for a realm that we can support. The UAS can also send multiple realms, especially when it's a proxy that has forked the request in which case the proxy will aggregate all of the Authenticate headers and then send them all back to the UAC. It doesn't stop there though... Each realm can require a different username from the others. There's also nothing preventing each digest algorithm from having a unique password although I'm not sure if that adds any benefit. So now... For each Authenticate header we encounter, we have to determine if we support the digest algorithm and, if not, just skip the header. We then have to find an auth object that matches the realm AND the digest algorithm or find a wildcard object that matches the digest algorithm. If we find one, we add it to the results vector and read the next Authenticate header. If the next header is for the same realm AND we already added an auth object for that realm, we skip the header. Otherwise we repeat the process for the next header. In the end, we'll have accumulated a list of credentials we can pass to pjproject that it can use to add Authentication headers to a request. NOTE: Neither we nor pjproject can currently handle digest algorithms other than MD5. We don't even have a place for it in the ast_sip_auth object. For this reason, we just skip processing any Authenticate header that's not MD5. When we support the others, we'll move the check into the loop that searches the objects. Changes: * Added a new API ast_sip_retrieve_auths_vector() that takes in a vector of auth ids (usually supplied on a call to ast_sip_create_request_with_auth()) and populates another vector with the actual objects. * Refactored res_pjsip_outbound_authenticator_digest to handle multiple Authenticate headers and set the stage for handling additional digest algorithms. * Added a pjproject patch that allows them to ignore digest algorithms they don't support. This patch has already been merged upstream. * Updated documentation for auth objects in the XML and in pjsip.conf.sample. * Although res_pjsip_authenticator_digest isn't affected by this change, some debugging and a testsuite AMI event was added to facilitate testing. Discovered during OpenSIPit 2021. ASTERISK-29397 Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281	5 years ago
Joseph Nadiv	48851279a8	res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml ... RFC 4235 Section 4.1.6 describes XML elements that should be sent to subscribed endpoints to identify the local and remote participants in the dialog. This patch adds this functionality to PJSIP by iterating through the ringing channels causing the NOTIFY, and inserts the channel info into the dialog so that information is properly passed to the endpoint in dialog-info+xml. ASTERISK-24601 Patch submitted: Joshua Elson Modified by: Joseph Nadiv and Sean Bright Tested by: Joseph Nadiv Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b	5 years ago
Naveen Albert	10aa4f9876	app_voicemail: Configurable voicemail beep ... Hitherto, VoiceMail() played a non-customizable beep tone to indicate the caller could leave a message. In some cases, the beep may not be desired, or a different tone may be desired. To increase flexibility, a new option allows customization of the tone. If the t option is specified, the default beep will be overridden. Supplying an argument will cause it to use the specified file for the tone, and omitting it will cause it to skip the beep altogether. If the option is not used, the default behavior persists. ASTERISK-29349 Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280	5 years ago
Naveen Albert	ea117be4c6	AMI: Add AMI event to expose hook flash events ... Although Asterisk can receive and propogate flash events, it currently provides no mechanism for doing anything with them itself. This AMI event allows flash events to be processed by Asterisk. Additionally, AST_CONTROL_FLASH is included in a switch statement in channel.c to avoid throwing a warning when we shouldn't. ASTERISK-29380 Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81	5 years ago
Naveen Albert	859cd2a56b	main/file.c: Don't throw error on flash event. ... AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c where it should be ignored. Adding this to the switch ensures a warning isn't thrown on RFC2833 flash events, since nothing's amiss. ASTERISK-29372 Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc	5 years ago
Joshua C. Colp	8f8b4e7e41	pjsip: Add patch for resolving STUN packet lifetime issues. ... In some cases it was possible for a STUN packet to be destroyed prematurely or even destroyed partially multiple times. This patch provided by Teluu fixes the lifetime of these packets and ensures they aren't partially destroyed multiple times. https://github.com/pjsip/pjproject/pull/2709 ASTERISK-29377 Change-Id: Ie842ad24ddf345e01c69a4d333023f05f787abca	5 years ago
Naveen Albert	11e61217a0	chan_sip: Expand hook flash recognition. ... Some ATAs send hook flash events as application/hook-flash, rather than a DTMF event. Now, we also recognize hook-flash as a flash event. ASTERISK-29370 Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725	5 years ago
Sean Bright	f557126998	chan_pjsip: Correct misleading trace message ... ASTERISK-29358 #close Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647	5 years ago
Ben Ford	0b4b207076	STIR/SHAKEN: Switch to base64 URL encoding. ... STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	5 years ago
Ben Ford	d00bedba81	STIR/SHAKEN: OPENSSL_free serial hex from openssl. ... We're getting the serial number of the certificate from openssl and freeing it with ast_free(), but it needs to be freed with OPENSSL_free() instead. Now we duplicate the string and free the one from openssl with OPENSSL_free(), which means we can still use ast_free() on the returned string. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab	5 years ago
Ben Ford	184a027eaa	STIR/SHAKEN: Fix certificate type and storage. ... During OpenSIPit, we found out that the public certificates must be of type X.509. When reading in public keys, we use the corresponding X.509 functions now. We also discovered that we needed a better naming scheme for the certificates since certificates with the same name would cause issues (overwriting certs, etc.). Now when we download a public certificate, we get the serial number from it and use that as the name of the cached certificate. The configuration option public_key_url in stir_shaken.conf has also been renamed to public_cert_url, which better describes what the option is for. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d	5 years ago
George Joseph	63c25d3821	Updates for the MessageSend Dialplan App ... Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a	5 years ago
Sean Bright	d72c7d2d66	translate.c: Avoid refleak when checking for a translation path ... Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6	5 years ago
Sean Bright	0eb5c58bd4	res_rtp_asterisk: More robust timestamp checking ... We assume that a timestamp value of 0 represents an 'uninitialized' timestamp, but 0 is a valid value. Add a simple wrapper to be able to differentiate between whether the value is set or not. This also removes the fix for ASTERISK~28812 which should not be needed if we are checking the last timestamp appropriately. ASTERISK-29030 #close Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7	5 years ago
Joshua C. Colp	d2b8141d26	chan_local: Skip filtering audio formats on removed streams. ... When a stream topology is provided to chan_local when dialing it filters the audio formats down. This operation did not skip streams which were removed (that have no formats) resulting in calling being aborted. This change causes such streams to be skipped. ASTERISK-29407 Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6	5 years ago
Asterisk Development Team	0fd9c6588b	Update CHANGES and UPGRADE.txt for 16.18.0	5 years ago
Sean Bright	5a2939a5d6	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	5 years ago
Jean Aunis	080d0eb72c	translate.c: Take sampling rate into account when checking codec's buffer size ... Up/down sampling changes the number of samples produced by a translation. This must be taken into account when checking the codec's buffer size. ASTERISK-29328 Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055	5 years ago
Joshua C. Colp	c43094f1cb	svn: Switch to https scheme. ... Some versions of SVN seemingly don't follow the redirect to https. Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253	5 years ago
George Joseph	5ace782583	res_pjsip: Update documentation for the auth object ... Change-Id: I2f76867ce02ec611964925159be099de83346e38	5 years ago
George Joseph	a4a63db253	bridge_channel_write_frame: Check for NULL channel ... There is a possibility, when bridge_channel_write_frame() is called, that the bridge_channel->chan will be NULL. The first thing bridge_channel_write_frame() does though is call ast_channel_is_multistream() which had no check for a NULL channel and therefore caused a segfault. Since it's still possible for bridge_channel_write_frame() to write the frame to the other channels in the bridge, we don't want to bail before we call ast_channel_is_multistream() but we can just skip the multi-channel stuff. So... bridge_channel_write_frame() only calls ast_channel_is_multistream() if bridge_channel->chan is not NULL. As a safety measure, ast_channel_is_multistream() now returns false if the supplied channel is NULL. ASTERISK-29379 Reported-by: Vyrva Igor Reported-by: Ross Beer Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce	5 years ago
Sean Bright	84115fe657	loader.c: Speed up deprecation metadata lookup ... Only use an XPath query once per module, then just navigate the DOM for everything else. Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92	5 years ago
Joshua C. Colp	a9e1e0e1c8	loader: Output warnings for deprecated modules. ... Using the information from the MODULEINFO XML we can now output useful information at the end of module loading for deprecated modules. This includes the version it was deprecated in, the version it will be removed in, and the replacement if available. ASTERISK-29339 Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a	5 years ago
Kevin Harwell	bd426397bb	res_rtp_asterisk: Fix standard deviation calculation ... For some input to the standard deviation algorithm extremely large, and wrong numbers were being calculated. This patch uses a new formula for correctly calculating both the running mean and standard deviation for the given inputs. ASTERISK-29364 #close Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f	5 years ago
Kevin Harwell	473e648fca	res_rtp_asterisk: Don't count 0 as a minimum lost packets ... The calculated minimum lost packets represents the lowest number of lost packets missed during an RTCP report interval. Zero of course is the lowest, but the idea is that this value contain the lowest number of lost packets once some have been missed. This patch checks to make sure the number of lost packets over an interval is not zero before checking and setting the minimum value. Also, this patch updates the rtp lost packet test to check for packet loss over several reports vs one. Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008	5 years ago
Kevin Harwell	07f8456dcb	res_rtp_asterisk: Statically declare rtp_drop_packets_data object ... This patch makes the drop_packets_data object static. Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b	5 years ago
Joshua C. Colp	3d4564dc16	res_pjsip: Give error when TLS transport configured but not supported. ... Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	5 years ago