5213 Commits (ec16d2ecbd70df3d99edb7c05bb390e020c738c4)

Author	SHA1	Message	Date
Joshua C. Colp	ec16d2ecbd	AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. ... If a re-INVITE is received after we have sent a BYE request then it is possible for no channel to be present on the session. If this occurs we allow PJSIP to produce the offer instead. Since the call is being hung up if it produces an incorrect offer it doesn't actually matter. This also ensures that code which produces SDP does not need to handle if a channel is not present. ASTERISK-29381 Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042	5 years ago
Andre Barbosa	f4d3f021f9	res_stasis_playback: Check for chan hangup on play_on_channels ... Verify `ast_check_hangup` before looping to the next sound file. If the call is already hangup we just break the cycle. It also ensures that the PlaybackFinished event is sent if the call was hangup. This is also use-full when we are playing a big list of file for a channel that is hangup. Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played. With the patch we just break the playback cycle when the chan is hangup. ASTERISK-29501 #close Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8	5 years ago
Sean Bright	d5bb27a06f	res_http_media_cache.c: Fix merge errors from 18 -> master ... ASTERISK-27871 #close Change-Id: I6624f2d3a57f76a89bb372ef54a124929a0338d7	5 years ago
Sean Bright	237285a9a8	res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup. ... From RFC 8225 Section 5.2.1: The "dest" claim is a JSON object with the claim name of "dest" and MUST have at least one identity claim object. The "dest" claim value is an array containing one or more identity claim JSON objects representing the destination identities of any type (currently "tn" or "uri"). If the "dest" claim value array contains both "tn" and "uri" claim names, the JSON object should list the "tn" array first and the "uri" array second. Within the "tn" and "uri" arrays, the identity strings should be put in lexicographical order, including the scheme-specific portion of the URI characters. Additionally, make it clear that there was a failure to sign the JWT payload and not necessarily a memory allocation failure. Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9	5 years ago
Sean Bright	d568326807	res_http_media_cache.c: Parse media URLs to find extensions. ... Use cURL's URL parsing API, falling back to the urlparser library, to parse playback URLs in order to find their file extensions. For backwards compatibility, we first look at the full URL, then at any Content-Type header, and finally at just the path portion of the URL. ASTERISK-27871 #close Change-Id: I16d0682f6d794be96539261b3e48f237909139cb	5 years ago
Igor Goncharovsky	99d44f0c5a	res_ari: Fix audiosocket segfault ... Add check that data parameter specified when audiosocket used for externalMedia. ASTERISK-29514 #close Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617	5 years ago
Sean Bright	0ac9c83561	res_pjsip_config_wizard.c: Add port matching support. ... In f8b0c2c9 we added support for port numbers in 'match' statements but neglected to include that support in the PJSIP config wizard. The removed code would have also prevented IPv6 addresses from being successfully used in the config wizard as well. ASTERISK-29503 #close Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d	5 years ago
Andre Barbosa	a47308ccb2	res_stasis_playback: Send PlaybackFinish event only once for errors ... When we try to play a list of sound files in the same Play command, we get only one PlaybackFinish event, after all sounds are played. But in the case where the Play fails (because channel is destroyed for example), Asterisk will send one PlaybackFinish event for each sound file still to be played. If the list is big, Asterisk is sending many events. This patch adds a failed state so we can understand that the play failed. On that case we don't send the event, if we still have a list of sounds to be played. When we reach the last sound, we send the PlaybackFinish with the failed state. ASTERISK-29464 #close Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322	5 years ago
Bernd Zobl	c30f68a57b	res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress ... With the fix for ASTERISK_28754 channels are no longer put on hold if an outbound INVITE is answered with a "Session Progress" containing "inactive" audio. The previous change moved the evaluation of the media attributes to `negotiate_incoming_sdp_stream()` to have the `remotely_held` status available when building the SDP in `create_outgoing_sdp_stream()`. This however means that an answer to an outbound INVITE, which does not traverse `negotiate_incoming_sdp_stream()`, cannot set the `remotely_held` status anymore. This change moves the check so that both, `negotiate_incoming_sdp_stream()` and `apply_negotiated_sdp_stream()` can do the checks. ASTERISK-29479 Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75	5 years ago
George Joseph	b7027de195	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	f160725fc4	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
Sean Bright	c0fc8adbb6	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	1b38e89734	res_pjsip_dtmf_info: Hook flash ... Adds hook flash recognition support for application/hook-flash. ASTERISK-29460 Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea	5 years ago
George Joseph	c3654a9959	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
Ben Ford	12e8600849	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	44fde9f428	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
Evgenios_Greek	2193cf1b26	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	98e4119642	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	a985e5069c	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
Jeremy Lainé	d162789c4d	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	9cc1d6fc22	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	3cccdf6d98	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
Ben Ford	0564d12280	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	05f7bc9c66	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	259ecfa289	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	09303e8e22	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	b1807d440e	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
Sean Bright	4a843e00ef	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	5 years ago
George Joseph	512d38868c	res_pjsip: Update documentation for the auth object ... Change-Id: I2f76867ce02ec611964925159be099de83346e38	5 years ago
Ben Ford	45a1977de4	res_aeap: Add basic config skeleton and CLI commands. ... Added support for a basic AEAP configuration read from aeap.conf. Also added 2 CLI commands for showing individual configurations as well as all of them: aeap show server <id> and aeap show servers. Only one configuration option is required at the moment, and that one is server_url. It must be a websocket URL. The other option, codecs, is optional and will be used over the codecs specified on the endpoint if provided. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453 Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3	5 years ago
George Joseph	53c702e1cc	res_prometheus: Clone containers before iterating ... The channels, bridges and endpoints scrape functions were grabbing their respective global containers, getting the count of entries, allocating metric arrays based on that count, then iterating over the container. If the global container had new objects added after the count was taken and the metric arrays were allocated, we'd run out of metric entries and attempt to write past the end of the arrays. Now each of the scape functions clone their respective global containers and all operations are done on the clone. Since the clone is stable between getting the count and iterating over it, we can't run past the end of the metrics array. ASTERISK-29130 Reported-By: Francisco Correia Reported-By: BJ Weschke Reported-By: Sébastien Duthil Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af	5 years ago
Kevin Harwell	0fc906a5e1	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	c4a376aac2	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	65b68fd060	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	8bd13a995a	res_rtp_asterisk: Only raise flash control frame on end. ... Flash in RTP is conveyed the same as DTMF, just with a specific digit. In Asterisk however we do flash as a single control frame. This change makes it so that only on end do we provide the flash control frame to the core. Previously we would provide a flash control frame on both begin and end, causing flash to work improperly. ASTERISK-29373 Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226	5 years ago
Kevin Harwell	b86f1ef54c	res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command ... This patch makes it so when Asterisk is compiled in DEVMODE a CLI command is available that allows someone to drop incoming RTP packets. The command allows for dropping of packets once, or on a timed interval (e.g. drop 10 packets every 5 seconds). A user can also specify to drop packets by IP address. Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024	5 years ago
Joshua C. Colp	623abc2b6a	res_pjsip: Give error when TLS transport configured but not supported. ... Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b	5 years ago
George Joseph	a03a05195a	res_pjsip_session: Make reschedule_reinvite check for NULL topologies ... When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	5 years ago
Joshua C. Colp	71dfbdc7b9	res_pjsip: Add support for partial transport reload. ... Some configuration items for a transport do not result in the underlying transport changing, but instead are just state we keep ourselves and use. It is perfectly reasonable to change these items. These include local_net and external_* information. ASTERISK-29354 Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a	5 years ago
Joshua C. Colp	cce5ee5b7a	res_rtp_asterisk: Force resync on SSRC change. ... When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	5 years ago
Joshua C. Colp	149e5e5b86	xml: Embed module information into core XML documentation. ... This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	5 years ago
Joshua C. Colp	7438586d8e	documentation: Fix non-matching module support levels. ... Some modules have a different support level documented in their MODULEINFO XML and Asterisk module definition. This change brings the two in sync for the modules which were not matching. ASTERISK-29336 Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35	5 years ago
Jaco Kroon	41389bfdbd	func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds ... Change-Id: I75152cece8a00b7523d542e5ac22796f9595692b Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Alexander Traud	1ae40e502d	res_format_attr_*: Parameter Names are Case-Insensitive. ... see RFC 4855: parameter names are case-insensitive both in media type strings and in the default mapping to the SDP a=fmtp attribute. This change is required for H.263+ because some implementations are known to use even mixed-case. This does not fix ASTERISK~29268 because H.264 was not fixed. This approach here lowers/uppers both parameter names and parameter values. H.264 needs a different approach because one of its parameter values is not case-insensitive: sprop-parameter-sets is Base64. Change-Id: Idf2a73457be231647aed3c87b1da197afba86892	5 years ago
Sean Bright	df37b8181c	res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse. ... ao2_replace() bumps the reference count of the object that is doing the replacing, which is not what we want. We just want to drop the old ref on the old object and update the pointer to point to the new object. Pointed out by George Joseph in #asterisk-dev Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d	5 years ago
Torrey Searle	8c247e2a94	res/res_rtp_asterisk: generate new SSRC on native bridge end ... For RTCP to work, we update the ssrc to be the one corresponding to the native bridge while active. However when the bridge ends we should generate a new SSRC as the sequence numbers will not continue from the native bridge left off. ASTERISK-29300 #close Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10	5 years ago
Joshua C. Colp	304f8ddfb2	sorcery: Add support for more intelligent reloading. ... Some sorcery objects actually contain dynamic content that can change despite the underlying configuration itself not changing. A good example of this is the res_pjsip_endpoint_identifier_ip module which allows specifying hostnames. While the configuration may not change between reloads the DNS information of the hostnames can. This change adds the ability for a sorcery object to be marked as having dynamic contents which is then taken into account when reloading by the sorcery file based config module. If there is an object with dynamic content then a reload will be forced while if there are none then the existing behavior of not reloading occurs. ASTERISK-29321 Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1	5 years ago
George Joseph	607603cf89	res_pjsip_refer: Move the progress dlg release to a serializer ... Although the dlg session count was incremented in a pjsip servant thread, there's no guarantee that the last thread to unref this progress object was one. Before we decrement, we need to make sure that this is either a servant thread or that we push the decrement to a serializer that is one. Because pjsip_dlg_dec_session requires the dialog lock, we don't want to wait on the task to complete if we had to push it to a serializer. Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41	5 years ago
Joshua C. Colp	6f67f24afd	res_pjsip_registrar: Include source IP and port in log messages. ... When registering it can be useful to see the source IP address and port in cases where multiple devices are using the same endpoint or when anonymous is in use. ASTERISK-29325 Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0	5 years ago
Ben Ford	fd560ad9fa	AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. ... When Asterisk sends a reinvite negotiating T38 faxing, it's possible a crash can occur if the response contains a m=image and zero port. The reinvite callback code now checks session_media to see if it is null or not before trying to access the udptl variable on it. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	5 years ago