4654 Commits (77a5383957504d58889d4347ce72792aac6e51ea)

Author	SHA1	Message	Date
sungtae kim	52e1e362f0	res_pjsip.c: Added disable_rport option for pjsip.conf ... Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	6 years ago
Nickolay Shmyrev	f323ac334b	res_http_websocket.c: Continue reading after ping/pong ... Do not return error if the client received ping frame while looking for a string and just wait for another frame. ASTERISK-28958 #close Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922	6 years ago
Kevin Harwell	ef19d69c6b	PJSIP_MEDIA_OFFER: override what's specified on configuration ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP definition no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	6 years ago
Joshua C. Colp	df509590a1	res_pjsip: Apply AOR outbound proxy to static contacts. ... The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	6 years ago
Università di Bologna - CESIA VoIP	09370b0a2b	res_corosync: Fix crash in huge distributed environment. ... 1) Fix memory-leaks Added code to release ast_events extracted from corosync and stasis messages 2) Clean stasis cache when a member of the corosync cluster leaves the group Added code to remove from the stasis cache of the members remained on the group all the messages with the EID of the left member. If the device states of the left member remain in the stasis cache of other members, they will not be updated anymore and high priority cached values, like BUSY, will take precedence over current device states. 3) Stop corosync event propagation when node is not joined to the group Updated dispatch_thread_handler code to detect when asterisk is not joined to the corosync group and added some condition in publish_event_to_corosync code to send corosync messages only when joined. When a node is not joined its corosync daemon can't send messages: the cpg_mcast_joined function append new messages to the FIFO buffer until it's full and then it blocks indefinitely. In this scenario if the stasis_message_cb callback, registered by res_corosync to handle stasis messages, try to send a corosync messages, the thread of the stasis thread-pool will be blocked until the node join the corosync cluster. ASTERISK-28888 Reported by: Università di Bologna - CESIA VoIP Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2	6 years ago
Moises Silva	635885ec33	res_http_websocket: Add payload masking to the websocket client ... ASTERISK-28949 Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca	6 years ago
Joshua C. Colp	1a40a6d55a	res_sorcery_memory_cache: Disallow per-object expire with full backend. ... The AMI action and CLI command did not take into account the properties of full backend caching. This resulted in an expired object remaining removed until a full backend update occurred, instead of having the object updated when needed. This change makes it so that the AMI action and CLI command for object expire will now fail instead of putting the cache into an undesired state. If full backend caching is enabled then only operations which act on the entire cache are available. ASTERISK-28942 Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4	6 years ago
Walter Doekes	50ba662b6a	res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h" ... Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78	6 years ago
Kevin Harwell	58eaf41e28	pjproject: Upgrade bundled version to pjproject 2.10 ... This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81	6 years ago
George Joseph	dd983edafe	res_fax: Don't start a gateway if either channel is hung up ... When fax_gateway_framehook is called and a gateway hasn't already been started, the framehook gets the t38 state for both the current channel and the peer. That call trickles down to the channel driver which determines the state. If either channel is hung up (or in the process of being hung up), the channel driver's tech_pvt is going to be NULL which, in the case of chan_pjsip, will cause a segfault. * Added a hangup check for both the channel and peer channel before starting a fax gateway. * Added a check for NULL tech_pvt to chan_pjsip_queryoption so we don't attempt to reference a tech_pvt that's already gone. ASTERISK-28923 Reported by: Yury Kirsanov Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c	6 years ago
Kevin Harwell	654ec4b908	Compiler fixes for gcc 10 ... This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	6 years ago
Walter Doekes	1e5ea08e0c	pjsip: Prevent invalid memory access when attempting to contact a non-sip URI ... You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri, without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S). ASTERISK-28936 Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a	6 years ago
Joshua C. Colp	f30a61022a	res_fax: Don't consume frames given to fax gateway on write. ... In a particular fax gateway scenario whereby it would have to translate using the read translation path on a channel the frame being translated would be consumed. When the frame is in the write path it is not permitted to free the frame as the caller expects it to continue to exist. This change makes it so that the frame is only consumed on the read path where it is acceptable to free it. ASTERISK-28900 Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d	6 years ago
Pirmin Walthert	84da8622f6	res_pjsip_logger: use the correct pointer when logging tx_messages to pcap ... When writing tx messages to pcap files, Asterisk is using the wrong pointer resulting in lots of wasted space. This patch fixes it to use the correct pointer. ASTERISK-28932 #close Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23	6 years ago
Joshua C. Colp	e7f7c8c9de	res_rtp_asterisk: Re-order RTP destruction. ... The destructor for RTP deallocated transport resources before terminating the ICE support. This could result in a crash as the thread handling ICE would access already freed parts of the RTP data. This change re-orders the destruction so that ICE is stopped before destroying things. ASTERISK-28885 Change-Id: Ie71add549f12b6cdea7e9dbf976d1bd1d2fc0bdc	6 years ago
Pirmin Walthert	81b86702fb	res_pjsip_logger.c: correct the return value checks when writing to pcap ... files fwrite() does return the number of elements written and not the number of bytes. However asterisk is currently comparing the return value to the size of the written element what means that asterisk logs five WARNING messages on every packet written to the pcap file. This patch changes the code to check for the correct value, which will always be 1. ASTERISK-28921 #close Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2	6 years ago
Joshua C. Colp	c8cabb08aa	res_pjsip: Use correct pool for storing the contact_user value. ... When replacing the user portion of the Contact URI the code was using the ephemeral pool instead of the tdata pool. This could cause the Contact user value to become invalid after a period of time. The code will now use the tdata pool which persists for the lifetime of the message instead. ASTERISK-28794 Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5	6 years ago
Pirmin Walthert	4b7293f44e	res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header ... While asterisk is filtering out the x-ast-orig-host parameter from the contact on response messages, it is not filtering it out from the request URI and the to header on SIP requests (for example INVITE). ASTERISK-28884 #close Change-Id: Id032b33098a1befea9b243ca994184baecccc59e	6 years ago
Joshua C. Colp	0d86f49f7f	res_sorcery_config: Always reload configuration on errors. ... When a configuration file in Asterisk is loaded information about it is stored such that on a reload it is not reloaded if nothing has changed. This can be problematic when an error exists in a configuration file in PJSIP since the error will be output at start and not subsequently on reload if the file is unchanged. This change makes it so that if an error is encountered when res_sorcery_config is loading a configuration file a reload will always read in the configuration file, allowing the error to be seen easier. Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed	6 years ago
Joshua C. Colp	4fba0f678b	res_pjsip_logger: Expand functionality to improve logging. ... The PJSIP packet logger now has the following CLI commands: pjsip set logger pcap <filename> When used this will create a pcap file containing the incoming and outgoing SIP packets, in unencrypted form. pjsip set logger verbose <on / off> This allows you to toggle logging to verbose on and off. pjsip set logger host <IP/subnet mask> add This allows you to add an additional IP address or subnet mask to logging, allowing you to log multiple instead of just a single IP address or all traffic. The normal "pjsip set logger host" CLI command has also been expanded to allow subnet masks as well. ASTERISK-28895 Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e	6 years ago
sungtae kim	5de3f22ee0	res_rtp_asterisk.c: Fixed memory leak ... Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	6 years ago
Roger James	3df1b65dd0	res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered. ... Changed source and destination address fields in struct pjsip_history_entry so that they are long enough to hold an IPv6 address. ASTERISK-28854 Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e	6 years ago
Guido Falsi	5c74f22977	res_rtp_asterisk: Protect access to nochecksums with #ifdef ... Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. ASTERISK-28852 #close Change-Id: I381718893b80599ab8635f2b594a10c1000d595e	6 years ago
sungtae kim	f2d9d31ef5	res_ari_channels: Fixed endpoint 80 characters limit ... Fixed it to copy the entire string from the requested endpoint body except tech-prefix. ASTERISK-28847 Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25	6 years ago
DanielYK	e4508fcc6e	res_pjsip: Fixed format of IPv6 addresses for external media addresses ... ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	6 years ago
Alexander Traud	f91dc188dd	curl: Add build-time dependency. ... ASTERISK-28838 Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52	6 years ago
Alexander Traud	bb5c0f30c4	res_pjsip_refer: Add build-time dependency. ... ASTERISK-28838 Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25	6 years ago
Alexander Traud	59f2ad2019	res_pjsip: Add build-time dependency. ... ASTERISK-28838 Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074	6 years ago
bernard merindol	43dfd0d300	res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0 ... When the first DTMF receive in RF2833 codec have TimeStamp at 0 is not processed. ASTERISK-28812 Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504	6 years ago
Jaco Kroon	ce1213e72f	res_rtp_asterisk: iterate all local addresses looking to populate ICE. ... By using pjproject to give us a list of candidates, and then filtering, if the host has >32 addresses configured, then it is possible that we end up filtering out all 32 of those, and ending up with no candidates at all. Instead, get getifaddrs (which pjsip is using underlying anyway) to retrieve all local addresses, and iterate those, adding the first 32 addresses not excluded by the ICE ACL. In our setup at any point in time We've got between 6 and 328 addresses on any given system. The lower limit is the lower limit but the upper limit is growing on a near daily basis currently. Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Jaco Kroon	a142a1455c	res_pjsip: document legal dtls_verify endpoint options. ... Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b	6 years ago
Alexander Traud	92efcc9047	res_rtp_asterisk: Build without PJProject. ... Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	6 years ago
Sean Bright	b92bd5783b	Revert "res_config_odbc: Preserve empty strings returned by the database" ... This reverts commit a3a2fbaec6. Reason for revert: There is a lot of code that relies on the broken behavior that this fixes. Change-Id: I410c395a0168acbdaf89e616e3cb5e1312d190cb	6 years ago
Joshua C. Colp	69805a772e	res_pjsip: Don't set endpoint to unavailable in all cases. ... When an AOR is modified endpoints are updated that reference the AOR so they can start receiving updates and reflect the correct state. If this is the case then we shouldn't change the endpoint to be offline if it does not reference the AOR but instead only when the endpoint is completely updated for all its AORs. ASTERISK-28056 patches: pjsip_options-aor.diff submitted by jhord (license 6978) Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164	6 years ago
Torrey Searle	ad7be12f0d	res_pjsip_session: implement processing of Content-Disposition ... RFC5621 requires any content type with a Content-Disposition with handling=required to be rejected with a 415 response ASTERISK-28782 #close Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4	6 years ago
Sungtae Kim	0daf62f501	res_pjsip_session: Fixed wrong session termination ... When the Asterisk receives 200 OK with invalid SDP, the Asterisk/PJPROJECT terminating the session. But if the channel was in the Bridge, Asterisk tries send the Re-Invite before terminating the session. And when the Asterisk sending the Re-Invite, it doesn't check the SDP is NULL or not. This crashes the Asterisk. Fixed it to close the session correctly if the UAS sends the 200 OK with wrong SDP. ASTERISK-28743 Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da	6 years ago
Michael Neuhauser	0bd1420064	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active ... Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	6 years ago
Jaco Kroon	f4a8c71dc2	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. ... A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Torrey Searle	85786d75c7	res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use ... bridge_p2p_rtp_write will forward rtp to the bridged rtp instance without modifying the ssrc. However, it is not updating the SSRC in the bridged rtp. Thus, when SSRC packets are generated, they have the correct SSRC for the sender. ASTERISK-28773 #close Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478	6 years ago
Joshua C. Colp	793f312199	res_rtp_asterisk: Fix unused warning for "ice". ... Change-Id: I31e022f722509214cd600c428939c91ace0c59fd	6 years ago
George Joseph	348819093d	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated" into 13	6 years ago
Rodrigo Ramírez Norambuena	b999447494	res_rtp_asterisk: Add 'rtp show settings' cli command ... This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	6 years ago
Torrey Searle	04f2fad021	res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated ... If ICE support is enabled but not negotiated, the rtp->ice structure is not being destroyed. This leads to Asterisk waiting for ICE to complete instead of immediately starting the DTLS handshake, resulting in the call leg having no RTP. ASTERISK-28769 #close Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27	6 years ago
lvl	4b242d4280	res_pjsip_refer: ensure refer progress is still sent after Proceeding() ... ASTERISK-28766 #close Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1	6 years ago
Joshua Colp	994853658b	Merge "message & stasis/messaging: make text message variables work in ARI" into 13	6 years ago
Kevin Harwell	c5ababbd2b	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions" into 13	6 years ago
Kevin Harwell	0fb06121cb	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Torrey Searle	ed670ca0c6	res/res_pjsip_sdp_rtp: Fix MOH transitions ... Update the state of remote_hold immediately on receipt of remote SDP so that the information is available when building the SDP answer ASTERISK-28754 #close Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f	6 years ago
Kevin Harwell	ff87627140	Merge "pjsip: Update ACLs on named ACL changes." into 13	6 years ago
George Joseph	66f2c78513	Merge "RTP/ICE: Send on first valid pair." into 13	6 years ago