5030 Commits (16.11)

Author	SHA1	Message	Date
sungtae kim	95a0b7c9ff	res_ari: Fix create channel request channelId parameter parsing ... If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. Fixed it to parse the channelId, other_channel_id parameter correclty. ASTERISK-28948 Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157 (cherry picked from commit 0b133a32ec)	6 years ago
sungtae kim	f5a6e5d65c	res_ari: Fix create request body parameter parsing. ... If parameters were passed in the body as JSON to the create route they were not being parsed before checking to ensure that required fields were set. This change moves the parsing so it occurs before checking. ASTERISK-28940 Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf	6 years ago
Pirmin Walthert	04ae0a5ff8	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
Pirmin Walthert	cbf5d12576	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	b37d7b5d0a	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	81929d478c	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	dcad322562	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
Alexander Traud	c00b032bbf	res_srtp: Set all possible flags while selecting the Crypto Suite. ... The flags of a previous selection could have been set within the object 'srtp', for example, when the previous selection returned failure after setting just 'some' flags. Now, not to clutter the code, all possible flags are cleared first, and then the selected flags are set as before. ASTERISK-28903 Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee	6 years ago
Joshua C. Colp	229cb16d52	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
Nicholas John Koch	7949343efb	res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings ... A warning was triggered that there may be a problem regarding file extension (which is correct and should not be set anyway). The warning also appeared if there was dot within the path itself. E.g. [sales-queue-hold] mode=playlist entry=/var/www/domain.tld/moh/funky_music The music played correctly but you get a warning message. Now there will be a check if the position of a potential dot character is after the last position of a slash character. This dot charachter will be treated as a extension naming. Dots within the path then ignored. ASTERISK-28892 Reported-By: Nicholas John Koch Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e	6 years ago
sungtae kim	862b9fcce8	res_rtp_asterisk.c: Fixed memory leak ... Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	6 years ago
Joshua C. Colp	ed75fd14a3	ari: Allow variables to be set on channel create. ... This change adds the same variable functionality that is available for originating a channel to the create call. Now when creating a channel you can specify dialplan variables to set instead of having to do another API call. ASTERISK-28896 Change-Id: If13997ba818136d7c070585504fc4164378aa992	6 years ago
Roger James	110d563452	pjsip_resolver.c: Ensure AAAA dns requests are made. ... 1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups when an IPV6 transport type has been requested. 2. Rename all occurrences of pjsip_transport_get_type_name to pjsip_transport_get_type_desc. This ensures that the log/debug info shows whether the transport is IPv6 or IPv4. 3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport types. This results in invalid values. Use a bitwise or instead. ASTERISK-26780 Patches: pjsip_resolver.c uploaded by Peter Sokolov (License #7070) Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58	6 years ago
Roger James	1249f911f3	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	858d3216dd	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
Joshua C. Colp	2128eb1f47	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
sungtae kim	64e0ce67fc	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
Joshua C. Colp	f3ac42b9bd	fax: Fix crashes in PJSIP re-negotiation scenarios. ... This change fixes a few re-negotiation issues uncovered with fax. 1. The fax support uses its own mechanism for re-negotiation by conveying T.38 information in its own frames. The new support for re-negotiating when adding/removing/changing streams was also being triggered for this causing multiple re-INVITEs. The new support will no longer trigger when transitioning between fax. 2. In off-nominal re-negotiation cases it was possible for some state information to be left over and used by the next re-negotiation. This is now cleared. ASTERISK-28811 ASTERISK-28839 Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2	6 years ago
DanielYK	c36f34aa90	res_pjsip: Fixed format of IPv6 addresses for external media addresses ... ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	6 years ago
Alexander Traud	63f57e044d	res_pjsip: Sync load- and build-time deps. ... MODULEINFO is checked while buidling/linking the module. AST_MODULE_INFO is checked while loading/running the module. ASTERISK-28838 Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042	6 years ago
Alexander Traud	905f8ddc3e	res_pjsip_refer: Add build-time dependency. ... ASTERISK-28838 Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25	6 years ago
Alexander Traud	ea102bd1a7	curl: Add build-time dependency. ... ASTERISK-28838 Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52	6 years ago
Alexander Traud	5c5f7223ec	res_pjsip: Add build-time dependency. ... ASTERISK-28838 Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074	6 years ago
Pirmin Walthert	a7c4579979	res_rtp_asterisk: Resolve loop when receive buffer is flushed ... When the receive buffer was flushed by a received packet while it already contained a packet with the same sequence number, Asterisk never left the while loop which tried to order the packets. This change makes it so if the packet is in the receive buffer it is retrieved and freed allowing the buffer to empty. ASTERISK-28827 Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3	6 years ago
Pirmin Walthert	9a928cae86	res_rtp_asterisk: Free payload when error on insertion to data buffer ... When the ast_data_buffer_put rejects to add a packet, for example because the buffer already contains a packet with the same sequence number, the payload will never be freed, resulting in a memory leak. The data buffer will now return an error if this situation occurs allowing the caller to free the payload. The res_rtp_asterisk module has also been updated to do this. ASTERISK-28826 Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1	6 years ago
bernard merindol	4af84c5675	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	7a00db673a	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	c9e1b4d142	res_pjsip: document legal dtls_verify endpoint options. ... Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b	6 years ago
Alexander Traud	2b2fe947ce	res_rtp_asterisk: Build without PJProject. ... Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	6 years ago
traud	fdb6370759	res_rtp_asterisk: Avoid absolute value on unsigned subtraction. ... ASTERISK-28809 Change-Id: I269731715347c8e5ef7db1b6ffd3f8d15fc04be4	6 years ago
Sean Bright	bfc9337ab4	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	06aa51d14e	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	bd091949cc	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
Joshua C. Colp	1c5129bca4	res_pjsip_session: Apply intention behind requested formats. ... When an outgoing channel is created a list of formats may optionally be provided which is used as a request that the formats be used if possible. If an endpoint is not configured for any of the formats we ignore this request and use what is configured. This has the side effect of also including other stream types (such as video) that were not present in the requested formats. This change makes it so that the intention of the request is preserved - that is if only an audio format is requested then even if there is no joint audio format between the request and the configuration we will still only place an audio stream in the outgoing call. ASTERISK-28787 Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f	6 years ago
Joshua C. Colp	7771a198d5	res_rtp_asterisk: Ensure sufficient space for worst case NACK. ... ASTERISK-28790 Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99	6 years ago
Joshua C. Colp	0b92aa4c49	res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream. ... When examining a stream to determine hold/unhold information we only care about the default audio stream. Other streams aren't used for hold/unhold. ASTERISK-28784 Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4	6 years ago
Sungtae Kim	ac3a81992a	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
Joshua C. Colp	c86af00ce1	res_pjsip_session: Don't restrict non-audio default streams to sendrecv. ... The state of the default audio stream is used for hold/unhold so we restrict it to sendrecv as the core does not handle when it changes as a result of hold/unhold. This restriction does not apply to other media types though so we now only restrict it to audio. This allows the other default streams to store their state at all values, and not just sendrecv and removed. ASTERISK-28783 Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625	6 years ago
Michael Neuhauser	b2e0c6cacc	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	7 years ago
Jaco Kroon	351b2be00a	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>	7 years ago
Torrey Searle	f2ba1919e6	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	7 years ago
George Joseph	90e5dc2959	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated" into 16	7 years ago
Torrey Searle	1efd90b72b	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	7 years ago
Rodrigo Ramírez Norambuena	7dcabaef08	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	7 years ago
Kevin Harwell	44a8615e22	Merge "res_pjsip_refer: ensure refer progress is still sent after Proceeding()" into 16	7 years ago
Joshua Colp	1affc17d95	Merge "message & stasis/messaging: make text message variables work in ARI" into 16	7 years ago
Kevin Harwell	47658a0a2b	Merge "res_rtp_asterisk: Improve video performance in certain networks." into 16	7 years ago
lvl	f529d56f76	res_pjsip_refer: ensure refer progress is still sent after Proceeding() ... ASTERISK-28766 #close Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1	7 years ago
Joshua C. Colp	f295af447d	res_rtp_asterisk: Improve video performance in certain networks. ... The receive buffer will now grow if we end up flushing the receive queue after not receiving the expected packet in time. This is done in hopes that if this is encountered again the extra buffer size will allow more time to pass and any missing packets to be received. The send buffer will now grow if we are asked for packets and can't find them. This is done in hopes that the packets are from the past and have simply been expired. If so then in the future with the extra buffer space the packets should be available. Sequence number cycling has been handled so that the correct sequence number is calculated and used in various places, including for sorting packets and for determining if a packet is old or not. NACK sending is now more aggressive. If a substantial number of missing sequence numbers are added a NACK will be sent immediately. Afterwards once the receive buffer reaches 25% a single NACK is sent. If the buffer continues to grow and reaches 50% or greater a NACK will be sent for each received future packet to aggressively ask the remote endpoint to retransmit. ASTERISK-28764 Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41	7 years ago
Kevin Harwell	9617e530b6	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions" into 16	7 years ago