32942 Commits (033c2a22839698be0c0f54419169c76c4eac6046)

 

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Author	SHA1	Message	Date
traud	da9554d925	chan_sip: TCP/TLS client without server. ... It is possible to configure a TCP/TLS client without having a TCP/TLS server. In that case, no error or warning was printed but the headers Contact and Via in SIP REGISTER were "(null)". ASTERISK-28798 Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2	6 years ago
Alexander Traud	52ecbbd014	_pjsua: Build even with Clang. ... Change-Id: Iebf7687613aa0295ea3c82256460b337f1595be2	6 years ago
Alexander Traud	ee1c7f465b	res_rtp_asterisk: Build without PJProject. ... Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	6 years ago
Kevin Harwell	fa3c8f94e0	chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet ... If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a digit begin before media, or rtp has been setup then it's possible the outgoing channel will hear a constant DTMF tone upon answering. This happens because when there is no media, or rtp chan_pjsip notifies the core to initiate inband DTMF. However, upon digit end if media, and rtp become available then chan_pjsip does not notify the core to stop inband DTMF. Thus the tone continues playing. This patch makes it so chan_pjsip only notifies the core to start inband DTMF in only the required cases. Now if there is no media, or rtp availabe upon digit begin chan_pjsip does nothing, but tells the core it handled it. ASTERISK-28817 #close Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5	6 years ago
Alexander Traud	7febd22304	bridge_softmix_binaural: Show state in menuselect. ... ASTERISK-28819 Change-Id: Iba7ee7bc7936d7a156171c8fc0f1670e864e7600	6 years ago
traud	7cdb493a1e	BuildSystem: Remove doc/tex and doc/pdf leftovers. ... Furthermore, the nowhere used compress is removed. ASTERISK-28816 Change-Id: I77daab80cfabb56d51c3ea6b1d14bd9b9fbc577c	6 years ago
Alexander Traud	7a04947abd	BuildSystem: Allow space in path. ... ASTERISK-28818 Change-Id: Ib7f246896457d9e3b14d7f5199136d6545ce0b6f	6 years ago
traud	1ef1b1b0c2	res_rtp_asterisk: Avoid absolute value on unsigned subtraction. ... ASTERISK-28809 Change-Id: I269731715347c8e5ef7db1b6ffd3f8d15fc04be4	6 years ago
Sebastien Duthil	d40e343710	func_channel: allow reading 4 fields from dialplan ... The following fields return an error when read from dialplan: - exten - context - userfield - channame ASTERISK-28796 #close Change-Id: Ieacaac629490f8710fdacc9de80ed5916c5f6ee2	6 years ago
traud	b38f664250	chan_unistim: Avoid tautological warnings with clang. ... ASTERISK-28803 Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af	6 years ago
traud	bb28ed0d1b	test_stasis: Avoid always true warning with clang. ... ASTERISK-28808 Change-Id: I5e76831373532d7b8065d024e66cd1fb75dedd80	6 years ago
Sean Bright	60925c68e8	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
Jaco Kroon	c5f3836bcc	main/backtrace: binutils-2.34 fix. ... My tester missed this one previously, have confirmed a positive build this time round. Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf	6 years ago
Joshua C. Colp	d845464c76	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
George Joseph	7ba6d43083	test_res_pjsip_session_caps: Create unit test ... This unit test runs through combinations of... * Local codecs * Remote Codecs * Codec Preference * Incoming/Outgoing A few new APIs were created to make it easier to test the functionality but didn't result in any actual functional change. ASTERISK_28777 Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164	6 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Ben Ford	57a457c26c	res_stir_shaken: Implemented signing of JSON payload. ... This change provides functions that take in a JSON payload, verify that the contents contain all the mandatory fields and required values (if any), and signs the payload with the private key. Four fields are added to the payload: x5u, attest, iat, and origid. As of now, these are just placeholder values that will be set to actual values once the logic is implemented for what to do when an actual payload is received, but the functions to add these values have all been implemented and are ready to use. Upon successful signing and the addition of those four values, a ast_stir_shaken_payload is returned, containing other useful information such as the algorithm and signature. Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393	6 years ago
Kevin Harwell	3c345ec56d	channel: write to a stream on multi-frame writes ... If a frame handling routine returns a list of frames (vs. a single frame) those frames are never passed to a tech's write_stream handler even if one is available. For instance, if a codec translation occurred and that codec returned multiple frames then those particular frames were always only sent to the tech's "write" handler. If that tech (pjsip for example) was stream capable then those frames were essentially ignored. Thus resulting in bad audio. This patch makes it so the "write_stream" handler is appropriately called for all cases, and for all frames if available. ASTERISK-28795 #close Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d	6 years ago
traud	fc07eeaba1	test_utils: Avoid incorrect error message on load. ... In case of no OpenSSL headers, the module was built but did not load. ASTERISK-28789 Change-Id: Ie007e84296bcf2bd4237f19d68ba5f932b84cd02	6 years ago
traud	cd8cbf7384	func_aes: Avoid incorrect error message on load. ... In case of no OpenSSL headers, the module func_aes was built but did not load. ASTERISK-28788 Change-Id: I0b99b8468cbeb3b0eab23069cbd64062ef885ffc	6 years ago
sungtae kim	dbddb6725d	dial.c: Removed dial string 80 character limitation ... The dial application had 80 characters of destination length limitation. But this limitation causes unexpected dial string cut if the dial string is long. Removed unnecessary limited buffer to support longer dial destination. ASTERISK-27946 Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330	6 years ago
Torrey Searle	e12244153a	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
Jaco Kroon	d32e559e8a	acl: implement a centralized ACL output mechanism for HAs and ACLs. ... named_acl.c (which is really a named_ha) now uses ast_ha_output. I've also updated main/manager.c to output the actual ACL on "manager show user <username>" if one is set. If this works then we can add similar to other modules as required. Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f	6 years ago
Joshua C. Colp	1b6c58896f	chan_sip: Send 403 when ACL fails. ... Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	6 years ago
Joshua C. Colp	3ed80fc57b	CHANGES: Change md file extension to txt. ... Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7 (cherry picked from commit c9cd681261)	6 years ago
Joshua C. Colp	21e9051461	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	96e8d411e1	res_rtp_asterisk: Ensure sufficient space for worst case NACK. ... ASTERISK-28790 Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99	6 years ago
Kevin Harwell	26713dc88b	ast_coredumper: add Asterisk information dump ... This patch makes it so ast_coredumper now outputs the following information to a *-info.txt file when processing a core file: asterisk version and "built by" string BUILD_OPTS system start, and last reloaded date/time taskprocessor list equivalent of "bridge show all" equivalent of "core show channels verbose" Also a slight modification was made when trying to obtain the pid(s) of a running Asterisk. If it fails to retrieve any it now reports an error. Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b	6 years ago
Jaco Kroon	6f731f153b	netsock2: compile fixes. ... This fixes ast_addressfamily_to_sockaddrsize to reference the provided argument, and ast_sockaddr_from_sockaddr to not use the name of a structure as argument. Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7	6 years ago
Ben Ford	211bb8a79c	res_stir_shaken: Initial commit and reading private key. ... This commit sets up some of the initial framework for the module and adds a way to read the private key from the specified file, which will then be appended to the certificate object. This works fine for now, but eventually some other structure will likely need to be used to store all this information. Similarly, the caller_id_number is specified on the certificate config object, but in the end we will want that information to be tied to the certificate itself and read it from there. A method has been added that will retrieve the private key associated with the caller_id_number passed in. Tab completion for certificates and stores has also been added. Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f	6 years ago
Jaco Kroon	4f92dcd66b	dahdiras: Only set plugin dahdi.so to pppd if we're running as root. ... Users of this should set plugin dahdi.so in their options file. ASTERISK-16676 Change-Id: I6d01ad0a10e9fea477876d0941c3f38aac357e91	6 years ago
Jaco Kroon	40e93b0240	dundi: fix NULL dereference. ... If a negative (error) return is received from dundi_lookup_internal, this is not handled correctly when assigning the result to the buffer. As such, use a signed integer in the assignment and do a proper comparison. ASTERISK-21205 Change-Id: I5214ebb6491e2bd14f90c7d3ce229da86888f739	6 years ago
Joshua C. Colp	34750d2068	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	8147f43756	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
Jaco Kroon	a699e016dd	build: enable building with uClibc ... This patch has been included in Gentoo distribution for at least since asterisk 1.8, but there are references in the logs going back as far as 1.0.0 - not sure if this is still required in any way, it does apply, and it doesn't (as far as we can determine) cause build failures. Change-Id: I46d8845e30200205e80580680bf060aa3012ba54	6 years ago
Jaco Kroon	f824cd6a13	build: search from newest to oldest for gmime. ... We (Gentoo distribution) reckon that in the case of multiple versions of gmime installed we should prefer the newest one. Change-Id: Idf7be613230232eb1d573d93c4a5a8297f4ecd2d	6 years ago
Joshua C. Colp	9620ecbf80	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	5562fb2ea0	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	82c3939c38	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
Jaco Kroon	2ad64e97c0	Update main/backtrace.c to deal with changes in binutils 2.34. ... binutils 2.34 merged this commit: https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\ h=fd3619828e94a24a92cddec42cbc0ab33352eeb4 Which effectively does things like: -#define bfd_section_size(bfd, ptr) ((ptr)->size) -#define bfd_get_section_size(ptr) ((ptr)->size) +#define bfd_section_size(sec) ((sec)->size) So in order to remain backwards compatible we need to detect this API change, and adjust accordingly. The simplest is to notice that the bfd_get_section_size and bfd_get_section_vma MACROs are no longer defined, and define then onto the new API. The alternative is to litter the code with a number of #ifdef #else #endif splatters right through the code. Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f	6 years ago
Sean Bright	c4e0983742	func_odbc.conf.sample: Clarify sample documentation ... ASTERISK-20325 #close Change-Id: I06cb9b467b0fd06c8af2a5aee049f872c09cc4b6	6 years ago
Sean Bright	49cf84578e	chan_vpb: Fix 'catching polymorphic type ... by value' error ... Fixes the following compile error: chan_vpb.cc:2688:26: error: catching polymorphic type ‘class std::exception’ by value Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649	6 years ago
Sean Bright	d68f940f6e	dns_txt: Add TXT record parsing support ... Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5	6 years ago
Joshua C. Colp	98d10d0a16	audiohook: Don't allow audiohooks to attach to hung up channels. ... Given a scenario where MixMonitor was initiated over AMI it was possible for the channel and MixMonitor thread to remain alive past hang up of the channel. This scenario required the AMI initiated MixMonitor to retrieve the channel, a hangup to occur on the channel in another thread, and then for MixMonitor to actually start. If this occurred the MixMonitor thread would remain alive indefinitely and the channel reference would remain. This change ensures that audiohooks are never able to be attached to channels that have been hung up. An additional fix has also been done in app_mixmonitor to properly release the channel reference if this occurs. ASTERISK-28780 Change-Id: I8044c06daa06f0f16607788c596f55623be26f58	6 years ago
George Joseph	00a7e4b51d	CI: Create generic jenkinsfile ... This is a generic jenkinsfile to build Asterisk and optionally perform one or more of the following: * Publish the API docs to the wiki * Run the Unit tests * Run Testsuite Tests This job can be triggered manually from Jenkins or be triggered automatically on a schedule based on a cron string. Change-Id: Id9d22a778a1916b666e0e700af2b9f1bacda0852	6 years ago
Torrey Searle	a1dba820cf	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
George Joseph	0cde95ec89	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated"	6 years ago
George Joseph	e828a596c5	Merge "chan_pjsip: Check audio frame when remote SSRC changes."	6 years ago
George Joseph	99efe1f868	Merge "codec negotiation: add incoming_call_offer_prefs option"	6 years ago
George Joseph	7b09d9c5aa	Merge "enum.c: Make ast_get_txt() actually do something."	6 years ago