asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	23211669a7	BuildSystem: Search for Python/C API when possibly needed only. The Python/C API is used only if the Test Framework was enabled in Asterisk 'make menuselect'. The Test Framework is available only if the Developer Mode was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API is used only if the PJProject was found and not disabled in Asterisk; the user did not go for './configure --without-pjproject'. Furthermore, because version 2 of that Python/C API is required (currently) and because some platforms do not offer a generic version 2, the script searches for 2.7 explicitly as well. To avoid version mismatch between the Python/C API and the Python environment, the script searches for the latter in the same versions, in the same the order as well. Because this Python/C API is just for (some) Asterisk contributors, the script also goes for the Python 3 environment as a last resort for all other Asterisk users. This allows 'make full' even on minimal installations of Ubuntu 18.04 LTS and newer. Because the Python/C API is Asterisk contributor specific, the Python packages are removed from the script './contrib/scripts/install_prereq' as this script is intended for Asterisk users. Asterisk contributors have to install much more packages in any case, like: sudo apt install autoconf automake git git-review python2.7-dev ASTERISK-28824 ASTERISK-27717 Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876	5 years ago
traud	2bcc3df533	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	5 years ago
Alexander Traud	a0d5f7ccff	_pjsua: Build even with Clang. Change-Id: Iebf7687613aa0295ea3c82256460b337f1595be2	5 years ago
Alexander Traud	92efcc9047	res_rtp_asterisk: Build without PJProject. Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec	5 years ago
Kevin Harwell	b2e7fc6972	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	5 years ago
traud	9b259c6514	BuildSystem: Remove doc/tex and doc/pdf leftovers. Furthermore, the nowhere used compress is removed. ASTERISK-28816 Change-Id: I77daab80cfabb56d51c3ea6b1d14bd9b9fbc577c	5 years ago
Alexander Traud	785225c5a0	BuildSystem: Allow space in path. ASTERISK-28818 Change-Id: Ib7f246896457d9e3b14d7f5199136d6545ce0b6f	5 years ago
traud	9082f90217	stasis: Avoid always true warnings with clang. ASTERISK-28801 Change-Id: I63ba125226b9fe8a018bd28825c877603eb8f398	5 years ago
Sebastien Duthil	bb974877fa	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	5 years ago
traud	05440f3ef6	chan_unistim: Avoid tautological warnings with clang. ASTERISK-28803 Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af	5 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	5 years ago
traud	55dacdce56	test_stasis: Avoid always true warning with clang. ASTERISK-28808 Change-Id: I5e76831373532d7b8065d024e66cd1fb75dedd80	5 years ago
Jaco Kroon	42989a285b	main/backtrace: binutils-2.34 fix. My tester missed this one previously, have confirmed a positive build this time round. Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf	5 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	5 years ago
traud	b338da8a19	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	5 years ago
traud	151a27fa81	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	5 years ago
sungtae kim	59e193b295	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	5 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	5 years ago
Jaco Kroon	5ddac396f0	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	5 years ago
Joshua C. Colp	305227dd38	chan_sip: Send 403 when ACL fails. Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 years ago
Kevin Harwell	e96e5cc3d4	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	5 years ago
Jaco Kroon	6b2aea9ce2	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	5 years ago
Jaco Kroon	96efd52460	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	5 years ago
Jaco Kroon	9065d95254	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	5 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	5 years ago
Jaco Kroon	18b058d65b	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	5 years ago
Jaco Kroon	5d5f05ab61	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	5 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	5 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>	5 years ago
Jaco Kroon	345e072a6d	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	5 years ago
Sean Bright	8b7071994e	func_odbc.conf.sample: Clarify sample documentation ASTERISK-20325 #close Change-Id: I06cb9b467b0fd06c8af2a5aee049f872c09cc4b6	5 years ago
Sean Bright	3e4e016a6c	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	5 years ago
Joshua C. Colp	6636fff2cf	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	5 years ago
George Joseph	4990a6e2fb	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	5 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	5 years ago
Joshua C. Colp	793f312199	res_rtp_asterisk: Fix unused warning for "ice". Change-Id: I31e022f722509214cd600c428939c91ace0c59fd	5 years ago
George Joseph	348819093d	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated" into 13	5 years ago
George Joseph	6dc064b84f	Merge "chan_pjsip: Check audio frame when remote SSRC changes." into 13	5 years ago
George Joseph	8f33d14d00	Merge "enum.c: Make ast_get_txt() actually do something." into 13	5 years ago
George Joseph	8943624bf7	Merge "enum.c: Add support for regular expression flag in NAPTR record" into 13	5 years ago
Joshua Colp	d6248d5f5c	Merge "res_rtp_asterisk: Add 'rtp show settings' cli command" into 13	5 years ago
Paulo Vicentini	f0b6d24704	chan_pjsip: Check audio frame when remote SSRC changes. If the SSRC of a received RTP packet differed from the previous SSRC an SSRC change control frame would be queued ahead of the media frame. In the case of audio this would result in the format of the audio frame not being checked, and if it differed or was not allowed then it could cause the call to drop due to failure to set up a translation path. The chan_pjsip module will now no longer assume the first frame will be the audio frame and instead goes through the complete list to find it. ASTERISK-28759 Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec	5 years ago
Sean Bright	58b8c24c40	enum.c: Add support for regular expression flag in NAPTR record A regular expression in a NAPTR response record can have a trailing 'i' flag to indicate that the expression should be evaluated in a case-insensitive way. We were not checking for that flag which caused the record parsing to fail on otherwise valid input. Although this change will initially go into Asterisk 13, 16, and 17, it is my intention to replace the majority of this code in 16 and up - including this fix - by changing enum.c to consume the new DNS API which duplicates most of this logic already. Asterisk 13 doesn't have the DNS API, so this fix will be as good as it gets. ASTERISK-26711 #close Reported by: Vitold Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0	5 years ago
Jared Smith	74718010d0	indications.conf.sample: Add indication tones for Indonesia These tones come from http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf ASTERISK-23407 Change-Id: I48e2285f1e5bb29b3335f762006f66c423d0fcb8	5 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	5 years ago
Asterisk Development Team	077cc24d66	Update CHANGES and UPGRADE.txt for 13.32.0	5 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	5 years ago
Sean Bright	0ee0134cc9	enum.c: Make ast_get_txt() actually do something. The ast_get_txt() API function (and by extension, the TXTCIDNAME dialplan function) were broken in `65b8381550` such that we would never actually make a DNS TXT query as described. This patch restores the documented behavior. ASTERISK-19460 #close Reported by: George Joseph Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376	5 years ago
lvl	4b242d4280	res_pjsip_refer: ensure refer progress is still sent after Proceeding() ASTERISK-28766 #close Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1	5 years ago
Joshua Colp	30f718a598	Merge "check_expr2: fix cross-compile/hardening issues" into 13	5 years ago

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