8631 Commits (17d5ef3a94e453050ca6397c53a9ce0754572212)

Author	SHA1	Message	Date
Alexander Traud	c29bf45863	chan_iax2: Fix for Doxygen. ... ASTERISK-29737 Change-Id: I282003cc553989fd5c19ceeac9e478fa4ee06cec	4 years ago
Josh Soref	636d3dc4be	channels: Spelling fixes ... Correct typos of the following word families: appease permanently overriding residue silliness extension channels globally reference japanese group coordinate registry information inconvenience attempts cadence payloads presence provisioning mimics behavior width natively syslabel not owning unquelch mostly constants interesting active unequipped brodmann commanding backlogged without bitstream firmware maintain exclusive practically structs appearance range retransmission indication provisional associating always whether cyrillic distinctive components reinitialized initialized capability switches occurring happened outbound ASTERISK-29714 Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339	4 years ago
Naveen Albert	6d242723bb	sig_analog: Fix truncated buffer copy ... Fixes compiler warning caused by a truncated copy of the ANI2 into a buffer of size 10. This could prevent the null terminator from being copied if the copy value exceeds the size of the buffer. This increases the buffer size to 101 to ensure there is no way for truncation to occur. ASTERISK-29702 #close Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0	4 years ago
Naveen Albert	530e89655e	chan_iax2: Allow both secret and outkey at dial time ... Historically, the dial syntax for IAX2 has held that an outkey (used only for RSA authenticated calls) and a secret (used only for plain text and MD5 authenticated calls, historically) were mutually exclusive, and thus the same position in the dial string was used for both values. Now that encryption is possible with RSA authentication, this poses a limitation, since encryption requires a secret and RSA authentication requires an outkey. Thus, the dial syntax is extended so that both a secret and an outkey can be specified. The new extended syntax is backwards compatible with the old syntax. However, a secret can now be specified after the outkey, or the outkey can be specified after the secret. This makes it possible to spawn an encrypted RSA authenticated call without a corresponding peer being predefined in iax.conf. ASTERISK-29707 #close Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e	4 years ago
Mike Bradeen	8031b6c9e6	various: Fix GCC 11 compilation issues. ... test_voicemail_api: Use empty char* for empty_msg_ids. chan_skinny: Fix size of calledParty to be maximum extension. menuselect: Change Makefile to stop deprecated warnings. Added comments test_linkedlist: 'bogus' variable was manually allocated from a macro and the test fails if this happens but the compiler couldn't 'see' this and returns a warning. memset to all 0's after allocation. chan_ooh323: Fixed various indentation issues that triggered misleading indentation warnings. ASTERISK-29682 Reported by: George Joseph Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe	4 years ago
Naveen Albert	8ad4cb901b	chan_iax2: Add encryption for RSA authentication ... Adds support for encryption to RSA-authenticated calls. Also prevents crashes if an RSA IAX2 call is initiated to a switch requiring encryption but no secret is provided. ASTERISK-20219 Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40	4 years ago
Naveen Albert	7feee9c65b	res_pjsip_caller_id: Add ANI2/OLI parsing ... Adds parsing of ANI II digits (Originating Line Information) to PJSIP, on par with what currently exists in chan_sip. ASTERISK-29472 Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847	4 years ago
Naveen Albert	38cc3160d3	chan_iax2: Add ANI2/OLI information element ... Adds an information element for ANI2 so that Originating Line Information can be transmitted over IAX2 channels. ASTERISK-29605 #close Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2	4 years ago
Sarah Autumn	b731b810b9	sig_analog: Changes to improve electromechanical signalling compatibility ... This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	4 years ago
Sean Bright	b02538de2e	mgcp: Remove dead debug code ... ASTERISK-20339 #close Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5	4 years ago
Joshua C. Colp	68adc2304f	policy: Add deprecation and removal versions to modules. ... app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. cdr_mysql was deprecated in 1.8, to be removed in 19. app_mysql was deprecated in 1.8, to be removed in 19. app_ices was deprecated in 16, to be removed in 19. app_macro was deprecated in 16, to be removed in 21. app_fax was deprecated in 16, to be removed in 19. app_url was deprecated in 16, to be removed in 19. app_image was deprecated in 16, to be removed in 19. app_nbscat was deprecated in 16, to be removed in 19. app_dahdiras was deprecated in 16, to be removed in 19. cdr_syslog was deprecated in 16, to be removed in 19. chan_oss was deprecated in 16, to be removed in 19. chan_phone was deprecated in 16, to be removed in 19. chan_sip was deprecated in 17, to be removed in 21. chan_nbs was deprecated in 16, to be removed in 19. chan_misdn was deprecated in 16, to be removed in 19. chan_vpb was deprecated in 16, to be removed in 19. res_config_sqlite was deprecated in 16, to be removed in 19. res_monitor was deprecated in 16, to be removed in 21. conf2ael was deprecated in 16, to be removed in 19. muted was deprecated in 16, to be removed in 19. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29554 ASTERISK-29555 ASTERISK-29557 ASTERISK-29558 ASTERISK-29559 ASTERISK-29560 ASTERISK-29561 ASTERISK-29562 ASTERISK-29563 ASTERISK-29564 ASTERISK-29565 ASTERISK-29566 ASTERISK-29567 ASTERISK-29568 ASTERISK-29569 ASTERISK-29570 ASTERISK-29571 ASTERISK-29572 ASTERISK-29573 ASTERISK-29574 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Kevin Harwell	dbfde090ea	AST-2021-008 - chan_iax2: remote crash on unsupported media format ... If chan_iax2 received a packet with an unsupported media format, for example vp9, then it would set the frame's format to NULL. This could then result in a crash later when an attempt was made to access the format. This patch makes it so chan_iax2 now ignores/drops frames received with unsupported media format types. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	4 years ago
Naveen Albert	11e61217a0	chan_sip: Expand hook flash recognition. ... Some ATAs send hook flash events as application/hook-flash, rather than a DTMF event. Now, we also recognize hook-flash as a flash event. ASTERISK-29370 Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725	4 years ago
Sean Bright	f557126998	chan_pjsip: Correct misleading trace message ... ASTERISK-29358 #close Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647	4 years ago
George Joseph	63c25d3821	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	4 years ago
Joshua C. Colp	e42417e910	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	4 years ago
Alexander Traud	10c0d9e711	chan_iax2: System Header strings is included via asterisk.h/compat.h. ... The system header strings was included mistakenly with commit 3de0204. That header is included via asterisk.h and there via the compat.h. Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a	4 years ago
Holger Hans Peter Freyther	4ceb07434c	pjsip: Generate progress (once) when receiving a 180 with a SDP ... ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	4 years ago
Alexander Traud	99666117be	chan_sip: Filter pass-through audio/video formats away, again. ... Instead of looking for pass-through formats in the list of transcodable formats (which is going to find nothing), go through the result which is going to be the jointcaps of the tech_pvt of the channel. Finally, only with that list, ast_format_cap_remove(.) is going to succeed. This restores the behaviour of Asterisk 1.8. However, it does not fix ASTERISK_29282 because that issue report is about chan_sip and PJSIP. Here, only chan_sip is fixed because PJSIP does not even call ast_rtp_instance_available_formats -> ast_translate_available_format. Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34	4 years ago
Alexander Traud	2bcec10db8	chan_sip: Allow [peer] without audio (text+video). ... Two previous commits, 620d9f4 and 6d980de, allow to set up a call without audio, again. That was introduced originally with commit f04d5fb but changed and broke over time. The original commit missed one scenario: A [peer] section in sip.conf, which does not allow audio at all. In that case, chan_sip rejected the call, although even when the requester offered no audio. Now, chan_sip does not check whether there is no audio format but checks whether there is no format in general. In other words, if there is at least one format to offer, the call succeeds. However, to prevent calls with no-audio, chan_sip still rejects calls when both call parties (caller = requester of the call *and* callee = [peer] section in sip.conf) included audio. In such a case, it is expected that the call should have audio. ASTERISK-29280 Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e	4 years ago
George Joseph	10c2e1749b	chan_iax2.c: Require secret and auth method if encryption is enabled ... If there's no secret specified for an iax2 peer and there's no secret specified in the dial string, Asterisk will crash if the auth method requested by the peer is MD5 or plaintext. You also couldn't specify a default auth method in the [general] section of iax.conf so if you don't have static peers defined and just use the dial string, Asterisk will still crash even if you have a secret specified in the dial string. * Added logic to iax2_call() and authenticate_reply() to print a warning and hanhup the call if encryption is requested and there's no secret or auth method. This prevents the crash. * Added the ability to specify a default "auth" in the [general] section of iax.conf. ASTERISK-29624 Reported by: N A Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254	4 years ago
Alexander Traud	4a341f288d	chan_sip: Set up calls without audio (text+video), again. ... The previous commit 6d980de fixed this issue in the core of Asterisk. With that, each channel technology can be used without audio theoretically. Practically, the channel-technology driver chan_sip turned out to have an invalid check preventing that. chan_sip tested whether there is at least one audio format. However, chan_sip has to test whether there is at least one format. More cannot be tested while requesting chan_sip because only the [general] capabilities but not the [peer] caps are known yet. And the [peer] caps might not be a subset or show any intersection with the [general] caps. This change here fixes this. The original commit f04d5fb, thirteen years ago, contained a software bug as it passed ANY audio capability to the channel-technology driver. Instead, it should have passed NO audio format. Therefore, this addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. Then, Asterisk 10 changed that from ANY to NO, but nobody reported since then. ASTERISK-29265 Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f	4 years ago
Dan Cropp	a5364ac69b	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Alexander Traud	7dc0cded86	chan_sip: SDP: Reject audio streams correctly. ... This completes the fix for ASTERISK_24543. Only when the call is an outgoing call, consult and append the configured format capabilities (p->caps). When all audio formats got rejected the negotiated format capabilities (p->jointcaps) contain no audio formats for incoming calls. This is required when there are other accepted media streams. ASTERISK-29258 Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9	4 years ago
Ben Ford	0520f8320c	chan_pjsip.c: Add parameters to frame in indicate. ... There are a couple of parameters (datalen and data) that do not get set in chan_pjsip_indicate which could cause an Invalid message to pop up for things such as fax. This patch adds them to the frame. Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8	4 years ago
Alexander Traud	34b0960310	chan_sip: SDP: Sidestep stream parsing when its media is disabled. ... Previously, chan_sip parsed all known media streams in an SDP offer like video (and text) even when videosupport=no (and textsupport=no). This wasted processor power. Furthermore, chan_sip accepted SDP offers, including no audio but just video (or text) streams although videosupport=no (or textsupport=no). Finally, chan_sip denied the whole offer instead of individual streams when they had encryption (SDES-sRTP) unexpectedly enabled. ASTERISK-29238 ASTERISK-29237 ASTERISK-29222 Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755	4 years ago
Ivan Poddubnyi	97afc9055f	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	4 years ago
Ivan Poddubnyi	17fa9c93d0	chan_pjsip: Stop queueing control frames twice on outgoing channels ... The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. This resulted in extra noise in logs (for example, "is making progress" and "is ringing" get logged twice by app_dial), as well as in noise in signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. This change splits the response handler into 2 functions: - one for updating HANGUPCAUSE, which is still called twice, - another that does the rest, which is called only once as before. ASTERISK-28016 Reported-by: Alex Hermann ASTERISK-28549 Reported-by: Gant Liu ASTERISK-28185 Reported-by: Julien Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940	4 years ago
Dan Cropp	e127a57761	chan_pjsip: Incorporate channel reference count into transfer_refer(). ... Add channel reference count for PJSIP REFER. The call could be terminated prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY occurred several minutes later, it would attempt to access a session which was no longer valid. Terminate event subscription if pjsip_xfer_initiate() or pjsip_xfer_send_request() fails in transfer_refer(). ASTERISK-29201 #close Reported-by: Dan Cropp Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435	4 years ago
Joshua C. Colp	ea744ca7c2	pjsip: Match lifetime of INVITE session to our session. ... In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	5 years ago
Alexander Traud	f667c5a781	chan_sip: Remove unused sip_socket->port. ... 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port vanished. However, the struct member itself and all seven set/uses remained as dead code. ASTERISK-28798 Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59	5 years ago
Alexander Traud	334661601a	Compiler fixes for GCC when printf %s is NULL ... ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	951ce0524d	chan_sip: On authentication, pick MD5 for sure. ... RFC 8760 added new digest-access-authentication schemes. Testing revealed that chan_sip does not pick MD5 if several schemes are offered by the User Agent Server (UAS). This change does not implement any of the new schemes like SHA-256. This change makes sure, MD5 is picked so UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can still be used. This should have worked since day one because SIP/2.0 already envisioned several schemes (see RFC 3261 and its augmented BNF for 'algorithm' which includes 'token' as third alternative; note: if 'algorithm' was not present, MD5 is still assumed even in RFC 7616). Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd	5 years ago
George Joseph	6abf6f345d	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	5 years ago
Kevin Harwell	17258e0cdf	chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution ... This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function is called on a channel prior to answering a warning is issued and the function returns unsuccessful. ASTERISK-28878 #close Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb	5 years ago
Dennis Buteyn	9c6d14b26b	chan_sip: Clear ToHost property on peer when changing to dynamic host ... The ToHost parameter was not cleared when a peer's host value was changed to dynamic. This causes invites to be sent to the original host. ASTERISK-29011 #close Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c	5 years ago
Kevin Harwell	c356187969	PJSIP_MEDIA_OFFER: override configuration on refresh ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh 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 for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	5 years ago
Kevin Harwell	2148505f09	chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ ... A patch made a reference to the PJSIP_SC_NULL enumeration value, which was added to pjproject 2.8 and above thus making it so Asterisk would fail to compile with prior versions of pjproject. This patch removes the reference, and instead initializes the value to '0'. ASTERISK-28886 #close Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7	5 years ago
Frederic LE FOLL	5415606846	chan_sip: chan_sip does not process 400 response to an INVITE. ... chan_sip handle_response() function, for a 400 response to an INVITE, calls handle_response_invite() and does not generate ACK. handle_response_invite() does not recognize 400 response and has no default response processing for unexpected responses, thus it does not generate ACK either. The ACK on response repetition comes from handle_response() mechanism "We must re-send ACKs to re-transmitted final responses". According to code history, 400 response specific processing was introduced with commit "channels/chan_sip: Add improved support for 4xx error codes" This commit added support for : - 400/414/493 in handle_response_subscribe() handle_response_register() and handle_response(). - 414/493 only in handle_response_invite(). This fix adds 400 response support in handle_response_invite(). ASTERISK-28957 Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad	5 years ago
Guido Falsi	10bc2d40e5	chan_dadhi: Fix setvar in dahdi channels ... The change to how setvar works for various channels performed in ASTERISK~23756 missed some required change in the dahdi channel, where the variables are actually set while reading configuration. This change should fix the issue. ASTERISK-28955 Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274	5 years ago
George Joseph	efeff59152	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	5 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	5 years ago
Alexander Traud	d53925c98a	chan_sip: externhost/externaddr with non-default TCP/TLS ports. ... ASTERISK-28372 Reported by: Anton Satskiy ASTERISK-24428 Reported by: sstream Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c	5 years ago
Alexander Traud	338990d973	chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. ... ASTERISK-27195 Reported by: Joshua Roys Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de	5 years ago
traud	9e0995b1b7	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
Kevin Harwell	48669ea81b	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	8021924a46	chan_unistim: Avoid tautological warnings with clang. ... ASTERISK-28803 Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af	5 years ago
Joshua C. Colp	ebe7749127	chan_sip: Send 403 when ACL fails. ... Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2	5 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	5 years ago
Sean Bright	2a3b2d5781	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