asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	dd20beedb9	Merge "res_pjsip_sdp_rtp: Fix ICE candidates leak." into 13	6 years ago
George Joseph	e39dbc4909	Merge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected" into 13	6 years ago
Joshua Colp	9721f3908d	res_pjsip_sdp_rtp: Fix ICE candidates leak. Given the non-default configuration of enabling ICE support on an endpoint that does not result in an ICE negotiation occurring the ICE candidates would be leaked. This change makes it so that the ICE candidates are only retrieved if ICE negotiation is occurring. ASTERISK-28460 Change-Id: I7b3f76f031c41fb8a3dc3ef1a84b77e2a8cb969f	6 years ago
George Joseph	f22dedc597	Merge "app_confbridge: Attended transfer event fixup" into 13	6 years ago
George Joseph	5d45686c28	Merge "translate.c do not log WARNING on empty audio frame" into 13	6 years ago
Alexei Gradinari	d5db7473e7	res_fax: gateway sends T.38 request to both endpoints if V.21 detected According T.38 Gateway 'Use case 3' https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway T.38 Gateway should send T.38 negotiation request to called endpoint if FAX preamble (using V.21 detector) generated by called endpoint. But it does not, because fax_gateway_detect_v21 constructs T.38 negotiation request, but forwards it only to other channel, not to the channel on which FAX preamble is detected. Some SIP endpoints could be improperly configured to rely on the other side to initiate T.38 re-INVITEs. With this patch the T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38. Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39	6 years ago
Alexei Gradinari	3bfe1f3a1b	translate.c do not log WARNING on empty audio frame There is WARNING "no samples for ..." on each Playtones. The function ast_playtones_start calls ast_activate_generator, which calls ast_prod. The function ast_prod calls ast_write with empty audio frame. In this case it's spam log. Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660	6 years ago
George Joseph	d0f01af913	chan_dahdi: Address gcc9 issues Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
Joshua Colp	d5fed38ab5	Merge "res_rtp_asterisk: Add support for DTLS packet fragmentation." into 13	6 years ago
George Joseph	41f5d15763	app_confbridge: Attended transfer event fixup When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	6 years ago
Joshua Colp	1ea9bad34d	res_rtp_asterisk: Add support for DTLS packet fragmentation. This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	6 years ago
George Joseph	83c353c650	Merge "app_attended_transfer: new application AttendedTransfer" into 13	6 years ago
George Joseph	dd4f0c94e8	Merge "app_blind_transfer: new application BlindTransfer" into 13	6 years ago
George Joseph	0e30d17293	Merge "chan_pjsip.c: Check for channel and session to not be NULL in hangup" into 13	6 years ago
Alexei Gradinari	45a9ee4c53	app_attended_transfer: new application AttendedTransfer AttendedTransfer queues up attended transfer to the given extension. This application can be useful with Custom Dynamic Features. For example to make attended transfer to a predefined number. features.conf ;;; [applicationmap] my_atxfer => 7,self,GoSub,"my_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_atxfer TRANSFER_CONTEXT=my_transfer [my_atxfer] exten => s,1,AttendedTransfer(1234567890) same => n,Return() [my_transfer] include => default ;;; This application also can be used to completly redefine Attended transfer feature using dialplan. For example: features.conf ;;; [featuremap] atxfer => 7 [applicationmap] custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_atxfer TRANSFER_CONTEXT=my_transfer [custom_atxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,AttendedTransfer(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
agupta	67841b8f55	chan_pjsip.c: Check for channel and session to not be NULL in hangup We have seen some rare case of segmentation fault in hangup function and we could notice that channel pointer was NULL. Debug log shows that there is a 200 OK answer and SIP timeout at the same time. It looks that while the SIP session was being destroyed due to timeout call hangup due to answer event lead to race condition and channel is being destroyed from two different places. The check ensures we check it not to be NULL before freeing it. ASTERISK-25371 Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778	6 years ago
Alexei Gradinari	dd12e1cbd3	app_blind_transfer: new application BlindTransfer BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Chris-Savinovich	45c1159c62	cdr_pgsql: fix error in connection string Fixes an error occurring in function pgsql_reconnect() caused when value of hostname is blank. Which in turn will cause the connection string to look like this: "host= port=xx", which creates a sintax error. This fix now checks if the corresponding values for host, port, dbname, and user are blank. Note that since this is a reconnect function the database library will replace any missing value pairs with default ones. ASTERISK-28435 Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423	6 years ago
Joshua Colp	1c665ae39b	Merge "res_fax: fix segfault on inactive "reserved" fax session" into 13	6 years ago
Friendly Automation	6df25921ca	Merge "app_readexten: new option 'p' to stop reading on '#' key" into 13	6 years ago
Friendly Automation	066ff4d3a8	Merge "res_fax: add channel name to CLI 'fax show session'" into 13	6 years ago
Friendly Automation	e4d91ab4b2	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed" into 13	6 years ago
Alexei Gradinari	e306c62ff1	res_fax: fix segfault on inactive "reserved" fax session The change #10017 "Handle fax gateway being started more than once" introdiced a bug which leads to segfault in res_fax_spandsp. The res_fax_spandsp module does not support reserving sessions, so fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE. The fax_gateway_start does not create a real fax session if the fax session is already present and the state is not AST_FAX_STATE_RESERVED. But the "reserved" session created for res_fax_spandsp has state AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting. Then when fax_gateway_framehook is called and gateway T.38 state is NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to segfault, because session tech_pvt is not set, i.e. the tech session was not initialized/started. This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved" session created for res_fax_spandsp will start. This patch also adds extra check and log ERROR if tech_pvt is not set before call tech->write. ASTERISK-27981 #close Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803	6 years ago
Asterisk Development Team	fba341af8b	Update CHANGES and UPGRADE.txt for 13.27.0	6 years ago
Friendly Automation	7dd93bfe9a	Merge "build: Fix file format in CHANGES-staging." into 13	6 years ago
Alexei Gradinari	dfa513c565	res_fax: add channel name to CLI 'fax show session' This patch adds a channel name to output of CLI 'fax show session' and also expands the channel name field up to 30 characters on CLI 'fax show sessions' Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953	6 years ago
Ben Ford	6aeab9d5e7	build: Fix file format in CHANGES-staging. One of the change files doesn't conform to the format that the release scripts need in order to parse it. Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c	6 years ago
Guido Falsi	ac4921c373	chan_dahdi: add missing include. After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	6 years ago
Alexei Gradinari	6ded762dbf	app_readexten: new option 'p' to stop reading on '#' key This patch adds the 'p' option. The extension entered will be considered complete when a # is entered. Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1	6 years ago
George Joseph	d6fb8abd84	Merge "res_rtp_asterisk: Add ability to propose local address in ICE" into 13	6 years ago
Joshua Colp	fc49632bbc	pjproject-bundled: Add upstream timer fixes Fixed #2191: - Stricter double timer entry scheduling prevention. - Integrate group lock in SIP transport, e.g: for add/dec ref, for timer scheduling. ASTERISK-28161 Reported-by: Ross Beer Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5	6 years ago
George Joseph	90fe830a77	res_rtp_asterisk: Add ability to propose local address in ICE You can now add the "include_local_address" flag to an entry in rtp.conf "[ice_host_candidates]" to include both the advertized address and the local address in ICE negotiation: [ice_host_candidates] 192.168.1.1 = 1.2.3.4,include_local_address This causes both 192.168.1.1 and 1.2.3.4 to be advertized. Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db	6 years ago
Alexei Gradinari	595d60846a	pjsip: replace 180 by 183 if SDP negotiation has completed The caller endpoint hears dead silence if a callee replies 180 (without SDP) and the caller already received 183 (with SDP). It happens because Asterisk sends 180 (WITH SDP) to the caller, there are not incoming RTP packets from the callee and Asterisk does not generate inband ringing, so there are not any outgoing RTP packets to the caller. This patch replaces 180 by 183 if SDP negotiation has completed, as if the caller endpoint is configured with "inband_progress=yes". In this case Asterisk will generate inband ringing untill Asterisk receive incoming RTP packets from the callee. ASTERISK-27994 #close Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73	6 years ago
Friendly Automation	fa98e8cacb	Merge "Fixes for GCC 9" into 13	6 years ago
Friendly Automation	7067177be4	Merge "build: Pass --fno-partial-inlining to third-party when appropriate" into 13	6 years ago
Joshua Colp	e6cedc77a4	Merge "pjsip_options.c: Allow immediate qualifies for new contacts." into 13	6 years ago
George Joseph	4337895aee	Fixes for GCC 9 Various fixes for issues caught by gcc 9. Mostly snprintf trying to copy to a buffer potentially too small. ASTERISK-28412 Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e	6 years ago
Friendly Automation	92832bf176	Merge "Revert "pjproject-bundled: Add upstream timer fixes"" into 13	6 years ago
George Joseph	c41e3184e3	Revert "pjproject-bundled: Add upstream timer fixes" This reverts commit `cfeb8a59eb`. The fixes in question cause assert failures when pjproject asserts are enabled. Reverting in 13 until a solution is found for all branches. Change-Id: Iae5bd340e0543613185fecb63f9c86fa985fe664	6 years ago
Ben Ford	f71a0e3f60	pjsip_options.c: Allow immediate qualifies for new contacts. When multiple endpoints try to register close together using the same AOR with qualify_frequency set, one contact would qualify immediately while the other contacts would have to wait out the duration of the timer before being able to qualify. Changing the conditional to check the contact container count for a non-zero value allows all contacts to qualify immediately. Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415	6 years ago
Friendly Automation	8b0029692f	Merge "pjproject-bundled: Add upstream timer fixes" into 13	6 years ago
George Joseph	7646e2257f	build: Pass --fno-partial-inlining to third-party when appropriate When the gcc version is >= 8.2.1, we were already setting the --fno-partial-inlining flag for Asterisk source files to get around a gcc bug but we weren't passing the flag down to the bundled builds of pjproject and jansson. ASTERISK-28392 Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704	6 years ago
George Joseph	cfeb8a59eb	pjproject-bundled: Add upstream timer fixes Fixed #2191: - Stricter double timer entry scheduling prevention. - Integrate group lock in SIP transport, e.g: for add/dec ref, for timer scheduling. ASTERISK-28161 Reported-by: Ross Beer Change-Id: I02a791fd1570a1e594a132b36c4ff72441108c17	6 years ago
George Joseph	9d8a093a94	res_pjsip: Check return from pjsip_parse_uri calls Updated ast_sip_create_rdata_with_contact and registrar_find_contact to check the return from pjsip_parse_uri before attempting to use the uri returned. ASTERISK-28402 Reported-by: Ross Beer Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7	6 years ago
George Joseph	6edef49525	Merge "mwi core: Move core MWI functionality into its own files" into 13	6 years ago
Friendly Automation	66aa081bb3	Merge "app_amd: Fix infinite loop on silent calls" into 13	6 years ago
Friendly Automation	57a9935ae0	Merge "stasis: Fix crash at shutdown." into 13	6 years ago
agupta	188b1d3e68	app_amd: Fix infinite loop on silent calls The total time logic will now be executed on calls which do not pass any media. ASTERISK-28143 Change-Id: I24726bd29d7e467fc721ca265363417234b22855	6 years ago
Friendly Automation	10b46eea9e	Merge "app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings" into 13	6 years ago
Ben Ford	4589260961	stasis: Fix crash at shutdown. When compiling in dev mode, stasis statistics are enabled and can cause a crash at shutdown due to the following: - Containers are freed - Topics and subscriptions remain - When those topics and subscriptions are deallocated, they go to do things with the container This changes the containers to global ao2 objects, and whenever needed in the code, a reference must be obtained and checked before any operations can be done. ASTERISK-28353 #close Change-Id: Ie7d5e907fcfcb4d65bd36d5e4eb923126fde8d33	6 years ago

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