asterisk

Commit Graph

Author	SHA1	Message	Date
Friendly Automation	25c6890864	Merge "translate.c do not log WARNING on empty audio frame" into 16	6 years ago
Nasir Iqbal	52a3d4a761	app_amd: issue with silence suppression fixed Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual wait time instead of `framelength` is enough to fix timeout / TOOLONG issue. ASTERISK-28419 #close Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb	6 years ago
George Joseph	01712bbdc9	CI: New way to determnine libdir We were using the presence of /usr/lib64 to determine where shared libraries should be installed. This only existed on Redhat based systems and was safe. If it existed, use it, otherwise use /usr/lib. Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is installed there, it won't work. The new method, just looks for $ID in /etc/os-release and if it's centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib. NOTE: This applies only to the CI scripts. Normal asterisk build and install is not affected. Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3	6 years ago
George Joseph	1ee2f01f62	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
Alexei Gradinari	8b77318a2c	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	ccc92b6ecb	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
Sean Bright	694097ee68	pjproject: Update to 2.9 release Relies on https://github.com/asterisk/third-party/pull/4 Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3	6 years ago
Joshua Colp	82789aafd6	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	ca462f6e15	Merge "app_attended_transfer: new application AttendedTransfer" into 16	6 years ago
George Joseph	5c2bf122ed	Merge "app_blind_transfer: new application BlindTransfer" into 16	6 years ago
George Joseph	da5874ec47	Merge "chan_pjsip.c: Check for channel and session to not be NULL in hangup" into 16	6 years ago
Joshua Colp	5c32680bb7	Merge "cdr_pgsql: fix error in connection string" into 16	6 years ago
George Joseph	e7d0ca1591	Merge "pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi" into 16	6 years ago
agupta	72f26aa8eb	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
Kirsty Tyerman	a1c84709b8	pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi ASTERISK-28234 Reported-by: Kirsty Tyerman Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1	6 years ago
Chris-Savinovich	c2621aa190	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
Alexei Gradinari	86cd77ec0a	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
Alexei Gradinari	6321b559b9	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
Joshua Colp	de38c9c3b3	Merge "res_fax: fix segfault on inactive "reserved" fax session" into 16	6 years ago
Friendly Automation	eefd53b718	Merge "app_readexten: new option 'p' to stop reading on '#' key" into 16	6 years ago
Friendly Automation	c945d3d9c8	Merge "res_fax: add channel name to CLI 'fax show session'" into 16	6 years ago
Friendly Automation	53df0ff200	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed" into 16	6 years ago
Asterisk Development Team	d2c07aceca	Update CHANGES and UPGRADE.txt for 16.4.0	6 years ago
Friendly Automation	7053104222	Merge "build: Fix file format in CHANGES-staging." into 16	6 years ago
Alexei Gradinari	e77704f45c	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
Alexei Gradinari	e0a574253e	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
Ben Ford	ec74fd56a7	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	86fb72c4d0	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
Friendly Automation	0fc6617246	Merge "res_rtp_asterisk: timestamp should be unsigned instead of signed int" into 16	6 years ago
George Joseph	a99f814414	Merge "res_rtp_asterisk: Add ability to propose local address in ICE" into 16	6 years ago
Morten Tryfoss	9351aa3f0e	res_rtp_asterisk: timestamp should be unsigned instead of signed int Using timestamp with signed int will cause timestamps exceeding max value to be negative. This causes the jitterbuffer to do passthrough of the packet. ASTERISK-28421 Change-Id: I9dabd0718180f2978856c50f43aac4e52dc3cde9	6 years ago
Alexei Gradinari	db5bc0fabf	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
Alexei Gradinari	9516fb64c9	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
Joshua Colp	33ed2e1bb8	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	79b15d0b30	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
Joshua Colp	2aa9bc6d2c	Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count." into 16	6 years ago
Friendly Automation	2de4f668be	Merge "conversions.c: Add conversions for largest max sized integer" into 16	6 years ago
Friendly Automation	a1a8c47f53	Merge "Fixes for GCC 9" into 16	6 years ago
Friendly Automation	790b9223ed	Merge "build: Pass --fno-partial-inlining to third-party when appropriate" into 16	6 years ago
Alexei Gradinari	de82bdd746	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
Joshua Colp	cf2f8db1b7	Merge "pjsip_options.c: Allow immediate qualifies for new contacts." into 16	6 years ago
George Joseph	e7734476c6	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
Joshua Colp	ece29db9bd	res_rtp_asterisk: Fix sequence number cycling and packet loss count. This change fixes two bugs which both resulted in the packet loss count exceeding 65,000. The first issue is that the sequence number check to determine if cycling had occurred was using the wrong variable resulting in the check never seeing that cycling has occurred, throwing off the packet loss calculation. It now uses the correct variable. The second issue is that the packet loss calculation assumed that the received number of packets in an interval could never exceed the expected number. In practice this isn't true due to delayed or retransmitted packets. The expected will now be updated to the received number if the received exceeds it. ASTERISK-28379 Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6	6 years ago
Ben Ford	941dead08d	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
Kevin Harwell	edc3e0df1a	conversions.c: Add conversions for largest max sized integer Added a conversion for umax (largest maximum sized integer allowed). Adjusted the other current conversion functions (uint and ulong) to be derivatives of the umax conversion since they are simply subsets of umax. Also made the negative check move the pointer on spaces since strtoumax does it anyways. Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08	6 years ago
agupta	9a0fa51443	stasis: Hangup channel for Local channel No such extension error When we use early bridge with create and dial from stasis using Local channel and the dialplan does not any entry the it is returned from core_local.c with No such extension . In such case asterisk locks up till the channel is not hangup with the error Exceptionally long voice queue length * Found that in such case app_control_dial fails on ast_call method and return -1 * Since it is called from stasis_app_send_command_async and return -1 does not cause resources to be freed and since no PBX exist it is not able to read from channel causing exceptionally long queue * After putting this code found that the channel was releasing immediately and resources were freed. ASTERISK-28399 Reported by: Abhay Gupta Tested by: Abhay Gupta Change-Id: I0a55c923fc6995559f808d63b9488762b4489318	6 years ago
George Joseph	543d487746	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
Joshua Colp	8357ab7e9a	Merge "app_confbridge: Add "all" variants of REMB behavior." into 16	6 years ago
Friendly Automation	27696cbda6	Merge "stasis: Only place stasis created and dialed channels into dial bridge." into 16	6 years ago
Friendly Automation	748f1d64a1	Merge "stasis: Call callbacks when imparting fails" into 16	6 years ago

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