asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	015cdc9f64	sig_pri: Fix deadlock caused by sig_pri_queue_hangup The change to add setting hangupsource to sig_pri_queue_hangup() made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued deadlocks when a hangup request was received from the core at the same time a hanguprequest was received from the remote end via the D channel. Although the PRI's channel private structure was being unlocked before setting the hangupsource, the PRI's own lock was still being held during the process. If channel actions were also coming from the core, a deadlock on the PRI could result. This deadlock could then escalate to the entire DAHDI subsystem via DAHDI's global interface list lock, especially if someone used the PRI CLI commands. Fix: * We now unlock the PRI as well as the PRI's channel private structure before setting the hangupsource, then relock both afterwards. ASTERISK-28605 Reported by: Dirk Wendland Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9	5 years ago
Friendly Automation	e5446d7ffc	Merge "func_odbc: acf_odbc_read() and cli_odbc_read() unicode support" into 17	6 years ago
Friendly Automation	b401891036	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 17	6 years ago
Boris P. Korzun	6d2eced3d3	func_odbc: acf_odbc_read() and cli_odbc_read() unicode support Added ast_odbc_ast_str_SQLGetData() considers SQL_DESC_OCTET_LENGTH column attribute for correct allocating the buffer. ASTERISK-28497 #close Change-Id: I50e86c8a277996f13d4a4b9b318ece0d60b279bf	6 years ago
George Joseph	321b0c5041	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP" into 17	6 years ago
Jean Aunis	811c7bdabe	chan_sip: voice frames are no longer transmitted after emitting a COLP The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Sean Bright	843d601e2c	db: Initialize condition primitive before use The db_init() function ultimately calls db_sync() which signals the condition before it is initialized. Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab	6 years ago
George Joseph	86550a59f3	Merge "config.c: Skip UTF-8 BOMs if present when reading config files" into 17	6 years ago
Joshua C. Colp	0c36ef530b	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR." into 17	6 years ago
Joshua C. Colp	7aa2445264	Merge "confbridge: Add support for specifying maximum sample rate." into 17	6 years ago
Friendly Automation	2dcb96a868	Merge "main/file.c: Limit media cache usage to remote files." into 17	6 years ago
Friendly Automation	7d0757d660	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors." into 17	6 years ago
Sean Bright	8c8989f205	config.c: Skip UTF-8 BOMs if present when reading config files ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
Kevin Reeves	45f7f9bb85	main/file.c: Limit media cache usage to remote files. When testing for the existance of a file, the media cache is searched even if the file has no chance of being in it. This can cause performance issues as the media cache size increases. As a result, calls to applications like Read and Playback using local files must scan through the media cache before playing. Under load and with a large cache, this can delay the playback of those files. This patch updates the function that checks for the existance of a file to only consult the media cache database if the requested file is a remote path. It introduces a new is_remote_path() function in main/file.c. ASTERISK-28625 #close Reported-by: kevin@phoneburner.com Change-Id: If91137493732d9034dafa381c081c69274a7dcc9	6 years ago
George Joseph	2634e63f4d	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read" into 17	6 years ago
Friendly Automation	2725c885e8	Merge "sip_to_pjsip.py: Fix trustrpid typo" into 17	6 years ago
Joshua C. Colp	3104c03476	Merge "configure: Add check for MySQL client bool and my_bool type usage." into 17	6 years ago
Joshua C. Colp	58a6d4ecf5	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP." into 17	6 years ago
Pascal Cadotte Michaud	69c9b74836	sip_to_pjsip.py: Fix trustrpid typo ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	6 years ago
Frederic LE FOLL	a0e31cc14e	app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. Temporary channel lifespan is very short and CDR deactivation request through ast_cdr_set_property() may happen when CDR is not available yet. Use CDR_PROP() dialplan function instead, it will first wait for pending CDR insertion requests to be processed. ASTERISK-28636 Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074	6 years ago
Joshua C. Colp	5a80c36d62	configure: Add check for MySQL client bool and my_bool type usage. Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Friendly Automation	c7966269f5	Merge "res_pjsip_nat: Restore original contact for REGISTER responses" into 17	6 years ago
Joshua C. Colp	41fdcf7a65	confbridge: Add support for specifying maximum sample rate. ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Friendly Automation	a7a5bf0129	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
Joshua Colp	895a66f772	Merge "ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging." into 17	6 years ago
Joshua C. Colp	c9147a759e	res_pjsip_session: Set stream state on created streams for incoming SDP. A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Kevin Harwell	68724425e4	res_fax: wrap v21 detected Asterisk initiated negotiation with config option A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
Asterisk Development Team	08a6e8c553	Update CHANGES and UPGRADE.txt for 17.1.0	6 years ago
Jaco Kroon	6041fc8b9c	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Pascal Cadotte Michaud	39aff53cff	PJSIP_CONTACT: add missing argument documentation add missing argument "rtt" and "status" to the documentation The change to the dtd file allow an enumlist to contain one or many configOptionToEnum or enum. This is different from the previous patch I submitted when you could have a configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or (one or many enums) ASTERISK-28626 Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d	6 years ago
George Joseph	87d5fb1196	Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" into 17	6 years ago
Joshua Colp	6a89784b78	Revert "PJSIP_CONTACT: add missing argument documentation" This reverts commit `450173a0ae`. Reason for revert: Regression in XML validation. validity error : Content model of enumlist is not determinist: (configOptionToEnum \| (configOptionToEnum , enum+) \| enum+) As we are preparing to do releases and this is not critical I am reverting this for now until resolved. Change-Id: I47a6714bb555b35a371861d434daf5b97c0cdcbd	6 years ago
George Joseph	86822722b6	res_rtp_asterisk: Add frame list cleanups to ast_rtp_read In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a	6 years ago
Friendly Automation	5fb0d1d562	Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found" into 17	6 years ago
Jaco Kroon	32ce8fa046	chan_sip: in case of tcp/tls, be less annoying about tx errors. chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to 213.150.203.60:1492 returned -2: Interrupted system call returned -2 implies this wasn't actually an OS error, so errno makes no sense either. Internal error was already logged higher up, and -2 generally means that either there isn't a valid connection available, or the pipe notification failed, and that is already correctly logged. ASTERISK-28651 #close Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Friendly Automation	ffbfc09fc1	Merge "app_queue: Fix old confusing comment about when the members are called" into 17	6 years ago
George Joseph	fd14d7302f	res_pjsip_nat: Restore original contact for REGISTER responses RFC3261 Section 10 "Registrations", specifically paragraph "10.2.4: Refreshing Bindings", states that a user agent compares each contact address (in a 200 REGISTER response) to see if it created the contact. If the Asterisk endpoint has the rewrite_contact option set however, the contact host and port sent back in the 200 response will be the rewritten one and not the one sent by the user agent. This prevents the user agent from matching its own contact. Some user agents get very upset when this happens and will not consider the registration successful. While this is rare, it is acceptable behavior especially if more than 1 user agent is allowed to register to a single endpoint/aor. This commit updates res_pjsip_nat (where rewrite_contact is implemented) to store the original incoming Contact header in a new "x-ast-orig-host" URI parameter before rewriting it, and to restore the original host and port to the Contact headers in the outgoing response. This is only done if the request is a REGISTER and rewrite_contact is enabled. pjsip_message_filter was also updated to ensure that if a request comes in with any existing x-ast-* URI parameters, we remove them so they don't conflict. Asterisk will never send a request with those headers in it but someone might just decide to add them to a request they craft and send to Asterisk. NOTE: If a device changes its contact address and registers again, it's a NEW registration. If the device didn't unregister the original registration then all existing behavior based on aor/remove_existing and aor/max_contacts apply. ASTERISK-28502 Reported-by: Ross Beer Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e	6 years ago
George Joseph	2526c8e562	Merge "channel.c: Resolve issue with receiving SIP INFO packets for DTMF" into 17	6 years ago
Friendly Automation	af079f5085	Merge "res_pjsip_outbound_registration: add support for SRV failover" into 17	6 years ago
Friendly Automation	6685a1862c	Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases" into 17	6 years ago
Friendly Automation	e17bde758b	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
Friendly Automation	aa2765d366	Merge "chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime." into 17	6 years ago
Friendly Automation	478a602e70	Merge "res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled" into 17	6 years ago
Friendly Automation	77b5ec3e7a	Merge "parking: Fall back to parker channel name even if it matches parkee." into 17	6 years ago
Sean Bright	4e057eb9d2	res_pjsip_registrar.c: Prevent potential double free if AOR is not found The simple fix here is simply to NULL out username and password after we call ast_free on them. Unfortunately, I noticed that we weren't checking for allocation failures for username and password, and adding those checks made things noisy and cumbersome. So instead we partially rollback the recent LGTM patch, and move the alloca calls into find_aor_name(). ASTERISK-28641 #close Reported by: Ross Beer Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2	6 years ago
Sean Bright	f26e5bacc0	res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases We're appropriately sizing the id_domain_alias buffer, but then copying the data into the id_domain one. We were then using the uninitialized id_domain_alias buffer we just allocated. This is ASTERISK~28641 adjacent, but significant enough to warrant its own patch. Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9	6 years ago
Walter Doekes	e1eb5e8dc2	app_queue: Fix old confusing comment about when the members are called ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b	6 years ago
Frederic LE FOLL	3e73893e53	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Sean Bright	4d56adf8fb	res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled We need to copy the endpoint name before we call ao2_cleanup() on it, otherwise we might try to access memory that has been reclaimed. ASTERISK-28445 #close Reported by: Bernhard Schmidt Change-Id: I404b952608aa606e0babd3c4108346721fb726b3	6 years ago
George Joseph	c0ab241cc0	Merge "media_cache.c: Various CLI improvements" into 17	6 years ago

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