asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	a036c35903	res_stasis: Handle re-enter stasis bridge with swap channel. We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee	9 years ago
Richard Mudgett	9942d50aa5	bridge: Hold off more than one imparting channel at a time. An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1	9 years ago
Joshua Colp	b1b3460783	Merge "pjproject: Add patch for removing strip of '[]' from header params" into 13	9 years ago
Joshua Colp	08f6408dc6	Merge "PJSIP: Remove PJSIP parsing functions from uri length validation." into 13	9 years ago
Joshua Colp	ded3794fc6	app_talkdetect: Make the module core supported. This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88	9 years ago
Mark Michelson	efae187217	PJSIP: Remove PJSIP parsing functions from uri length validation. The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60	9 years ago
Joshua Colp	9a22ef81af	Merge "app_queue: Frequent segfaults in function can_ring_entry()" into 13	9 years ago
Joshua Colp	47adff8977	Merge "stasis_bridge.c: Update stasis bridge push diagnostic messages." into 13	9 years ago
Joshua Colp	a048a0ffbd	Merge "res_pjsip_transport_management: Allow unload to occur." into 13	9 years ago
Joshua Colp	c922846c6d	Merge "bridge_channel.c: Ignore role setup failure in channel push." into 13	9 years ago
Mark Michelson	f436b9ab11	res_pjsip_registrar: Fix bad memory-ness with user_agent. Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08	9 years ago
Joshua Colp	49bfdc9ac0	res_pjsip_transport_management: Allow unload to occur. At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a	9 years ago
Richard Mudgett	f4693d1897	bridge_channel.c: Ignore role setup failure in channel push. We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00	9 years ago
Jaco Kroon	22335fe18a	chan_sip: Don't verify table if rtupdate=no If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d	9 years ago
Joshua Colp	c7732a2600	Merge "Codecs: strip codec name while parsing allow/disallow options" into 13	9 years ago
ibercom	3b9d8b60b2	app_queue: Frequent segfaults in function can_ring_entry() ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117	9 years ago
Richard Mudgett	724acb6ce7	stasis_bridge.c: Update stasis bridge push diagnostic messages. Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a	9 years ago
Joshua Colp	56c8182913	Merge "app_voicemail/IMAP: function 'save_to_folder' creates wrong folder" into 13	9 years ago
Mark Michelson	5f78801859	transport management: Register thread with PJProject. The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283	9 years ago
Joshua Colp	13cb5ea73f	Merge "res_pjsip_transport_management: Kill idle TCP connections." into 13	9 years ago
Joshua Colp	120493d5c0	Merge "Rename res_pjsip_keepalive res_pjsip_transport_management" into 13	9 years ago
Joshua Colp	6c9c714bb6	Merge "AST-2016-004: Fix crash on REGISTER with long URI." into 13	9 years ago
Mark Michelson	7fb3724a77	res_pjsip_transport_management: Kill idle TCP connections. "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6	9 years ago
Mark Michelson	707fd4dcd0	Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433	9 years ago
Mark Michelson	0b4bb19e0b	AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55	9 years ago
Richard Mudgett	f6e080c6a4	bridge_softmix.c: Fix crash if could not allocate the dsp. Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372	9 years ago
Joshua Colp	1f853df29c	Merge "app_voicemail: Fix test_voicemail_notify_endl test." into 13	9 years ago
George Joseph	cf15a2f2d3	pjproject: Add patch for removing strip of '[]' from header params From the patch submitted to Teluu on 4/12/2016 <<<<<<<<< The wholesale stripping of '[]' from header parameters causes issues if something (like a port) occurs after the final ']'. '[2001🅰️:b]' will correctly parse to '2001🅰️:b' '[2001🅰️:b]:8080' will correctly parse to '2001🅰️:b' but the scanner is left with ':8080' and parsing stops with a syntax error. I can't even find a case where stripping the '[]' is a good thing anyway. Even if you continued to parse and resulted in a string that looks like this... '2001🅰️🅱️8080', it's not valid. This came up in Asterisk because Kamailio sends us a Contact with an alias URI parameter that has an IPv6 address in it like this: Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6> which should be legal but causes a syntax error because of the characters after the final ']'. Even if it didn't, the '[]' should still not be stripped. I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 enabled. No issues were caused by removing the code that strips the '[]'. >>>>>>>>>>> ASTERISK-25123 #close Reported-by: Anthony Messina Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a	9 years ago
Joshua Colp	4ab4fc9141	Merge "res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event" into 13	9 years ago
Joshua Colp	daa086fae4	app_voicemail: Fix test_voicemail_notify_endl test. The test_voicemail_notify_endl test checks the end-of-line characters of an email message to confirm that they are consistent. The test wrongfully assumed that reading from the email message into a buffer will always result in more than 1 character being read. This is incorrect. If only 1 character was read the test would go outside of the buffer and access other memory causing a crash. The test now checks to ensure that 2 or more characters are read in ensuring the test stays within the buffer. ASTERISK-25874 #close Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710	9 years ago
zuul	70c788ec5e	Merge "res_pjsip: Add headers to AMI Event ContactStatusDetail" into 13	9 years ago
Alexei Gradinari	f896136460	app_voicemail/IMAP: function 'save_to_folder' creates wrong folder If try to move message to Cust1 (number 5) the function 'save_to_folder' tries to create Greeting folder instead of Cust1. This patch fixed it by setting GREETINGS_FOLDER = -1 ASTERISK-24927 #close Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51	9 years ago
Alexei Gradinari	70b7673f09	res_pjsip: Add headers to AMI Event ContactStatusDetail * Added Useragent and RegExpire headers to AMI Event ContactStatusDetail with associated documentation. ASTERISK-25903 #close Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239	9 years ago
zuul	cf1c0277b5	Merge "res_pjsip_outbound_publish: Add transport for outbound PUBLISH" into 13	9 years ago
zuul	600148b5b0	Merge "alembic: Remove batch operations (and sqlite support)" into 13	9 years ago
Joshua Colp	5eec2386cf	Merge "core_unreal: Fix hangupcauses not getting set on Local channels" into 13	9 years ago
zuul	df40173a00	Merge "res_pjsip contact: Lock expiration/addition of contacts" into 13	9 years ago
Alexei Gradinari	64ecd41c8f	Codecs: strip codec name while parsing allow/disallow options Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d	9 years ago
Jaco Kroon	3f6c4667b8	core_unreal: Fix hangupcauses not getting set on Local channels ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa	9 years ago
zuul	f2edcfe62e	Merge "app_voicemail/IMAP: IMAP access FATAL error: Out of memory" into 13	9 years ago
George Joseph	fe7e48db03	res_pjsip contact: Lock expiration/addition of contacts Contact expiration can occur in several places: res_pjsip_registrar, res_pjsip_registrar_expire, and automatically when anyone calls ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar may also be attempting to renew or add a contact. Since none of this was locked it was possible for one thread to be renewing a contact and another thread to expire it immediately because it was working off of stale data. This was the casue of intermittent registration/inbound/nominal/multiple_contacts test failures. Now, the new named lock functionality is used to lock the aor during contact expire and add operations and res_pjsip_registrar_expire now checks the expiration with the lock held before deleting the contact. ASTERISK-25885 #close Reported-by: Josh Colp Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059	9 years ago
zuul	736a2c303d	Merge "lock: Add named lock capability" into 13	9 years ago
George Joseph	0c414eaf35	pjproject: Add patch to fix Via IPv6 parsing There's a bug in pjproject's sip_parser where the ":" wasn't correctly interpreted. This is causing IPv6 addresses in the "received" parameter of the Via header to cause a syntax check failure. This patch was submitted to Teluu on 4/10/2016. ASTERISK-25910 #close Reported-by: Anthony Messina Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922	9 years ago
George Joseph	772ff3048f	lock: Add named lock capability Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45	9 years ago
Alexei Gradinari	fd601f26f7	res_pjsip_outbound_publish: Add transport for outbound PUBLISH The first available transport of the appropriate type is used now. This patch adds new config option 'transport' for outbound-publish. If transport is set then outbound PUBLISH requests will use this transport. ASTERISK-25901 #close Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151	9 years ago
zuul	e7e16b7465	Merge "pbx.c: Minor code rearangements." into 13	9 years ago
Alexei Gradinari	5f768d2a9c	res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event BLF pickup isn't working on Cisco SPA and Snom phones if the direction="recipient" attribute is missing in 'dialog' tag. This patch adds direction="recipient" if extension state is Ringing. ASTERISK-24601 #close Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c	9 years ago
Richard Mudgett	82638fb0c7	pbx.c: Minor code rearangements. * Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95	9 years ago
Alexei Gradinari	bc320df173	app_voicemail/IMAP: IMAP access FATAL error: Out of memory Sometimes uw-imap function 'mail_fetchbody' returns huge len which then pass to uw-imap function 'rfc822_base64'. uw-imap tries to allocate huge memory and abort() on fail. This patch check the len. If the len more than max size (128 Mbytes) log error. This patch also set variables len, newlen to avoid uninizialezed len. This patch also check pointer returned by rfc822_base64. ASTERISK-25899 #close Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca	9 years ago
Richard Mudgett	2ef8a954b3	pbx: Update doxygen for extension state watchers. Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e	9 years ago

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