asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	6b1a632290	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	10 years ago
Richard Mudgett	1c5248c383	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	10 years ago
zuul	cc9b72208f	Merge "Dial: Combine frame handling functions."	10 years ago
Joshua Colp	7bfa1784c1	Merge "pjproject: Add patch for removing strip of '[]' from header params"	10 years ago
zuul	c60f8e947f	Merge "PJSIP: Remove PJSIP parsing functions from uri length validation."	10 years ago
Joshua Colp	d95512a7dd	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	10 years ago
Mark Michelson	0235a66532	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	10 years ago
Joshua Colp	64b03ec702	Merge "app_queue: Frequent segfaults in function can_ring_entry()"	10 years ago
Joshua Colp	d268fe527d	Merge "stasis_bridge.c: Update stasis bridge push diagnostic messages."	10 years ago
Joshua Colp	2b6764d8b9	Merge "res_pjsip_transport_management: Allow unload to occur."	10 years ago
Joshua Colp	4b798cc99a	Merge "bridge_channel.c: Ignore role setup failure in channel push."	10 years ago
Mark Michelson	b8b60135ec	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	10 years ago
Joshua Colp	6cfa02394f	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	10 years ago
Richard Mudgett	6365f0018f	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	10 years ago
Jaco Kroon	f06ce7f90a	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	10 years ago
Joshua Colp	6fe38d60aa	Merge "Codecs: strip codec name while parsing allow/disallow options"	10 years ago
ibercom	dbb47e0a47	app_queue: Frequent segfaults in function can_ring_entry() ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117	10 years ago
Richard Mudgett	af114edb8b	stasis_bridge.c: Update stasis bridge push diagnostic messages. Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a	10 years ago
Joshua Colp	f7a434fa6d	Merge "app_voicemail/IMAP: function 'save_to_folder' creates wrong folder"	10 years ago
Mark Michelson	5e64d7e7a3	Dial: Combine frame handling functions. There is a good amount of repetition in the two frame handling routines in the Dial API. This commit combines the two functions into one. This is in preparation for an upcoming commit that adds the ability to handle frames for a channel in a bridge. ASTERISK-25925 Reported by Mark Michelson Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58	10 years ago
Alexei Gradinari	a6e2ba187a	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	10 years ago
Mark Michelson	be4333ddad	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	10 years ago
Joshua Colp	d3e4d10f04	Merge "res_pjsip_transport_management: Kill idle TCP connections."	10 years ago
Joshua Colp	0c239112bb	Merge "Rename res_pjsip_keepalive res_pjsip_transport_management"	10 years ago
Joshua Colp	d76f29a547	Merge "AST-2016-004: Fix crash on REGISTER with long URI."	10 years ago
Mark Michelson	216f22fd0f	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	10 years ago
Mark Michelson	d9fba46016	Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433	10 years ago
Mark Michelson	7b8b6e2e4f	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: I3ea7cee16f29c8088794de3085ca7523c1c4833d	10 years ago
Richard Mudgett	ff3af764de	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	10 years ago
Joshua Colp	fddec0c266	Merge "app_voicemail: Fix test_voicemail_notify_endl test."	10 years ago
George Joseph	bd3671b397	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	10 years ago
Joshua Colp	c714d0006b	Merge "res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event"	10 years ago
Joshua Colp	5a0534dc62	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	10 years ago
Alexei Gradinari	c00c298a0e	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	10 years ago
Alexei Gradinari	49813bc9e5	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	10 years ago
zuul	74951bd591	Merge "res_pjsip_outbound_publish: Add transport for outbound PUBLISH"	10 years ago
Joshua Colp	11ec5801d9	Merge "alembic: Remove batch operations (and sqlite support)"	10 years ago
Joshua Colp	4a7ed92308	Merge "core_unreal: Fix hangupcauses not getting set on Local channels"	10 years ago
zuul	9480ec2bd8	Merge "res_pjsip contact: Lock expiration/addition of contacts"	10 years ago
Alexei Gradinari	4e00e31ef1	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	10 years ago
Jaco Kroon	2cc56573de	core_unreal: Fix hangupcauses not getting set on Local channels ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa	10 years ago
zuul	c3339816f1	Merge "app_voicemail/IMAP: IMAP access FATAL error: Out of memory"	10 years ago
George Joseph	a621dd5e96	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	10 years ago
Joshua Colp	44fba00ca4	Merge "lock: Add named lock capability"	10 years ago
George Joseph	8637f29d24	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	10 years ago
Joshua Colp	8610f4344f	Merge "pbx.h: Make ast_state_cb_type take more const."	10 years ago
George Joseph	216abb0ae7	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	10 years ago
Alexei Gradinari	f9dab80816	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	10 years ago
Joshua Colp	b47dfd1c6e	Merge "pbx.c: Minor code rearangements."	10 years ago
Alexei Gradinari	b3be945415	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	10 years ago

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