asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	acc7f3c243	Merge "chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND."	8 years ago
Joshua Colp	38b4189643	Merge changes from topic 'ASTERISK-26672' * changes: res_rtp_asterisk.c: Fix uninitialized memory crash. chan_rtp.c: Fix uninitialized memory crash. res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().	8 years ago
Alexander Traud	e220c11bec	chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND. After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats but remember the joint format. Cached formats contain default parameters, often create an empty fmtp line. However, a joint format might have passed format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and contain the resulting format parameters from a SDP negotiation. ASTERISK-26691 #close Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc	8 years ago
Joshua Colp	ae57652983	chan_pjsip: Use session for retrieving CHANNEL() information. The CHANNEL() dialplan function implementation for PJSIP allows querying of PJSIP specific information. This used the channel passed in to get the PJSIP session and associated information. It is possible for this channel to be masqueraded and end up as a different channel type by the time the information request is actually acted upon. This change retrieves the PJSIP session safely and accesses data from it (including channel). This provides a guarantee that the session and channel will not be altered when the request is being acted upon. ASTERISK-26673 Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6	8 years ago
Richard Mudgett	67b47191e9	chan_rtp.c: Fix uninitialized memory crash. unicast_rtp_request() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort the UnicastRTP channel request if it fails. ASTERISK-26672 Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0	8 years ago
Joshua Colp	1943ece514	Merge "chan_dahdi.c: Fix bounds check regression."	8 years ago
Corey Farrell	8fbb384ea2	chan_sip: Reorder unload_module to deal with stuck TCP threads. In some situations TCP threads may become frozen. This creates the possibility that Asterisk could segfault if they become unfrozen after chan_sip has been dlclose'd. This reorders the unload_module process to allow abort if threads do not exit within 5 seconds. High level order as follows: 1) Unregister from the core to stop new requests. 2) Signal threads to stop 3) Clear config based tables (but do not free the table itself). 4) Verify that threads have shutdown, cancel unload if not. 5) Clean all remaining resources. ASTERISK-26586 Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882	8 years ago
Richard Mudgett	9404efa6f4	chan_dahdi.c: Fix bounds check regression. Caused by ASTERISK-25494 Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb	8 years ago
Joshua Colp	963735dfce	Merge "Fix typo in chan_sip"	9 years ago
Joshua Colp	b8a0770d74	Merge "chan_sip: Delete unneeded check"	9 years ago
Badalyan Vyacheslav	4c6ba1dbba	Fix typo in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb	9 years ago
Badalyan Vyacheslav	51118e7d70	chan_sip: Delete unneeded check P is always true. We check it before Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb	9 years ago
Badalyan Vyacheslav	fe5be81821	Small code cleanup in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a	9 years ago
Walter Doekes	c796f00c35	chan_sip: Do not allow non-SP/HTAB between header key and colon. RFC says SIP headers look like: HCOLON = ( SP / HTAB ) ":" SWS SWS = [LWS] ; sep whitespace LWS = [WSP CRLF] 1WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234 chan_sip implemented this: HCOLON = ( LOWCTL / SP ) ":" SWS LOWCTL = %x00-1F ; CTL without DEL This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with \x00-\x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To\x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change fixes so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. ASTERISK-26433 #close AST-2016-009 Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b	9 years ago
Joshua Colp	2a415187c5	Merge "res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter"	9 years ago
zuul	a0c0b1c9cb	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no"	9 years ago
Alexei Gradinari	e5e887be53	chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d	9 years ago
Matt Jordan	0e15760795	res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise 'ws' when WebSockets are to be used as the transport. This applies to both secure and insecure WebSockets. There were two bugs in Asterisk with respect to this: (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for insecure websockets and 'wss' for secure websockets. While this would seem to make sense - since 'WS' and 'WSS' are used for the Via Transport parameter - this is not the case for the SIP URI. This patch corrects that by registering the secure websockets with pjproject using the shorthand 'WS', and by returning 'ws' when asked for the transport parameter. Note that in pjproject, it is perfectly valid to have multiple transports use the same shorthand. (2) In chan_sip, we return an upper-case version of the transport 'WS' instead of 'ws'. Since we should be strict in what we send and liberal in what we accept (within reason), this patch lower-cases the transport before appending it to the parameter. ASTERISK-24330 #close Reported by: cervajs, Inaki Baz Castillo Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42	9 years ago
Michael Kuron	0b588778c0	chan_sip: Fix segfault during module unload If a TCP/TLS connection was pending (not accepted and not timed out) during unload of chan_sip, Asterisk would segfault when trying to send a signal to a thread whose thread ID hadn't been recorded yet. This commit fixes that by recording the thread ID before calling the blocking connect() syscall. This was a regression introduced by `776a14386a`. The above wasn't enough to fix the segfault, which was now delayed to the point where connect() timed out. Therefore, it was necessary to also remove the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be used to interruput the connect() syscall. This was a regression introduced by `5d313f51b9`. ASTERISK-26586 #close Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
Igor Goncharovskiy	dfb951817f	Fix closing rtp ports after call finished in chan_unistim. Fix ASTERISK-26565 by adding ast_rtp_instance_stop before rtp instance destroy for chan_unistim. Also several fixes for displayed text translation. Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc	9 years ago
C.J. Collier	73524bde9c	chan_sip: Fix typo and re-wrap surrounding docs Correct typo of end-pints to end-points Re-wrap session timer parameter docs to max 80 chars wide; this eases reading on terminals with lower resolution, commonly the case for those with visual impairments. ASTERISK-26573 Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>	9 years ago
Kevin Harwell	bf01ff53f8	Revert "chan_sip: Fix lastrtprx always updated" This reverts commit `93332cb1d0`. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d	9 years ago
Joshua Colp	a1bbdabb8e	Merge "chan_sip: add missing account code"	9 years ago
zuul	673964d330	Merge "chan_dahdi: remove by_name support"	9 years ago
Sebastian Gutierrez	0904c1f4cc	chan_sip: add missing account code Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482	9 years ago
Grachev Sergey	2526dff94d	chan_sip: Incorrect display option Outbound reg. retry 403 If in sip.conf (general section) set option register_retry_403=no, the command "sip show settings" return value: Outbound reg. retry 403:0 If in sip.conf (general section) set option register_retry_403=yes, the command "sip show settings" return value: Outbound reg. retry 403:-1 * In static char "sip show settings" for "Outbound.reg. retry 403" option use AST_CLI_YESNO ASTERISK-26476 #close Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9	9 years ago
zuul	0ec5abe592	Merge "Remove ASTERISK_REGISTER_FILE."	9 years ago
Joshua Colp	24d0907849	Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads."	9 years ago
Tzafrir Cohen	0646b48ece	chan_dahdi: remove by_name support Support for referring to DAHDI channels by logical names was added in (FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support of refering to channels by name. While technically usable, it has never been properly supported in dahdi-tools, as using it would require many changes at the Asterisk level. Instead logical mapping was added at the kernel level. Thus it seems that refering to DAHDI channels by name is not really used by anyone, and therefore should probably be removed. Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Alexei Gradinari	2b9ad3a5f7	chan_pjsip: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk. This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref to inform pjproject that an INVITE session is in use. ASTERISK-26482 #close Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33	9 years ago
Joshua Colp	1843b7fa0c	Merge "chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia."	9 years ago
Joshua Colp	8b2b8be4e3	Merge "chan_rtp: Set a sane default rtp engine for unicast."	9 years ago
Moises Silva	2b03017022	chan_rtp: Set a sane default rtp engine for unicast. ASTERISK-26439 Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011	9 years ago
Michael Kuron	e9315791b3	chan_sip: Only send video on outgoing channel if incoming channel supports it Previously, the settings videosupport=always and videosupport=yes behaved identically and unconditionally caused a video offer to be sent in the SDP on an outgoing call. This was a regression introduced with commit `5a1d90e1fb` in Asterisk 1.6.1. This commit restores correct behavior: videosupport=always causes a video offer to be sent unconditionally, while videosupport=yes will only offer video on an outbound channel if the incoming channel it is bridged to also supports video. That way, the device receiving the outgoing call can display the correct user interface elements for audio or video and will not unnecessarily show a blank video window on an audio-only call. ASTERISK-17470 #close Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae	9 years ago
Alexander Traud	4f7f8a7e95	chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. In the SIP channel driver chan_sip, auto_comedia was expected to be used in tandem with auto_force_rport. Or stated differently: Only when auto_force_rport was chosen (the default), auto_comedia worked. This change allows auto_comedia to be set independently of the state of (auto_)force_rport. For example, nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments when IPv6 clients are behind a Firewall. ASTERISK-26457 #close Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2	9 years ago
Alexander Traud	c4268ec734	chan_sip: Honor support of Symmetric Response (rport) for SIP requests. In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6	9 years ago
zuul	3f62485ba7	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4."	9 years ago
zuul	eeeff9487f	Merge "chan_sip: Address runaway when realtime peers subscribe to mailboxes"	9 years ago
Alexander Traud	5dd99465d3	chan_sip: Resolve externhost not to IPv6; instead go for IPv4. For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac	9 years ago
George Joseph	d425971009	chan_sip: Address runaway when realtime peers subscribe to mailboxes Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0	9 years ago
Aaron An	18a8ca06eb	channels/chan_pjsip: fix HANGUPCAUSE function bug. HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61	9 years ago
zuul	544fe73811	Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets"	9 years ago
Steve Davies	6ba68b486e	chan_sip: Fix session timeout on retransmit of non-UDP packets Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204	9 years ago
Walter Doekes	740292e6ae	chan_sip: Allow target refresh (Contact update) on re-INVITE. Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from us the Contact could get updated, but after re-INVITEs from the peer, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435	9 years ago
Joshua Colp	82a3d659dc	chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73	9 years ago
Alexander Traud	7a12355dbd	chan_sip: Allow Preferred sRTP. Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd	9 years ago

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8341 Commits (fb02cc5a8bae22f2eca4f60d84550b30a19d723a)