3701 Commits (b259ac95acefd67a90a451f55acab7fd6ce77d97)

Author	SHA1	Message	Date
Richard Mudgett	e2d3215b83	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5a17ed7a38	channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB ... When a SIP device that has its registration stored in RealTime unregisters, the entry for that device is updated with blank values, i.e., "", indicating that it is no longer registered. Unfortunately, one of those values that is 'blanked' is the device's port. If the column type for the port is not a string datatype (the recommended type is integer), an ODBC or database error will be thrown. MariaDB does not coerce empty strings to a valid integer value. This patch updates the query run from chan_sip such that it replaces the port value with a value of '0', as opposed to a blank value. This is the value that other database backends coerce the empty string ("") to already, and the handling of reading a RealTime registration value from a backend already anticipates receiving a port of '0' from the backends. ASTERISK-24772 #close Reported by: Richard Miller patches: chan_sip.diff uploaded by Richard Miller (License 5685) ........ Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431674 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	fe76d4829f	Use SIPS URIs in Contact headers when appropriate. ... RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in chan_sip. ASTERISK-24646 #close Reported by Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/4346 ........ Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	3b0f03ef7b	chan_sip: stale nonce causes failure ... When refreshing (with a small expiration) a registration that was sent to chan_sip the nonce would be considered stale and reject the registration. What was happening was that the initial registration's "dialog" still existed in the dialogs container and upon refresh the dialog match algorithm would choose that as the "dialog" instead of the newly created one. This occurred because the algorithm did not check to see if the from tag matched if authentication info was available after the 401. So, it ended up assuming the original "dialog" was a match and stopped the search. The old "dialog" of course had an old nonce, thus the stale nonce message. This fix attempts to leave the original functionality alone except in the case of a REGISTER. If a REGISTER is received if searches for an existing "dialog" matching only on the callid. If the expires value is low enough it will reuse dialog that is there, otherwise it will create a new one. ASTERISK-24715 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4367/ ........ Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431194 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	874cb5615d	chan_sip: Case insensitive comparison of "defaultuser" parameter. ... All the other configuration options are case insensitive, so this one should be too. ASTERISK-24355 #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions 430993 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430994 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5835bf7a7f	channels/chan_sip: Fix registration leak during reload ... When the SIP registrations were migrated to using ao2 in what was then trunk, the explicit destruction of the registrations on module reload was removed and not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the issue reporter, on ASTERISK-24673 confirmed that the reference in the registry_list container was being leaked. Since the purpose of cleanup_all_regs is to prep a registration for destruction, this function now calls an ao2_callback function callback with the OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations. This cleans up each registration, and also removes it from the registration container registry_list. Review: https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan Engström Tested by: Stefan Engström ........ Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c7ea108e02	Revert -r430452 It needs to be redone for the next major AMI version change instead. ... ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ef34a05f21	AMI: Remove no longer used parameter from astman_send_listack(). ... Follow-up issue to -r430435 from reviewboard review. ASTERISK-24049 Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	52a7cdb101	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	264a50c52a	chan_sip: Send CANCEL via original INVITE destination even after UPDATE request ... Given the following scenario: * Three SIP phones (A, B, C), all communicating via a proxy with Asterisk * A call is established between A and B. B performs a SIP attended transfer of A to C. B sets the call on hold (A is hearing MOH) and dials the extension of C. While phone C is ringing, B transfers the call (that is, what we typically call a 'blond transfer'). * When the transfer completes, A hears the ringing of phone C, while B is idle. In the SIP messaging for the above scenario, a REFER request is sent to transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an UPDATE request to phone C to update party information. This update is sent directly to phone C, not through the intervening proxy. This has the unfortunate side effect of providing route information, which is then set on the sip_pvt structure for C. If someone (e.g. B) is trying to get the call back (through a directed pickup), Asterisk will send a CANCEL request to C. However, since we have now updated the route set, the CANCEL request will be sent directly to C and not through the proxy. The phone ignores this CANCEL according to RFC3261 (Section 9.1). This patch updates reqprep such that the route is not updated if an UPDATE request is being sent while the INVITE state is INV_PROCEEDING or INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent to the correct location. Review: https://reviewboard.asterisk.org/r/4279 ASTERISK-24628 #close Reported by: Karsten Wemheuer patches: issue.patch uploaded by Karsten Wemheuer (License 5930) ........ Merged revisions 429982 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429983 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8b6ecc449c	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429675 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	58095d2486	chan_sip: Allow T.38 switch-over when SRTP is in use. ... Previously when SRTP was enabled on a channel it was not possible to switch to T.38 as no crypto attributes would be present. This change makes it so it is now possible. If a T.38 re-invite comes in SRTP is terminated since in practice you can't encrypt a UDPTL stream. Now... if we were doing T.38 over RTP (which does exist) then we'd have a chance but almost nobody does that so here we are. ASTERISK-24449 #close Reported by: Andreas Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523) ........ Merged revisions 429632 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429633 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	03c94ef761	res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. ... Frames with a payload length of 0 were incorrectly handled in res_http_websocket. Provided a frame with a payload had been received prior it was possible for a double free to occur. The realloc operation would succeed (thus freeing the payload) but be treated as an error. When the session was then torn down the payload would be freed again causing a crash. The read function now takes this into account. This change also fixes assumptions made by users of res_http_websocket. There is no guarantee that a frame received from it will be NULL terminated. ASTERISK-24472 #close Reported by: Badalian Vyacheslav Review: https://reviewboard.asterisk.org/r/4220/ Review: https://reviewboard.asterisk.org/r/4219/ ........ Merged revisions 429270 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429272 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429273 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	c17cef1c38	Direct Media calls within private network sometimes get one way audio ... When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media. ASTERISK-24563 Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ ........ Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429196 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1106e8fd0f	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7c9f4c668	ast_str: Fix improper member access to struct ast_str members. ... Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4cea5fd4ba	chan_sip: Fix theoretical leak of p->refer. ... If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ ........ Merged revisions 428117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428118 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428119 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	bc02cbabd9	chan_sip: Fix bug where DTLS configuration from general would copy dtlsenable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d0523b4b3c	chan_sip: Add support for setting DTLS configuration in the general section. ... Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0ddc3bde24	channels/chan_sip: Add improved support for 4xx error codes ... This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426600 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426601 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426602 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ff83ff564c	channels/chan_sip: Support mutltiple Supported and Required headers ... A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426596 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426597 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	86eea19c8f	channels/chan_sip: Respect outboundproxy setting when sending qualify requests ... The outboundproxy setting is currently ignored when sending OPTIONS requests as a result of the qualify setting. This means that if an Asterisk server is unable to send the packet directly to a peer, it is unable to qualify any non-inbound registered peer (e.g. a peer SIP Trunk). This patch grabs the outboundproxy information for a peer when a qualify attempt is being constructed and, if it finds the information, uses it when sending the OPTIONS request. Review: https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close Reported by: Damian Ivereigh patches: outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632) ........ Merged revisions 425818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425819 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425820 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425821 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9e72c74db5	chan_sip: Fix so asterisk won't send reINVITE after a BYE. ... After a reINVITE glare situation, Asterisk would re-send the reINVITE even though the call had been hung up in the mean time. This patch unschedules the reinvite when handling the BYE. ASTERISK-22791 #close Reported by: Paolo Compagnini Tested by: Paolo Compagnini Review: https://reviewboard.asterisk.org/r/4056/ (testcase is in review r4055) ........ Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425297 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425298 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425299 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	d3f525fd8f	chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE. ... If a device re-INVITEs at the same time as the dialog is hung up, and if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would fail to destroy the dialog after a while. This resulted in (most prominently) file handle leaks. (Patch reindented by me.) ASTERISK-20784 #close ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) Reviewboard: https://reviewboard.asterisk.org/r/4052/ (testcase can be found at r4051) ........ Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425069 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425070 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425071 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	chan_pjsip: Fix deadlock when masquerading PJSIP channels. ... Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	c3a7524457	chan_sip: Simplify some unref code by removing unlink_peer_from_tables. ... ASTERISK-22945 #related Reported by: ibercom Patches: asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) ........ Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424182 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424183 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424184 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	841d978a30	chan_sip: Remove excess ref of realtime peer before sip_poke_peer. ... The peer is referenced at the end of sip_poke_peer, it should not get an extra ref before the call to sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: asterisk11.patch uploaded by ibercom (License #6599) Review: https://reviewboard.asterisk.org/r/4031/ ........ Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424177 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424178 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	39fada4dc9	chan_sip: Unref outbound proxy structure on dialog/pvt destruction. ... Make sure outbound proxy refs are always unreffed on dialog destruction. Review: https://reviewboard.asterisk.org/r/4016/ ........ Merged revisions 423800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423801 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423802 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	593455621b	chan_sip: On INVITE retransmission, don't add an extra 503 response. ... INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is retransmitted, asterisk would generate a 503 in addition to the 486. Thanks Torrey Searle for providing a working regression test. ASTERISK-24335 #close Review: https://reviewboard.asterisk.org/r/4003/ Patches: retrans_486_invite.patch uploaded by Torrey Searle (License #5334) ........ Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423721 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423722 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423723 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	2df2d785b7	The assertion that peer was not found on final event ... message was being triggered on configuration reload. This patch changes that case to just return instead. Review: https://reviewboard.asterisk.org/r/3953/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting ... Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bf85018107	CallerID: Fix parsing of malformed callerid ... This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	cee660dadf	chan_sip: Use the server reflexive ICE candidate RTCP port as provided. ... This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) ........ Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	77ddc5b713	chan_sip: Don't use port derived from fromdomain if it isn't set ... If a user does not provide a port in the fromdomain setting, chan_sip will set the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will then get used unilaterally in certain places. This causes issues with TLS, where the default port is expected to be 5061. This patch modifies chan_sip such that fromdomainport is only used if it is not the standard SIP port; otherwise, the port from the SIP pvt's recorded self IP address is used. Review: https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close Reported by: Elazar Broad patches: fromdomainport_fix.diff uploaded by Elazar Broad (License 5835) ........ Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421718 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421719 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421720 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	969982b878	chan_sip: Fix type mismatch when the format is changed. ... Symptom is most likely an invalid ao2 object bad magic number message or a less likely crash. ........ Merged revisions 420881 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	91f7b66183	chan_sip: Mark chan_sip and its files as extended support ... ........ Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a1424a2f1a	chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue. ... Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ ........ Merged revisions 420434 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420435 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420436 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	47bf7efc4d	Multiple revisions 420089-420090,420097 ... ........ r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5bea6c1b1c	chan_sip: complete upgrade to ao2 ... This change upgrades sip_registry and sip_subscription_mwi to astobj2. ASTERISK-24067 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3759/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb87796f67	ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech ... This patch serves two purposes: (1) It fixes some bugs with endpoint subscriptions not reporting all of the channel events (2) It serves as the preliminary work needed for ASTERISK-23692, which allows for sending/receiving arbitrary out of call text messages through ARI in a technology agnostic fashion. The messaging functionality described on ASTERISK-23692 requires two things: (1) The ability to send/receive messages associated with an endpoint. This is relatively straight forwards with the endpoint core in Asterisk now. (2) The ability to send/receive messages associated with a technology and an arbitrary technology defined URI. This is less straight forward, as endpoints are formed from a tech + resource pair. We don't have a mechanism to note that a technology that *may* have endpoints exists. This patch provides such a mechanism, and fixes a few bugs along the way. The first major bug this patch fixes is the forwarding of channel messages to their respective endpoints. Prior to this patch, there were two problems: (1) Channel caching messages weren't forwarded. Thus, the endpoints missed most of the interesting bits (such as channel creation, destruction, state changes, etc.) (2) Channels weren't associated with their endpoint until after creation. This resulted in endpoints missing the channel creation message, which limited the usefulness of the subscription in the first place (a major use case being 'tell me when this endpoint has a channel'). Unfortunately, this meant another parameter to ast_channel_alloc. Since not all channel technologies support an ast_endpoint, this patch makes such a call optional and opts for a new function, ast_channel_alloc_with_endpoint. When endpoints are created, they will implicitly create a technology endpoint for their technology (if one does not already exist). A technology endpoint is special in that it has no state, cannot have channels created for it, cannot be created explicitly, and cannot be destroyed except on shutdown. It does, however, have all messages from other endpoints in its technology forwarded to it. Combined with the bug fixes, we now have Stasis messages being properly forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar), where bar has a single channel associated with it and foo has two channels associated with it. The messages would be forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the applications resource, can: - subscribe to endpoint:PJSIP/foo and get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - subscribe to endpoint:PJSIP and get notifications for channels PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, it never has events itself. It merely provides an aggregation point for all other endpoints in its technology (which in turn aggregate all channel messages associated with that endpoint). This patch also adds endpoints to res_xmpp and chan_motif, because the actual messaging work will need it (messaging without XMPP is just sad). Review: https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	media formats: re-architect handling of media for performance improvements ... In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	f91989d44e	media formats: fix ref leak of peer for mwi subscription ... Holding a reference to the peer during mwi subscriptions resulted in a circular reference because the final event message would not be sent until destruction of the peer. Instead, pass the name of the peer to the event callback so that it can fail gracefully after the peer has gone. ASTERISK-23959 Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged revisions 418636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1ce23d4534	chan_sip: Make progressinband=never really mean 'never' ... progressinband=never in sip.conf is easily defeated if an onward trunk sends a progress indication of its own. This is almost certain to happen if the onward trunk is ISDN or IAX as these technologies send a progress indication even if early media is not required. This progress message is passed to the caller, and causes the "never" option to be rather badly named. This patch changes the behaviour of this setting in the following ways: 1) In sip_write(), do not pass the media unless we have either progressed beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early media is both set-up and wanted. This helps resolve double-ringing on some buggy handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly enabling early media. Avoid sending double ring indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch to also encapsulate the fact that a channel has *sent or received* a 183 Progress indication. This makes the updated code in sip_write() much more simple. Review: https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close Reported by: Steve Davies patches: inband_never_present_early_media2 uploaded by Steve Davies (License 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	97834718c2	Remove many deprecated modules ... Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6e60f5d317	Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 ... ........ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip and chan_pjsip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ Review: https://reviewboard.asterisk.org/r/3686/ ........ Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	44dba37bd1	chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line ... This patch is essentially a backport of a small portion of r397526 from ASTERISK-21981. In that patch, pass through support and format attribute negotiation was added for Opus. Part of that included being more tolerant to whitespace in the fmtp line of an SDP; that part of the patch is being applied here. As the author of the backport pointed out, in SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916 #close Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 417587 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417588 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	365ae7523b	res_http_websocket: Close websocket correctly and use careful fwrite ... When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d171e0b2e9	chan_sip: Fix handling of "From" headers longer than 256 characters ... From headers were processed using a 256 character buffer on the stack. This change replaces that with a heap allocation by ast_strdup. ASTERISK-23790 #close Reported by: uniken1 Tested by: uniken1 Review: https://reviewboard.asterisk.org/r/3669/ Patches: chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 417248 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417249 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417250 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8313964d72	channels/chan_sip: Forbid remote bridging if T.38 is negotiated ... When a framehook is removed - such as the fax gateway framehook - the bridge framework will re-evaluate the bridge mixing technologies to see if it can improve the bridging. When this occurs, get_rtp_info will be called to determine if local or remote bridging can be used. Using remote bridging will cause a fax to fail, as direct media negotiation will cause some small number of packets to not arrive at the remote endpoint. This patch forces local native bridging if T.38 negotiation is in progress or has been established. ........ Merged revisions 416318 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	13e697f8c0	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ca5745dbe	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	5ca495ed2f	chan_sip: Fix order of variables specified in SIPNotify action ... Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415390 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415410 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	d14983dbce	chan_sip: Start session timer at 200, not at INVITE. ... Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ ........ Merged revisions 414620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414628 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d00882108f	res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfers ... PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e81b873fa2	chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ... ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ ........ Merged revisions 413876 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	119599407b	res_pjsip_refer: Add Referred-By header on INVITE for blind transfers. ... Per rfc3892, the Referred-By header in a REFER must be copied into the referenced request (IE. The outgoing INVITE to the transfer target). * Automatically put the Referred-By header in the outgoing INVITE message if the SIPREFERREDBYHDR channel variable is defined with a value. * Made chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance so chan_pjsip has a better chance to interoperate. * Fixed refer_blind_callback() and refer_incoming_refer_request() to not modify the data in the pointer returned by pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data since the calling routine doesn't own the buffer. ASTERISK-23501 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3514/ ........ Merged revisions 413210 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	20750e261b	chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues. ... * Fixed early exit in sip_msg_send() not destroying the message iterator. * Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() tolerant of a NULL iter parameter in case ast_msg_var_iterator_init() fails. * Made ast_msg_var_iterator_destroy() clean up any current message data ref. * Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() use iter instead of i. * Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers(). ........ Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ae21162a69	chan_sip: Add sendrpid trust options ... In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ ........ Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7d26eefce4	chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL URIs ... This patch is a continuation of https://reviewboard.asterisk.org/r/3349/, committed in r412303. It resolves a finding oej had that the phone-context be available in a channel variable separate from SIPDOMAIN. This patch adds that variable as SIPURIPHONECONTEXT. It also allows a local number (or global number specified in the TEL URI) to be used to look up as a peer. (issue ASTERISK-17179) Review: https://reviewboard.asterisk.org/r/3349/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d28af99e65	chan_sip.c: Fix channel staging assertion failure. ... The failing assertion ensures that the final snapshot gets generated so CDR records can get finalized. The only place where a channel staging snapshot flag could be left set is in chan_sip.c:handle_request_bye(). The function could return before clearing the flag because the channel could dissappear while the function had to have the channel unlocked. * Fixed handle_request_bye() channel snapshot staging coverage area to not have a return in the middle of it and be unable to clear the staging flag. * Pushed the channel snapshot staging coverage area into ast_rtp_instance_set_stats_vars() to ensure that the staging is not interrutped. * Made callers of ast_rtp_instance_set_stats_vars() not call it with any channels or channel driver private locks held to eliminate the deadlock potential. The callers must hold references to the passed in channel and rtp objects. * Eliminated sip_hangup() trying to get the bridge peer. It is futile at this point because the channel could never be in a bridge. Review: https://reviewboard.asterisk.org/r/3431/ ........ Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c6a2a513c2	chan_sip.c: Moved some sip_pvt unrefs after their last use. ... * Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end of the function. The unref needs to happen after the last use of the pointer. ........ Merged revisions 412348 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	cc4a0a7fc9	Reverting r411189 so that it can be put up for public review ... --- r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- ........ Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	eed03fc01a	chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests ... This patch adds support for handling TEL URIs in inbound INVITE requests. This includes the Request URI and the From URI. The number specified in the Request URI will be the destination of the inbound channel in the dialplan. The phone-context specified in the Request URI will be stored in the TELPHONECONTEXT channel variable. Review: https://reviewboard.asterisk.org/r/3349 ASTERISK-17179 #close Reported by: Geert Van Pamel Tested by: Geert Van Pamel patches: asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140) asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4f30c7e91f	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output ... This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. (5) The old refcounter implementation in utils/ has been removed. Review: https://reviewboard.asterisk.org/r/3377/ ........ Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	03beadb6e9	internal_timing: Remove the option and always make it enabled if a timing module is loaded. ... The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	fa3a2f8eca	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) ... Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a4890eddfb	chan_sip: Fix incorrect use of timers ... If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) ........ Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411089 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411091 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6d81951f0d	chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted. ... (closes issue ASTERISK-20841) Reported by: Kelly Goedert ........ Merged revisions 411021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411022 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411023 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c300f7e5a8	AST-2014-002: chan_sip: Exit early on bad session timers request ... This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 410308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410311 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	0291965f79	chan_sip: Fix deadlock of monlock between unload_module and do_monitor ... Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ ........ Merged revisions 410224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's ... Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d3ac8b8a0e	chan_sip: Allow static realtime members to be qualified during module load. ... When a static realtime peer with qualify=yes is loaded, Asterisk will fail to send an OPTIONS request due to the lastms being equal to 0. This results in the peer being unable to receive calls from Asterisk because the status is permanently UNKNOWN. This patch allows an OPTIONS request to be sent during module load by ignoring the lastms value on startup only. Review: https://reviewboard.asterisk.org/r/3294/ (closes issue ASTERISK-17523) Reported by: Maciej Krajewski Tested by: wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112) ........ Merged revisions 410105 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410106 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410107 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	f5b2f1333f	Minor whitespace change to 'sip show peers' output. ... (closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom ........ Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409473 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409474 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d820685e83	chan_sip: Add precautionary p->owner checks. ... * Add precautionary p->owner checks in sip_hangup(), get_refer_info(), get_also_info(), and interpret_t38_parameters(). * Simplify some tangled logic in get_refer_info(), get_also_info(), and add_rpid(). * Removed some dead code in handle_request_invite(). (closes issue ASTERISK-23323) Reported by: Walter Doekes Patches: issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified) ........ Merged revisions 409207 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	954a3cf26f	chan_sip: Fix crash in ast_channel_hangupcause_set(). ... * Fix crash in ast_channel_hangupcause_set() because p->owner not checked before calling. Regression introduced by the fix for ASTERISK-22621. (closes issue ASTERISK-23135) Reported by: OK (issue ASTERISK-23323) Reported by: Walter Doekes ........ Merged revisions 409156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409157 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409158 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	3cfa1c8826	chan_sip: prevent add_route from adding empty header. ... Fix regression caused by ASTERISK-22582. Empty Route headers were added when the route had a single strict hop. (closes issue ASTERISK-23306) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3236/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f001981862	chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling bridge blind transfer ... This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the NOTIFY request that informs the transferor if the transfer succeeded or failed. This patch also clears said flag from the off nominal NOTIFY paths in the local_attended_transfer code, as once we've sent the NOTIFY request it is safe to send by the BYE request. This was caught by the blind-transfer-accountcode test in the Asterisk Test Suite. (closes issue ASTERISK-23290) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3214/ ........ Merged revisions 408069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	cb4e210773	chan_sip: Isolate code that manages struct sip_route. ... * Move route code to sip/route.c + sip/include/route.h * Rename functions to sip_route_* * Replace ad-hoc list code with macro's from linkedlists.h * Create sip_route_process_header() to processes Path and Record-Route headers (previously done with different code in build_route and build_path) * Add use of const where possible * Move struct uriparams, struct contact and contactliststruct from sip.h to reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was a problem. These moved declares are not used outside of reqresp_parser. * While modifying reqprep() the lack of {} caused me trouble. I added them. * Code outside route.c treats sip_route as an opaque structure, using macro's or procedures for all access. (closes issue ASTERISK-22582) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0fbffdb3b2	chan_sip: Decline image streams on unsupported transports ... This change allows chan_sip to decline individual image streams over unsupported transports in the SDP of the 200 response. Previously, an image stream offer with RTP/AVP as the transport would cause chan_sip to respond with a 488. (closes issue ASTERISK-22988) Reported by: adomjan Original patch by: adomjan ........ Merged revisions 406170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406171 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	778d74cacf	Make sure the maxptime attribute is added to the correct offers. ... ........ Merged revisions 405877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 ... Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	5516cda6af	chan_sip: fix Local From tag on outbound register regression ... In ASTERISK-12117, an improvement to insure consistant local from tags on outbound registrations resulted in an undesirable behavior - caused by leftover unexpired sip_pvt dialogs (with the previous cseq number), resulting in many uncessary REGISTER requests. Instead of significant rework of transmit_register(), this change deletes the dialogs after a 200 OK response indiciating a successful registration, keeping the old dialogs from interfering with normal operation. (closes issue ASTERISK-22946) Reported by: Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/3109/ ........ Merged revisions 405433 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405434 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405435 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	522593f901	Add the missing part of r400140 ... When the patch to add retry-on-forbidden-response was committed, part of the patch for chan_sip was not committed which caused the feature to be entirely nonfunctional. This corrects the code in question. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874 ........ Merged revisions 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405081 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3ccd5dee18	udptl: Dead code elimination. ast_udptl_bridge was not used. ... Removing dead code starting with ast_udptl_bridge() eliminated the code in this change. Note: This code has actually been dead since Asterisk v1.4 when it was first put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e4803bbd9e	Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. ... This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e2630fcd51	channels: Return allocated channels locked. ... This change makes ast_channel_alloc return allocated channels locked. By doing so no other thread can acquire, lock, and manipulate the channel before it is completely set up. (closes issue AST-1256) Review: https://reviewboard.asterisk.org/r/3067/ ........ Merged revisions 404204 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	84e1790beb	bridge_native_rtp: Deadlock during 4-way conference creation ... The change contains a slightly adjusted patch that was on the issue (submitted by kmoore). A fix was made by adding in a bridge lock while calling bridge_start/stop from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. (closes issue ASTERISK-22749) Reported by: Kinsey Moore Review: https://reviewboard.asterisk.org/r/3066/ Patches: lock_inversion.diff uploaded by kmoore (license 6273) ........ Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	90108b15a0	Reset peer outboundproxy on sip.conf reload ... If you set a peer's outboundproxy and then removed it from the config, this would not get picked up in a config reload. This patch fixes that by resetting it in set_peer_defaults(). Closes ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/ ........ Merged revisions 403634 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403635 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1212906351	Reverting r403311. It's causing ARI tests to hang. ... ........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. ... This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	094db82a73	chan_sip: keep same local (from) tag for outgoing register requests ... For outbound register requests the tag on the From line was updated every 20 seconds prior to a successful registration and also once for each registration renewal. That behavior can possibly cause the registration to be denied because of the different tag, and is not aligned with the intention of RFC 3261 8.1.3.5 "... request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request...". This updates chan_sip to have a field to keep the local tag in the registration structure and use that tag for registration requests where the callid is also unchanged. (closes issue ASTERISK-12117) Reported by: Pawel Pierscionek Review: https://reviewboard.asterisk.org/r/2988/ ........ Merged revisions 402604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 402605 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402606 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	029ce1e962	chan_sip: Use AST_AF* defined constant when calling ast_get_ip ... While the structure passed to ast_get_ip should be set memset to 0, thus initializing the ss_family member to 0, explicitly setting it to AST_AF_UNSPEC is more portable. ........ Merged revisions 402507 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	fe47684b43	chan_sip: notify dialog info ignores presentation indicator in callerid ... The presentation indicator in a callerid (e.g. set by dialplan function Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies are generated during extension monitoring. Added a check to make sure the name and/or number presentations on the callee (remote identity) are set to allow. If they are restricted then "anonymous" is used instead. (closes issue AST-1175) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2976/ ........ Merged revisions 402450 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402452 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	98dea21bc1	chan_sip: Fix RTCP port for SRFLX ICE candidates ... This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ ........ Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	230141d677	chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers" ... While looking at ASTERISK-22236, Walter Doekes pointed out that when running "sip show peers", the setting being displayed can be confusing. The display of "N" used to mean NAT (i.e. yes). The NAT setting has gone through many different changes resulting in the display of different characters to try and convey what the current setting is for 'Forcerport' (A for Auto and Forcerport is currently on, a for Auto but Forcerport is off, Y for yes, and N for no). During the initial code review to try and clarify these settings (especially since "N" no longer meant what it used to mean in prior versions of Asterisk), Mark Michelson suggested using the full space available to display the settings which helped to make the settings very clear. That was a great suggestion. Therefore, this patch does the following: * The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No. * A column for the 'Comedia' setting has been added. It too will display the setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No. * UPGRADE.txt has been updated to document this change. (closes issue ASTERISK-22728) Reported by: Walter Doekes Tested by: Michael L. Young Patches: asterisk-forcerport-display-clarification_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2941 ........ Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402112 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	9ba7742431	chan_sip: Allow a sip peer to accept both AVP and AVPF calls ... Adapts the behaviour of avpf to only impact the format of outgoing calls. For inbound calls, both AVP and AVPF calls will be accepted regardless of the value of avpf in the configuration. (closes issue ASTERISK-22005) Reported by: Torrey Searle Patches: optional_avpf_trunk.patch uploaded by tsearle (license 5334) ........ Merged revisions 401884 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401885 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago