asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	d97e6c1401	Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	7d9e0158c3	AST-2012-010: Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e0883154cf	Re-fix how local tag is generated when sending a 481 to an INVITE. Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	bf6ef69702	Allow chan_sip to decline unwanted media streams This change replaces the static array of four representable media streams with an AST_LIST so that chan_sip can keep track of offered media streams. This allows chan_sip to deal with offers containing multiple same-type streams and many other situations without rejecting the SDP offer in its entirety, yet still generating a valid response. This also covers cases where Asterisk can not comprehend the offer if it is in the correct format. Previously, chan_sip would reject SDP offers or entirely ignore individual stream offers in an effort to be more compatible which would often result in invalid SDP responses. Review: https://reviewboard.asterisk.org/r/1988/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	14a985560e	Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	463f9d729a	Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. Review: https://reviewboard.asterisk.org/r/1954 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f454dceaf3	Re-add LastMsgsSent value for SIP peers Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. The value was previously removed for Asterisk 10; however, since it was still present in Asterisk 1.8 and still useful for reporting purposes, it was decided to re-add the value. This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show peer [peer]' command, and makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 ........ Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	cd37bec058	logger: Adds additional support for call id logging and chan_sip specific stuff This patch improves the handling of call id logging significantly with regard to transfers and adding APIs to better handle specific aspects of logging. Also, changes have been made to chan_sip in order to better handle the creation of callids and to enable the monitor thread to bind itself to a particular call id when a dialog is determined to be related to a callid. It then unbinds itself before returning to normal monitoring. review: https://reviewboard.asterisk.org/r/1886/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ae1502be33	Add support for lightweight NAT keepalive. If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	9ac6938e09	Missed lastinvite CSeq int to uint32_t change from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	299dd5d4fc	Adds an option to sip.conf that prevents diversion headers from being added. send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ebe2c33b72	Fix worker thread resource leak in SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356690 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	e5c51ee44c	Add auto_force_rport and auto_comedia NAT options This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	3342183016	Add callbackextension matching & realtime callbackextensions This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	0ccc1f5274	Merged revisions 353321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 \| alecdavis \| 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) \| 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 \| alecdavis \| 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) \| 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	e0ca02fe21	Merged revisions 352863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352863 \| alecdavis \| 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) \| 19 lines Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 \| alecdavis \| 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) \| 12 lines rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	19a4928fee	INFO/Record request configurable to use dynamic features Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	03596bcb47	chan_sip autocreatepeer=persist option for auto-created peers to survive reload This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	113612b9d6	Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c225800646	Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04f04ff39d	Don't treat a host:port string as a domain The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e28d306387	Make room for the fax detect flags The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343277 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7f883ef495	Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10de040b6e	More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c4a7d0e2c7	Merged revisions 338417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 \| irroot \| 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) \| 19 lines Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 \| irroot \| 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) \| 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8493c46308	Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 \| irroot \| 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) \| 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ea5262eeb5	Merged revisions 332042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 \| mnicholson \| 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) \| 2 lines fix a code comment AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1858e274e3	Merged revisions 332027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 \| mnicholson \| 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) \| 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 \| mnicholson \| 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) \| 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	5c627eba2b	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Richard Mudgett	14d510c5b7	Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 \| rmudgett \| 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) \| 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1d93d217f0	Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 \| kmoore \| 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) \| 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	0bd877621e	Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f90bc95f0d	Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 \| jrose \| 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) \| 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	32d43ebe19	When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	48a9694ed0	Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 \| mnicholson \| 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) \| 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 \| mnicholson \| 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) \| 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 \| mnicholson \| 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) \| 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 \| jpeeler \| 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) \| 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	1b2039e7db	Merged revisions 285006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 \| dvossel \| 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) \| 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d17eded2e9	Merged revisions 284950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 \| dvossel \| 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) \| 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	848135748f	Merged revisions 283559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 \| dvossel \| 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) \| 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 \| dvossel \| 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) \| 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bcf5988caf	Merged revisions 283493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 \| dvossel \| 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) \| 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	1990c4347e	Merged revisions 281650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 \| russell \| 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) \| 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4a98994542	Merged revisions 279568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279568 \| dvossel \| 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) \| 21 lines transaction matching using top most Via header This patch modifies the way chan_sip.c does transaction to dialog matching. Asterisk now stores information in the top most Via header of the initial incoming request and compares that against other Requests that have the same call-id. This results in Asterisk being able to detect a forked call in which it has received multiple legs of the fork. I completely stripped out the previous matching code and made the comparisons a little more explicit and easier to understand. My comments in the code should offer all the details involving this patch. This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to find multiple dialogs with the same call-id. Since the callback function was returning (CMP_MATCH \| CMP_STOP) only the first item found was being returned. I fixed this by making a new callback function for finding multiple dialogs that only returns (CMP_MATCH) on a match allowing for multiple items to be returned. Review: https://reviewboard.asterisk.org/r/776/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	57a92a6a7c	Allow IPv6 addresses for UDPTL streams. Review: https://reviewboard.asterisk.org/r/795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cb5892bb67	Fix port setting of external address in SIP. There are two changes here: 1. Since the externip setting can now have a port attached to it, calling it "externip" is misleading. The option is now documented and parsed as "externaddr." This also extends to the "matchexterniplocally" setting. It is now documented and parsed as "matchexternaddrlocally." The old names for the options may still be used, but they are no longer used in the sip.conf.sample file. 2. If no port is set for the externaddr, and UDP is the transport to be used, then we will set the port of the externaddr to that of the udpbindaddr. This was how things worked prior to the IPv6 merge, so this is a regression fix. (closes issue #17665) Reported by: mmichelson Patches: 17665.diff#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	e129b31fc6	Add ability to configure the Max-Forwards header in the dialplan, as well as in sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	23b6e621d2	chan_sip: RFC compliant retransmission timeout Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	243d87038e	correct handling of get_destination return values A failure when calling the get_destination can mean multiple things. If the extension is not found, a 404 error is appropriate, but if the URI scheme is incorrect, a 404 is not approperiate. This patch adds the get_destination_result enum to differentiate between these and other failure types. The only logical difference in this patch is that we now send a "416 Unsupported URI scheme" response instead of a "404" when the scheme is not recognized. This indicates to the initiator of the INVITE to retry the request with a correct URI. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	8a07dbf95d	rfc compliant sip option parsing + new unit test RFC 3261 section 8.2.2.3 states that if any unsupported options are found in the Require header field, a "420 (Bad Extension)" response should be sent with an Unsupported header field containing only the unsupported options. This is not currently being done correctly. Right now, if Asterisk detects any unsupported sip options in a Require header the entire list of options are returned in the Unsupported header even if some of those options are in fact supported. This patch fixes that by building an unsupported options character buffer when parsing the options that can be sent with the 420 response. A unit test verifying this functionality has been created. Some code refactoring was required. Review: https://reviewboard.asterisk.org/r/680/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9bbeb945e8	Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 \| mnicholson \| 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) \| 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a1fe641a38	retransmit response to BYE requests until timer J expires According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T164) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T164 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8b790e4f06	Mailbox list would previously grow at each reload, containing duplicates. Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	77a96c5a93	do all sip registry parsing before transmit_register This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c7303d840e	Add support for direct media ACLs directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address. In some networks not all phones are fully routed, i.e. not all phones can ping each other. This patch adds a way to restrict directmedia for certain peers between certain networks. (closes issue #16645) Reported by: raarts Patches: directmediapermit.patch uploaded by raarts (license 937) Tested by: raarts Review: https://reviewboard.asterisk.org/r/467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ba9b0d95e6	Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	9e1b6c7236	Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	4b8f1c8cac	Add routines for parsing SIP URIs consistently. From the original issue report opened by Nick Lewis: Many sip headers in many sip methods contain the ABNF structure name-andor-addr = name-addr / addr-spec Examples include the to-header, from-header, contact-header, replyto-header At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success. I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure Nick has also included unit tests for verifying these routines as well, so...heck yeah. (closes issue #16708) Reported by: Nick_Lewis Patches: reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657 Review: https://reviewboard.asterisk.org/r/549 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	42577406fd	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame can fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f199367b93	Tweak formatting and add minor updates to some comments. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a9932363a9	-----Changes ----- New files - channels/sip/sip.h – A new header for shared #define, enum, and struct definitions. - channels/sip/include/sip_utils.h – sip util functions shared among the all the sip APIs - channels/sip/include/config_parser.h – sip config-parser API - channels/sip/config_parser.c – Contains sip.conf parsing helper functions with unit tests. - channels/sip/include/reqresp_parser.h – sip request response parser API - channels/sip/reqresp_parser.c – Contains sip request and response parsing helper functions with unit tests. New Unit Tests - sip_parse_uri_test - sip_parse_host_test - sip_parse_register_line_test Code Refactoring - All reusable #define, enum, and struct definitions were moved out of chan_sip.c into sip.h. During this process formatting changes were made to comments in both sip.h and chan_sip.c in order to better adhere to the coding guidelines. - The beginnings of three new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h using existing chan_sip.c functions. - parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c along with unit tests for both functions. - sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to config-parser.c along with unit tests for both functions. Changes to parse_uri() -removal of the options parameter. It was never used and did not behave correctly. -additional check for [?header] field. When this field was present, the transport type was not being set correctly. ----- Overview ----- This patch is introduced with the hope that unit tests for all our sip parsing functions will be written soon. chan_sip is a huge file, and with the addition of each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing we begin refactoring chan_sip, starting with the parsing functions. With each parsing function we move into a separate helper file, a unit test should accompany it. I've attempted to lay down the ground work for this change by creating two new parser helper files (config-parser.c and reqresp-parser.c) and moving all shared structs, enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything in Asterisk using unit tests, but string parsing is one area where unit tests make the most sense. By beginning to restructure the code in this way, chan_sip not only becomes less bloated, but Asterisk as a whole will become more stable. Review: https://reviewboard.asterisk.org/r/477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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