asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	f50bb3bfa4	SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7b4eeed257	Add basic support for handling connected line-related UPDATE requests. SIP purists may want to look the other way... When COLP/CONP support for SIP was committed, there was a condition under which Asterisk may transmit a SIP UPDATE in order to communicate the change in connected line information. The issue here is that while we could send a SIP UPDATE message, we were not prepared to receive such an UPDATE and would always responde with a 501 when we received an UPDATE. The situation was a bit rough. We really want to be able to receive UPDATEs having to do with connected line changes, but the amount of effort involved in properly supporting RFC 3311 was staggering. This commit represents a compromise. First, it was decided that it is important to only send a SIP UPDATE to an endpoint that is able to handle one. So, now we have added parsing of the Allow header into SIP. We store the allowed methods on SIP peers so that when we communicate with them, we already will know what we can and cannot send to them. We will parse the peer's allowed methods when he registers with us. If the peer is not the type to register with us, but the qualify option is enabled, then we will use the response to the OPTIONS request we send the peer to determine the peer's allowed methods. When the peer's registration expires, or when qualify deems the peer to be unreachable, we clear the allowed methods from the peer. For an actual call, we will copy the peer's allowed methods to the sip_pvt representing the call leg. If we are communicating with an endpoint which is not a peer, then we will just parse the Allow header from the first message we receive during the call and store the information in the sip_pvt. If, during communication with a peer, we receive a 501 response, then we will make sure to save the fact that we cannot use that method when communicating with that peer. Now, with all that infrastructure in place, the only actual place we use this information currently is when attempting to send a connected line change using an UPDATE request. If we cannot send the change immediately using an UPDATE, we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon as it is allowed. The second part of the changes here is for Asterisk to accept UPDATE requests that have connected line changes. Since we are not fully supporting RFC 3311, Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead, if you are communicating with what you know to be another Asterisk box, you may set the rpid_update parameter in sip.conf so that we will send UPDATEs to that Asterisk box. When we send a connected line update, we set a custom header called "X-Asterisk-rpid-update." On the receiving end, if Asterisk receives an UPDATE that does not have the "X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501 since media-changing UPDATEs are not supported. We should never get such UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow header. If the custom header is present in the received UPDATE, though, then we will check the incoming request for connected line updates and queue the update on the channel where the change occurred. ABE-1840 ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a6adc84e69	SIP option to specify outbound TLS/SSL client protocol. chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3b68be6aaa	Remove nonexistent option from sip.conf.sample. The option to choose which connected line header to use is not 'rpid_header' but 'sendrpid' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4d74179f20	Add a new option, mwi_from, to sip.conf. This allows for you to change the From header for outgoing MWI NOTIFY requests. Prior to this, the best you could do was to set a callerid in the general section of sip.conf. The problem was that this was used for all outbound requests, not just MWI NOTIFY requests. AST-201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08971ce205	Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 \| tilghman \| 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) \| 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 \| tilghman \| 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) \| 2 lines Fix for AST-2009-003 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	da2230adf0	SIP preferred codec only feature Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1ae8e9f3b	Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	0685c4b281	Update documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	aca43d126a	Add some more notes about device matching. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2c4f19eb2c	Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 \| oej \| 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) \| 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	d4736e9897	Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	453b4cb8fb	Allow specifying a port number in the user portion of a register => line in sip.conf With this commit, a register => line in sip.conf may contain a port number in the "user" section of the line. Please see CHANGES and sip.conf.sample for more details regarding this. (closes issue #14198) Reported by: Nick_Lewis Patches: chan_sip.c-domainport2.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	91192e30c5	This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fd62012a31	Qualify trumps poke per lmadsen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	92a4edc593	Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time. (closes issue #13217) Reported by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	f9b6507796	If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated. Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes\|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7bd3ce358b	If you enabled 'notifycid' one of the limitations is that the calling channel is only found if it dialed the extension that was subscribed to. You can now specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as it's value implies, ignore the current context of the caller when doing the lookup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6ba4e7853e	Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID of the calling party when subscribed to the state of an extension that is ringing. This has some limitations which are documented in sip.conf.sample. (closes issue #13827) Reported by: seanbright Patches: issue13827.patch uploaded by seanbright (license 71) Reviewed by: russellb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	007807bf41	Updating docs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	d3517de987	Spaces to replace tabs... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	204845843e	Adding a separation of remote authentication and our authentication. remotesecret => our password for a remote service secret => our authentication when someone calls us Secret => still has both functions if remotesecret is not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	0327f37d34	The default in chan_sip for notifyringing is yes, so update the sample conf to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f6c78aa0fe	whistle git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	cebd2c1df2	Add support for subscribing to a voice mailbox on a remote SIP server and making the new/old message count available to local devices. (issue #AST-77) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	aada13230f	Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 \| tilghman \| 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) \| 11 lines Create rules for disallowing contacts at certain addresses, which may improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8b6dd2ad43	Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 \| tilghman \| 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) \| 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	35a37e6724	Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 \| russell \| 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) \| 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6cb6583475	SIP should use the transport type set in the Moved Temporarily for the next invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	853f6a8b3e	Move implementation of an attended-transfer-complete sound from one channel driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5a1d90e1fb	Additional option for videosupport (always) that disables the optimization to fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	e18e813814	- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	638234f146	- Fixing issues with "sip show settings" - Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0fd94cb93d	Make TCP disabled by default (it's considered experimental) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	90098f3cc9	Reformatting the config sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	1b07e87538	Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	1626397996	Merged revisions 126844 via svnmerge from	18 years ago
Brett Bryant	12d5cebea2	Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	48a9e5cada	Merged revisions 123883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 \| tilghman \| 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) \| 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e4d1b39bd8	Merged revisions 118646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 \| file \| 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) \| 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	f4237076bf	Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	41fd7a6a21	(closes issue #6113 ) Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	5fb4b1bbe5	This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	738e4ec94e	Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer. (closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a567b41083	Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0de4eba640	Add manager peerstatus events when peer can't authenticate. (closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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238 Commits (be4798f0b3afd2c8732b6766a5832cf94ef7617a)