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path: root/configs/sip.conf.sample
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2012-06-01Help mitigate potential reinvite glare scenarios.Mark Michelson
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
2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28Add support for lightweight NAT keepalive.Joshua Colp
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
2012-02-28Adding transport=udp to sample sip.conf - Also changes version of Asterisk ↵Jonathan Rose
1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-27Adds an option to sip.conf that prevents diversion headers from being added.Jonathan Rose
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
2012-02-09Add auto_force_rport and auto_comedia NAT optionsTerry Wilson
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
2012-01-25Redocuments sip types peer, user, friend in sip.conf.sampleJonathan Rose
There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352512 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23INFO/Record request configurable to use dynamic featuresJonathan Rose
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
2011-12-23chan_sip autocreatepeer=persist option for auto-created peers to survive reloadJonathan Rose
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
2011-12-18Correct two flaws in sip.conf.sample related to AST-2011-013.Kevin P. Fleming
* The sample file listed *two* values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348517 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-12Update sample configs to put incoming calls into context public.Richard Mudgett
* Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ ........ Merged revisions 347953 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-21Default to nat=yes; warn when nat in general and peer differTerry Wilson
It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-14Restore SIP DTMF overlap dialing method.Richard Mudgett
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
2011-09-21Merged revisions 337263 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | 1 line Whitespace fixup from SRTP patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 336936 via svnmerge from Gregory Nietsky
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
2011-09-12New sip.conf option for setting default tonezone for channel or individual ↵Olle Johansson
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
2011-09-07Implement the '!' negation element to negate codecs directly in the allow ↵Tilghman Lesher
keyword. This permits the list of codecs to be specified in one configuration line, instead of two or more, generally with the aim of either allowing all codecs with the exception of a few or disallowing most but permitting a few. Review: https://reviewboard.asterisk.org/r/1411/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332022 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30Merged revisions 325935 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-13Addition of "outofcall_message_context" sip.conf option.David Vossel
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-01Support routing text messages outside of a call.Russell Bryant
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
2011-05-20Merged revisions 319938 via svnmerge from Jonathan Rose
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
2011-04-21Merged revisions 314628 via svnmerge from Matthew Nicholson
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
2011-04-13Add 'description' field for CLI and Manager outputLeif Madsen
(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
2011-03-07Merged revisions 309765 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-24Merged revisions 308679 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replacing doc/* and asterisk.pdf with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replacing doc/* with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01SIP Configuration DocumentationAndrew Latham
sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305247 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines Add alternative name for config option. The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-17Merged revisions 302005 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines Document "encryption" option in sip.conf.sample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-17Merged revisions 298773 via svnmerge from Brad Watkins
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) | 10 lines Fix parsing of mwi => lines in sip.conf Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (closes issue #18350) Reported by: gbour Tested by: Marquis, gbour Review: https://reviewboard.asterisk.org/r/1053/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 285006 via svnmerge from David Vossel
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
2010-09-03Merged revisions 284950 via svnmerge from David Vossel
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
2010-08-24Merged revisions 283493 via svnmerge from David Vossel
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
2010-08-19Merged revisions 282740 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282302 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282269 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines res_stun_monitor for monitoring network changes behind a NAT device Review: https://reviewboard.asterisk.org/r/854 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-11Merged revisions 281687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines Fix parsing of IPv6 address literals in outboundproxy (closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-10Merged revisions 281650 via svnmerge from Russell Bryant
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
2010-08-09Merged revisions 281356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6 addresses and the output of netstat. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Merged revisions 280777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 (Tue, 03 Aug 2010) | 8 lines Better documentation related to IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff uploaded by sperreault (license 252) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Reverted r280706 and r280707. Will commit in branch 1.8 and merge to trunk ↵
properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Better documentation related to IPv6.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-19Fix port setting of external address in SIP.Mark Michelson
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
2010-07-19Make ACLs IPv6-capable.Mark Michelson
ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Clarify syntax changesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inOlle Johansson
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
2010-07-08Add IPv6 to Asterisk.Mark Michelson
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
2010-07-06Merged revisions 274283 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines Correct sip.conf.sample comments for prematuremedia option. (closes issue #17513) Reported by: festr Patches: patch uploaded by festr (license 443) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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