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2011-07-05Merged revisions 326411 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-05Merged revisions 326291 via svnmerge from Richard Mudgett
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
2011-07-01Merged revisions 326144 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30Fixes warning message caused by confbridge playback chan not being answered.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325937 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
2011-06-29Merged revisions 325740 via svnmerge from Kinsey Moore
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
2011-06-28Merged revisions 325416 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines Fix random misspelling noticed on asterisk-users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-28Merged revisions 325339 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-28Merged revisions 325212 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-28Fixes issue with video and text not being reinvited correctly with directmediaDavid Vossel
If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-28Don't forget to build the Via when sending MESSAGETerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-27Merged revisions 324914 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-23Merged revisions 324678 via svnmerge from Kinsey Moore
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324708 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-23Merged revisions 324685 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines Fixes sip crash when calling remove_uri_parameters with NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-23Merged revisions 324652 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-22Merged revisions 324491 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line Use correct variable for text SRTP media. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-22Merged revisions 324484 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-22Merged revisions 324481 via svnmerge fromRichard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a reference leak in an error path in sip_msg_send(). ........ r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines Timout or error on INFO or MESSAGE transaction causes call to be lost. When exchanging INFO messages within a call, 4xx error causes the call to be disconnected although RFC 2976 explicitly states that such transactions do not modify the state of the dialog. When exchanging MESSAGE messages within a call, 4xx error causes the call to be disconnected. To provide least surprise, we should not disconnect the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428 Section 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review: https://reviewboard.asterisk.org/r/1257/ Review: https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-22Merged revisions 324479 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line Comments and whitespace in chan_sip.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-21Fixes issue with finding correct extension when message context is used.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-20Merged revisions 324237 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines Ignore media offers with a port of 0 Section 5.1 of RFC3264 states: A port number of zero in the offer indicates that the stream is offered but MUST NOT be used. (closes issue ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch uploaded by jacco (license 1277) Tested by: jacco, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-17Merged revisions 324174 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines Add header string to libpri debug output. Add header string to libpri debug output so the libpri output can be found/extracted easier from huge debug trace files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-16Merged revisions 324048 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-14Merged revisions 323370 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-14Merged revisions 323371 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-14Store sip peer name as var data on a outofcall msg.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323325 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
2011-06-10Merged revisions 323040 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-09Merged revisions 322807 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-09Add autoanswer to skinny.Damien Wedhorn
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-08Remove skinny do_monitor and use ast_sched_start insteadDamien Wedhorn
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-08Merged revisions 322322 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-07Correct some whitespace and a reference debug message.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-03Merged revisions 321812-321813 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-02Fix message destination extension.Russell Bryant
Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 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
2011-05-31Merged revisions 321515 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-31Merged revisions 321511 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-27Merged revisions 321273 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-26Merged revisions 321155 via svnmerge from Mark Murawki
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-26Merged revisions 321100 via svnmerge from Mark Murawki
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-26Merged revisions 320947 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines Remove some variables that were set but unused. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320883 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC request has incorrect To/From SIP headers. They are reversed and the dialog tags are the same when they should not be. If pedantic mode was disabled, then the cancel would have succeeded despite the incorrect message. * The SIP_OUTGOING flag was not set correctly for the dialog and I had to move some CC subscribe handling code as a result. * Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes in and the CC instance is not found, the 404 response was duplicated. JIRA AST-568 JIRA SWP-3493 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-23Merged revisions 320504 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines Fixes segfault occuring in chan_sip.c at __set_address_from_contact Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve which is where the segfault was occuring due to null str. (closes issue #18857) Reported by: sybasesql Review: https://reviewboard.asterisk.org/r/1225/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 320180 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines This commit modifies the way polling is done on TLS sockets. Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320181 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
2011-05-18Merged revisions 319654 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-18Merged revisions 319552 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-17Merged revisions 319469 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines Merged revision 319468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-17Remove extraneous line variables.Damien Wedhorn
The vars were either explicitly or implicitly not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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