asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	fe6dd5b23a	Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f99e9b5bad	Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	91a65d8b3f	Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 \| kmoore \| 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) \| 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e372d326ca	Only send MWI Notify on register if the registration is successful. lastmsgssent was removed from chan_sip and the old behavior of sending a mwi notify on register [except when subscribemwi is set] was restored but this must only happen when registration succeeds. leaking information for unsuccessful registrations is not secure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	cdae7c1714	Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 \| schmidts \| 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) \| 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e00bacf5aa	Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	925bcb92b4	Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 \| twilson \| 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) \| 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	0482a757c3	Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 \| schmidts \| 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) \| 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	287c621613	Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	baa5ff86e5	Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f0d579f1d5	Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 \| mjordan \| 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) \| 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1c6e0d278d	Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 \| rmudgett \| 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) \| 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	cd49ac4da3	Merged revisions 339147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 \| lmadsen \| 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) \| 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	050f405fbc	Merged revisions 339086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 \| twilson \| 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) \| 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	67749ae7bb	Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	dd73844f72	Merged revisions 338492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 \| lmadsen \| 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) \| 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	540b7a38a9	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/branches/10@338417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	857e4fdb14	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	a31b5ce87e	Allow Setting Auth Tag Bit length Based on invite or config option	14 years ago
Terry Wilson	17124a2510	Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 \| twilson \| 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) \| 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	dd12759127	Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 \| oej \| 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) \| 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	0cfc3c6e9e	Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 \| oej \| 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) \| 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1701266596	Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 \| jrose \| 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) \| 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7c7e697c3a	Merged revisions 335319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 \| oej \| 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) \| 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	0f50837b20	Merged revisions 335259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 \| schmidts \| 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) \| 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4e57652651	Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 \| mjordan \| 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) \| 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	8c5c537997	Merged revisions 334156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 \| mnicholson \| 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) \| 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1180e8d803	Merged revisions 334006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 \| kmoore \| 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) \| 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	02d2a80332	Merged revisions 333836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 \| twilson \| 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) \| 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6ae104a4b2	Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 \| jrose \| 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) \| 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	0059096c5e	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/branches/10@332027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b83a26eefa	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/branches/10@332022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	132a7e75a0	Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 \| dvossel \| 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) \| 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	678ece77d5	Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 \| kmoore \| 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) \| 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9bd1af5e42	Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 \| rmudgett \| 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) \| 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	252a251ac2	Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 \| dvossel \| 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) \| 2 lines Optimization to buffer initialization fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	03fd1aca68	Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 \| dvossel \| 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) \| 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	8cf16c1c15	Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 \| qwell \| 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) \| 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	2fb05fb506	Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 \| seanbright \| 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) \| 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	aacc128812	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Kinsey Moore	af58e75ce1	Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 \| kmoore \| 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) \| 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/2.0@328936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	98c4fee4cb	Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 \| kmoore \| 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) \| 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	c4b8e21079	Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 \| markm \| 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) \| 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ee2096fe55	Make hint watcher callback take const strings for context and exten parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cc8daa1385	Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 \| rmudgett \| 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) \| 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e613fd544	Merged revisions 327211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 \| rmudgett \| 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) \| 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ba1cc98f1a	Merged revisions 326683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 \| mnicholson \| 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) \| 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	14553512ee	Merged revisions 326681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 \| mnicholson \| 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) \| 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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3171 Commits (fe6dd5b23a5c76b836cdd5ef2c387cde954a274b)