aboutsummaryrefslogtreecommitdiffstats
path: root/channels/chan_sip.c
AgeCommit message (Collapse)AuthorFilesLines
2010-10-02Merged revisions 289840 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289841 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289701 via svnmerge from jpeeler1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289702 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01don't iterate through all dialogs to find and delete old subscribesschmitds1-35/+5
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289623 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revisions 289554 via svnmerge from mnicholson1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289555 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-28Merged revisions 289054-289055 via svnmerge from rmudgett1-3/+28
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) | 1 line Break up long ast_manager_event_multichan() event lines. ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) | 1 line Revert stuff not ready for commit in -r289054. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289056 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-27For an INVITE transaction, treat all 2XX responses the same as a 200.dvossel1-3/+15
ABE-2305 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@289023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-27Formatting fixesoej1-12/+25
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288993 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-27Merged revisions 288961 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) | 5 lines Still build SIP, even if res_crypto cannot be built (use, not depend). (closes issue #18062) Reported by: a user on the mailing list ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288962 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288852 via svnmerge from dvossel1-2/+17
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) | 5 lines Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288853 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288821 via svnmerge from dvossel1-2/+19
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) | 4 lines Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3. ABE-2293 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288418 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288419 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288345 via svnmerge from dvossel1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288346 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288159 via svnmerge from tilghman1-9/+18
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288160 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Instead of iterate through all dialogs, add two separte container for ↵schmitds1-37/+90
needdestroy and rtptimeout adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks. both container will be checked on every loop of do_monitor instead of iterate through all dialogs. (closes issue #17912) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/917/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288063 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 287929 via svnmerge from dvossel1-3/+17
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) | 4 lines Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header. ABE-2258 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287930 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 287895 via svnmerge from russell1-13/+19
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) | 10 lines Don't use ast_strdupa() from within the arguments to a function. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287896 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 287893 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Merged revisions 287645 via svnmerge from dvossel1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) | 9 lines Fixes issue with registrations not working properly with pedantic=yes. (closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287646 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incomingoej1-0/+2
SIP messages. Adding error based on RFC 3398 recommendations. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287087 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Merged revisions 286931 via svnmerge from jpeeler1-9/+16
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286939 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Merged revisions 286868 via svnmerge from mnicholson1-3/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286869 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286834 via svnmerge from dvossel1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) | 2 lines Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286835 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286758 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286759 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Merged revisions 286457 via svnmerge from qwell1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286457 | qwell | 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286458 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-11Formatting changes.oej1-23/+45
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@286271 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-08Merged revisions 285568 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-08Merged revisions 285564 via svnmerge from dvossel1-14/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines Fixes interoperability problems with session timer behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ |SITUATIONS | 'session-timer' MODES | |___________|________________________| | | originate | accept | |-----------|------------|-----------| |1. | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | -------------------------------------- (closes issue #17005) Reported by: alexrecarey ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285565 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-07Merged revisions 285455 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | 8 lines Don't automatically add domains for wildcard bindaddrs. (closes issue #17832) Reported by: oej Patches: 17832-wildcard.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-07Merged revisions 285369 via svnmerge from qwell1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) | 7 lines Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress. (closes issue #17831) Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license 4) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285394 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03Merged revisions 285017 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) | 4 lines Call correct lock function as transferer is a sip_pvt not a channel Both functions are #defined to ao2_lock, but still... ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285029 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03Merged revisions 285006 via svnmerge from dvossel1-3/+3
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: http://svn.digium.com/svn/asterisk/trunk@285007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03Merged revisions 284952 via svnmerge from dvossel1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) | 2 lines During OPTIONS authentication, the authpeer does not need to be returned for any reason. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284953 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03Merged revisions 284950 via svnmerge from dvossel1-14/+35
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: http://svn.digium.com/svn/asterisk/trunk@284951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284705 via svnmerge from dvossel1-4/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284706 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284610 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-01Merged revisions 284561 via svnmerge from dvossel1-3/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines During request to dialog matching, verify init_ruri is present before comparing. During request to dialog matching, we attempt a best effort routine for fork detection which requires several elements to be in place. The dialog's initial request uri is one of those elements. Since it is best effort, if the init_ruri is not present for some reason we can not proceed with that routine. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284562 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-01Merged revisions 284477 via svnmerge from twilson1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284479 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-31Merged revisions 284415 via svnmerge from tilghman1-4/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284440 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-31Add trustrpid and sendrpid global values to 'sip show settings'lmadsen1-0/+2
(closes issue #17860) Reported by: jtodd Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10) Tested by: lmadsen, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284315 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-27Merged revisions 284032 via svnmerge from dvossel1-26/+67
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284033 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-26Merged revisions 283692 via svnmerge from dvossel1-2/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283693 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-25Merged revisions 283595 via svnmerge from dvossel1-3/+9
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283596 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-25Merged revisions 283559 via svnmerge from dvossel1-6/+20
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283560 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-25Merged revisions 283527 via svnmerge from russell1-26/+16
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-24Merged revisions 283457 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines Fix issue where TOS is no longer set on RTP packets. Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk. (closes issue #17890) Reported by: elguero Patches: qos_18.diff uploaded by elguero (license 37) Review: https://reviewboard.asterisk.org/r/868 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283458 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-24Merged revisions 283382 via svnmerge from dvossel1-1/+17
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283383 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-19Merged revisions 282895 via svnmerge from dvossel1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282896 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-19Merged revisions 282891 via svnmerge from dvossel1-17/+30
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines fixes sip peer memory leaks in the peer_by_ip table (issue #17798) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282892 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-19Merged revisions 282860 via svnmerge from mnicholson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282861 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-18Merged revisions 282639 via svnmerge from mnicholson1-12/+13
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests. This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received. (issue #17486) Reported by: davidw Tested by: mnicholson (issue #12713) Reported by: davidw Review: https://reviewboard.asterisk.org/r/860/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282640 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-26 11:30:50 +0000