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2011-01-12Don't reject all SUBSCRIBE auth requeststwilson1-3/+3
When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301682 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Fix backwards and broken XML documentation.lmadsen1-2/+2
(closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300520 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300216 via svnmerge from twilson1-14/+20
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300298 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Merged revisions 299194,299198,299220 via svnmerge from mnicholson1-25/+26
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Typos: recieved => receivedtzafrir1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299003 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Merged revisions 297959 via svnmerge from twilson1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297960 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Merged revisions 297603 via svnmerge from jpeeler1-7/+30
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297605 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297185 via svnmerge from oej1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297186 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 297072 via svnmerge from jpeeler1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297073 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Merged revisions 295628 via svnmerge from twilson1-3/+19
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294688 via svnmerge from jpeeler1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294733 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Go off hold when we get an empty reinvite telling us to.mnicholson1-31/+40
(closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293722 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293723 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-22Update the LDIF file for LDAP.lmadsen1-0/+6
The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@292786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291392 via svnmerge from russell1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Make exit from handle_request_do() consistent.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291111 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291109 via svnmerge from rmudgett1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291110 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289797 via svnmerge from jpeeler1-1/+1
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/branches/1.6.2@289798 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289699 via svnmerge from jpeeler1-3/+8
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/branches/1.6.2@289700 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. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Properly handle channel allocation failures duing invites with replaces.mnicholson1-2/+3
ABE-2588 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289553 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288416 via svnmerge from dvossel1-1/+1
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/branches/1.6.2@288417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288343 via svnmerge from dvossel1-2/+3
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/branches/1.6.2@288344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288112 via svnmerge from tilghman1-9/+22
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/branches/1.6.2@288113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286756 via svnmerge from mnicholson1-1/+1
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/branches/1.6.2@286757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Remove "Internal IP" from sip show settings, as it's not at all useful to ↵qwell1-1/+0
display. (closes issue #17840) Reported by: oej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant1-136/+233
pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Change SIP NOTIFY requests to expect a response so authentication will work.jpeeler1-1/+1
This changes the request to be sent with the transmit type XMIT_RELIABLE so that sip_ack doesn't return false and cause the 401 to be ignored in cases where authentication is required. (closes issue #14255) Reported by: zktech git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29fixes wrong SRV query for TLS connectiondvossel1-5/+33
(closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Fix bad behavior of dynamic_exclude_static option in sip.conf.mmichelson1-7/+7
We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279784 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Backport sip_uri_params_cmp() fix from trunk to 1.6.2.mmichelson1-39/+25
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277530 via svnmerge from mnicholson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul 2010) | 11 lines Merged revisions 277497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested. FAX-128 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@277563 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-15Merged revisions 276788 via svnmerge from jpeeler1-35/+33
https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 | jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines Correct not setting the bindport before attempting to open the socket. Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@276809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Merged revisions 276571 via svnmerge from jpeeler1-50/+49
https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 | jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines Fix MWI notification transmission problems over SIP. MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@276572 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10Merged revisions 245192 via svnmerge from russell1-34/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 | mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 lines Remove useless sip options related to hash table size. First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. ........ (merge to 1.6.2 inspired by issue #17553) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275469 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Merged revisions 275249 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines Fix logging message for stale nonce. (closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275260 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274284 via svnmerge from twilson1-11/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines Merged revisions 274280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@274360 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix typos reported by Lintiantzafrir1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273642 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 273078 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010) | 17 lines Merged revisions 273060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273087 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Merged revisions 272805 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun 2010) | 11 lines Merged revisions 272804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines Decode URI in contact header of 302 response. ABE-2352 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272806 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271903 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271904 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271690 via svnmerge from mnicholson1-125/+124
https://origsvn.digium.com/svn/asterisk/trunk ................ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271691 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Merged revisions 270658 via svnmerge from twilson1-21/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r270658 | twilson | 2010-06-15 15:18:04 -0500 (Tue, 15 Jun 2010) | 20 lines Make contactdeny apply to src ip when nat=yes chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 Review: [full review board URL with trailing slash] ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270693 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Merged revisions 268817 via svnmerge from tilghman1-4/+38
https://origsvn.digium.com/svn/asterisk/trunk ........ r268817 | tilghman | 2010-06-07 17:47:13 -0500 (Mon, 07 Jun 2010) | 9 lines Mailbox list would previously grow at each reload, containing duplicates. Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268819 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Merged revisions 266292 via svnmerge from dvossel1-11/+25
https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines fixes crash when creation of UDPTL fails (closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266293 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Merged revisions 266006 via svnmerge from dvossel1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines fixes failed SIP Directed pickup resulting in dead channel (closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@266007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Recorded merge of revisions 265842 via svnmerge from mmichelson1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, 26 May 2010) | 9 lines Re-enable "always" option for videosupport option in sip.conf. (closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265890 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265698 via svnmerge from mmichelson1-16/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 lines Properly use peer's outboundproxy for outbound REGISTERs. The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265699 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merged revisions 265449 via svnmerge from mmichelson1-11/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 lines Allow type=user SIP endpoints to be loaded properly from realtime. (closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 265087 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 lines Be sure to set the sin_family on the proxy when allocating. (closes issue #17157) Reported by: stuarth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265088 f38db490-d61c-443f-a65b-d21fe96a405b
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