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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
2010-09-10Merged revisions 286222 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10 Sep 2010) | 1 line Return -1 if chan_local doesn't support an option ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286223 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286114 via svnmerge from pabelanger1-19/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286117 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286113 via svnmerge from rmudgett1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286116 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286059 via svnmerge from twilson1-0/+67
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant8-402/+528
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
2010-08-03Merged revisions 280811 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines Prevent DAHDI channels from overriding the callerid, once it's been set by the user. (closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280812 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
2010-07-29Implement support for ast_channel_queryoption on local channels. Currently ↵mnicholson1-0/+54
only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280306 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Add missing enum value "unknown" to the SS7 called_nai and calling_nai ↵rmudgett1-0/+4
config options. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280229 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
2010-07-23Two more typos to cancell.tzafrir1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@278934 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22DNID does not get cleard on a new call when using immediate=yes with ISDN ↵rmudgett1-4/+4
signaling. When you are using chan_dahdi ISDN signaling with immediate=yes and a call comes in without a DNID then you get the DNID of a previous call. Chan_dahdi does not touch the DNID field on a new call if it does not have a DNID. Made always copy the DNID from the new call. The patches backport the relevant changes from trunk -r210387. (closes issue #17568) Reported by: wuwu Patches: issue17568_v1.4.patch uploaded by rmudgett (license 664) issue17568_v1.6.2.patch uploaded by rmudgett (license 664) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@278703 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Fix invalid test for rxisoffhook in FXO channelstzafrir1-15/+17
This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@278524 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
2010-07-16Merged revisions 277467 via svnmerge from rmudgett1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500 (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@277485 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
2010-07-07Merged revisions 274639 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010) | 1 line Add missing conditional around chan_dahdi mfcr2_skip_category config parameter. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@274640 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Merged revisions 274595 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500 (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@274627 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
2010-07-06Merged revisions 274281 via svnmerge from tilghman1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010) | 2 lines Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@274282 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-03Merged revisions 273830 via svnmerge from tilghman3-3/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273831 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix typos reported by Lintiantzafrir3-5/+5
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
2010-06-24Merged revisions 272447 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500 (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272459 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272370 via svnmerge from russell1-45/+62
https://origsvn.digium.com/svn/asterisk/trunk ........ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010) | 23 lines Resolve some errors produced during module unload of chan_iax2. The external test suite stops Asterisk using the "core stop gracefully" command. The logs from the tests show that there are a number of problems with Asterisk trying to cleanly shut down. This patch addresses the following type of error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 (iax2_process_thread_cleanup): Error destroying mutex &thread->lock: Device or resource busy For an example in the context of a build, see: http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary purpose of this patch is to change the thread pool shutdown procedure to be more explicit to ensure that the thread exits from a point where it is not holding a lock. While testing that, I encountered various crashes due to the order of operations in unload_module() being problematic. I reordered some things there, as well. Review: https://reviewboard.asterisk.org/r/736/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272371 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
2010-06-18Merged revisions 269307 via svnmerge from jpeeler1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010) | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271338 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270981 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r270981 | qwell | 2010-06-16 16:10:48 -0500 (Wed, 16 Jun 2010) | 11 lines Merged revisions 270980 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270982 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270867 via svnmerge from dvossel1-6/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r270867 | dvossel | 2010-06-16 12:36:51 -0500 (Wed, 16 Jun 2010) | 28 lines Merged revisions 270866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines fixes chan_iax2 race condition There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@270868 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
2010-06-09Merged revisions 269497 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r269497 | russell | 2010-06-09 17:19:20 -0500 (Wed, 09 Jun 2010) | 9 lines Merged revisions 269495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269502 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
2010-06-07Fix some chan_oss build system problems (these changes were already in trunk).russell1-0/+1
This is what caused a bunch of tests to fail on 1.6.2. They expected a console channel driver, but chan_oss was failing to load. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268815 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-06Merged revisions 268495 via svnmerge from tilghman1-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r268495 | tilghman | 2010-06-05 19:37:30 -0500 (Sat, 05 Jun 2010) | 2 lines Finally track down and eliminate the "FRACK! warnings from chan_iax2". ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267537 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r267537 | russell | 2010-06-03 12:31:41 -0500 (Thu, 03 Jun 2010) | 2 lines Don't stop Asterisk if chan_usbradio isn't configured. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267490 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r267490 | russell | 2010-06-03 12:05:30 -0500 (Thu, 03 Jun 2010) | 2 lines Remove a line that was killing Asterisk on startup. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267491 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Merged revisions 267445 via svnmerge from russell1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r267445 | russell | 2010-06-03 09:48:09 -0500 (Thu, 03 Jun 2010) | 2 lines Comment out a rule that likes to run implicitly unnecessarily, breaking builds ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267446 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Merged revisions 267352 via svnmerge from russell2-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r267352 | russell | 2010-06-02 17:46:37 -0500 (Wed, 02 Jun 2010) | 7 lines try to fix some random chan_h323 compilation failures After some debugging, the random chan_h323 build failures appear to be due to complications introduced by some chan_h323 specific build stuff getting triggered during a clean. Simplify this by moving the h323 clean commands down into channels/makefile. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267353 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
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