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2009-07-16Merged revisions 206808 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206809 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Merged revisions 206768 via svnmerge from dvossel1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206775 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Blocked revisions 206767 via svnmergejpeeler0-0/+0
........ r206767 | jpeeler | 2009-07-15 17:02:55 -0500 (Wed, 15 Jul 2009) | 10 lines The dialing flag was mistakingly removed from sig_pri. This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206769 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Merged revisions 206707 via svnmerge from rmudgett2-19/+52
https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206762 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Merged revisions 206702 via svnmerge from dvossel1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206705 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Merged revisions 206636 via svnmerge from seanbright1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Blocked revisions 206566 via svnmergejpeeler0-0/+0
........ r206566 | jpeeler | 2009-07-14 15:01:10 -0500 (Tue, 14 Jul 2009) | 8 lines Restore some missing functionality to sig_analog. The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206600 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Recorded merge of revisions 206567 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206585 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Merged revisions 206489 via svnmerge from rmudgett3-343/+461
https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206555 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Merged revisions 206386 via svnmerge from russell1-3/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206387 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Merged revisions 206341 via svnmerge from rmudgett2-23/+57
https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205985 via svnmerge from dvossel1-5/+27
https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206017 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205939 via svnmerge from kpfleming1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205940 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Fix build.mmichelson1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205880 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205878 via svnmerge from mmichelson1-4/+18
https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205879 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205840 via svnmerge from dvossel1-7/+25
https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205843 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205776 via svnmerge from mmichelson1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205777 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-10Merged revisions 205770 via svnmerge from kpfleming1-9/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205771 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205728 via svn merge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205729 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205696 via svnmerge from kpfleming3-10/+27
https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205697 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205600 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205608 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205479 via svnmerge from dvossel3-27/+34
https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205597 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Blocked revisions 205562 via svnmergemvanbaak0-0/+0
........ r205562 | mvanbaak | 2009-07-09 16:10:01 +0200 (Thu, 09 Jul 2009) | 2 lines make this compile again under devmode ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205563 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205532 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205533 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205412 via svnmerge from dvossel4-34/+35
https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205415 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205350 via svnmerge from mmichelson1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205351 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205291 via svnmerge from qwell2-103/+225
https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205296 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08oops, fixing buildtilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205224 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205216 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205220 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205196 via svnmerge from tilghman1-0/+72
https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205200 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205151 via svnmerge from russell1-29/+29
https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205152 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205120 via svnmerge from russell6-7/+107
https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205139 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08SIP Dialog ref countingdvossel2-83/+169
This patch adds reference counting for sip dialogs into 1.6.0. When proc_session_timer() is called from the scheduler thread it has no guarantee the session timer's dialog won't be freed from underneath it. Now the session timer holds a reference to the dialog, preventing it from being destroyed during the middle of proc_session_timer(). (closes issue #13623) Reported by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205117 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-06Restore Hungarian (mistakenly removed during merge)tilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204980 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-06Merged revisions 204948 via svnmerge from kpfleming1-3/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204949 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-02Merged revisions 204835 via svnmerge from rmudgett1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-02Merged revisions 204710 via svnmerge from dvossel3-61/+212
https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204754 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-01removes fake dialog_unref and dialog_ref function calls.dvossel1-45/+18
dialog_unref() and dialog_ref() in 1.6.0 where only place holders for reference counting once it was implemented. The functions did nothing but return the pointer on ref and NULL on unref. These calls have been removed to make way for a patch that actually does dialog ref counting. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204652 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Merged revisions 204563 via svnmerge from tilghman2-214/+208
https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204581 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Merged revisions 204475 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204476 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Recorded merge of revisions 204470 via svnmerge from tilghman3-35/+42
https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204471 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29Merged revisions 204301 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204302 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29Merged revisions 204247 via svnmerge from mmichelson1-44/+43
https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204248 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29Blocked revisions 204013 via svnmergemmichelson0-0/+0
................ r204013 | mmichelson | 2009-06-29 10:04:39 -0500 (Mon, 29 Jun 2009) | 11 lines Blocked revisions 204012 via svnmerge ........ r204012 | mmichelson | 2009-06-29 10:04:17 -0500 (Mon, 29 Jun 2009) | 6 lines Place unlock of mutex in an else block so that it does not get unlocked twice. (closes issue #15400) Reported by: aragon ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204015 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-27Merged revisions 203909 via svnmerge from rmudgett1-8/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203910 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203853 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203855 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203802 via svnmerge from russell1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203818 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203779 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203780 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203721 via svnmerge from dbrooks1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines Fixing voicemail's error in checking max silence vs min message length Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26reverse whitespace change 203711 that was based on looking at sig_analog ↵jpeeler1-5/+5
(which has about a 1000 line indentation change that is not worth doing here) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203717 f38db490-d61c-443f-a65b-d21fe96a405b
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