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2010-02-04Change channel state on local channels for busy,answer,ring.jpeeler1-0/+32
Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/#9700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did not exten => #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@244785 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-01Revert previous chan_local fix (r236981) and fix instead by destroying ↵tilghman1-3/+1
expired frames in the queue. (closes issue #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt uploaded by tilghman (license 14) 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: kobaz, atis (closes issue #16581) Reported by: ZX81 (closes issue #16681) Reported by: alexr1 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@244070 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-28Fix a bogus third argument to ast_copy_string().russell1-4/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243779 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-22Initialize notify_types to NULLoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@242226 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Fix deadlock in agent_read by removing call to agent_logoff.jpeeler1-1/+22
One must always lock the agents list lock before the agent private. agent_read locks the private immediately, so locking the agents list lock is not an option (which is what agent_logoff requires). Because agent_read already has access to the agent private all that is necessary is to do the required hanging up that agent_logoff performed. (closes issue #16321) Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler (license 325) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@241227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07fixes crash in "scheduled_destroy" in chan_iaxdvossel1-1/+1
A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238411 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Change in sip show channels display format allowing more digits for CIDdvossel1-3/+3
(closes issue 0016459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238409 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04It's also possible for the Local channel to directly execute an Application.tilghman1-1/+1
Reviewboard: https://reviewboard.asterisk.org/r/452/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@237318 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-02Release memory of the contact acl before unloading moduleoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@237135 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-30Don't queue frames to channels that have no means to process them.tilghman1-1/+3
(closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236981 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-23Properly set T.38 attributes and don't return before T.38 ports are ↵mnicholson1-31/+29
configured when T.38 is found but no audio stream is found. (closes issue #16318) Reported by: bird_of_Luck Patches: t38-sdp-parsing-fix3.diff uploaded by mnicholson (license 96), written by vrban and mnicholson Tested by: vrban, mihaill git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236261 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-22fixes issue with p->method incorrectly set to ACKdvossel1-2/+9
It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236062 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-14Stop sending 183's after call hangup.oej1-0/+1
There where still cases where the 183 keep-alive mechanism would not stop sending 183's even though the Asterisk server had sent a final reply to the invite. EDVX-28 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234492 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10When we receive no response at all to our INVITE, allow the channel to be ↵tilghman1-3/+7
destroyed. (closes issue #15627) Reported by: falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14) 20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14) Tested by: falves11 Review: https://reviewboard.asterisk.org/r/446/ (closes issue #15716) Reported by: dant (closes issue #16270) Reported by: corruptor (closes issue #15356) Reported by: falves11 (issue #16382) Reported by: lftsy git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234095 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07fixes missing Contact header angle bracketsdvossel1-1/+3
(closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233471 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Allow SDP packets with only video session information.mnicholson1-1/+1
(closes issue #16387) Reported by: zalex1953 Tested by: mnicholson, zalex1953 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233392 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Do not modify the gain settings on data calls.jpeeler1-1/+5
(The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-25fixes conditional jump or move depending on uninitialised STACK valuedvossel1-2/+2
(closes issue #16261) Reported by: edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231233 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23When 'sip set debug' is enabled, and the last line of an incoming SIP messagekpfleming1-0/+2
is not properly newline terminated, ensure that that line is included in the debug output. (part of issue #16268) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230875 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Correct fix for issue #16268... the reporter's original patch was very close ↵kpfleming1-10/+15
to correct. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230839 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Ensure that SDP parsing does not ignore the last line of the SDP.kpfleming1-4/+5
(closes issue #16268) Reported by: sgimeno git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230772 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-15Correct mistaken option name in error message.kpfleming1-1/+1
The configuration option for allowing hosts to make non-token-based calls is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230246 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Respect the maddr parameter in the Via header.file1-2/+13
(closes issue #14446) Reported by: frawd Patches: via_maddr.patch uploaded by frawd (license 610) Tested by: frawd git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230144 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Fix a crash caused by two threads thinking they should both free thefile1-1/+1
chan_local private structure when only one should. (closes issue #15314) Reported by: sroberts Patches: Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780) Tested by: davidw, lottc git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230038 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10don't crash on log message in solarisdvossel1-1/+1
AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229167 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10Reverted revision 202022.mnicholson1-14/+2
(closes issue #16175) Reported by: paul-tg git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229091 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Don't overwrite caller ID name on a trunk with the configured fullname when ↵file1-0/+6
using users.conf (issue ABE-1989) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228547 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Fix crash on VPB exception when no hardware is present.qwell1-1/+1
(closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228079 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05chan_misdn Asterisk 1.4.27-rc2 crashdbrooks1-0/+1
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Modify the SDP parsing code to parse session and media level items separately.mnicholson1-379/+490
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd Tested by: frawd, mnicholson, file Review: https://reviewboard.asterisk.org/r/385/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227758 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Fix a security issue where sending a REGISTER with a differing username in ↵file1-2/+0
the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227700 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Make sure the outgoing flag is cleared if a new channel fails to get created ↵rmudgett1-0/+3
for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227275 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Fix a bug where an RPID header could be generated with a blank username in ↵file1-1/+1
the URI. (closes issue #15909) Reported by: kobaz git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227166 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Fixing bug before someone reports it...oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Adding IP address in Contact ACL log message and removing redundant messageoej1-2/+1
(based on kpfleming's feedback) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227089 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Use proper response code when violating Contact ACL's.oej1-2/+15
Review: https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02SIP channel name uniquenessdbrooks1-1/+3
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226972 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02fixes crash on iterator_destroy on uninitialized iteratordvossel1-1/+0
(closes issue #16162) Reported by: krn git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226736 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02changes calltoken debug messages from LOG_NOTICE to LOG_DEBUG like they are ↵dvossel1-6/+6
supposed to be (closes issue #16144) Reported by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226688 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-29Add an option to enabling passing music on hold start and stop requests ↵file1-2/+5
through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226531 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21IAX2: VNAK loop caused by signaling frames with no destination call numberdvossel1-10/+77
It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225243 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21IAX/SIP shrinkcallerid optiondvossel2-10/+32
The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225032 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17Fix stale caller id data from being reported in AMI NewChannel eventjpeeler1-2/+9
The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224330 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-16Never released PRI channels when using Busy() or Congestion() dialplan apps.rmudgett1-4/+12
When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224260 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-13Fix PRI timer T309 operationjeang1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223955 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Remove automatic switching from T.38 to voice mode in chan_sip.kpfleming1-2/+1
chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover. (issue #16025) Reported by: jamicque git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09fixes sip registration using authuser in user.confdvossel1-3/+11
(closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223205 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09'auth=' did not parse md5 secret correctlydvossel1-12/+10
(closes issue https://issues.asterisk.org/view.php?id=15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223142 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Fix memory leak if chan_misdn config parameter is repeated.rmudgett1-4/+15
Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222797 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07chan_misdn.c:process_ast_dsp() memory leakrmudgett1-2/+6
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222691 f38db490-d61c-443f-a65b-d21fe96a405b
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