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2010-04-02Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()russell1-24/+8
(closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256014 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-30Don't kill Asterisk if the H323 listener does not start.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255409 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve a number of FreeBSD build issues.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253631 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-17Revert API change in release branchestwilson4-8/+8
This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-13Merged revisions 252089 via svnmerge from twilson4-28/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252175 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Forward declaring dahdi_pri was already done.rmudgett1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251997 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Make chan_dahdi wakeup_sub() prototype not conditional.rmudgett1-5/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251986 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Make sure to clear red alarm after polarity reversal.jpeeler1-0/+12
From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250480 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03fixes problem with duplicate TXREQ packetsdvossel1-3/+8
When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250394 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Modify queued frames from local channels to not set the other side to upjpeeler1-30/+1
In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249536 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27overlap receiving: automatically send CALL PROCEEDING when dialplan startsalecdavis1-1/+14
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis (closes issue #16789) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249365 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27add a reference to the now-published IAX2 RFCkpfleming1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249234 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26For T.38 reINVITEs treat a 606 the same as a 488.mmichelson1-0/+2
(closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249100 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-23fixes invite with replaces deadlockdvossel1-14/+46
(closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248396 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revision 247904 fromrmudgett1-27/+5
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247910 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-09Fixes iaxs and iaxsl size off by one issue.dvossel1-3/+3
2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@245792 f38db490-d61c-443f-a65b-d21fe96a405b
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
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