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2010-07-06Correct sip.conf.sample comments for prematuremedia option.jpeeler1-6/+8
(closes issue #17513) Reported by: festr Patches: patch uploaded by festr (license 443) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274283 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Add option to not do a call forward on 482 Loop Detectedtwilson2-10/+22
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/1.4@274280 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Fix problem with RFC 2833 DTMF not being accepted.mmichelson1-1/+5
A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue #17571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274157 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Make get_member_status return QUEUE_NO_MEMBERS instead of ↵mnicholson1-4/+6
QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274093 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-05Command 'stop gracefully' doesn't.tilghman2-16/+44
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-05Remove extra line breaks from 'core show config mappings'pabelanger1-3/+1
(closes issue #17583) Reported by: pabelanger Patches: issue17583.patch uploaded by pabelanger (license 224) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273884 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch ↵tilghman9-6395/+19675
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/1.4@273793 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Autoservice loop optimization causes a busy loop, when channels are serviced ↵tilghman1-0/+5
while in hangup. (closes issue #17564) Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273717 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix various typos, reported by Lintiantzafrir6-14/+14
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273640 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02If all members are paused, the wrong status is indicated.tilghman1-0/+7
(closes issue #17576) Reported by: ramonpeek Patches: diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Don't return a partially initialized datastore.russell1-1/+4
If memory allocation fails in ast_strdup(), don't return a partially initialized datastore. Bad things may happen. (related to ABE-2415) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273565 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Allow admin user to join conference without using admin mode and no user pin.jpeeler1-2/+1
Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273474 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Ensure channel placed in meetme in ringing state is properly hung up.jpeeler1-0/+14
An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273354 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Allow the "useragent" value to be restored into memory from the realtime ↵tilghman1-6/+8
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/1.4@273060 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29_Really_ skip the channel... don't just retry for another 200 cycles.tilghman1-1/+1
(Closes issue SWP-1652, ABE-2240) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273057 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Remove properties that were erroneously merged to 1.4 from one of my branches.russell0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273017 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Don't change ownership/group/permissions on run directory, if it already exists.tilghman1-4/+8
(closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272925 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Also trim trailing blanks on #includestilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272922 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Change the way that we read include files, to accommodate for changes in GCC ↵tilghman1-13/+6
4.4. (closes issue #17472) Reported by: seandarcy Patches: config2.patch uploaded by nivan (license 1066) Tested by: nivan git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272921 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Backport applicable parts of test_astobj2.russell1-0/+329
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272881 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Backport unit test API to 1.4.russell10-2/+1270
Review: https://reviewboard.asterisk.org/r/750/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272878 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Decode URI in contact header of 302 response.mmichelson1-0/+1
ABE-2352 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272804 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Force SILENTMAKE where it is needed.russell1-5/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272763 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Backport method of setting SUBMAKE from trunk.russell1-1/+5
By setting the PRINT_DIR variable, SUBMAKE will print out the directories it descends into, which is important for editors (like vim) that watch the build output so that they can take you to the file where an error occurred. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272688 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-25Make the structure of the table specified before match the queries and results.tilghman1-5/+5
(closes issue #17557) Reported by: cmaj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272562 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-24ss_thread calls pri_grab without lock during overlap dialrmudgett1-0/+2
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/1.4@272446 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Send AgentComplete manager events in the event of blind and attended transfers.mnicholson1-0/+29
(closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272367 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23First caller into a dynamic conference now enter pin once.pabelanger1-3/+4
If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272255 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Backport part of revision 136715 to fix callerid in voicemail text files ↵tilghman1-2/+3
(IMAP only). (closes issue #16945) Reported by: mneuhauser git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272147 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER ↵mnicholson1-0/+1
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/1.4@271902 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Allow users to specify a port for dundi peers.mnicholson1-5/+8
(closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271761 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Modify chan_sip's packet generation api to automatically calculate the ↵mnicholson2-75/+61
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271689 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-21Do not use sizeof to calculate size of a heap allocated character array.jpeeler1-60/+62
Change left out from 271399. (closes issue #16053) Reported by: diLLec git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271552 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Check for newly added memory allocation failures gracefully during AEL2 parsing.jpeeler1-25/+69
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271444 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Fix crash when parsing some heavily nested statements in AEL on reload.jpeeler1-3/+6
Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271399 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Remove an unnecessary assignment that causes a DEBUG_THREADS build failure ↵russell1-4/+0
on mac os x. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271340 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Fix a build problem on Mac OS X with DEBUG_THREADS enabled.russell4-19536/+6433
This set of changes was already in trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271339 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-18Eliminate deadlock potential in dahdi_fixup().jpeeler1-2/+2
(This is a backport of 269307, committed to trunk by rmudgett.) 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. (closes issue #17261) Reported by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271335 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ↵mnicholson2-6/+3
ast_dnsmgr_entry struct. (closes issue #15827) Reported by: DennisD Patches: (modified) dnsmgr_15827.patch uploaded by chappell (license 8) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271123 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Need to lock the agent chan before access its internal bits.qwell1-0/+2
Pointed out by russellb on asterisk-dev mailing list. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270980 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fixed typo in macro-pagepabelanger1-1/+1
Reported to #asterisk-dev by a student of jsmith. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270979 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16fixes chan_iax2 race conditiondvossel1-6/+4
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/1.4@270866 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Merged revisions 270658 via svnmerge from twilson1-18/+20
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/1.4@270724 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Variables have always been case-sensitive, so we should not be removing ↵tilghman1-1/+1
case-insensitive matches. Bug reported via the -dev list. See http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270583 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Move information about zonemessages into the [zonemessages] section.lmadsen1-33/+32
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270442 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15fixes FXS port still ringing when answered, as reported by Tzafrir on dev-list.alecdavis1-0/+1
(issue #17067) Reported by: tzafrir Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270404 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Properly play first file in sort list.pabelanger1-4/+9
When using sort=alpha we would always skip the first file in the list first time through. We now check for that properly. (closes issue #17470) Reported by: pabelanger Patches: sort.aplha.patch uploaded by pabelanger (license 224) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/703/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270331 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-12Fix typo in examplepabelanger1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270078 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11For SpeeX, 0 bits remaining is valid and does not need an emitted warning.tilghman1-1/+3
(closes issue #15762) Reported by: nblasgen Patches: issue15672.patch uploaded by pabelanger (license 224) Tested by: nblasgen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269960 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.mmichelson1-3/+22
The issue here was that the frame created when adjusting for PLC had no offset to its audio data. If this frame were translated to another format prior to being sent out an RTP socket, all went well because the translation code would put an appropriate offset into the frame. However, if the SLIN audio were not translated before being sent out the RTP socket, bad things would happen. Specifically, the ast_rtp_raw_write makes the assumption that the frame has at least enough of an offset that it can accommodate an RTP header. This was not the case. As such, data was being written prior to the allocation, likely corrupting the data the memory allocator had written. Thus when the time came to free the data, all hell broke loose. ....Well, Asterisk crashed at least. The fix was just what one would expect. Offset the data in the frame by a reasonable amount. The method I used is a bit odd since the data in the frame is 16 bit integers and not bytes. I left a big ol' comment about it. This can be improved on if someone is interested. I was more interested in getting the crash resolved. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269821 f38db490-d61c-443f-a65b-d21fe96a405b
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