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2007-12-19Make the 1.4 builders happy, ensure var is NULL.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93955 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Avoid segfault in chan_iax when peer isn't definedtilghman1-1/+1
(Closes issue #11602) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93949 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18FreeBSD also does not have byte swap functions.qwell1-1/+1
Issue 11586, patch by sobomax. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93764 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18Merged revisions 93667 via svnmerge from tilghman2-9/+93
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93668 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17If a call is received with a called number IE containing nothing go to the ↵file1-1/+6
's' extension. (closes issue #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93250 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17fix some copy-and-paste leftoverskpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17Issue 11574: Add dependencies on res_monitor and res_features. oej4-0/+9
I wonder if Asterisk can run at all without res_features. My guess is that there's propably a lot of more modules and the core that depends on it. Reported by: caio1982 (closes issue #11574) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93182 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,kpfleming1-2/+8
rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@93180 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Up the length of the format on the SIP channel since it can now be rather long.file1-2/+2
(closes issue #11552) Reported by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92937 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread ↵crichter1-1/+1
starting. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92934 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-13Properly initialize polarity statuses, so that they are detected properly.tilghman1-8/+20
Reported by: julianjm Patch by: julianjm (Closes issue #10238) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92815 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-13If a typo is found in a config file, we previous continued on with what was ↵qwell3-103/+115
already loaded. We do not want to do this (see bug below for details). This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded. Isue #10690. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92696 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10It is possible for nativeformats to contain more then one codec, so print ↵file1-2/+3
out multiple ones. (closes issue #11366) Reported by: ovi git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10Avoid reinvite race situations with two Asterisks tryingoej1-5/+36
to reinvite each other in 1.4 and trunk. This patch implements support for the 491 error code that Asterisk 1.4 generates on situations where we get an incoming INVITE and already has one in progress. Thanks to mavetju for reporting and to Raj Jain for an excellent explanation of the problem. Patch by myself. Tested with 8 Asterisk servers connected to each other in a training network. Closes issue #10481 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@92158 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06Add support for accepting and sending T.38 in the initial INVITE.file1-0/+18
(closes issue #9402) Reported by: thdei git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@91439 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03A big one...mmichelson1-0/+1
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90735 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Changing some bad logic when calculating the interdigit timeout.mmichelson1-1/+1
(closes issue #11402, reported and patched by eferro) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90639 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Only log debug messages if debug is enabled.qwell1-33/+54
Closes issue #11416, patch by casper. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90546 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Change the behavior of ao2_link(). Previously, in inherited a reference.russell1-6/+6
Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90348 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Fix locking issues under one legged replaces scenarios.file1-3/+6
(closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90269 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Clear the DTMF buffer if the call times out.mmichelson1-0/+1
(closes issue #11418, reported and patched by eferro) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90231 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27If we get a codec offer using a well-known payload type, but using it for ↵oej1-8/+29
another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89630 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20added RR patch from iroot #10908, thanks.crichter1-5/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89445 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Fix an uninitialized memory read found by valgrindtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89301 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Yet another memory corruption issue.tilghman1-0/+12
Reported by: atis Patch by: tilghman Fixes issue #10923 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89298 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Don't send re-invites during pending INVITE transactions.oej1-6/+19
Patch by one47 - thanks! Closes issue #9305 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89281 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Improve support for multipart messages. Code by gasparz, changesoej1-18/+25
by me (mostly formatting). Thanks, gasparz! Closes issue #10947 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89280 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Fix building on newer systems which require a third arg to open() when using ↵qwell2-2/+2
O_CREAT. Issue 11238, reported by puzzled. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89254 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13If we set a value for qualify, we should actually pay attention to it, ↵tilghman1-2/+2
instead of overriding the value git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89246 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Fix two cases of memory corruption caused by background threads.tilghman1-0/+8
Reported by: atis Patch by: tilghman Fixes issue #10923 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12if we're NT and no number was dialed and overlapdial is set, we wait for the ↵crichter1-0/+11
ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12added restart all interfaces Restart_Indicator, to automatically send a ↵crichter4-29/+41
RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12fixed a state/event issue with overlapdial=yes when no extension matched. ↵crichter2-32/+44
removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12fixed the support for CW and therefore for the reject_cause option.crichter1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89170 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on ↵crichter5-2/+28
a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89169 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Rework of the commit I made yesterday to use the already built-inmmichelson1-27/+14
ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89119 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Do not add a sip: to the beginning of the To URI unless needed.file1-1/+1
(closes issue #10756) Reported by: goestelecom git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89101 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Improve the devicestate logic for multiple devices. If any are available ↵file1-6/+15
then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89099 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Add support for allowing one outgoing transaction. This means if a response ↵file1-2/+5
comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89097 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07If callerid is configured in sip.conf use that for checking the presence of ↵file1-2/+2
an extension in the dialplan. (closes issue #11185) Reported by: spditner git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89095 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07This patch makes it possible for SIP phones to dial extensions defined with ↵mmichelson1-2/+24
'#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89090 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Make it so that if a peer is determined to be unreachable using qualify ↵file1-1/+1
their devicestate will report back unavailable. (closes issue #11006) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89032 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Fix improbable but possible memory leaks in chan_zap.file1-2/+10
(closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88994 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05When traversing the list of channel variables here in transmit_invite(), the russell1-2/+8
asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Fix up some indentation.russell1-26/+26
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88765 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05If a SIP channel is put on hold multiple times do not keep incrementing the ↵file1-1/+1
onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88671 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Make sure we destroy the config structure on configuration failure.qwell1-0/+2
Issue 11163, patch by eliel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88585 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-04Rename ast_string_field_free_pool to ast_string_field_free_memory,rizzo2-8/+8
and ast_string_field_free_all to ast_string_field_reset_all to avoid misuse (due to too similar names and an error in documentation). Fix two related memory leaks in app_meetme. No need to merge to trunk, different fix already applied there. Not applicable to 1.2 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88471 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Make subscribecontext behave as advertised. It will now look for the ↵file1-18/+18
presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88366 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02If an INFO request within a dialog is received with a content length of 0 ↵file1-0/+5
simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88328 f38db490-d61c-443f-a65b-d21fe96a405b
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