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2009-08-21fixes sip register parsing when user@domain is useddvossel1-13/+30
(issue #15008) (issue #15672) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213635 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21increment the mfcr2 monitor count when clearing the call requestmoy1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213454 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-20fixed bug caused by calling ast_request without calling ast_call on an R2 ↵moy1-1/+8
channel, ie, CHANISAVAIL git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213216 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Better parsing for the "register" linetilghman1-68/+126
Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19If we have realtime caching enabled, 'sip reload' must purge users/peers, ↵tilghman1-1/+1
even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213093 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Add COLP support to chan_dahdi/sig_pri.rmudgett2-62/+543
Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP support is now available for any switch for which libpri supports COLP (currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/340/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213007 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Merged revisions 212727 via svnmerge from rmudgett1-26/+28
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212758 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Correct spelling of AGENTACCEPTDTMF in chan_agent.seanbright1-1/+1
(closes issue #15668) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212581 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212498 via svnmerge from jpeeler1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212506 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212430 via svnmerge from rmudgett2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212431 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-14Add braces where missing and a few whitespace fixes in sig_analogjpeeler1-301/+455
(closes issue #15678) Reported by: alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212291 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-14More code that somehow got left out of sig_analogjpeeler3-18/+119
* confirmanswer option now respected * check and set waiting for dialtone timer * unneeded needcallerid flag removed from analog_subchannel * ss_astchan does not need to be a void pointer * swap_channels callback updated to trunk * analog_hangup now resets channel to default law git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212287 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Send a generic return result when we receive a CallDeflection facility ↵rmudgett1-0/+9
message in chan_misdn. ETSI 300-196 implies that a facility return result without arguments does not have the operation-value. This fact implies for ETSI that you can only use the invoke-id to match requests with responses. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Ensure that T38FaxVersion is put into outgoing SDP in the proper case.kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212113 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Check an actual populated variable when seeing if we need to do video or not.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212067 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12This patch adds support for choosing a realm based on the domain in the From ↵mnicholson1-2/+87
or To header in the incoming request. Eligible domains are taken from the domains list in the config file. This functionality is enabled when domainsasrealm is enabled in the config file. (closes issue #11361) Reported by: arkadia Patches: sip_realm_mnich_to_added_2.patch uploaded by arkadia (license 233) Tested by: arkadia git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12Fix chan_dahdi option ringtimeoutjpeeler3-6/+23
dahdi_read relies on the dahdi_pvt copy of ringt which was not getting set in sig_analog. This patch adds a callback to do so. (closes issue #15288) Reported by: alecdavis Patches: chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211908 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12Make asterisk handle 423 Interval Too Short messages better.mnicholson1-3/+6
This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten. (closes issue #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211876 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12Always specify which RTP engine is desired for a new RTP instance.russell6-7/+7
This fixes a crash reported in #asterisk-dev where chan_mgcp unexpectedly allocated an RTP instance from res_rtp_multicast, since by not specifying an engine, you get the first one in the list of engines. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211732 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10Encapsulate testing for which signaling styles are used by sig_pri.rmudgett1-70/+143
Created the dahdi_sig_pri_lib_handles() function and SIG_PRI_LIB_HANDLE_CASES macro to simplify testing for which signaling styles are handled by sig_pri. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211675 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman14-139/+145
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10add manager events when a skinny device registers/unregistersmvanbaak1-0/+4
like we have in chan_sip (closes issue #15499) Reported by: arifzaman Patches: 2009072600-skinnymanagerevents.diff.txt uploaded by mvanbaak (license 7) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211475 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10Fix PRI/BRI channels when in alarm condition to only be marked for hangup if jpeeler2-8/+4
T309 is not enabled. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211435 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10Restoring some code to sig_pri. Not sure if it is really needed.rmudgett3-3/+24
Putting some DSP code back into sig_pri that was removed by the chan_dahdi/sig_pri reorganization. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211392 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10Fix retrieval of the port used for the video stream when adding SDP to a SIP ↵file1-10/+10
message. (closes issue #15121) Reported by: jsmith git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211347 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-07Fixed some unsafe down cast pointer operations for sig_pri.rmudgett1-29/+32
You cannot cast the struct dahdi_pvt.sig_pvt pointer to a specific signaling private pointer without first checking that it is in fact pointing to the correct signaling private structure. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211197 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-07Fix static on line when PRI does overlap dialing.rmudgett1-2/+10
The wrong encoding law was used because = was used when it should have been ==. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211191 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-06Miscellaneous minor fixes to sig_analog.rmudgett1-6/+9
* Sanity adjustments to __analog_ss_thread for sig_analog environment. * Deleted some duplicated code. * Fixed analog_ss_thread_start passing the wrong pointer. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210869 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-06Sanity adjustments to pri_ss_thread for sig_pri environment.rmudgett1-2/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210866 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-06Accept additional T.38 reinvites after an initial one has been handled.file1-91/+93
Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210817 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-05Fix potential deadlock issue with USERUSERINFO channel variable.rmudgett1-33/+87
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210732 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-05More changes from chan_dahdi that did not make it into sig_pri.rmudgett3-19/+19
* Q.SIG channel mapping option. * discardremoteholdretrieval option. * libPRI debug defines. * pri_set_overlapdial() now set correctly. * pthread creation of pri_ss_thread now matches. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210696 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-05Merged revisions 210575 via svnmerge from rmudgett1-38/+41
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210640 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-04Fix CALLERID() values for sig_pri on incoming calls.rmudgett3-32/+169
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210387 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-04Removed some dead code.rmudgett1-28/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210353 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Rename 'canreinvite' option to 'directmedia', with backwards compatibility.kpfleming3-31/+32
It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210190 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Changes from chan_dahdi that did not make it into sig_pri.rmudgett2-123/+134
* Moved SUPPORT_USERUSER to sig_pri.c * Fix PRI_DEADLOCK_AVOIDANCE parameter. * Whitespace changes. * Added missing unlock in pri_dchannel():PRI_EVENT_RING case. * Balanced curly braces. * ast_debug/ast_log changes from chan_dahdi. * sig_pri_indicate() should default to return -1 if the indication is not handled. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210154 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Trim trailing whitespace.rmudgett3-100/+100
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-01Merged revisions 209759 via svnmerge from kpfleming3-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209760 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-31Improve chan_sip's ability to determine what methods should and should not ↵mmichelson1-67/+46
be used in a dialog. The previous effort here was to store what a peer is capable of receiving by parsing REGISTER requests from the peer and keeping that information for as long as the registration was active. The problem with this is that there are a great number of SIP devices which give no indication of the methods allowed in their REGISTER requests, and it is unreasonable to try to guess what the device may or may not support. In addition, some SIP devices have been found to claim support for a specific method, but their handling the method is less than ideal, or they are actually lying. With this patch, we now determine what methods a device supports by parsing the Allow header we receive from them, and we do this with each new dialog. In addition, a configuration option has been added so that an administrator can essentially blacklist certain methods from being used with certain peers if the admin knows that support for a specific method is dodgy or nonexistent. ABE-1822 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209673 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Add missing ifdef-s for service maintenance message functionalityjpeeler2-0/+4
(closes issue #15614) Reported by: fabled git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209619 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Fixes numerous spelling errors. Patch submitted by alecdavis.dbrooks3-5/+5
(closes issue #15595) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Fix a crash that can result if text codecs are allowed but textsupport is ↵mmichelson1-1/+1
disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209516 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-28Define side-effect-safe MIN and MAX macros and remove duplicate definitions ↵kpfleming5-28/+0
from various files. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209400 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Fixing typos. Replaces "recieved" with "received" and "initilize" with ↵dbrooks2-4/+4
"initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 208923 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208924 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-25Merged revisions 208746 via svnmerge from jpeeler2-4/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208749 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Merged revisions 208587 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208588 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Resolve a T.38 negotiation issue left over from the udptl-updates merge.kpfleming1-4/+4
The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23Rework of T.38 negotiation and UDPTL API to address interoperability problemskpfleming1-287/+123
Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
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