6037 Commits (d806ae0da0811f26d23fd241ede8840d09cf74e4)

Author	SHA1	Message	Date
Mark Michelson	a4dc276ed9	Merged revisions 204300 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	200f1dc19e	Merged revisions 204243,204246 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	f45133674d	Merged revisions 203908 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203909 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	5606db2224	Merged revisions 203848 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	92f0cdfce7	Ensure the TCP read buffer is fully initialized before handling each packet. ... (closes issue #14452) Reported by: umberto71 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	48f7381af0	Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control. ... (closes issue #8855) Reported by: mikma Tested by: klaus3000, file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	519f1dd7d6	moving debug message from level 0 to 1. ... (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	59c1998d67	Improve T.38 negotiation by exchanging session parameters between application and channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	5ebf0f3c50	Check if polarityonanswerdelay has elapsed before setting a channel as answered ... after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	6fad61406c	make sure chan_dahdi compiles with only libss7 and not libpri installed ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3930f83be6	Picking nits ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	bbfe6967ab	Remove some unnecessary code and update sample config file with respect to GR-303. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	5c7da226e4	New signaling module to handle PRI/BRI operations in chan_dahdi ... This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into sig_pri.c. Functionality in theory should not change (mostly). A few trivial changes were made in sig_analog with verbose messages and commenting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	afa8db54a0	Unmute when we get a dtmfup (we muted on dtmfdown) event. ... This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ae87ba45b5	Add support for multicast RTP paging. ... (closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	ce70b28f38	Insure ring cadence is set for fxs ports ... Moved SETCADENCE ioctl call to before call into new analog signal module to insure that it gets set. (closes issue #15381) Reported by: alecdavis Patches: fix15381.diff uploaded by dbailey (license 819) Tested by: dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c6a986222e	Merged revisions 203115 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	80822297d4	Merged revisions 203036 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0a915a84e6	Merged revisions 202966 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4c07c7a6b2	Ensure the default settings are applied for T.38 when we set it up for a peer. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	2a68d05b96	I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	2ce62b35cf	Make outgoing_colp=2 misdn.conf port parameter not send redirecting or transfer messages. ... If the outgoing_colp parameter is set to not send COLP information, then it does not make sense to send redirecting or transfer messages announcing new COLP information that is blocked. The service provider may supply the listed number for that line when it passes the messages to the next hop. Why tell the switch that these events happened when the information is otherwise suppressed? Also blocked the number of previous redirects that may have occurred to calls going out the port when outgoing_colp is 2. Follow on to JIRA ABE-1853. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5f73ab9f4e	Merged revisions 202671 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e2bfdbac0a	Merged revisions 202414 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f142cbe10c	Merged revisions 202341-202342 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e68e6f9d75	Merged revisions 202336 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	55c6789f74	Use sched_yield() instead of usleep(1) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	05da5f14d9	Merged revisions 201993 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e85296e244	Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	21ad428d0d	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. ... Review: https://reviewboard.asterisk.org/r/285/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dcfe69ec64	fixes some memory leaks and redundant conditions ... (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. ... This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a11ac5ae2f	parsing extension correctly from sip register lines ... If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	68ba81dfe6	Add rtsavesysname to chan_iax ... chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax. (closes issue #14837) Reported by: barthpbx Patches: iax2-rtsavesysname.patch uploaded by barthpbx (license 744) rt_iax.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	99b98d8f9a	Fix problem with no audio due to ignoring the SDP. ... A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	ecfbab0782	Merged revisions 201380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9bf67151c9	SIP registry ref count error ... During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	940accbd99	update chan_iax to use 64bit feature flags. ... (closes issue #15335) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/284/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9a66b1dcdf	fix issue with build_contact introduced by the "SIP trasnport type issues" commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f1dc620467	Enable applications to enable/disable digit and tone detection. ... Some applications (notably app_fax) do not need digit detection nor FAX tone detection while they are running, and if Asterisk is using software DSPs to provide the detection, this consumes extra CPU cycles that could be better spent on the actual application. This patch allows applications to query and control the state of digit and tone detection on a channel, and modifies app_fax to disable them while the FAX operations are occurring (and re-enable digit detection afterwards). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ee8cdd555f	SIP transport type issues ... What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	2c8cd1db92	keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	85e57521ab	Accept T.38 re-INVITE responses with invalid SDP versions. ... This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4379249674	Convert a number of global module variables to 'static'. ... These modules all contained variables that are module-global but not system-global, but were not marked 'static'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	78ee46f13f	Some minor structure size improvements in sip_pvt and sip_peer. ... Using the 'pahole' tool, it is now quite easy to see where structure fields could be organized differently to keep the compiler from having to add padding to satisfy alignment requirements. These changes reduced the sizes of sip_pvt and sip_peer by a few bytes each (on 64-bit platforms), and also fixed a spelling error in a field name. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d224f78dd5	Merged revisions 200513 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	b52abf3d21	added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways ... also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e1c03cbf1a	Fix some bad locking stemming from trying to forward a call to a non-existent ... extension from a queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	616e85c95f	Fix a crash due to a potentially NULL p->options. ... Thanks to mnicholson for pointing it out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago