asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	38e98f42bc	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@208587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1c46ba9635	Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	594a236e12	Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	94bc859e81	Remove inaccurate XXX comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	eb5f3170fc	Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e07afa4876	Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	dca651b85d	Revert r207573, this approach could potentially block for an unacceptable amount of time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	75f1eaf2a1	Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker after all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided either in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8b940dbeb7	Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	423a444c0b	Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch only helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d162e4b055	Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	1e30dcf61c	Enhance configuration option for overlapdial allowing direction choice Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	98a6820737	sip option flags handled incorrectly (issue #15376) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	7c82de7d7e	SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7782df0963	Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6db6a73b8d	Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8d5516a153	Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 \| russell \| 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) \| 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9f3cd22c7a	Fix some memory leaks in chan_misdn. JIRA ABE-1911 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	12b5e7706c	Properly ACK 487 responses to canceled INVITEs. From the review board request: The fix from review 298 has exposed a new bug in chan_sip. When we hang up an outgoing call, we first will dump all the outstanding packets on the sip_pvt using __sip_pretend_ack. Then, if we can, we send a CANCEL. The problem with this is that since destroyed all the outstanding packets on the dialog, we cannot match the incoming 487 response to our INVITE. Because we cannot match the response, we do not send an ACK. To correct this, instead of using __sip_pretend_ack, I have changed the code to loop through the list of packets and call __sip_semi_ack on each one instead. This causes us to stop retransmitting the requests, but we still have them around in case we get responses for them. When discussing this earlier today with Josh Colp, we both agreed that in the majority of cases, this would be enough of a fix. However, we also agreed that we should have a safety net in place in case we never receive a response to our initial INVITE. To handle this, I have modified __sip_autodestruct to behave similar to the way it does in Asterisk 1.4. If there are outstanding packets on the sip_pvt, the needdestroy flag is not set, and the last request on the dialog was either a CANCEL or BYE, then we set the needdestroy flag and reschedule destruction for 10 seconds in the future. If, though, the needdestroy flag is set, then we use __sip_pretend_ack to kill the remaining outstanding packets so that the monitor thread can destroy the sip_pvt. I ran two separate tests. First, I placed a call from my Aastra phone to my Polycom phone. I hung up the Aastra before the Polycom answered. I verified through sip debug output that Asterisk properly ACKed the 487 received from the Polycom. For my second test, I set up a SIPp UAS scenario so that it would send a 200 OK in response to a CANCEL but would not send a 487 for the original INVITE. I verified that after about 40 seconds, Asterisk properly cleans up the outgoing sip_pvt for the call. Review: https://reviewboard.asterisk.org/r/308 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	1678f43bfa	SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	43a5245325	Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	fb1c512a40	No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	beaf6217b3	Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	202f9967c6	Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e5bef05d8f	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/branches/1.4@204300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	439ce618c5	Fix build oops. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9589d9fb2e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4f3580b882	segfault after SPINLOCK schedule delete Using the SPINLOCK schedule delete macro can result in the iax_pvt lock being given up. This makes it possible for the iax_pvt to dissappear when we thought we held the mutex the entire time. To resolve this, the iax_pvt's ref count is incremented. (closes issue #15377) Reported by: aragon Patches: iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested by: aragon, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	f65dccafb6	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/branches/1.4@203908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fc73897bbd	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/branches/1.4@203848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c05d6ceccd	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/branches/1.4@203115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	1ac27cf7ec	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/branches/1.4@203036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a1fa4f0391	Use the handy UNLINK macro instead of hand-coding the same thing in-line. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d6106936cb	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/branches/1.4@202671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f76b499923	Fix more memory leaks that may result if rtp is not successfully allocated. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b0c0c17764	Fix potential memory leak in chan_sip when video rtp is not allocated properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dcfd8d7c7c	Make Polycom subscription type override check more explicit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	26ba38b8f4	Remove an extra debug line left from previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d31f78a172	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1f7d3e9a01	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/branches/1.4@202336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f543251260	Since we don't have sip_pvt_lock() in 1.4, we need to use ast_mutex_* directly. (closes issue #15366) Reported by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	e735cdc36b	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f17d5d22d2	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/branches/1.4@201993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	ebe2c1829b	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/branches/1.4@201380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c946be82a9	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/branches/1.4@200513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ed94be12f0	Additional updates to AST-2009-001 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	67928d88a9	'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	590408dca3	Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	26cec158af	Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	eb2a672328	Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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4757 Commits (38e98f42bc656c514b321a55bdfd5bb1c0a0fb55)