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2007-12-18Fixing AST-2007-027 (Closes issue #11119)tilghman1-3/+30
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@93667 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-23(closes issue #10236)file1-0/+5
Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76560 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21Backport a fix for a memory leak that was fixed in trunk in reivision 76221russell1-0/+1
by rizzo. The memory used for the localaddr list was not freed during a configuration reload. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76226 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-20(closes issue #10247)file1-0/+3
Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@76080 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06If a sip_pvt struct has already registered an extension state callback,russell1-1/+4
remove the old one before adding a new one. If this isn't done, Asterisk will crash. (issue #10120) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-06(closes issue #10125)russell1-2/+6
Reported by: makoto Patches submitted by: makoto This fixes a crash in chan_sip that happens when the bindaddr setting is not valid on Asterisk startup, gets fixed, and then a reload gets issued. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73678 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support ↵kpfleming1-0/+3
it for G.729 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73547 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-05Copy language information to the dialog structure when calling a peer for ↵file1-0/+2
situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@73466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25Ignore other URIs after the first in a 300 Multiple Choice response. (issue ↵file1-1/+3
#10041 reported by homesick) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@71414 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-20Don't overwrite the configured username setting upon a REGISTER. (issue ↵file1-3/+1
#8565 reported by jsmith) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@70551 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Backport fix for crashes related to subscriptions from 1.4 ...russell1-0/+5
Fix a crash that could occur when handing device state changes. When the state of a device changes, the device state thread tells the extension state handling code that it changed. Then, the extension state code calls the callback in chan_sip so that it can update subscriptions to that extension. A pointer to a sip_pvt structure is passed to this function as the call which needs a NOTIFY sent. However, there was no locking done to ensure that the pvt struct didn't disappear during this process. (issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use the sip_pvt lock wrappers by eliel) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@69990 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-18Set the peer name on the dialog to the one configured in sip.conf and NOT ↵file1-1/+1
the username to be used for authentication attempts. (issue #9967 reported by achauvin) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@69765 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-07Only notify the devicestate system of a peer state change when the peer is ↵file1-1/+0
built from the config file. (issue #9900 reported by arkadia) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@67938 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31It is now possible for this path of execution to have the frame pointer be ↵file1-1/+1
NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@66764 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29Issue #9802 - Change inuse counter on CANCELoej1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@66349 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is ↵file1-0/+1
not matched to a user or peer. (issue #9546 reported by mcrawford) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65837 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-23ensure that variables are set on a newly created channel before we start a ↵kpfleming1-4/+6
PBX on it git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65682 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Not getting an ACK to a 200 OK in the initial invite is critical to the call.oej1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65122 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Issue 9235 - part of the problem, maybe not all. Please retry with this ↵oej1-2/+7
patch (and no other patch) if you have problems with hanging SIP channels. Thank you. A special Thank You to WeBRainstorm that gave me access to his system. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65075 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Fixing possible bug in auth of BYEoej1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64603 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield ↵oej1-15/+15
on the mailing list. Thanks!) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64535 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Issue #9726 - rlister - Better logging for ACL denialsoej1-11/+27
While at it, also added better logging and handling of peers that are not supposed to register. My patch, stole the issue report from Russell. My apologies, Russell :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64514 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-10Do not allocate SIP pvt's for PEERs we can not reach. oej1-0/+4
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63748 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Properly handle hints that point to multiple devices in chan_sip. Why ↵file1-5/+8
chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63610 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03When a peer is seeded or built tell the devicestate core to update it's ↵file1-0/+2
status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62987 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by ↵oej1-0/+2
PPYY, failed to fix by OEJ final fix by wojtekka - THANKS!!!! THis was a hard one to catch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62126 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Allow RFC2833 to be sent in the response SDP when an INVITE comes in without ↵file1-0/+2
SDP. (issue #9546 reported by mcrawford) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61771 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-19Send NOTIFY to Contact: in SUBSCRIBE oej1-0/+4
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61685 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-16Don't stop RTP on errors on INFO messages.oej1-8/+11
Disclaimer: This patch was needed for Edvina AstHoloApp and was meant to be included in 1.2, but never made it in time so I felt I could add it now. No, just joking, patching error found while testing T.140 with Omnitor earlier this spring. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61663 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11If someone sets the "useragent" option in sip.conf to be empty, then don't addrussell1-4/+8
the User-Agent header at all. It is an optional header, anyway. Also, the bug report says that some of Japan's SIP providers don't allow it for some weird reason. (issue #9488, reported by makoto, fixed by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61476 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11Fix a bug with switching between host=dynamic and using specific hosts forrussell1-5/+5
peers. The code would only reset the peer's address when it is dynamic if it was a new peer structure. Now, it will also reset the address if it was already in the peer list, but before the reload, it was not dynamic. (issue #9515, reported by caio1982, fixed by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61426 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11Remove the attempt at reporting configuration errors in sip.conf. This canrussell1-7/+4
cause a bunch of improper messages when using realtime. I give up. As oej tried to convince me when I put this in, there is just no easy way to do it. (inspired by a message on the -dev list) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61376 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09- Don't send ActionID before Response: header. oej1-6/+7
- Don't use a blank in an AMI header git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61038 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-05Only unlock our pvt and net locks if we are actually going to try to lock ↵file1-6/+9
the owner again. (issue #9472 reported by zoa) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@60213 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-04Fix the return value of handle_common_options() so that it always properlyrussell1-6/+3
indicates whether it handled the option or not. (issue #9455, reported by Netview, fixed by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@60083 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-03Add a missing "\r\n" in the body of the NOTIFY that is sent to indicate therussell1-1/+1
status of a transfer. (issue #9388, reported by rarritt) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@60016 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-03Don't attempt to report configuration errors in build_user(). oej pointed outrussell1-2/+4
that for a "friend" entry, this won't work, because all user options are valid for peers, but not the other way around. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@59938 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-03Make chan_sip report when it encounters an unknown option.russell1-14/+9
(issue #9440, reported by nightcrawler) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@59916 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-24Only try to handle a response if it has a response code. (ASA-2007-011)file1-10/+10
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@59194 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-13Issue #9229 - No port in request URI on register to non default SIP ports ↵oej1-4/+11
(neelakantan) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@58847 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-09If we are unable to lookup the host in a c line we have to abort, otherwise ↵file1-0/+2
the previous data is gone and we will (potentially) have no data when all is said and done. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@58579 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: ↵murf1-2/+15
re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@58115 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Change error message to proper messageoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@58052 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02If a SIP message comes in and goes to a method handler that requires ↵file1-0/+6
additional values that may not be present then send back an error. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@57475 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Only change the original or clone channel if it's the channel behind the ↵file1-1/+1
proxy channel, not if it's just a regular bridged channel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@56230 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22If we receive a frame that is not in any of the negotiated formats, then droprussell1-0/+5
it. (potentially issue #8781 and SPD-12) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@56010 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-18Issue #9020 - SIP message retransmission time too short.oej1-1/+3
Backporting fix implemented in 1.4, where we have a minimum level for the T1 timer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@55249 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17Allow chan_sip to handle attended transfers from a SIP phone that is sitting ↵file1-1/+1
behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@55073 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@54438 ↵oej1-1/+1
f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-13Make sure that outbound calls are applied to the peer.oej1-2/+1
This fixes some issues with "hints not working", but only in 1.2. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@54179 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-01 10:16:44 +0000