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asterisk
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14226 Commits
140 Branches
1567 Tags
344 MiB
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2058 Commits (11e86ae8f0be26bd371b4a33e4cd48fb2ac97025)

Author SHA1 Message Date
Jeff Peeler 11e86ae8f0 Ensure user portion of SIP URI matches dialplan when using encoded characters.
15 years ago
Stefan Schmidt 4d84c4d68a don't iterate through all dialogs to find and delete old subscribes
15 years ago
David Vossel a0579869a7 RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.
15 years ago
Matthew Nicholson ffeacadcea Handle media specific T.38 SDP information
15 years ago
David Vossel abf7e809c6 During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.
15 years ago
Tilghman Lesher 3fec8c28ad Try both the encoded and unencoded subscription URI for a match in hints.
15 years ago
Matthew Nicholson b34044bd3a Don't clear the username from a realtime database when a registration expires.
15 years ago
David Vossel 5f8e639d61 Fixes sip extension state update DEADLOCK
15 years ago
David Vossel 49e083523d In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
15 years ago
David Vossel f9927f1291 Removed relatedpeer code from sip_autodestruct
15 years ago
Tilghman Lesher a8742f4a9a Don't send a devstate change on poke_noanswer if the state did not change.
15 years ago
David Vossel 0e9fb94288 Parse all "Accept" headers for SIP SUBSCRIBE requests.
15 years ago
David Vossel c26e47f289 Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response.
15 years ago
David Vossel 3550593f30 This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
15 years ago
David Vossel d04e866925 tos_sip option was not being set correctly
15 years ago
David Vossel 4f90553a45 chan_sip: fixes provisional keepalive scheduled item crash
15 years ago
Mark Michelson 7bc66e1d77 Backport fixes for sip_uri_params_cmp() from trunk.
15 years ago
Paul Belanger b48d6c3dbe Regression with T.38 negotiation
15 years ago
Matthew Nicholson 1c8d76145b Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
15 years ago
Paul Belanger dca914c43b Fix logging message for stale nonce.
15 years ago
Terry Wilson 79d795c383 Add option to not do a call forward on 482 Loop Detected
15 years ago
Tzafrir Cohen 97f61244f9 Fix various typos, reported by Lintian
15 years ago
Tilghman Lesher 65458abbe8 Allow the "useragent" value to be restored into memory from the realtime backend.
15 years ago
Mark Michelson 2cc343faa2 Decode URI in contact header of 302 response.
15 years ago
Matthew Nicholson 7eabbade0a Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct.
15 years ago
Matthew Nicholson 2d31f18ae9 Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
15 years ago
Terry Wilson 21bdb89a2b Merged revisions 270658 via svnmerge from
15 years ago
Paul Belanger b3076dd0f5 Fix formatting issue with previous patch.
15 years ago
Paul Belanger abc4bceec2 Missing fallback to audio fax feature when T.38 re-INVITE failed
15 years ago
Paul Belanger 07dbb0f074 Registration fix for SIP realtime.
15 years ago
Tilghman Lesher 244f519a25 Don't recreate peer, when responding to a repeated deregistration attempt.
15 years ago
Terry Wilson 1b2dbf6c1a Revert API change in release branches
15 years ago
Terry Wilson 529e8af144 Merged revisions 252089 via svnmerge from
15 years ago
Mark Michelson 99e1a0f967 For T.38 reINVITEs treat a 606 the same as a 488.
15 years ago
David Vossel 8d9b578022 fixes invite with replaces deadlock
15 years ago
Russell Bryant fe49526a56 Fix a bogus third argument to ast_copy_string().
16 years ago
Olle Johansson abc434d772 Initialize notify_types to NULL
16 years ago
David Vossel a6bc57fe40 Change in sip show channels display format allowing more digits for CID
16 years ago
Olle Johansson 789554011a Release memory of the contact acl before unloading module
16 years ago
Matthew Nicholson 77c0c82575 Properly set T.38 attributes and don't return before T.38 ports are configured when T.38 is found but no audio stream is found.
16 years ago
David Vossel af1319f400 fixes issue with p->method incorrectly set to ACK
16 years ago
Olle Johansson a3c42b73de Stop sending 183's after call hangup.
16 years ago
Tilghman Lesher bd6cde63d3 When we receive no response at all to our INVITE, allow the channel to be destroyed.
16 years ago
David Vossel 55e6ac62e9 fixes missing Contact header angle brackets
16 years ago
Matthew Nicholson 6bf84d94b7 Allow SDP packets with only video session information.
16 years ago
David Vossel 8730e034d3 fixes conditional jump or move depending on uninitialised STACK value
16 years ago
Kevin P. Fleming 851de5edaa When 'sip set debug' is enabled, and the last line of an incoming SIP message
16 years ago
Kevin P. Fleming afd2b5e203 Correct fix for issue #16268... the reporter's original patch was very close to correct.
16 years ago
Kevin P. Fleming 800a1df304 Ensure that SDP parsing does not ignore the last line of the SDP.
16 years ago
Joshua Colp 4950fbeebc Respect the maddr parameter in the Via header.
16 years ago