Asterisk
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asterisk
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2107 Commits
150 Branches
1605 Tags
304 MiB
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Commit Graph

292 Commits (02858c574c38b0775b21c53f8d447f16f6754da8)

Author SHA1 Message Date
Mark Spencer d21c6b7c13 Properly decode strings in REPLACES and ignore leading whitespace
22 years ago
Martin Pycko 9d2c55e62a Fix SIP_CODEC support
22 years ago
Mark Spencer a2303e3d0f Don't reinvite if we got a refer
22 years ago
Mark Spencer 027d5bbf30 Add ${SIPDOMAIN} and "dynamic peer" mode
22 years ago
Mark Spencer 9175d0a1ee If somehow we don't have an answer, set need to destroy on CANCEL
22 years ago
Mark Spencer b0f482d9b3 Handle 401 for INVITE as well (should we add support for BYE too?)
22 years ago
Mark Spencer 1e95c3a4ac Remove really broke MP3 stuff in favor of G.726 in the near future
22 years ago
Martin Pycko 0c5ee9486b Make SIP_CODEC warn about not supported/not configured codec
22 years ago
Martin Pycko da93b54687 Add externip keyword so that it's possible to use asterisk behind a NAT through port forwarding
22 years ago
Mark Spencer 77003fea94 Fix *8# magically (bug #116)
22 years ago
Mark Spencer c6569f5177 Check that there really is a tag
22 years ago
Mark Spencer 37ff334508 Set joint capability as well when checking users
22 years ago
Mark Spencer 67d2608db9 Handle capability on inbound too
22 years ago
Mark Spencer a1fe10de33 Add language supporrt for SIP (bug #411)
22 years ago
Mark Spencer 5e25770a83 Okay, '*' and '#' are still okay in INFO messages
22 years ago
Mark Spencer bc69cfe5a4 Minor typo fix
22 years ago
Martin Pycko f59f259586 In the SIP -> SIP if the called party disconnects with 486 make sure that the calling party gets 486 also
22 years ago
Mark Spencer a32f02162b Actually add md5secret option to CVS (bug #288)
22 years ago
Mark Spencer c8a93d08eb Properly strip "from" line
22 years ago
Mark Spencer 1682b2ced8 Strip "sip:" from callerid
22 years ago
Mark Spencer 4966cfa119 Fix typo
22 years ago
Mark Spencer f537e2e0ec Consider "from" when getting destination (bug #368)
22 years ago
Mark Spencer 1343c2638c Increase sizes in SIP (bug #352)
22 years ago
Mark Spencer e4b8d32001 Make sip show channel display a given callid when only partly specified
22 years ago
Martin Pycko c0dde4ac5b Add 486 message (SIP) to 17 cause (PRI) translation when you call Dial app and then Hangup:
22 years ago
Mark Spencer 9e68716e0e Check registry carefully before unlinking
22 years ago
Mark Spencer d3e0c54f7d Fix leaking sip channels
22 years ago
Mark Spencer 80f05b38b0 Fix SIP for outbound too
22 years ago
Mark Spencer 694d3435e1 Add sayunixtime, chan_sip updates for codec negotiation
22 years ago
Mark Spencer b9643771af Destroy channels on BYE when no owner is present
22 years ago
Mark Spencer eca42807c3 Fix counter in SIP (bug #291)
22 years ago
Mark Spencer 83e9734908 Fix capability lookup failure
22 years ago
Mark Spencer 273e1b07d6 Keep voicemail from segging on a permissions problem (bug #245)
22 years ago
Mark Spencer 90f6d836ee Fix deadlock pointed to by Martin
22 years ago
Mark Spencer c511b967ed Outgoing limit updates (bug #98)
22 years ago
Mark Spencer faeb2e28f2 Add extended Matching functionality (bug #97)
22 years ago
Martin Pycko 78ef0d0ebe Add SIPDtmfMode application
22 years ago
Mark Spencer 76b99bb025 Fix typo
22 years ago
Mark Spencer 979dc97f36 Fix "potential" deadlock and make not-a-deadlock
22 years ago
Mark Spencer 479a67e629 BSD portability enhancements (bug #234)
22 years ago
Martin Pycko 6e8a074912 Add restrictcid=[yes|no] for chan_zap and chan_sip. You can specify if the sip/zap users will have the callerid transmitted as ANI or CALLERID.
22 years ago
Mark Spencer 687a0e76b5 Take URI from responses (bug #215)
22 years ago
Mark Spencer 174af33085 Actually use the "get_also" function
22 years ago
Mark Spencer eeceb89d31 First pass at BYE/Also transfer
22 years ago
Mark Spencer 1d7e348b8f Make sipdebug default to 0
22 years ago
Mark Spencer dcb9b8f9a2 Make SIP registration persistent (bug #159)
22 years ago
Mark Spencer 2d9561427e Enable Asterisk to register with itself (with authentication)
22 years ago
Mark Spencer f148fe7e4f Send registration info on 407 too
22 years ago
Mark Spencer d6cfcc994d Use URI in sip, update chan_vpb
22 years ago
Mark Spencer 8d4afb9c19 Fix call pickup with SIP
22 years ago
Mark Spencer 6b8998b8ec Store password for later authentication needs
22 years ago
Mark Spencer bc46ece67f Use "domain" instead of "uri" if supplied
22 years ago
Mark Spencer 438ed9e5f8 Minor fix
22 years ago
Mark Spencer 46059b0714 Reliably transmit BYE always
22 years ago
Mark Spencer ddb5ae08e8 Small changes that could be useful
22 years ago
Mark Spencer 7d8d7837b4 Never create do initreq with BYE
22 years ago
Mark Spencer 436f02d38a Don't actually destroy in handle_request
22 years ago
Mark Spencer 4b1baccb2e In some cases, no need for an initreqprep
22 years ago
Mark Spencer a869aae156 Only suggest destroy
22 years ago
Mark Spencer b79a237186 Prefer bind address if specified over our actual route address
22 years ago
Mark Spencer 913a5331b7 Only half-ack provisional responses
22 years ago
Mark Spencer b938de4ea7 Check for 407 on both outbound and non-outbound calls
22 years ago
Mark Spencer 312760c381 First attempt at fixing bug #131 (authenticate BYE)
22 years ago
Mark Spencer 9140063401 Uh, don't lock
23 years ago
Mark Spencer 341c78552d If we can't transfer, be sure we hangup
23 years ago
Martin Pycko 501c800c5a Show the names of the codecs instead of the numbers (bug #92)
23 years ago
Mark Spencer 4e82233874 Don't lock when soft hanging up
23 years ago
Mark Spencer 3e211c91a2 Make sure strncat is > 0
23 years ago
Mark Spencer 37b7bccc61 Asterisk:
23 years ago
Mark Spencer 1bb58646de Totally revamp thread debugging to support locating and removing deadlocks
23 years ago
Mark Spencer e85c9cbc45 Fix typo
23 years ago
Mark Spencer e4614cf487 Add optional pedantic SIP checking
23 years ago
Mark Spencer b12891297f Handle 501 right
23 years ago
Mark Spencer 56f1a46121 Fix TOS bit in SIP when verbose < 2, add support for mgcp TOS bits
23 years ago
Mark Spencer d407e8e25e Small SIP simplification
23 years ago
Mark Spencer 5ea641b2d3 If local user not found, allow call to pass
23 years ago
Mark Spencer 70f306e8d8 Revision to SIP
23 years ago
Mark Spencer ecf50fcee8 Add per-user limits to chan_sip
23 years ago
Mark Spencer 3f1d92ad68 Merge Matteo's INFO patch
23 years ago
Martin Pycko f2d573fe85 Fix two little things
23 years ago
Mark Spencer 53fab94828 Minor fix for cid name
23 years ago
Martin Pycko 09a80571a5 Fix a segfault with the calls that need to register with the remote gateway (eg: when asterisk has to send REGISTER message in order to place a call)
23 years ago
Mark Spencer 817fc344e7 Default to allowing H.263 but not H.261
23 years ago
Mark Spencer 9befc69684 Properly handle broken up packets
23 years ago
Mark Spencer f5e13431a5 Add SIP/RTP video support, video enable app_echo, start on RTCP
23 years ago
Mark Spencer 733f251743 Turn off moh on SIP transfer
23 years ago
Mark Spencer 0e46787829 Make names more rational
23 years ago
Mark Spencer 4c021fb727 Merge (for the time being) the alert code...
23 years ago
Mark Spencer 2685819181 Minor fix for SIP authentication
23 years ago
Mark Spencer b04542e11e Handle registration setups better when username contains an @
23 years ago
Mark Spencer 3664ca0cbb Remove duplicate define
23 years ago
Mark Spencer fc8d3631b4 Add SRV code to SIP, cleanup ENUM and make IAX2 do the right thing on dials
23 years ago
Martin Pycko c680adba74 Fix the segfault in get_calleridname if the From: string starts with "<sip:..."
23 years ago
Martin Pycko 79c8c611db Add intercepting the calleridname from incoming SIP calls
23 years ago
Mark Spencer 1ba2edcfb2 Fix INFO on SIP with weird hardware
23 years ago
Mark Spencer acc40fafeb Enforce mailboxes checked in increasing order
23 years ago
Mark Spencer 38c28eb6cb Restart faster on checking NOTIFY messages on mailboxes
23 years ago
Mark Spencer 921ffb5dd7 Only send 180 ringing once, fix CTRL+D in main code
23 years ago
Mark Spencer bbdb142d2e Always lock call in find_call
23 years ago
Mark Spencer f345e8a77c Handle progress in chan_sip properly
23 years ago