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path: root/channels/chan_sip.c
AgeCommit message (Expand)AuthorFilesLines
2004-01-13comment out getsipuri to stop compiler warningjeremy1-2/+4
2004-01-13via z9hG4bK issue. Bug #663jeremy1-1/+4
2004-01-13fix copy/paste errorjeremy1-2/+2
2004-01-13Implement NAT trick for outbound SIP connectionsjeremy1-27/+54
2004-01-12fix typo. Bug #813jeremy1-1/+1
2004-01-12clean up sip debug output. Bug #674jeremy1-1/+1
2004-01-12Add new sound, fix nit picky detail in channel registrationmarkster1-1/+1
2004-01-12unregister cli functions and fix monitor thread. Bug #798jeremy1-1/+10
2004-01-11Fix ast-db seeding. Bug #767jeremy1-4/+10
2004-01-11More buffer overflow checks from Corydon76.jeremy1-14/+21
2004-01-11check buffer for possible overflow. Thanks Corydon76 Bug #788jeremy1-1/+5
2004-01-10Cleanup ACL parsing, handle properly reload on sip with ACLmarkster1-1/+10
2004-01-09Bug #765: Modifying Auto-Congestion to p->maxtime * 4 instead of * 2malcolmd1-1/+1
2003-12-19Send date with invitemarkster1-1/+1
2003-12-19Fix reload race in SIPmarkster1-3/+8
2003-12-13The SIP motto is "There's More Than One Standard for Doing It"markster1-1/+9
2003-12-09Don't retransmit INVITE on 407 retransmissionmarkster1-5/+8
2003-12-09Minor sip fixesmarkster1-6/+9
2003-12-09Minor mgcp debuggin changes, sip deadlock improvementmarkster1-1/+1
2003-11-28Properly build Contact on INVITE when asking for authenticationmarkster1-0/+11
2003-11-27Temporarily revert changes, prepare for proper fix (sorry for the cvs spam)markster1-7/+3
2003-11-27Always save the extensionmarkster1-2/+2
2003-11-27Build contact, don't copy itmarkster1-3/+5
2003-11-26Have a contact line in responses, merge logging patchesmarkster1-0/+2
2003-11-26limit fixes in SIPmarkster1-6/+33
2003-11-26Use global nat settings in auto peersmarkster1-0/+1
2003-11-25Minor 407 improvementsmarkster1-12/+24
2003-11-24Apply host access rules in SIPmarkster1-5/+5
2003-11-24Increment session version each time we send a new SDPmarkster1-1/+8
2003-11-24Allow SIP/peer/exten like IAXmarkster1-2/+9
2003-11-21Delete on unregister (bug #527)markster1-0/+1
2003-11-15Don't do reinvite if both parties talk diffrent codecsmartinp1-0/+7
2003-11-14Do the last thing the right way (thanks Mark)martinp1-1/+1
2003-11-14Don't allow to run SIPDtmfMode on other channels than SIP ones.martinp1-0/+4
2003-11-14A little oops ...martinp1-3/+3
2003-11-13Fix typomarkster1-2/+2
2003-11-13Properly decode strings in REPLACES and ignore leading whitespacemarkster1-4/+20
2003-11-12Fix SIP_CODEC supportmartinp1-1/+1
2003-11-12Don't reinvite if we got a refermarkster1-2/+7
2003-11-11Add ${SIPDOMAIN} and "dynamic peer" modemarkster1-2/+66
2003-11-10If somehow we don't have an answer, set need to destroy on CANCELmarkster1-0/+2
2003-11-10Handle 401 for INVITE as well (should we add support for BYE too?)markster1-14/+22
2003-11-04Remove really broke MP3 stuff in favor of G.726 in the near futuremarkster1-0/+8
2003-11-04Make SIP_CODEC warn about not supported/not configured codecmartinp1-2/+2
2003-10-31Add externip keyword so that it's possible to use asterisk behind a NAT throu...martinp1-15/+27
2003-10-25Fix *8# magically (bug #116)markster1-0/+1
2003-10-24Check that there really is a tagmarkster1-1/+1
2003-10-23Set joint capability as well when checking usersmarkster1-0/+2
2003-10-23Handle capability on inbound toomarkster1-4/+20
2003-10-22Add language supporrt for SIP (bug #411)markster1-0/+5
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