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${ noResults }
6678 Commits (8ae2b6a6124cb0e73ffc7f30b935a41b49a33439)
Author | SHA1 | Message | Date |
---|---|---|---|
|
8ae2b6a612 |
Merged revisions 283558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
abca511f03 |
Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2787a14001 |
Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5c82781efe |
Fix issue where TOS is no longer set on RTP packets.
Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk. (closes issue #17890) Reported by: elguero Patches: qos_18.diff uploaded by elguero (license 37) Review: https://reviewboard.asterisk.org/r/868 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
6f3a4b0511 |
Merged revisions 283381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c453d72423 |
Merged revisions 283049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e9a51ba86b |
Merged revisions 282894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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af6e8a5abb |
Merged revisions 282890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines fixes sip peer memory leaks in the peer_by_ip table (issue #17798) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d4cc26fa1e |
Merged revisions 282859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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82c2cf5159 |
Use the correct type for aoce_delayhangup bit field.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2392b8ed1c |
Use the correct operator when calculating the PRI span devstate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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38a0c0849f |
Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests.
This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received. (issue #17486) Reported by: davidw Tested by: mnicholson (issue #12713) Reported by: davidw Review: https://reviewboard.asterisk.org/r/860/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d0235ab07e |
Split _all_ arguments before parsing them.
This fixes multicast RTP paging using linksys mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4aed988d66 |
Merged revisions 282607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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647a8f6edd |
Merged revisions 282576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines fixes no default transport for temp peer creation in chan_sip (closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c1a577848b |
ACCEPT message should respond with the new FORMAT2 ie
(closes issue #17804) Reported by: tpanton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3c0616589e |
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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593512960d |
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282334 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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22682c2eee |
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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48fb2c3276 |
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fbfafb59ba |
Merged revisions 282235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines only do magic pickup when notifycid is enabled A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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31d1c6d76b |
handle all possible responses to REFER requests
(closes issue #17486) Reported by: davidw Patches: Issue17486-counterbid.diff.txt uploaded by davidw (license 780) Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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72f370ecc1 |
Fix a call to analog_set_pulsedial() not setting 0 or 1 only.
* Also a couple minor tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ea920c7cd3 |
Avoid a deadlock in add_header_max_forwards().
Related to r276951 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281760 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5c1c1b35bd |
Fix parsing of IPv6 address literals in outboundproxy
(closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7011a94fc0 |
Change the default value for alwaysauthreject in sip.conf to "yes".
(closes issue #17756) Reported by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281650 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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83e01097b1 |
Ensure that the proper external address is used for the RTP destination.
(closes issue #17044) Reported by: ebroad Tested by: ebroad Review: https://reviewboard.asterisk.org/r/566/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c4d808e7e4 |
Add some more stuff to copy from 281429.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bbdbe1180d |
Merged revisions 281430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines fixes SIP peers memory leak We zeroed out the peer's addr before it was removed from the peers_by_ip container. This made it impossible to be removed from the container as the addr is the key used by the container to find the peer. (closes issue #17774) Reported by: kkm Patches: 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3da327e87d |
Merged revisions 281391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines Prevent loss of Caller ID information set on local channel after masquerade. Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ca2ace07aa |
Check cur value before attempting a deref.
(closes issue #17775) Reported by: svinson Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: svinson (closes issue #17743) Reported by: tgruenberg Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: tgruenberg git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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50cb08aefa |
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663) Reported by: oej Patches: diff uploaded by sperreault (license 252) diff2 uploaded by sperreault (license 252) get_domain.diff uploaded by sperreault (license 252) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f7a2194c58 |
Merged revisions 280551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines fixes wrong SRV query for TLS connection (closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e32da6f7a5 |
Fix compilation error in chan_dahdi (strdupa -> ast_strdupa).
(closes issue #17751) Reported by: b11d Patches: strdupa_oops.diff uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a09163e0ae |
Use PRIx64 instead of PRId64 in format string.
related to r280302 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bb4178a14a |
Merged revisions 280306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c62b0630de |
Use PRId64 with format_t
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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50f2b57276 |
Give test category missing leading slash
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3b7f592cc0 |
Merged revisions 280229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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613e102539 |
Resolve compiler warning about formatting
(closes issue #17732) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e7b5069c9f |
Fix inband DTMF detection on outgoing ISDN calls.
This is a regression from the sig_pri split from chan_dahdi. When a call is first initiated, the inband DTMF detector is not enabled if it's an outgoing ISDN call. However, it needs to be turned on once the media path starts up. This handling was put back in the open_media() callback of chan_dahdi. In sig_pri, open_media() calls were added to a few places where it was needed, including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279916 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cbba00f5d0 |
Fix parsing error in sip_sipredirect().
The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address. In this particular scenario, there was no value from parsing the port out, so I just removed that logic. And while I was messing around in the function, I changed some variable names to be more descriptive. (closes issue #17661) Reported by: oej Patches: 17661.diff uploaded by mmichelson (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ab374d0446 |
fix sip transaction match with authentication, fix confusing log message when using getaddrinfo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ccfad47983 |
Support "channels" in addition to "channel" in chan_dahdi.conf.
Review: https://reviewboard.asterisk.org/r/804 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279815 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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62330bc1c2 |
Merged revisions 279784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines Fix bad behavior of dynamic_exclude_static option in sip.conf. We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cd034c3dfc |
If dringXcontext is null, fallback to default context value.
(closes issue #17693) Reported by: iasgoscouk Patches: issue17693.patch uploaded by pabelanger (license 224) Tested by: iasgoscouk Review: https://reviewboard.asterisk.org/r/803/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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610151af27 |
transaction matching using top most Via header
This patch modifies the way chan_sip.c does transaction to dialog matching. Asterisk now stores information in the top most Via header of the initial incoming request and compares that against other Requests that have the same call-id. This results in Asterisk being able to detect a forked call in which it has received multiple legs of the fork. I completely stripped out the previous matching code and made the comparisons a little more explicit and easier to understand. My comments in the code should offer all the details involving this patch. This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to find multiple dialogs with the same call-id. Since the callback function was returning (CMP_MATCH | CMP_STOP) only the first item found was being returned. I fixed this by making a new callback function for finding multiple dialogs that only returns (CMP_MATCH) on a match allowing for multiple items to be returned. Review: https://reviewboard.asterisk.org/r/776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279568 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bc3b185063 |
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279504 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d1ad460b3d |
SIP URI comparison fixes.
This initially was created to work around the issue of using a string comparison instead of a binary comparison for IP addresses. It evolved a bit when test cases were created and it was discovered that comparison of URI parameters was not working exactly as it should. sip_uri_cmp() and its helpers have been moved to reqresp_parser.c and a new test has been added. (closes issue #17662) Reported by: oej Review: https://reviewboard.asterisk.org/r/792 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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09206a7db8 |
... just kidding. Enable SIP by default. :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |