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${ noResults }
7185 Commits (e14f5721329e0ce00fb31d68bb9abff68083effd)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Walter Doekes
|
e14f572132 |
Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347167 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Jordan
|
b8dba87a4c |
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347068 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347069 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Richard Mudgett
|
938b642245 |
Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
........
Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
|
Walter Doekes
|
6209c5a894 |
For SIP REGISTER fix domain-only URIs and domain ACL bypass.
The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
fa4a7dcc45 |
Update SIP MESSAGE To parsing to correctly handle URI
The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346856 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alexandr Anikin
|
db0ed2e5c8 |
Merged revisions 346763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jonathan Rose
|
39424ebad2 |
Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.
183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346698 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
|
56b21b4683 |
Remove the few places where we try to ast_verbose() without a newline.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Jonathan Rose
|
9ef171ffe0 |
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346565 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Jonathan Rose
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fb4c483eb7 |
Reverting 346525 due to accidental patch against trunk instead of 1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Jonathan Rose
|
6fa827b5d0 |
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Tilghman Lesher
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77b670c4ab |
Allow each logging destination and console to have its own notion of the verbosity level.
Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
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7d9ba4875b |
Fix calls to ast_get_ip() not initializing the address family.
........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Walter Doekes
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2ee874178e |
Minor cleanup in chan_sip get_msg_text() function.
In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kinsey Moore
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e6ca768081 |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
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76394a727f |
Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Terry Wilson
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32d0faac9c |
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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4a125f45a0 |
Remove dead code since pri_grab() can never fail.
Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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bd4f81b51f |
Fix typo in sig_pri using wrong structure name.
It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.
(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
sig_pri.c.diff (License #5755) patch uploaded by zvision
........
Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345371 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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Richard Mudgett
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113612b9d6 |
Restore SIP DTMF overlap dialing method.
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Walter Doekes
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6ef49c3214 |
Update reqresp_parser parse_uri doxygen comments.
The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345161 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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818ac23b92 |
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345064 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Gregory Nietsky
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3845299e46 |
mISDN Round Robin break when no channel is available
Prevent channels been parsed repetitively. ........ Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344966 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Walter Doekes
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863d7189b9 |
Remove unneeded if(params) checks in reqresp_parser.
Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344839 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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a4365a8ae2 |
Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax transmission, fax teardown fails. Make sure capability is only touched if RTP is active. This regression was introduced in R344385. ........ Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344770 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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e48cecc848 |
Check sip.conf maxforwards parameter for range 1 <= x <= 255.
JIRA AST-710 ........ Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344716 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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c225800646 |
Fix several bugs with SDP parsing and well-formedness of responses
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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6d481420ce |
Fix deadlock during dialplan reload.
Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
........
Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344271 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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Terry Wilson
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04f04ff39d |
Don't treat a host:port string as a domain
The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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9537f22c54 |
Residual changes for Asterisk v10 branch from ASTERISK-18747.
Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ ........ Merged revisions 344004 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
David Vossel
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5cb719acec |
Merged revisions 343900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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cee432c9d8 |
Fixed reference to incorrect variable if unknown host configured crash.
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
........
Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 343852 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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Kinsey Moore
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5249a147b8 |
Make "sip show settings" CLI command get RPID flags from the right global page
The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) ........ Merged revisions 343743 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Nicholson
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2b6ebcb9e9 |
respect case changes in peer names on sip reload
ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343691 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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98e494d4a0 |
Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid.
Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343677 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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1c526d3d7d |
Prevent BLF subscriptions from causing deadlocks
Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343635 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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7a5f6684f0 |
Fix deadlock if peer is destroyed while sending MWI notice.
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343578 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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e28d306387 |
Make room for the fax detect flags
The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343277 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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7f883ef495 |
Remove registertrying option in chan_sip
This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Walter Doekes
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f7bdc835a4 |
Fix improper warning introduced by r342927 and more tweaks
Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 ........ Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343192 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Walter Doekes
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b41b49ea0e |
Several fixes to the chan_sip dynamic realtime peer/user lookup
There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Jordan
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9333071c1f |
Fixed invalid memory access when adding extension to pattern match tree
When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342770 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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6cf08ad550 |
Change D-channel warning to be less confusing on non-NFAS setups.
The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342485 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Jonathan Rose
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05c6628c55 |
Outbound SIP OPTIONS messages will now include fromuser of related peer.
This behavior matches up more closely with the way invite/register/etc are handled.
This patch also modifies some adjacent code for code style compliance. Pretty minor.
(closes issue ASTERISK-17616)
Reported by: Jeremy Kister
Patches:
chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232)
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Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 342062 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
Paul Belanger
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1ed8cd087a |
Outgoing calls with Google Voice
Google has recently make some changes (again) to their protocol. Rather then
patching asterisk to flip between the two different methods, we now allow both.
Lets hope this keeps Google Voice happy for a while.
(closes issue ASTERISK-18714)
Reported by: Iordan Iordanov
Patches:
chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311)
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Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 341436 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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Terry Wilson
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5f8648892f |
Don't use is_int() since it doesn't link well on all platforms
Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Stefan Schmidt
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2816ccc516 |
Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS
........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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b0076c5be1 |
Don't resolve numeric hosts or contact unresolved hosts
If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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10de040b6e |
More parking issues.
* Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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9f83c2b513 |
Initialize variables before calling parse_uri
If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341191 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |