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r151600 | mmichelson | 2008-10-22 15:05:14 -0500 (Wed, 22 Oct 2008) | 10 lines
Change some logical ands to bitwise ands and add
messages alerting that a channel is being ignored
if the PROC_DAHDI_NOCHAN option is set in process_dahdi.
(closes issue #13759)
Reported by: smurfix
Patches:
dahdi.patch uploaded by smurfix (license 547)
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r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
Make the sip_standard_port function more granular by allowing separate
type and port arguments. This is necessary because when building our From
and Contact headers, we need to be absolutely sure that we are placing our
source port there and not the peer's source port.
(closes issue #12761)
Reported by: asbestoshead
Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
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r151428 | mmichelson | 2008-10-21 18:27:45 -0500 (Tue, 21 Oct 2008) | 14 lines
If a peer uses any transport other than UDP, then MWI will
fail for that peer since sip_alloc will allocate a sip_pvt with
a default transport of UDP. This change resets the socket type
immediately after allocating the sip_pvt in sip_send_mwi_from_peer,
so that the proceeding call to create_addr_from_peer does not fail
right away. The socket data from the peer is properly copied to
the sip_pvt in create_addr_from_peer.
(closes issue #13710)
Reported by: andrew53
Patches:
sip_notify_use_tcp.patch uploaded by andrew53 (license 519)
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r151420 | mmichelson | 2008-10-21 18:08:56 -0500 (Tue, 21 Oct 2008) | 10 lines
When attempting to resolve hostnames, we need to be sure
to remove any parameters from the string so that name
resolution succeeds.
(closes issue #13727)
Reported by: fnordian
Patches:
resolvewithouturiparameter.patch uploaded by fnordian (license 110)
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r151101 | kpfleming | 2008-10-19 22:11:28 +0300 (Sun, 19 Oct 2008) | 13 lines
cleaup of the TCP/TLS socket API:
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
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r150207 | mmichelson | 2008-10-16 15:57:18 -0500 (Thu, 16 Oct 2008) | 12 lines
INVITES with proxy auth were sent with a different branch
than what was in the invite_branch of a sip_pvt, meaning
that if a CANCEL were sent later, the branch in the CANCEL
would not match the branch in the latest INVITE sent out, leading
to some endpoints responding to the CANCEL with a 481.
(closes issue #13714)
Reported by: fnordian
Patches:
invite_branch.patch uploaded by fnordian (license 110)
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r149981 | kpfleming | 2008-10-16 15:28:56 +0200 (Thu, 16 Oct 2008) | 3 lines
return this logic to where it used to be, *after* the dialog->needdestroy flag has been determined to be set; otherwise, we generate these debug messages every time we inspect every active dialog
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r149802 | mmichelson | 2008-10-15 15:55:42 -0500 (Wed, 15 Oct 2008) | 12 lines
Make the sip_proxy struct reference counted. This is
necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has
been freed.
(closes issue #13700)
Reported by: fnordian
Patches:
13700.patch uploaded by putnopvut (license 60)
Tested by: fnordian
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r148695 | kpfleming | 2008-10-14 11:31:50 +0200 (Tue, 14 Oct 2008) | 1 line
ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled
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r148376 | mmichelson | 2008-10-10 16:21:45 -0500 (Fri, 10 Oct 2008) | 13 lines
The logic used when checking a peer got changed subtly
in the "kill the user" commit and caused calls relying
on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that
commit.
(closes issue #13644)
Reported by: pj
Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60)
Tested by: pj
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r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines
Make sure that the inUse and inRinging fields for
a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.
(closes issue #13668)
Reported by: mjc
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r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
(closes issue #13557)
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines
Merged revisions 147681 via svnmerge from
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r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
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r146802 | tilghman | 2008-10-06 16:09:05 -0500 (Mon, 06 Oct 2008) | 15 lines
Merged revisions 146799 via svnmerge from
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
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r146713 | tilghman | 2008-10-06 11:52:14 -0500 (Mon, 06 Oct 2008) | 16 lines
Merged revisions 146711 via svnmerge from
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r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 Oct 2008) | 9 lines
Check whether an extension exists in the _call method, rather than the _alloc
method, because we need to evaluate the callerid (since that data affects
whether an extension exists).
(closes issue #13343)
Reported by: efutch
Patches:
20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
Tested by: efutch
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r144569 | murf | 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) | 14 lines
(closes issue #13557)
Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
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r144025 | mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16 lines
When a promiscuous redirect contained both a user and
host portion in the Contact URI and specifies a
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.
This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.
(closes issue #13523)
Reported by: mattdarnell
Patches:
13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell
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r143609 | mmichelson | 2008-09-19 10:43:28 -0500 (Fri, 19 Sep 2008) | 11 lines
We should only unsubscribe to the device state event
subscription if we have previously subscribed. Otherwise
a segfault will occur.
(closes issue #13476)
Reported by: jonnt
Patches:
13476.patch uploaded by putnopvut (license 60)
Tested by: jonnt
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r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines
Merged revisions 142865 via svnmerge from
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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r142080 | mmichelson | 2008-09-09 11:20:41 -0500 (Tue, 09 Sep 2008) | 29 lines
Merged revisions 142079 via svnmerge from
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r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines
When determining if codecs used by SIP peers allow
the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
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r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep 2008) | 22 lines
Merged revisions 141809 via svnmerge from
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r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines
Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.
(closes issue #11536)
Reported by: ibc
Patches:
11536v2.patch uploaded by putnopvut (license 60)
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