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r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines
Merged revisions 157503 via svnmerge from
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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
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r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines
Based on Russell's advice on the asterisk-dev list, I have
changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.
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r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines
* Add a lock to be used in the update_call_counter function.
* Revert logic to mirror 1.4's in the sense that it will not allow
the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.
(closes issue #13668)
Reported by: mjc
Patches:
hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
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r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines
Revision 155513 of chan_sip.c in trunk inadvertently
removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related
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r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines
Set the invite state to INV_CANCELLED in a place that
makes more sense. Where it was set before, it was impossible
to actually delay sending a CANCEL if we had not yet received
a provisional response to an INVITE.
(closes issue #13626)
Reported by: atis
Patches:
13626.patch uploaded by putnopvut (license 60)
Tested by: atis
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r155241 | russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines
Fix some code in chan_sip that was intended to unlink multiple objects from a
container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
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r152877 | russell | 2008-10-30 14:21:53 -0500 (Thu, 30 Oct 2008) | 9 lines
Modify the documentation of the sip_registry struct
- Remove a comment that says that the monitor thread is the only one that
ever touches these objects. This is no longer the case with TCP. Also,
I would eventually like to get the scheduler in its own thread, so this
is just a poor assumption to make.
- Note that reference counting of these objects with respect to scheduler
entries is not complete. There are some leaked references when deleting
scheduler entries.
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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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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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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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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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