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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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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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r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines
Merged revisions 141565 via svnmerge from
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r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line
This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
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r140489 | mmichelson | 2008-08-29 12:47:17 -0500 (Fri, 29 Aug 2008) | 30 lines
Merged revisions 140488 via svnmerge from
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
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r140418 | mmichelson | 2008-08-29 10:32:02 -0500 (Fri, 29 Aug 2008) | 18 lines
Merged revisions 140417 via svnmerge from
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r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug 2008) | 10 lines
Fix SIP's parsing so that if a port is specified
in a string to Dial(), it is not ignored.
(closes issue #13355)
Reported by: acunningham
Patches:
13355v2.patch uploaded by putnopvut (license 60)
Tested by: acunningham
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r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines
Merged revisions 140299 via svnmerge from
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r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines
Fix tag checking in get_sip_pvt_byid_locked when
in pedantic mode. The problem was that the wrong
tags would be compared depending on the direction
of the call.
(closes issue #13353)
Reported by: flefoll
Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244)
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r140061 | russell | 2008-08-26 11:10:06 -0500 (Tue, 26 Aug 2008) | 14 lines
Merged revisions 140060 via svnmerge from
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r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) | 6 lines
Fix some bogus scheduler usage in chan_sip. This code used the return value
of a completely unrelated function to determine whether the scheduler should
be run or not. This would have caused the scheduler to not run in cases where
it should have. Also, leave a note about another scheduler issue that needs
to be addressed at some point.
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r139563 | mmichelson | 2008-08-22 15:20:58 -0500 (Fri, 22 Aug 2008) | 15 lines
The -1 return value from incomplete or improper
headers for the SipNotify manager command was
causing the current manager session to become
disconnected. Change the return value to 0 for
these cases.
Also change a test for a NULL pointer to be
ast_strlen_zero instead.
(closes issue #13351)
Reported by: Laureano
Patches:
sipnotify_action_fix.patch uploaded by Laureano (license 265)
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r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line
While we're at it, make this machine parseable too.
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r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line
And remove code we don't need anymore.
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r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line
Let it compile now, too (woops)
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driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Use ast_strlen_zero in one place
- check for successful string comparison the way most of Asterisk code does it
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Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines
Don't hangup the call if we can't resolve the Contact if there's a proxy
route set for the call.
----
This comment was added a while ago and today it hit me badly.
/* OEJ: Possible issue that may need a check:
If we have a proxy route between us and the device,
should we care about resolving the contact
or should we just send it?
*/
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