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r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines
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r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
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r132883 | crichter | 2008-07-23 07:07:15 -0500 (Wed, 23 Jul 2008) | 9 lines
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r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line
another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
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r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul 2008) | 2 lines
use correct function name... please compile with --enable-dev-mode
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r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines
Merged revisions 132777 via svnmerge from
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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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r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines
Merged revisions 132645 via svnmerge from
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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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r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines
Merged revisions 132712 via svnmerge from
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r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines
ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'
(closes issue #12160)
Reported by: tzafrir
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r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) | 3 lines
Fix an issue in iax2 where a call that's been rejected still kept an open channel on the side that attempted to make the call (not the side of the
call that rejected the call). Changes were load tested and also approved by Russell.
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r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines
Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax:
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
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Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which
also solved problems with contexts having
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.
Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
(at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
was forked, making special note of all revs that
were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
determine if it was involved in the merge_and_delete
enhancements in trunk. 25 commits were done in 1.6.0,
all but one (106306) was a merge from trunk.
Trunk had 22 additional changes, of which 7 were
involved in the merge_and_delete enhancements:
106757
108894
109169
116461
123358
130145
130297
f. Go to trunk and collect patches, one by one,
of the changes made by each rev across the
entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and
resolve all failures and compilation problems
before proceding to the next patch.
h. test the stuff.
i. profit!
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r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines
(closes issue #13041)
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
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r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines
Merged revisions 130889 via svnmerge from
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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
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r130126 | tilghman | 2008-07-11 12:29:24 -0500 (Fri, 11 Jul 2008) | 17 lines
Merged revisions 130102 via svnmerge from
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r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines
Pass the devicestate from an underlying channel up through the Agent channel.
This should make the Agent always report the correct device state, even when
the underlying channel is used for other purposes.
(closes issue #12773)
Reported by: davidw
Patches:
20080710__bug12773.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw
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r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines
Merged revisions 130039 via svnmerge from
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines
Merged revisions 128950 via svnmerge from
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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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r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines
Merged revisions 128639 via svnmerge from
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r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines
By using the iaxdynamicthreadcount to identify a thread, it was possible
for thread identifiers to be duplicated. By using a globally-unique monotonically-
increasing integer, this is now avoided.
(closes issue #13009)
Reported by: jpgrayson
Patches:
chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492)
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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines
- Fixing issues with "sip show settings"
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines
Adding doxygen comments to missing parts, moving some #define
...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...
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