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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines
Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
Reported by: davidw
Patches:
20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw, jvandal, murf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133665 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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https://origsvn.digium.com/svn/asterisk/trunk
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
commands that conflicted with adding new features to the newer debug
commaands.
(closes issue #13103)
Reported by: mvanbaak
Patches:
2008071901__issue13103_iax2_set_debug_peer.diff uploaded by
mvanbaak (license 7)
Tested by: bbryant, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines
Revert part of issue #5620 (revision 6965) as it appears that it was in error.
This should fix talk call progress on analog lines.
(closes issue #12178)
Reported by: michael-fig
Patches:
20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
instead. DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is
the "real" state of that device across all servers sharing state. This would have
only been a problem with distributed device state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines
Ensure that a destination callno of 0 will not match for frames that do not
start a dialog (new, lagrq, and ping).
(closes issue #12963)
Reported by: russellb
Patches:
chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines
Fix handling of when a pvt disappears. Properly return the pvt locked
and don't hold the pvt lock while destroying the ast_channel.
(closes issue #13014)
Reported by: jpgrayson
Patches:
chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
...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...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Actually, kill the in-memory structure for type=user and start using the sip_peer
structure for every object. Have only one in-memory list and use them different
ways depending on type=user, type=peer and type=friend - like always.
The idea with this first patch is that configurations should work as before.
Some additional features for realtime peers. By adding a type= field, you
can now have multiple functions.
Let's test this for a while. Won't be integrated in 1.6.0, only in trunk,
for testing.
There's propably more to clean up and simplify here. Help is welcome
and encouraged!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's mixing peers and users in a strange way and should really not be a CLI command,
since it's not meant for human output. It should be done with an app connecting to
manager.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Kevin P. Fleming
Mark Michelson
Russell Bryant
Jason Parker
Brett Bryant
Christian Richter
Olle Johansson
Jeff Peeler
Matthew Fredrickson
Luigi Rizzo
Michiel van Baak
Sean Bright
Joshua Colp
Steve Murphy