https://origsvn.digium.com/svn/asterisk/trunk
................
r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines
Merged revisions 197562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
Use the address we already know when reloading a peer with nat=yes.
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
Merged revisions 197537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines
Merged revisions 197466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines
Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile.
Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
explicitly when running from the Makefile, otherwise we get errors during a
'make install.'
(closes issue #15209)
Reported by: seandarcy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines
Recorded merge of revisions 197194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
Use a different determinator on whether to print the delimiter, since leading fields may be blank.
(closes issue #15208)
Reported by: ramonpeek
Patch by me, though inspired in part by a patch from ramonpeek
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The bridge was terminating immediately after the attended transfer was
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines
Display an error message when chan_alsa fails to load due to a missing
or inaccessible configuration file.
Before this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed. With this
change, when chan_alsa fails to load due to a missing or inaccessible
configuration file, a message will be displayed.
(closes issue #14760)
Reported by: Nick_Lewis
Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines
Merged revisions 196826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines
Resolve a file handle leak.
The frames here should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became reference counted,
this code would leak the file stream for the file being read.
(closes issue #15181)
Reported by: jkroon
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines
Merged revisions 195991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines
Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer.
There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines
Merged revisions 195881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
Fix the CLI command 'manager show command' documentation and functionality.
The CLI command 'manager show command' supports passing multiple action names in
the same line, but it was not allowing that because of a incorrect check in the
argumentes counter. Also the documentation was updated to show that this usage
of the command is possible.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
Move the spawn of astcanary down, until after the call to daemon(3).
This avoids possible conflicts with the internal implementation of
daemon(3).
(closes issue #15093)
Reported by: tzafrir
Patches:
20090513__issue15093__2.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May 2009) | 18 lines
Fix externalivr's setvariable command so that it properly sets multiple variables.
The command had a for loop that was guaranteed to only execute once since
the continuation operation of the loop would set the input buffer NULL. I rewrote
the loop so that its operation was more obvious, and it would set multiple variables
correctly.
I also reduced stack space required for the function, constified the input string,
and modified the function so that it would not modify the input string while I was
at it.
(closes issue #15114)
Reported by: chris-mac
Patches:
15114.patch uploaded by mmichelson (license 60)
Tested by: chris-mac
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
Warn about the use of the application WaitExten() within a Macro().
Update applications documentation to warn the user about the use of the
WaitExten() application within a Macro(). Recommend the use of Read()
instead.
(closes issue #14444)
Reported by: ewieling
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines
Merged revisions 194557,194685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines
IAX2 "Ghost" Channels
There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away.
(closes issue #14207)
Reported by: clive18
Review: https://reviewboard.asterisk.org/r/246/
........
r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
Update to previous IAX2 "Ghost" Channels patch.
Fixed some comments made on reviewboard for the previous patch.
(issue #14207)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines
Merged revisions 194484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines
Fix a race condition where a reinvite could trigger a 482 response.
The loop detection/spiral detection code in chan_sip used the owner
channel's state as a criterion for determining if the incoming INVITE
is a looped request. The problem with this is that the INVITE-handling
code happens in a different thread than the thread that marks the owner
channel as being up. As a result, if a reinvite were to come in very quickly,
say from another Asterisk on the same LAN, it was possible for the reinvite
to arrive before the owner channel had been set to the up state.
This patch corrects the problem by using the invitestate of the sip_pvt
instead, since that can be guaranteed to be set correctly by the time
the reinvite arrives. Since there is a switch statement further in the
INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate
of the sip_pvt in case we should actually be treating the channel as if it were
up already.
(closes issue #12215)
Reported by: jpyle
Patches:
12215_confirmed.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r194357 | mmichelson | 2009-05-13 14:42:51 -0500 (Wed, 13 May 2009) | 18 lines
Blocked revisions 194356 via svnmerge
........
r194356 | mmichelson | 2009-05-13 14:41:44 -0500 (Wed, 13 May 2009) | 13 lines
Remove an extraneous unlocking operation from ast_channel_free.
In the case that we could not remove the desired channel from the
list of channels, there was an extra call to unlock the channel list.
Since we unlock the list later on in the function anyway, this results
in the list being unlocked twice yet only being locked once.
(closes issue #15098)
Reported by: tim_ringenbach
Patches:
remove_extra_unlock.diff uploaded by tim (license 540)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines
Update spiral support in trunk and 1.6.X to match what is in 1.4.
In 1.4, a SIP spiral is treated the same way as a call forward. This
works much better than what is currently in trunk and 1.6.X. The code
in trunk and 1.6.X did not create a new call to the recipient of the spiral,
instead trying to continue the same call. In addition to just being plain
wrong, this also had the side effect of only being able to spiral calls
to other SIP channels.
With this in place, as long as call forwards are honored, SIP spirals
will work properly. This means that it will work for outbound calls
made by the Queue, Dial, and Page applications. For originated calls and
spool calls, however, the spiral will not work properly until a generic
call forward mechanism is introduced into Asterisk.
(relates to issue #13630)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
