When receiving a 200 OK response to an INVITE, it was possible to transmit two
connected line updates instead of a single one. Furthermore, the second did not
have the proper information present.
Now the two have been combined into a single update and the correct information
is presented.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since we may have copied connected line info into the chanlist struct prior
to placing an outbound call, we need to be sure to free the allocated data
when we hang the call up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP purists may want to look the other way...
When COLP/CONP support for SIP was committed, there was a condition under
which Asterisk may transmit a SIP UPDATE in order to communicate the change
in connected line information. The issue here is that while we could send a
SIP UPDATE message, we were not prepared to receive such an UPDATE and would
always responde with a 501 when we received an UPDATE.
The situation was a bit rough. We really want to be able to receive UPDATEs
having to do with connected line changes, but the amount of effort involved
in properly supporting RFC 3311 was staggering. This commit represents a
compromise.
First, it was decided that it is important to only send a SIP UPDATE to
an endpoint that is able to handle one. So, now we have added parsing of
the Allow header into SIP. We store the allowed methods on SIP peers so
that when we communicate with them, we already will know what we can and
cannot send to them. We will parse the peer's allowed methods when he registers
with us. If the peer is not the type to register with us, but the qualify option
is enabled, then we will use the response to the OPTIONS request we send
the peer to determine the peer's allowed methods. When the peer's registration
expires, or when qualify deems the peer to be unreachable, we clear the allowed
methods from the peer.
For an actual call, we will copy the peer's allowed methods to the sip_pvt
representing the call leg. If we are communicating with an endpoint which is
not a peer, then we will just parse the Allow header from the first message
we receive during the call and store the information in the sip_pvt.
If, during communication with a peer, we receive a 501 response, then we will
make sure to save the fact that we cannot use that method when communicating
with that peer.
Now, with all that infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line change using
an UPDATE request. If we cannot send the change immediately using an UPDATE,
we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon
as it is allowed.
The second part of the changes here is for Asterisk to accept UPDATE requests
that have connected line changes. Since we are not fully supporting RFC 3311,
Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead,
if you are communicating with what you know to be another Asterisk box, you may
set the rpid_update parameter in sip.conf so that we will send UPDATEs to that
Asterisk box. When we send a connected line update, we set a custom header
called "X-Asterisk-rpid-update."
On the receiving end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501
since media-changing UPDATEs are not supported. We should never get such
UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow
header. If the custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and queue the update
on the channel where the change occurred.
ABE-1840
ABE-1822
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@195369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI command 'manager show command' supports passing multiple AMI actions
at a time. The issue with this command was in another place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When running the autocomplete of the CLI command 'manager show command <action>'
it was autocompleting everything else after the <action> argument, giving an error,
because this command doesn't support multiple AMI action names at a time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@195320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@195316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines
Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present.
(closes issue #15105)
Reported by: bamby
Patches:
process-vad-correctly.diff uploaded by bamby (license 430)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Up to now, cdr_custom would only accept a single filename/format from
cdr_custom.conf. This change allows you to specify multiple filename
& format directives.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@195162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Avoid duplicating xml documentation by allowing to include other parts of
the xml documentation using XInclude.
Example:
<xi:include xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function xml
documentation).
It is also possible to include documentation from other files in the
'documentation/' directory using the href="" attribute inside a xinclude
element.
(closes issue #15107)
Reported by: lmadsen
(issue #14444)
Reported by: ewieling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Always free the allocated memory for a string field, because
we are always using it (not only when xmldocs are enabled).
Also if there is an error allocating memory for the string field
remember to unlock the list of registered applications, before returning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was not possible to use an enumlist inside an enum:
<enumlist>
<enum name="aa">
<enumlist>
...
</enumlist>
</enum>
</enumlist>
Now we will be able to insert as many levels as we want.
(closes issue #15112)
Reported by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf.
Review: https://reviewboard.asterisk.org/r/244/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Select what to do with outgoing COLP information on this port.
0 - Send out COLP information unaltered. (default)
1 - Force COLP to restricted on all outgoing COLP information.
2 - Do not send COLP information.
outgoing_colp=0
Also fixed sending the EctInform message so it always has the
required redirectionNumber parameter when the status is active.
JIRA ABE-1853
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/trunk@194357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a typo in the structure being locked, and we were locking the
'sessions' container instead of the 'session' structure thar we are modifying.
Reported by seanbright on #asterisk-dev, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines
Fix logic for how to proceed with a single digit extension.
(closes issue #15091)
Reported by: andrew
Patches:
20090512__issue15091.diff.txt uploaded by tilghman (license 14)
Tested by: andrew
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) If MALLOC_DEBUG is used when concurrently using ast_backtrace, the free()
used in that routine will trigger an error, because the memory was allocated
internally to libc, where we could not intercept that call to wrap it.
Therefore, it's not memory we knew about, and the free is reported as an
error.
2) Now that channels are objects, the old hack of initializing a channel
to all zeroes no longer works, since we may try to call something like
ast_channel_lock() within a function on that reference. In that case, it's
reported as an error, because the pointer isn't an object reference.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the user logout the session was being destroyed twice and the file
descriptor was being closed twice. The sessions reference counter wasn't
used in a proper way.
The 'mansession' structure was being treated as an astobj2 and we were
calling ao2_lock/ao2_unlock causing astobj2 report a warning message and
not locking the structure.
Also we were using an ugly naming convention 'destroy_session',
'session_destroy', 'free_session', ... all this "duplicated" code was merged.
(closes issue #14974)
Reported by: pj
Patches:
manager.diff2 uploaded by eliel (license 64)
Tested by: dhubbard, eliel, mnicholson
(closes issue #15088)
Reported by: eliel
Review: http://reviewboard.asterisk.org/r/248/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines
This change modifies app_queue to properly generate CDR records in failure
situations.
This involves setting a proper cdr disposition coresponding to the given
failure condition and ensuring the proper information is stored in the cdr
record.
(closes issue #13691)
Reported by: dferrer
Tested by: mnicholson
(closes issue #13637)
Reported by: atis
Tested by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@193954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
