Olle Johansson [Wed, 27 May 2009 20:14:00 +0000 (20:14 +0000)]
Adding some generic handling of error codes sent to us in replys to requests.
Previously they always set hangupcause 0, which is generally wrong. With this
change, we're setting some generic hangup causes. For 5xx errors, which indicate
some sort of problem with the remote server, we're now setting CONGESTION.
Sean Bright [Wed, 27 May 2009 20:08:16 +0000 (20:08 +0000)]
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.'
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
........
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.
Fix handling of the 'state_interface' option of the 'queue add member' CLI
command.
This change relates to r184980, which was a backport of the state interface
changes to app_queue from trunk. trunk and all of the 1.6.x branches are not
affected.
'queue add member' allows for specifying an interface to use for device state
when adding a queue member via CLI, but the validation code was not properly
updated to reflect this optional argument.
Sean Bright [Wed, 27 May 2009 13:02:54 +0000 (13:02 +0000)]
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)
Sean Bright [Tue, 26 May 2009 22:38:05 +0000 (22:38 +0000)]
Add ActionID to CoreShowChannel event.
There is inconsistency in how we handle manager responses that are lists of
items and, unfortunately, third parties have come to rely on ActionID being on
every event within those lists instead of just keeping track of the ActionID for
the current response. This change makes CoreShowChannels include the ActionID
with each CoreShowChannel event generated as a result of it being called.
(closes issue #15001)
Reported by: sum
Patches:
patchactionid2.patch uploaded by sum (license 766)
Sean Bright [Tue, 26 May 2009 20:20:08 +0000 (20:20 +0000)]
Include startup and reload date in the CoreStatus manager message.
The CoreStartupTime and CoreReloadTime name/value pairs in the CoreStatus
response message only included the time and not the date. This patch,
inspired by the reporter's patch, adds 2 new fields - CoreStartupDate and
CoreReloadDate - which contain the date portion of these values.
Russell Bryant [Tue, 26 May 2009 18:20:57 +0000 (18:20 +0000)]
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.
Sean Bright [Sat, 23 May 2009 15:16:59 +0000 (15:16 +0000)]
Fix errors in cdr_custom that cause reference errors when non-CDR variable
substitution is done.
cdr_custom was creating a ast_channel struct directly and passing it into the
core for variable substition. This was fine as long as the format string
contained only calls to the CDR() function. Doing something like ${EPOCH} on
the other hand tried to lock the channel, which would fail and throw an error
because the passed channel hadn't been allocated as an ao2 object. So now we
create the dummy channel with ast_channel_alloc, and everything works as
expected.
David Vossel [Fri, 22 May 2009 21:09:45 +0000 (21:09 +0000)]
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.
Implement a new element in AstXML for AMI actions documentation.
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
Sean Bright [Fri, 22 May 2009 16:51:22 +0000 (16:51 +0000)]
Fix res_agi compilation after the const-ify the world merge.
Since we are dealing with a 'const char * const' now, we have to create a
temporary copy of the string to work on rather than the original. Fix inspired
by reporter. Reviewed by everyone-and-their-mother in #asterisk-dev.
Avoid using prototypes when not necessary (it is already defined in the header
file).
Make log_match_char_tree() static to main/pbx.c (only used there).
Kevin P. Fleming [Thu, 21 May 2009 21:13:09 +0000 (21:13 +0000)]
Const-ify the world (or at least a good part of it)
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
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.
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.
Mark Michelson [Wed, 20 May 2009 20:45:05 +0000 (20:45 +0000)]
Get rid of some duplicated code and correct a connected line error.
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.
Mark Michelson [Wed, 20 May 2009 20:14:28 +0000 (20:14 +0000)]
Plug a memory leak in app_dial.
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.
Mark Michelson [Tue, 19 May 2009 20:59:38 +0000 (20:59 +0000)]
Add basic support for handling connected line-related UPDATE requests.
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.
Add a similar dependency on SMDI for voicemail as already exists for ADSI.
(closes issue #14846)
Reported by: pj
Patches:
20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14)
20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14)
20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14)
........
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.
Avoid autocompleting passed the action name argument in the CLI command.
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.
Tilghman Lesher [Mon, 18 May 2009 19:17:15 +0000 (19:17 +0000)]
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
Mark Michelson [Mon, 18 May 2009 18:58:26 +0000 (18:58 +0000)]
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.
Sean Bright [Mon, 18 May 2009 14:54:43 +0000 (14:54 +0000)]
Allow cdr_custom to write to multiple files instead of just one.
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.
Allow to include sections of other parts of the xml documentation.
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.
Fix a missing unlock in case of error, and a missing free().
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.
IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames.
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.
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.
Kevin P. Fleming [Fri, 15 May 2009 13:13:47 +0000 (13:13 +0000)]
Add ability for modules to dynamically register logger levels
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.
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.
Richard Mudgett [Thu, 14 May 2009 22:03:49 +0000 (22:03 +0000)]
Add outgoing_colp misdn.conf port parameter.
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.
Tilghman Lesher [Thu, 14 May 2009 16:22:14 +0000 (16:22 +0000)]
If the timing ended on a zero, then we would loop forever.
(closes issue #14983)
Reported by: teox
Patches:
20090513__issue14983.diff.txt uploaded by tilghman (license 14)
Tested by: teox
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)
........
Do not lock the 'sessions' container, lock the allocated 'session'.
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!
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
........
Tilghman Lesher [Wed, 13 May 2009 00:13:43 +0000 (00:13 +0000)]
Two fixes found while debugging with ast_backtrace():
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.
Fix a crash when logging out from the AMI and avoid astobj2 warning messages.
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.
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.
Mark Michelson [Tue, 12 May 2009 20:28:13 +0000 (20:28 +0000)]
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.
Set the invitestate to INV_CANCELLED only if we are actually sending a SIP CANCEL.
The problem was that the hangup code was setting the invitestate too early. The result of
this was that we would always send a CANCEL request, even if it was not an appropriate
time to do so (e.g. we have not yet received a provisional response for our INVITE).
Note that this same fix had been applied to trunk and the 1.6.X branches starting with
revision 155467. This is why you will see this revision being blocked from those places.
projects / thirdparty / asterisk.git / log
