First of all, the code was unnecessary. The goal was to lock a channel
which was already locked. Second, the assumption of the deadlock avoidance
loop was that the sip_pvt was already locked and we were trying to get the
channel lock. The problem is that the sip_pvt was unlocked a few lines above.
Basically, I'm removing 5 lines of no-op.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
Add error message so that it is clear why a SIP peer was not processed when
a DNS lookup fails on a host or outboundproxy.
(closes issue #13432)
Reported by: p_lindheimer
Patches:
outboundproxy.patch uploaded by p (license 558)
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r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
Fix a problem where chan_sip would ignore "old" but valid responses.
chan_sip has had a problem for quite a long time that would manifest when
Asterisk would send multiple SIP responses on the same dialog before receiving
a response. The problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two requests out,
and a response arrived for the first request sent, then Asterisk would ignore
the response. The result was that Asterisk would continue retransmitting the
requests and ignoring the responses until the maximum number of retransmissions
had been reached.
The fix here is to rearrange the code a bit so that instead of simply comparing
the sequence number of the response to our latest outgoing sequence number, we
walk our list of outstanding packets and determine if there is a match. If there is,
we continue. If not, then we ignore the response.
In doing this, I found a few completely useless variables that I have now removed.
(closes issue #11231)
Reported by: flefoll
Review: https://reviewboard.asterisk.org/r/298
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r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
Fix build oops.
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
Resolve a crash related to a T.38 reinvite race condition.
This change resolves a crash observed locally during some T.38 testing.
A call was set up using a call file, and when the T.38 reinvite came in,
the channel state was still AST_STATE_DOWN. The reason is explained by
a comment in the code that previously lived in the handling of
AST_STATE_RINGING. This change modifies the logic to handle the same
race condition for any channel state that is not UP.
(closes ABE-1895)
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r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines
Fix a situation in which Asterisk would not stop retransmitting 487s.
If a CANCEL were received by Asterisk, we would send a 487 in response
to the original INVITE and a 200 OK for the CANCEL. If there were a network
hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating
with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used
to be to try sending another 487 to the canceled INVITE and another 200 OK to the
CANCEL.
The problem here is that the originally-sent 487 was sent "reliably" meaning that
it will be retransmitted until it is received properly. So when we receive the second
CANCEL it is likely that the first batch of 487s we sent is still going strong and
reaches the UA. The result was that the second set of 487s would be retransmitted
constantly until the maximum number of retries had been reached.
The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel
the retransmission of the first set of 487s and start a second set. This causes the
dialog to be terminated reasonably.
(closes issue #14584)
Reported by: klaus3000
Patches:
14584_v2.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
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r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines
Remove an extra debug line left from previous commit.
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r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines
Fix a possible infinite loop in SDP parsing during glare situation.
There was a while loop in get_ip_and_port_from_sdp which was controlled
by a call to get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is that
get_sdp_iterate never returns NULL. This means that if what we were searching
for was not present, the loop would run infinitely. This modification of the
loop fixes the problem.
(closes issue #15213)
Reported by: schmidts
(closes issue #15349)
Reported by: samy
(closes issue #14464)
Reported by: pj
(closes issue #15345)
Reported by: aragon
Patches:
sip_inf_loop.patch uploaded by mmichelson (license 60)
Tested by: aragon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'.
(closes issue #15111)
Reported by: ffs
Patches:
chan_sip.c_register-parser.patch uploaded by ffs (license 730)
Tested by: ffs, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to our SDP version comparison made audio not function
on some calls. This was because of a test wherein we were trying to
see if an unsigned value was less than 0. This is a dumb comparison
and arguably the compiler should have warned about it. Alas, though,
it slipped past. Now it's fixed by changing the variable to be a
signed type.
Found by several developers. Tested by mnicholson and dbrooks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines
Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read()
Zombie channels could be passed, and chan_sip.c wasn't checking for it.
Could crash Asterisk. Now checking for NULL pointer.
(closes issue #15330)
Reported by: okrief
Tested by: dbrooks
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During a sip reload, the list of sip_registry objects are
supposed to be traversed, unlinked, and destroyed, but
destruction never takes place due to a ref counting error.
This causes a memory leak when registry items are removed
from sip.conf and reloaded. While the registries are removed
from the global list, they are not removed from the scheduler.
Because of this, SIP register attempts continue to be sent
out for the item even though it may no longer be in the .conf.
(closes issue #15295)
Reported by: amorsen
Review: https://reviewboard.asterisk.org/r/282/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
What this patch addresses:
1. ast_sip_ouraddrfor() by default binds to the UDP address/port
reguardless if the sip->pvt is of type UDP or not. Now when no
remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
transport type, attempting to set the address and port to the
correct TCP/TLS bindings if necessary.
2. It is not necessary to send the port number in the Contact
header unless the port is non-standard for the transport type.
This patch fixes this and removes the todo note.
3. In sip_alloc(), the default dialog built always uses transport
type UDP. Now sip_alloc() looks at the sip_request (if present)
and determines what transport type to use by default.
4. When changing the transport type of a sip_socket, the file
descriptor must be set to -1 and in some cases the tcptls_session's
ref count must be decremented and set to NULL. I've encountered
several issues associated with this process and have created a function,
set_socket_transport(), to handle the setting of the socket type.
(closes issue #13865)
Reported by: st
Patches:
dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
13865.patch uploaded by mmichelson (license 60)
tls_port_v5.patch uploaded by vrban (license 756)
transport_issues.diff uploaded by dvossel (license 671)
Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel
Review: https://reviewboard.asterisk.org/r/278/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit changes the 'incoming SDP version' check logic a bit more; when
'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
switch to T.38, we'll always accept the peer's SDP response, even if they
don't properly increment the SDP version number as they should. If this situation
occurs, a warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion' configuration option
(although ideally they'd fix their SIP implementation to be RFC compliant).
AST-221
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These modules all contained variables that are module-global but not system-global,
but were not marked 'static'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Using the 'pahole' tool, it is now quite easy to see where structure fields
could be organized differently to keep the compiler from having to add
padding to satisfy alignment requirements. These changes reduced the sizes of
sip_pvt and sip_peer by a few bytes each (on 64-bit platforms), and also fixed
a spelling error in a field name.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer.
(closes issue #15283)
Reported by: jthurman
Patches:
sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
Tested by: jthurman, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.
* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.
* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.
ABE-1873
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes sure that we mark a method as being unallowed if we
receive a 405 response so that we don't continue to try to
send that same type of message.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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
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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.
EDVX002
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@196416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
