Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
David Vossel [Thu, 1 Jul 2010 16:40:17 +0000 (16:40 +0000)]
correct handling of get_destination return values
A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
........
David Vossel [Tue, 29 Jun 2010 20:44:05 +0000 (20:44 +0000)]
send a 400 Bad Request on malformed sip request
RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
David Vossel [Mon, 28 Jun 2010 18:38:47 +0000 (18:38 +0000)]
rfc compliant sip option parsing + new unit test
RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.
This is not currently being done correctly. Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test verifying
this functionality has been created. Some code refactoring was required.
David Vossel [Mon, 28 Jun 2010 14:55:25 +0000 (14:55 +0000)]
code guidelines cleanup for retrans_pkt() function
I am doing work in this function. I noticed a large number of
coding guidline fixes that needed to be made. Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.
David Vossel [Fri, 25 Jun 2010 19:39:53 +0000 (19:39 +0000)]
chan_sip: more accurate retransmissions
RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to. This occurs because
the scheduler runs in the do_monitor thread. If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms. To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits. The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.
Russell Bryant [Wed, 23 Jun 2010 23:09:28 +0000 (23:09 +0000)]
Resolve some errors produced during module unload of chan_iax2.
The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Terry Wilson [Wed, 23 Jun 2010 17:21:40 +0000 (17:21 +0000)]
Make sure reload updates SLA config
Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.
Mark Michelson [Wed, 23 Jun 2010 17:08:34 +0000 (17:08 +0000)]
Add extra protection for reinvite glare scenario.
Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.
The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.
The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.
Russell Bryant [Tue, 22 Jun 2010 23:20:37 +0000 (23:20 +0000)]
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.
This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.
David Vossel [Tue, 22 Jun 2010 20:37:05 +0000 (20:37 +0000)]
ignore CANCEL request after having already received final response to INVITE
RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response. This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".
Jeff Peeler [Tue, 22 Jun 2010 16:29:18 +0000 (16:29 +0000)]
Add regular expression filtering for manager events.
This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
David Vossel [Tue, 22 Jun 2010 15:46:22 +0000 (15:46 +0000)]
fixes attended transfer behavior when both transferee and transferer hung up
If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to. This mirrors the expected behavior used in 1.4.
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.
Fix crash when parsing some heavily nested statements in AEL on reload.
Due to the recursion used when compiling AEL in gen_prios, all the stack space
was being consumed when parsing some AEL that contained nesting 13 levels deep.
Changing a few large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used.
(This is a backport of 269307, committed to trunk by rmudgett.)
Calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.
David Vossel [Thu, 17 Jun 2010 18:45:32 +0000 (18:45 +0000)]
retransmit response to BYE requests until timer J expires
According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms. This results in some odd behavior. For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.
To resolve this I made a function called sip_scheddestroy_final(). This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards. It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time. This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.
Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy. sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.
There is code in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code works once
the thread is added to an active_list of threads, but we are not currently
guaranteed that a newly activated thread will enter the active_list immediately
because it is left up to the thread to add itself after frames have been
queued to it. This means that if two frames come in for the same call number
at the same time, it is possible for them to grab two separate threads because
the first thread did not add itself to the active_list fast enough. This
causes some pretty complex problems.
This patch resolves this race condition by immediately adding an activated
thread to the active_list within the network thread and only depending on
the thread to remove itself once it is done processing the frames queued to
it. By doing this we are guaranteed that if another frame for the same call
number comes in at the same time, that this thread will immediately be found
in the active_list of threads.
Terry Wilson [Tue, 15 Jun 2010 21:42:33 +0000 (21:42 +0000)]
Don't continue sending the file when there has been an error
If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.
Terry Wilson [Tue, 15 Jun 2010 21:10:15 +0000 (21:10 +0000)]
Don't send files twice and remove extra \r\n from header
After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.
Terry Wilson [Tue, 15 Jun 2010 20:18:04 +0000 (20:18 +0000)]
Make contactdeny apply to src ip when nat=yes
chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.
Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.
This commit makes contactdeny apply to the src ip when nat=yes instead.
projects / thirdparty / asterisk.git / log
