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r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines
Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491
Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well.
(closes issue #12013)
Reported by: alx
Review: http://reviewboard.digium.com/r/213/
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In my tests that exercised full frame handling in chan_iax2, the version with
these changes took 30% to 40% of the CPU time compared to the same test of
Asterisk trunk before these modifications.
While doing some profiling for <http://reviewboard.digium.com/r/205/>,
one function that caught my eye was network_thread() in chan_iax2.c.
After the things that I was working on there, it was the next target
for analysis and optimization. I used oprofile's source annotation
functionality and found that the loop traversing the frame queue in
network_thread() was to blame for the excessive CPU cycle consumption.
The frame_queue in chan_iax2 previously held all frames that either were
pending transmission or had been transmitted and are still pending
acknowledgment.
In network_thread(), the previous code would go back through the main
for loop after reading a single incoming frame or after being signaled
because a frame had been queued up for initial transmission. In each
iteration of the loop, it traverses the entire frame queue looking for
frames that need to be transmitted. On a busy server, this could easily
be quite a few entries.
This patch is actually quite simple. The frame_queue has become only a list
of frames pending acknowledgment. Frames that need to be transmitted are
queued up to a dedicated transmit thread via the taskprocessor API.
As a result, the code in network_thread() becomes much simpler, as its only
job is to read incoming frames.
In addition to the previously described changes, this patch includes some
additional changes to the frame_queue. Instead of one big frame_queue, now
there is a list per call number to further reduce wasted list traversals.
The biggest impact of this change is in socket_process().
For additional details on testing and test results, see the review request.
Review: http://reviewboard.digium.com/r/212/
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r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines
Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
To drill into the xmpp to find the capabilities between channels, chan_gtalk
calls iks_child() and iks_next(). iks_child() and iks_next() are functions in
the iksemel xml parsing library that traverse xml nodes. The bug here is that
both iks_child() and iks_next() will return the next iks_struct node
*regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG,
which in most cases, it is, but in this case (a call being made from the
Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data
(they are extraneous whitespaces), and chan_gtalk doesn't handle that case,
so capabilities don't match, and a call cannot be made.
iks_first_tag() and iks_next_tag(), on the other hand, will not return the
very next iks_struct, but will check to see if the next iks_struct is of
type IKS_TAG. If it isn't, it will be skipped, and the next struct of type
IKS_TAG it finds will be returned. This assures that chan_gtalk will find
the iks_struct it is looking for.
This fix simply changes all calls to iks_child() and iks_next() to become
calls to iks_first_tag() and iks_next_tag(), which resolves the capability
matching.
The following is a payload listing from Empathy, which, due to the extraneous
whitespace, will not be parsed correctly by iksemel:
<iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
<payload-type clockrate='8000' name='PCMA' id='8'/>
<payload-type clockrate='8000' name='PCMU' id='0'/>
<payload-type clockrate='90000' name='MPA' id='97'/>
<payload-type clockrate='16000' name='SIREN' id='98'/>
<payload-type clockrate='8000' name='telephone-event' id='99'/>
</description>
</session>
</iq>
Review: http://reviewboard.digium.com/r/181/
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r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines
Make chan_misdn BRI TE side normally defer channel selection to the NT side.
Channel allocation collisions are not handled by chan_misdn very well.
This patch simply avoids the problem for BRI only.
For PRI, allocation collisions are still possible but less likely since
there are simply more channels available and each end could use a different
allocation strategy.
misdn.conf options available:
te_choose_channel - Use to force the TE side to allocate channels.
method - Specify the channel allocation strategy.
(closes issue #13488)
Reported by: Christian_Pinedo
Patches:
isdn_lib.patch.txt uploaded by crich
Tested by: crich, siepkes, festr
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r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
Improve our handling of T38 in the initial INVITE from a device.
We now answer with matching media streams to what is requested. If an INVITE
is received with both a T38 and RTP media stream this means we answer with both.
For any outgoing calls created as a result of this inbound one no T38 is requested
in the initial INVITE. Instead if we start receiving udptl packets we trigger a
reinvite on the outbound side.
(closes issue #12437)
Reported by: marsosa
Tested by: pinga-fogo, okrief, file, afu
Review: http://reviewboard.digium.com/r/208/
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The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
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r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines
Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
If calls were placed using an IP address or hostname the global nat setting was copied over
but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
actions.
(closes issue #14546)
Reported by: acunningham
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This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
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If we receive a T38 request negotiate control frame we should only attempt to do so
if the option is enabled on the dialog.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The default codec configuration for chan_iax2 is bandwidth=low. I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
A user was having an issue where if an outgoing SIP call was canceled, the SIP device
would remain in use if we had not received any response to the initial INVITE we sent out.
The SIP device would remain in use until the autocongestion timer was exhausted.
I tracked down the cause of this to be the section of code I am removing here. I asked several
people what the purpose of this code was meant to be, but no one could give me any sort of
answer as to why this was here. The person who was having this issue has been using this patch
for several months and it has stopped the problems they have had.
AST-196
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Previously we reached across the channel bridge to get the other party's SIP dialog
structure in order to trigger an outgoing reinvite. This is extremely dangerous to do
and only works if bridged to another SIP channel. This patch changes this to use the
T38 control frame method of requesting a switchover. This change also causes the SIP
channel driver to propogate back whether the switchover worked or not instead of blindly
accepting the incoming T38 reinvite.
Review: http://reviewboard.digium.com/r/200/
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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Chan_h323 can now be compiled against both the previously supported versions of
OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
script has been modified to look in the default install location of h323 to
hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
Also, the CLI command "h323 show version" has been added which indicates which
version of h323 is in use.
(closes issue #11261)
Reported by: vhatz
Patches:
asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed OPENR2 using an incorrect string value if the extension
delimiter is not present in the Dial() function. This was fixed for
SS7 and PRI in trunk -r172400.
* Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others.
* Removed trailing whitespace that appeared with OPENR2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
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r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
Randomize IAX2 encryption padding
The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
Review: http://reviewboard.digium.com/r/193/
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The code responsible for sending the T38 reinvite did not check if an INVITE was
already being handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current INVITE is done being
handled.
(issue AST-191)
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r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
If we receive an INVITE from an endpoint and then later receive a BYE from that
same endpoint before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487.
There was logic in the code prior to this commit which seemed to exist solely to
handle this situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
channel. This made no sense since we created the owner channel when we received
the INVITE, meaning that the majority of the time we would never send the 487.
The 487 being sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial INVITE transaction.
With this commit, that logic is now correctly in place.
(closes issue #14149)
Reported by: legranjl
Patches:
14149.patch uploaded by mmichelson (license 60)
Tested by: legranjl
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r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
(closes issue #14607)
Reported by: stevenla
Tested by: dvossel
Review: http://reviewboard.digium.com/r/192/
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r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
Fix issue where an attended transfer could not be completed under a rare scenario.
When completing an attended transfer chan_sip does a check to make sure the extension
in the URI portion of the Refer-To header is a local valid extension. We don't actually
need to check this since we know for sure the other channel is already up and talking to
the extension. Some devices do not put the extension in the Refer-To header either, which
can cause the extension check to fail. We now no longer do this check if it is an attended
transfer.
(closes issue #14628)
Reported by: sverre
Patches:
14628.diff uploaded by file (license 11)
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r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
When dtmfmode was set to auto the inband DTMF detector was not setup
on outgoing SIP calls. This caused inband DTMF detection to fail.
The inband DTMF detector is now setup for both dtmfmode inband and auto.
(closes issue #13713)
Reported by: makoto
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The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
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Previously, chan_sip had both sip_peer and sip_user objects in memory. A
patch went in to remove sip_user to simplify the code, since everything
could be done with just sip_peer. This patch resolves some regressions
found that were introduced by those changes.
This code comes from svn/asterisk/team/group/sip-object-matching/.
Here is a list of the changes that have been made:
1) When doing a match by name with the find_peer() function, make it much
easier to specify which objects should be matched by having a parameter
that specifies exactly which object types should be considered. Also,
update find_by_name() to handle this parameter. Finally, update all
code to use the new option values.
2) When looking up an object for an outbound request by name, consider
peers only. (create_addr())
3) Only match peers on an incoming registration request.
4) When doing authentication (except for SUBSCRIBE), look up users
by name, instead of all objects by name.
5) When doing authentication (except for SUBSCRIBE), after looking for
a user by name, look for a peer by IP address, instead of all objects
by IP address.
6) When handling the SIP qualify CLI command or manager action, look for
a peer by name, instead of any object by name.
7) When handling the SIP unregister CLI command, look for a peer by name,
instead of any object by name.
9) In sip_do_debug_peer(), search for a peer by name, instead of any object
by name.
9) When handling the SIPPEER() dialplan function, search for a peer by name,
instead of any object by name.
10) In the following session timer related functions, st_get_se(),
st_get_refresher(), and st_get_mode(), when looking for an object for a
given sip_pvt using pvt->peername, look for a peer by name, instead of any
object by name.
11) Fix build_peer() to properly handle the case where separate type=peer and
type=user entries were specified in sip.conf.
(closes issue #14505)
Reported by: lmadsen
Review: http://reviewboard.digium.com/r/172/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the subscription context for a call pickup subscription differs
from the context of the call pickup target, there's not an easy way
to divine what context should be used for the pickup. The way to work
around this is to use PICKUPMARK as the context for the pickup.
This has been documented in the sip.conf.sample file
(ABE-1708)
closes issue #14567
submitted by: alecdavis
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Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.
XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.
(closes issue #14455)
Reported by: Nick_Lewis
Patches:
14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.
(closes issue #10297)
Reported by: DEA
Patches:
skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj
Review: http://reviewboard.digium.com/r/130/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
(closes issue #14531)
Reported by: festr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous implementation of T38 faxdetect resulted in both sides of the
call jumping to a fax extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current context.
This revision will jump to a 'fax' extension on incoming calls only.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is required to create a UDPTL structure in create_addr_from_peer() to handle the
scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but
is defined the peer's context. I tested this patch by enabling t38pt_udptl in the
[general] section on one system and only enabling t38pt_udptl in a peer's context on
the system sending a fax. Without the patch, the sending system will fail to initiate
T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
When this patch is applied the sending side will successfully initiate T38 negotiation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines
After a 'sip reload', qualifies for realtime peers weren't immediately
restarted, instead waiting until the next registration. We're now
caching the qualify across a reload/restart and starting the qualify
immediately upon loading the peer.
(closes issue #14196)
Reported by: pdf
Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines
Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that.
issue #13749
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
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r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
Fixed iax2 key rotation backwards compatibility
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines
Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
pool was used for the value while the old was left untouched/unused. If the current pool was full a new
pool was created. This would cause memory usage to increase steadily.
(issue #AA50-2332)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) It had numerous coding guidelines violations with regards to formatting.
2) It allocated memory using ast_calloc() that was never freed.
3) It didn't check for failure from the allocation.
4) It used sprintf() and strcat() to build the result, doing zero checking to
prevent writing past the end of the provided buffer.
The function also lacks API documentation, but that has not been addressed in
this commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
Review: http://reviewboard.digium.com/r/159/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf. This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones. The faxbuffers
buffer policy will be reverted during call teardown.
An example use of 'faxbuffers' is below. This example would switch to using
6 buffers with a full buffer policy.
faxbuffers=>6,full
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well.
(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines
Set the initiator attribute to lowercase in our replies when receiving calls.
This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.
(closes issue #13984)
Reported by: jcovert
Patches:
chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines
Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.
(closes issue #14419)
Reported by: klaus3000
Patches:
patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines
check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()
The reporter didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to determine the
changes, then modified the suggested changes to create a proper fix. The
summary above is a complete description of the changes.
(closes issue #13547)
Reported by: tecnoxarxa
Patches:
chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines
Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
(closes issue #14350)
Reported by: fhackenberger
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r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines
Remove a debug message I put in by accident.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread. chan_iax2 has been
updated to use this new code. Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.
Related to rev 171452 which resolved the same issues in 1.4.
Code from team/russell/sched_thread2
Review: http://reviewboard.digium.com/r/129/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2.
(closes issue #13468)
Reported by: nicox
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
supporting devices. The devices (snoms, specifically) need to receive a SIP
URI instead of just an extension. This adds that functionality.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines
Fix devicestate problems for "always-on" agent channels
A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.
The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.
The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.
The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.
In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.
(closes issue #14173)
Reported by: nathan
Patches:
14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines
Use the same branch tag in CANCEL as in INVITE
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
(closes issue #14346)
Reported by: oej
Patches:
bug14346.diff uploaded by oej (license 306)
Tested by: oej
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ccesario on IRC pointed out that his sip peers were not displayed
properly when he would issue the command "sip show peers." The problem
was that the onlymatchonip field was used to determine if the endpoint
was a "peer" or "user." The tricky part is that a "friend" is supposed
to be treated as both a "user" and a "peer" but the logic would not allow
"friends" to show up as "peers" since onlymatchonip was set to FALSE
for friends.
I have modified the sip_peer structure to more explicitly keep track of
what type endpoint it is so that the various manager and CLI commands
will display the expected information
Reported by ccesario via IRC
Tested by ccesario
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an ast_str expands to hold more data, any pointers that were pointing
to the data prior to the expansion will be pointing at invalid memory. This
change makes such pointers used in chan_sip.c instead be offsets from the
beginning of the string so that the same math may be applied no matter where
in memory the string resides.
To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added
to chan_sip.c so that given a sip_request and an offset, the string at that
offset is returned.
(closes issue #14220)
Reported by: riksta
Tested by: putnopvut
Review http://reviewboard.digium.com/r/126/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dan saw regular segfaults with the old implementation and
rewrote it to make it really eventbased.
I altered it to be trunk compatible and wedhorn gave some feedback
and ideas how to make it even better.
(closes issue #13821)
Reported by: DEA
Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3)
Tested by: mvanbaak, DEA
"no probs by me" from wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines
Prevent a crash in chan_local due to a potential NULL pointer dereference
Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.
(closes issue #14189)
Reported by: sascha
Patches:
14189.patch uploaded by putnopvut (license 60)
Tested by: sascha
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
David Vossel
Russell Bryant
David Brooks
Richard Mudgett
Joshua Colp
Kevin P. Fleming
Leif Madsen
Mark Michelson
Eliel C. Sardanons
Tilghman Lesher
Jeff Peeler
Olle Johansson
Michiel van Baak
Dwayne M. Hubbard
Philippe Sultan
Matthew Nicholson
Sean Bright
Steve Murphy
Matthew Fredrickson
Doug Bailey
Terry Wilson