https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
Reduce CPU usage related to building a peer merely for devicestates.
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.
(closes issue #15943)
Reported by: tpsast
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This way, we don't always write a null byte into
byte 1 of the buffer
(closes issue #15905)
Reported by: ebroad
Patches:
freadfix.patch uploaded by ebroad (license 878)
Tested by: ebroad
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This was preventing responses from being properly processed because the packet was not being found
causing handle_response to return prematurely.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Remove thread_spawned in handle_init_event since it was never used
* Always check handle_init_event in case a channel is destroyed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the course of this, I also found that the results of ast_gethostbyname
were being used incorrectly in both chan_iax2 and chan_sip, so both have
been fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed walking the iflist so it is always done with the iflock locked.
* Simplified iflist walking routines.
* Created chan_dahdi iflist insertion and extraction routines.
* Fixed duplicate_pseudo() malloc fail handling.
* Fixed infinite loop in action_dahdishowchannels() when showing a single channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Related to #12713
Patch by oej
A big thank you to file for finally fixing the transfer() dialplan application.
I've been waiting for years for this. Great work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
(closes issue #15839)
Reported by: ebroad
Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
