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sent back instead of 1 if the 's' extension did not exist.
(closes issue #12286)
Reported by: lmamane
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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.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
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the calling channel was answered.
(issue #13545)
Reported by: davidw
(issue #14244)
Reported by: mbnwa
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back to the caller call leg when reinvited.
(closes issue #13569)
Reported by: bkw918
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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.
(Related to Security fix AST-2009-001)
(closes issue #14867)
Reported by: aragon
Tested by: dvossel
(closes issue #14717)
Reported by: mobeck
Patches:
regauth_loop_update_patch.diff uploaded by dvossel (license 671)
Tested by: dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@194873 f38db490-d61c-443f-a65b-d21fe96a405b
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Fixed some comments made on reviewboard for the previous patch.
(issue #14207)
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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.
(closes issue #14207)
Reported by: clive18
Review: https://reviewboard.asterisk.org/r/246/
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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.
(closes issue #12215)
Reported by: jpyle
Patches:
12215_confirmed.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
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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.
AST-216
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responed yet.
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
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(closes issue #14976)
Reported by: alecdavis
Patches:
misdn_config.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, FabienToune
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not match.
Added the dialed extension and context to the chan_misdn messages warning
that the dialed number cannot be matched in the dialplan.
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If the fullcontact field appears in both the sippeers and the
sipregs table, then during reconstruction of the field, it will
otherwise be doubled.
(closes issue #14754)
Reported by: Alexei Gradinari
Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@192932 f38db490-d61c-443f-a65b-d21fe96a405b
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the core if rfc2833 is not enabled.
(closes issue #15036)
Reported by: dimas
Patches:
v1-15036.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@192633 f38db490-d61c-443f-a65b-d21fe96a405b
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mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
(closes issue #14728)
Reported by: dimas
Patches:
v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@192213 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14993)
Reported by: BigJimmy
Patches:
causepatch uploaded by BigJimmy (license 371)
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If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
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Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@189991 f38db490-d61c-443f-a65b-d21fe96a405b
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testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@189391 f38db490-d61c-443f-a65b-d21fe96a405b
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AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
(closes issue #14091)
Reported by: evandro
Patches:
autologoff.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/225/
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JIRA ABE-1835
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This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@188946 f38db490-d61c-443f-a65b-d21fe96a405b
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unlocked when it should have been.
(issue AST-210)
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(closes issue #14885)
Reported by: deepesh
Patches:
20090413__bug14885.diff.txt uploaded by tilghman (license 14)
Tested by: deepesh
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@188835 f38db490-d61c-443f-a65b-d21fe96a405b
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JIRA ABE-1835
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When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
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Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
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The sample configuration file has references to both spellings.
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RFC 5047 explains the proper course of action to take if a
reINVITE is received before the ACK from a previous invite
transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of
the sip_pvt representing this dialog.
(closes issue #13849)
Reported by: klaus3000
Patches:
13849_v2.patch uploaded by mmichelson (license 60)
Tested by: mmichelson, klaus3000
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@187484 f38db490-d61c-443f-a65b-d21fe96a405b
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(Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal")
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requested
Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
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https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
Fix for AST-2009-003
........
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corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.
this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized.
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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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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>
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@185120 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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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dvossel.
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(closes issue #13034)
Reported by: klaus3000
Patches:
20090316__bug13034.diff.txt uploaded by tilghman (license 14)
patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@183319 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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 0011261)
Reported by: vhatz
Patches:
asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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