Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.
Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.
The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction on this issue.
(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When performing a SIP transfer to a Park extension, if the Park fails, chan_sip
will currently not hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service frames,
resulting in stuck channels.
This patch immediately hangs up the two channels if a Park fails.
(closes issue ASTERISK-22834)
Reported by: rsw686
(closes issue ASTERISK-23047)
Reported by: Tommy Thompson
Review: https://reviewboard.asterisk.org/r/3107
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@402604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@402468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If preferred codecs included any non-audio format the code would
mistakenly add the audio format, even if it was not a joint capability
with the remote side.
(closes issue ASTERISK-21131)
Reported by: nbougues
Patches:
patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@401497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a 200 OK for an initial INVITE is received, we were doing
the right thing by ACKing and sending an immediate BYE. However,
we also were doing the wrong thing and queuing an answer frame,
thus causing the call to be answered. This would cause the call
to be hung up by the channel thread, thus resulting in a second
BYE being sent out.
In this fix, I also have set the hangupcause to be correct since
the initial BYE being sent by Asterisk had an unknown hangup
cause. I have changed to using "Bearer capabilty not available"
since the call was hung up due to an SDP offer/answer error.
(closes issue ASTERISK-22621)
reported by Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bumping the SDP version number can cause interoperability problems
since receivers of the responses will expect that a 200 SDP will
be identical to a previous 183 SDP.
(closes issue ASTERISK-21204)
reported by NITESH BANSAL
Patches:
dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a global option in chan_sip to allow it to continue
attempting registration if a 403 is received, clearing the cached nonce
and treating it as a non-fatal response. Normally, this would cause
registration attempts to that endpoint to stop.
(closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874/
Reported by: Rudi
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, Asterisk would incorrectly use the previous endpoint
addresses in SDP in spite of providing its own port. T38 is never meant to
be done through directmedia and Asterisk should always be in the media path
for these streams.
(closes issue ASTERISK-17273)
Reported by: Kevin Stewart
(closes issue ASTERISK-18706)
Reported by: Jeremy Kister
Review: https://reviewboard.asterisk.org/r/2853/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@399456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.
This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.
Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.
(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@397756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.
This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.
Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.
(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
issueA21064_fix.patch uploaded by wdoekes (License 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@397710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping
the old recv address since recv was already set. This has caused a problem when
a proxy is involved since responses to incoming requests from the proxy server,
after an outbound call is established, are never sent to the correct recv
address.
In 11, r382322 introduced this regression.
The fix is to revert that change and always store the recv address on incoming
requests.
Thank you Walter Doekes for helping to point out this error and Mark Michelson
for your input/review of the fix.
(closes issue ASTERISK-22071)
Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer
Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@397204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.
This patch sets the dialog's transport based on the transport that was defined
in the register line. If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@396240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The commit (r387133) for fixing ASTERISK-21466 accidentally removed an extra
line break between the peers returned by the AMI action SIPPeers. This
results in some parsers breaking because they expect this extra line break.
This patch restores that extra line break.
(closes issue ASTERISK-22239)
Reported by: Jacek Konieczny
Tested by: Jacek Konieczny, Michael L. Young
Patches:
asterisk-ami_sippeers_separator.patch by Jacek Konieczny (license 6298)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@396236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription
The problem is that the State Notify requests rely on the 200OK reponse for pacing control
and to not confuse the notify susbsystem.
The issue is, the pendinginvite isn't cleared if a response isn't received,
thus further notify's are never sent.
The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure.
(closes issue ASTERISK-21677)
Reported by: Dan Martens
Tested by: Dan Martens, David Brillert, alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2475/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 4028 Section 10
if the side not performing refreshes does not receive a
session refresh request before the session expiration, it SHOULD send
a BYE to terminate the session, slightly before the session
expiration. The minimum of 32 seconds and one third of the session
interval is RECOMMENDED.
Prior to this asterisk would refresh at 1/2 the Session-Expires interval,
or if the remote device was the refresher, asterisk would timeout at interval end.
Now, when not refresher, timeout as per RFC noted above.
(closes issue ASTERISK-21742)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2488/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 4028 Section 7.2
"UACs MUST be prepared to receive a Session-Expires header field in a
response, even if none were present in the request."
What changed
After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered
a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher.
Symptom:
After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device
may respond with a much lower Session-Expires (180 in our case) value that it is now using.
Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE.
After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the
refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response.
Fix:
handle_response_invite() when 200OK, remove check for outbound and reinvite.
(closes issue ASTERISK-21664)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2463/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When you have lots of SIP peers (according to the issue reporter, around 3500),
the 'sip show peers' CLI command or AMI action can crash due to a poorly placed
string duplication that occurs on the stack. This patch refactors the command
to not allocate the string on the stack, and handles the formatting of a single
peer in a separate function call.
(closes issue ASTERISK-21466)
Reported by: Guillaume Knispel
patches:
fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a BYE request is processed in chan_sip, the current SIP dialog is detached
from its associated Asterisk channel structure. The tech_pvt pointer in the
channel object is set to NULL, and the dialog persists for an RFC mandated
period of time to handle re-transmits.
While this process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no way of knowing
that the channel they've just obtained (which is still valid) and that they are
attempting to lock is about to have its tech_pvt pointer removed. By the time
they obtain the channel lock and call the channel technology callback, the
tech_pvt is NULL.
This patch adds a few checks to some channel callbacks that make sure the
tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
callbacks, which would crash if AMI initiated a DTMF on the channel at the
same time as a BYE was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this function), as well
as sip_indicate (as lots of things can queue an indication onto a channel).
Review: https://reviewboard.asterisk.org/r/2434/
(closes issue ASTERISK-20225)
Reported by: Jeff Hoppe
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
(closes issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2386/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-20904, the focus was around the changes to NAT that took place in
Asterisk 11. Since the report stated that 1.8 was fine, we didn't take a look
at 1.8 at the time.
While working on ASTERISK-21225, I could see that 1.8 would benefit from having
some of those changes applied to it.
This patch does the following:
* The important part of this patch is that it sets the peer's flags earlier in
build_peer so that the code properly uses the peer's flags based on the peer's
configuration.
* constify req parameter in check_via()
* update realtime schemas under the contrib directory to handle properly the NAT
settings available in 1.8 as well as to handle the changes made in 11 to make
upgrading easier when installing newer versions of Asterisk
(closes issue ASTERISK-21243)
Reported by: Michael L. Young
Patches:
asterisk-20904-changes_for_1.8.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2422/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
* A "407 Proxy Authentication Required" response is sent instead of a
"401 Unauthorized" response
* The presence or absence of additional tags occurs at the end of "403
Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403 Forbidden" response
after a retransmission
* Retransmission are sent when a matching peer did not exist, but not when a
matching peer did exist.
This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.
This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.
(closes issue ASTERISK-21013)
Reported by: wdoekes
Tested by: wdoekes, kmoore
patches:
AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r373424, several reentrancy problems in chan_sip were addressed. As a
result, the SIP channel driver is now properly locking the channel driver
private information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by functions called
by register_verify. This includes:
* Holding the private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel container lock.
This is a locking inversion, as any channel related lock must be obtained
prior to obtaining the SIP channel technology private lock.
* Holding the privat elock while calling sip_poke_peer. In the same vein as
sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
the same locking inversion.
Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.
(issue ASTERISK-21068)
Reported by: Nicolas Bouliane
(issue ASTERISK-20550)
Reported by: David Brillert
(issue ASTERISK-21314)
Reported by: Badalian Vyacheslav
(issue ASTERISK-21296)
Reported by: Gabriel Birke
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a session timer expires during a dialog that has re-negotiated to T.38
and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the poor fax
session under weigh.
This patch corrects that by sending T.38 parameters if we are in the middle of
a T.38 session.
(closes issue ASTERISK-21232)
Reported by: Nitesh Bansal
patches:
dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.
(issue ASTERISK-17888)
(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I added my extensive suite of session timer unit tests, apparently one of
them was failing and I never noticed. If neither Min-SE nor Session-Expires is
set in the header, it was responding with a Session-Expires of the global
maxmimum instead of the configured max for the endpoint.
(issue ASTERISK-20787)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, Asterisk only processed session timer information if both the
'Supported: timer' and 'Session-Expires' headers were present. However, the
Session-Expires header is optional. If we were to receive a request with a
Min-SE greater than our configured session-expires, we would respond with a
'Session-Expires' header that was too small.
This patch cleans the situation up a bit, always processing timer information
if the 'Supported: timer' header is present.
(closes issue ASTERISK-20787)
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2299/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC5347 section 2.5.2 states the following:
...
The attribute "T38MaxBitRate" was once incorrectly registered with
IANA as "T38maxBitRate" (lower-case "m"). In accordance with T.38
examples and common implementation practice, the form "T38MaxBitRate"
SHOULD be generated by implementations conforming to this package.
In general, it is RECOMMENDED that implementations of this package
accept lowercase, uppercase, and mixed upper/lowercase encodings of
all the T.38 attributes.
...
Asterisk currently does not perform case insensitive matching on the T.38
attributes. This causes the T38MaxBitRate attribute to be negotiated at
2400 baud instead of 14400 (or whatever value you actually wanted).
This patch makes it so that when we compare T.38 attributes, we do so in a case
insensitive fashion.
Note that while the issue reporter did not directly write the patch, they
contributed to it (and would have provided one themselves if the license had
gone through a tad faster), and hence get attribution for it.
(closes issue ASTERISK-20897)
Reported by: Eric Hill
Tested by: Eric Hill
patches:
-- uploaded by Eric Hill
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@380458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
