(closes issue ASTERISK-18573)
Patches:
sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
Thanks to irrot for the reminder, to Gregory for the patch!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
(Closes issue ASTERISK-18340)
Reported by: Thomas Arimont
(Closes issue ASTERISK-17725)
Reported by: kwk
Tested by: twilson, jrose
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
ended by using \r\n this confuses any interfacing script.
(closes issue ASTERISK-17486)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If Asterisk starts and no DNS is available, outbound registrations will fail
indefinitely. This patch copies the address from the sip_registry struct, which
will be updated, to the peer->addr when necessary.
If dnsmgr is enabled, the registration fails without the patch because even
though the address on the registry is updated via dnsmgr, the address is just
copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update
the address that is copied to the sip_pvt or peers.
Closes issue ASTERISK-18000
Review: https://reviewboard.asterisk.org/r/1335/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
