This is just a minor code cleanup change. These uses of ao2_callback() would
never return anything since the callbacks always returned 0. However, be more
explicit that no returned results are wanted by specifying OBJ_NODATA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".
* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking". This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.
* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c. This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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When Asterisk detects a hangup and cannot send a BYE due to a pending
INVITE, it sets the pendingbye flag and waits for the final response to that
INVITE. When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first send a CANCEL
request to terminate the pending re-INVITE. In that circumstance, Asterisk
was, in some scenarios, clearing the pendingbye flag after processing the
CANCEL request and not checking for a pending BYE when receiving the final
487 response to the INVITE.
This patch ensures that if the pendingbye flag is set, it is honored
regardless of the nature of the INVITE request currently in flight.
(closes issue ASTERISK-19365)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1807
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Added new chan_dahdi.conf colp_send option parameter to block connected
line updates per span.
(closes issue ASTERISK-17025)
Reported by: Michael Smith
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
* Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
* Other described in CHANGES file
Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.
(closes issue ASTERISK-16890)
Review: https://reviewboard.asterisk.org/r/1243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always
(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper
Review: https://reviewboard.asterisk.org/r/1737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.
* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.
* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.
(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
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The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive. The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.
* Remove ISDN hold restriction for calls connected to applications.
* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388)
Reported by: Birger Harzenetter
Tested by: rmudgett
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Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.
(closes ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
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chan_iax2 to pass in the correct types.
chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk
at this point, so this feels like a safe change to make.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.
(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
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Custom parking extensions may not be coded such that the first and only
extension priority is the Park application. These custom parking
extensions will not be recognized as parking extensions. When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan. Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time. The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.
* Fix handling of BLINDTRANSFER channel variable for call parking.
* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.
(closes issue ASTERISK-19322)
Reported by: aragon
Tested by: rmudgett, jparker
Review: https://reviewboard.asterisk.org/r/1730/
JIRA AST-766
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When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.
We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.
The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
(with some slight modifications prior to commit)
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