lp:~guilhem-fr/asterisk/11-fix

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Owner:
Guilhem Lettron
Project:
Asterisk
Status:
Development

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23500. By Guilhem Lettron

Fix colon in destdir

23499. By elguero

Remove Port Restriction When Checking For NAT

When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.

This patch removes the port from being checked and just useds the addresses
now.

(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    asterisk-remove-using-port-for-nat-check.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2927/

23498. By wdoekes

Properly copy/remove the device state cache flag over a masquerade.

In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.

In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.

(closes issue ASTERISK-22718)
Reported by: Walter Doekes

Review: https://reviewboard.asterisk.org/r/2925/
........

Merged revisions 401178 from http://svn.asterisk.org/svn/asterisk/branches/1.8

23497. By elguero

Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag

A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog. This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings. If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.

This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.

(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
    asterisk-2236-always-set-rport.diff uploaded
                                              by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2919/

23496. By kmoore

Reduce log level of a non-pubsub error message

Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.

(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
    asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
    asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
........

Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8

23495. By wdoekes

Don't check all realtime queues when doing "queue show some_queue".

When using realtime queues, queues have to be fetched from the database
every now and then to see if any info has been changed or to see if the
queue has been removed. When fetching info for an individual queue, the
pruning of other queues is unnecessarily costly.

Review: https://reviewboard.asterisk.org/r/2907/
........

Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8

23494. By rmudgett

chan_iax2: Fix channel left locked in off nominal code path.

23493. By mmichelson

Prevent chan_sip from sending duplicate BYEs.

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
........

Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8

23492. By rmudgett

chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.

* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.

* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.

(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
      jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
........

Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8

23491. By mmichelson

Do not increment the SDP version between 183 and 200 responses.

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)
........

Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8

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