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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
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(closes issue #10604)
Reported by: keepitcool
Patches:
branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
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(closes issue #12014)
Reported by: junky
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As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549.
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Issue 11130, patch by eliel.
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modifications by me.
Handle DTMF sent by Asterisk properly.
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structure (context)
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Closes issue 10548, reported by keepitcool.
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Googletalk calls are answered too early, which results in CDRs wrongly
stating that a call was ANSWERED when the calling party cancelled a
call before before being established.
We must not answer the call upon reception of a 'transport-accept' iq
packet, but this packet still needs to be acknowledged, otherwise the
remote peer would close the call (like in #8970).
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Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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payloads the other side shouldn't care. (issue #9426 reported by irroot)
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Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).
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(issue #9795, patch from zandbelt)
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don't build in OSP support unless we have found and are allowed to use SSL support
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found it
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did not support multithreading correctly.
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refinement, but this won't have as many folks bothered.
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This should solve issues 8970 and 8503.
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Issue 7764, patch by sailer
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55555!
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while in use will not result in crashes. (issue #8897 reported by junky)
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actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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(issue #8448 reported by phsultan)
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solve a fairly small problem... such is life.
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
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