This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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which event was received. What actually was happening was that it was reporting the number of bytes returned
from a call to read().
Thanks to Jared Smith for bringing the issue up on IRC
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in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
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could lead to a looped list, meaning it will traverse forever.
(closes issue #11818)
Reported by: michael-fig
Patches:
11818.patch uploaded by putnopvut (license 60)
Tested by: michael-fig
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it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
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or hostname was provided, which is the one part of the dial string that is
absolutely required. If it's not there, bail out.
(closes issue #11897)
Reported by sokhapkin
Patch by me
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Reported by: MVF
Patches:
bug11736-2.diff uploaded by oej (license 306)
Tested by: oej, MVF, revolution
(russellb: This was the showstopper for the release.)
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internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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