external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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r56505 | russell | 2007-02-23 17:24:18 -0600 (Fri, 23 Feb 2007) | 16 lines
Merged revisions 56504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2 lines
Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
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There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.
IAX2 media frames only carry the *source* call number. So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number. Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.
The way that chan_iax2 did this matching was extremely inefficient. It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it. It
would do this regardless of whether the call number was in use or not. So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations. (Note that the max call number is about 32k).
I have introduced a hash table of active IAX2 calls to improve this lookup
process. The hash is done on the IP address, port number, and call number.
So, for the previous example, a few lock/unlock operations may be done versus
25000 for each frame.
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines
Merged revisions 56406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) | 8 lines
The IMAP storage code uses the same code to build the email that is used when
voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage)
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performance of the GUI. This encodes the configuration into the JSON format
in a manager header, "JSON: ". The encoded information can be directly used
as a javascript object, so no parsing is needed. For large configuration
files, this can greatly improve loading times in the GUI. Furthermore, the
encoding takes up a lot less space when being transmitted than the other
alternatives. (Inspired by discussion with Pari)
Here is an example of what you get:
http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf
Response: Success
JSON: {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]}
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r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines
Merge changes from team/russell/sla_updates.
This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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r56006 | kpfleming | 2007-02-21 18:08:54 -0600 (Wed, 21 Feb 2007) | 2 lines
disable unloading of embedded modules... there is a fundamental problem with doing so that will not be fixed in this version of Asterisk due to its invasiveness
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r55957 | file | 2007-02-21 15:35:40 -0500 (Wed, 21 Feb 2007) | 10 lines
Merged revisions 55956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines
Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message.
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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
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