Posted: Thu May 07, 2009 5:07 am Post subject: [asterisk-dev] main/indications.c: Possibility to specify to
Hi!
Current implementation of tones in Asterisk lacks the posssibility to
specify tone levels. Even cheap VoIP gateways (Linksys/SPAxxxx) and VoIP
phones are able to set individual tone levels, but we currently are not.
To fix this, I've prepared a patch. Because I'm not currently running
SVN trunk at any place, the patch is for 1.6.1.0 branch. However, if it
will be found as useful, I'm willing to make a new patch for the SVN
trunk.
The patch adds a new, optional syntax element "@level" to all of the
possible tone formats. The level is entered in dB, so negative numbers
have to be used for most cases. However, the level can be set up to +3 dB,
which still generates an undistorted tone. This is similar to Linksys/Sipura
syntax.
For example, a new definition of SIT with levels of -16 dB is as follows:
info = 950@-16/330,0/30,1400@-16/330,0/30,1800@-16/330,0/1000
You are even able to specify a different levels for individual tone
fragments, so you can easily create a famous "hobbling" British congestion
tone:
congestion = 400@-10/400,0/350,400@-5/225,0/525
Of course the patch doesn't change the current behaviour: If a volume
is specified (non-zero) for ast_playtones_start(), it overrides the
levels in the tone spec and all the tone frags are played with the requested
level. It also accepts the original indications.conf file without any changes,
so if the user doesn't want to use this new feature, he doesn't notify any
change.
Please let me know, whether you are willing to accept the patch and whether
I should prepare a version for SVN trunk.
With regards, Pavel Troller
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Posted: Thu May 07, 2009 10:49 am Post subject: [asterisk-dev] main/indications.c: Possibility to specify to
----- "Pavel Troller" <patrol@sinus.cz> wrote:
Quote:
Current implementation of tones in Asterisk lacks the posssibility
to
specify tone levels. Even cheap VoIP gateways (Linksys/SPAxxxx) and
VoIP
phones are able to set individual tone levels, but we currently are
not.
To fix this, I've prepared a patch. Because I'm not currently
running
SVN trunk at any place, the patch is for 1.6.1.0 branch. However, if
it
will be found as useful, I'm willing to make a new patch for the SVN
trunk.
Thank you for your submission. Unfortunately, the Asterisk developers can't accept patches through the mailing list, they must be submitted through the bug tracker at http://bugs.digium.com/. This allows them to keep track of patches, who submitted them, where the code came from, etc. Also, the 1.6.1 branch of Asterisk is feature-frozen, so you will need to make a patch against SVN trunk so that it can be added to a future release of Asterisk. If you have any other questions, don't hesitate to ask here on the list.
--
Jared Smith
Training Manager
Digium, Inc.
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