[asterisk-users] PITCH_SHIFT()

2011-12-19 Thread John Jolly
This list is a great resource and I thank all the Asterisk Guru's who
actively contribute to it.

In Leif Madsen's AstriCon 2010 talk titled 5 Things You Didn't
Know Asterisk Could
Dohttps://docs.google.com/viewer?a=vq=cache:hCDfIk4pvngJ:leifmadsen.com/sites/default/files/AstriCon%25202010%2520-%25205%2520Things%2520You%2520Didn't%2520Know%2520Asterisk%2520Could%2520Do.pdf+asterisk+dialplan+func+PITCH+SHIFThl=engl=uspid=blsrcid=ADGEEShzSRqJl26lEybK-TvxHL4hKQrd-mBpAapRV6eyI8ST0E5AosCEqp2bm_h5eORZFwwEZDqzEKpT9Fg244nkCgX4BDEGL6bik4Non5_fgm62fzrBxyIXjm1hnqJx2-yGyVlbdXKdsig=AHIEtbQ2NyYajUzeJshmWKAgZEi0RprNjQpli=1
he
mentions that the PITCH_SHIFT() function is designed to be used dynamically
and can change the pitch of a channel on the fly using features.conf. Can
someone provide me with any information of how this would be accomplished
for dynamic use? I'm familiar with the dialplan syntax use examples such as:

exten = 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave
exten = 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more
exten = 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch

and so forth, but don't understand how these functions would be called
dynamically from features.conf.

any help is greatly appreciated...
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[asterisk-users] CDR ARI Question

2011-04-16 Thread John Jolly
I have a particular DID that when called will prompt the user to enter the
caller id that they want to be displayed followed by it prompting for the
phone number to dial. How would I go about getting thest calls logged in
both CDR and ARI? Currently, only the callerid information from the original
caller is populated in CDR and ARI. Is there a way to get this call detail
to be logged in the CDR and ARI databases?

My current dialplan looks like this -- the calls do get recorded but there
is no link in ARI to the actual recorded file(s).

exten =
s,6,Set(CALLFILENAME=${calleridnum}--${STRFTIME(${EPOCH},,%d-%m-%Y-%H:$
exten = s,7,MixMonitor(${CALLFILENAME}.wav)
exten = s,8,Dial(SIP/flowroute/${telnum})
exten = s,9,StopMonitor()

Any help/advice is greatly appreciated...

J. Jolly
jgjolly[at]gmail.com
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[asterisk-users] meetme conference playback of random sound file

2011-02-10 Thread John Jolly
i have been trying to find a way to accomplish the following but have not
found a method in which to do so:

i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself. i am able to successfully get it to play a
specific file prior to entering the conference unsure how to implement this
sort of randomization.

Is this possible? Any help will be greatly appreciated.

john jolly jgjolly[at]gmail.com
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