[asterisk-users] OC3/STM-1 Line Card

[Nick Khamis]

Hello Everyone,

Anyone know of a way of bypassing the 90K audiocodes mediant 3000
equipped for STM-1 interface using line cards and a linux box :).

Kind Regards,

Nick.

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Re: [asterisk-users] dCAP study recommendations

[Gregory Malsack]

I just took the dCap exam Friday. Good things to study is to take the dcaa exam 
online. If you can breeze thought that, you should be in good shape for the 
written. As for the practical exam. You'll want to make sure you know 
endpoints, dahdi, and trunking.

However, I also took the advanced class last week before the exam. I can't 
recommend this course enough! 3000.00 in my opinion if way too low of price! 
The instructors are beyond awesome, they included lunches every day, I received 
a d40, pri card, 4 port analog with 1fxo and 1fxs module, backpack, t shirt, 
awesome dialplan trix that after 10 years of selling, installing, and 
supporting asterisk systems I didn't know because they don't seem to be 
documented were I normally look for documentation... I could go on and on and 
on.

If you have the time to take a week for the class, I highly recommend you do. 
The value will be returned to you in no time! This course will give you more 
confidence and increase your proficiency!

I am more satisfied with this class than I thought possible!

Carlos Chavez  wrote:

>-BEGIN PGP SIGNED MESSAGE-
>Hash: SHA1
>
>The best guide is the Asterisk Fefinitive guide and a virtual machine
>so you can install several Asterisk servers and make them talk to each
>other.
>
>On 6/7/13 1:20 PM, Michael Gilleran wrote:
>> Greetings. Anyone have any recommendations for studying for the
>> dCAP Certification? Other than the expensive Digium courses, there
>> doesn?t seem to be anything online.
>> 
>> 
>> 
>> Thanks,
>> 
>> 
>> 
>> 
>> 
>> *Michael Gilleran  *
>> 
>> 
>> 
>> 
>> 
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>> 
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>
>- -- 
>Telecomunicaciones Abiertas de México S.A. de C.V.
>Carlos Chávez Prats
>Director de Tecnología
>+52-55-91169161 ext 2001
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Re: [asterisk-users] how to send dtmf after pause ?

[covici]

Sean Darcy  wrote:

> On 06/07/2013 01:48 PM, Asghar Mohammad wrote:
> > hi,
> > you can add more w (ww1234#) for more delay.
> >
> >
> >
> > On Fri, Jun 7, 2013 at 7:17 PM, Yves A.  > > wrote:
> >
> > This would be possible with an agi...
> > the agi can wait for silence or 10 seconds, as u like and then play
> > the dtmf tones and bridge the call to your extension afterwards.
> >
> > yves
> >
> > Am 07.06.2013 17:51, schrieb Sean Darcy:
> >
> >
> > I'm trying to call a conference service, wait 10 seconds, then
> > send the passcode.
> >
> > I've tried ww:
> >
> > Dial(SIP/18005551212ww12345#@s__ip.com ,60,r)
> >
> > The sip channel didn't like that. Added 'p' , still no help.
> >
> > I tried D:
> >
> > Dial(SIP/18005551...@sip.com
> > ,__60,rD(12345#)
> >
> > The dtmf is sent too soon. I tried inserting 'ww' but that was
> > just sent.
> >
> > I tried G:
> >
> > exten => 234.1.Dial(SIP/18005551212@__sip.com
> > ,60,rG(next))
> >   same=>n(next),Wait(10)
> >   same=>n,SendDTMF(12345#)
> >
> > but that didn't work at all,
> >
> > This is a common use case. There must be some simple answer I'm
> > missing.
> >
> > Thanks for any help.
> >
> > sean
> >
> >
> >
> 
> Thanks for the reply, but any 'w' s in the dial string cause
> CHAN_UNAVAILABLE.
> 
> I'm not sure I'm up for learning agi just yet. I was hoping for a
> dialplan solution.
> 
> sean

Those W's are only available in some dahdi drivers and they only wait
at the very beginning, if I remember correctly.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] how to send dtmf after pause ?

[Sean Darcy]

On 06/07/2013 01:48 PM, Asghar Mohammad wrote:

hi,
you can add more w (ww1234#) for more delay.



On Fri, Jun 7, 2013 at 7:17 PM, Yves A. mailto:yves...@gmx.de>> wrote:

This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play
the dtmf tones and bridge the call to your extension afterwards.

yves

Am 07.06.2013 17:51, schrieb Sean Darcy:


I'm trying to call a conference service, wait 10 seconds, then
send the passcode.

I've tried ww:

Dial(SIP/18005551212ww12345#@s__ip.com ,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com
,__60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was
just sent.

I tried G:

exten => 234.1.Dial(SIP/18005551212@__sip.com
,60,rG(next))
  same=>n(next),Wait(10)
  same=>n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm
missing.

Thanks for any help.

sean





Thanks for the reply, but any 'w' s in the dial string cause 
CHAN_UNAVAILABLE.


I'm not sure I'm up for learning agi just yet. I was hoping for a 
dialplan solution.


sean


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[asterisk-users] Pulse Audio "Motorboating" Audio with Asterisk

[Jerry Geis]

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
I get a motorboating sound or warble  - or - just not clear audio.

When I switch that to ALSA direct it sounds just fine.

What might be happening with pulse audio that it does not
sound clear???

asound.conf below.

Thanks,

Jerry

more /etc/asound.conf
#
# Place your global alsa-lib configuration here...
#

@hooks [
{
func load
files [
"/etc/alsa/pulse-default.conf"
]
errors false
}
]




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