Re: [asterisk-users] Gizmo SIP / Skype gateway

[Julian Lyndon-Smith]

David Quinton wrote:
> On Sun, 15 Feb 2009 15:01:42 +, Julian Lyndon-Smith
>  wrote:
>
>   
>> Anyone got any thoughts on this and how it compares to the chan_skype 
>> that's due soon ?
>>
>> "OpenSky is a free service provided by Gizmo5 which allows *any* mobile 
>> phone, web browser or IP aware phone network (SIP, asterisk, etc) to 
>> communicate with Skype users. OpenSky supports sending text messages and 
>> voice calls."
>> 
>
> If you read on I think you'll find that it's only free for the first 5
> mins.
>   
Yeah, was aware of that - but $20 per year is not what I would call a 
cost ... We also don't yet know the pricing structure of chan_skype ...

Just curious.

Julian
>
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Re: [asterisk-users] Gizmo SIP / Skype gateway

[Geoff Lane]

On Monday, February 16, 2009, Julian Lyndon-Smith wrote:

> We also don't yet know the pricing structure of chan_skype ...

I thought it was $99 per channel for corporate licenses or $19 for a
single, personal license ... or have I got the wrong ChanSkype?

http://www.chanskype.com follow the "buy" link for the prices.

HTH,

-- 
Geoff


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[asterisk-users] Faxing with asterisk

[Fabio Mosti]

Hi All,

I need to setup asterisk to receive fax.

I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.

Anyone have any idea or solution (Opensource or commercial) to suggest me ?

Best Regards

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Re: [asterisk-users] No such command 'core stop now'

[Jim Boykin]

Can anyone help?

On Sun, Feb 15, 2009 at 2:51 PM, Jim Boykin  wrote:
> It does not work at all even after long time. DNS resolution is not a
> problem, because if I load it from command line "asterisk -c",
> everything works fine.
>
> The problem is when it is configured to be loaded from /etc/inittab
> and the instance of asterisk was killed and init respawned it. After
> respawning, nothing seems to work properly
>
> Jim
>
> On Sun, Feb 15, 2009 at 2:35 PM, Michiel van Baak  
> wrote:
>> On 13:06, Sun 15 Feb 09, Jim Boykin wrote:
>>> This happens mysteriously & randomly. If asterisk was killed and
>>> restarted, it often gives this error
>>>
>>> myast*CLI> core stop now
>>> No such command 'core stop now' (type 'core show help core' for other
>>> possible commands)
>>
>> If you wait a bit, does it work then ?
>> It's possible asterisk is not fully loaded yet (dns resolution being the
>> main thing that can take some time).
>>
>> --
>>
>> Michiel van Baak
>> mich...@vanbaak.eu
>> http://michiel.vanbaak.eu
>> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>>
>> "Why is it drug addicts and computer aficionados are both called users?"
>>
>>
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>

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Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)

[Grygoriy Dobrovolskyy]

2009/2/13 Philipp Kempgen 

> Benny Amorsen schrieb:
>
> > Top posting is annoying. Gmail is broken; maybe I should just killfile
> > @gmail.com.
>
> Emails sent through Gmail's *web interface* are broken.  :-)
>
>
>   Philipp Kempgen
>
> --
> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
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Sorry about off-topic, but can you advise the mail client who is able to
organise the web mailing list topic as web interface does ? (i mean by
blocks/topics) I wold be glad to use something else with the same usability,
but dont see any alternative.
Thank you
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Re: [asterisk-users] Faxing with asterisk

[Grygoriy Dobrovolskyy]

2009/2/16 Fabio Mosti 

> Hi All,
>
> I need to setup asterisk to receive fax.
>
> I'm try Spandsp (opensource) and Attrafax (commercial) both on
> asterisk 1.4.23) but the results are disappointing.
> with spandsp many times the fax arrives cut.
> with Attrafax i have some problem.
>
> Anyone have any idea or solution (Opensource or commercial) to suggest me ?
>
> Best Regards
>
> Try hylafax with IAXmodem. The best results i had it the multitech modems
directly connected to FXS PCI card, you have a nice web interface if you
wish also (avantfax) You can find some nice install scripts at the elastix
forums.
Have a nice day
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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

[Grygoriy Dobrovolskyy]

2009/2/13 Tzafrir Cohen 

> On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
> >
> > I've been involved with getting better data for running Asterisk on
> > the Amazon EC2 cloud computing system.  Here are some calculations
> > I've made on costs based on current published prices on Amazon's
> > system.  Feel free to tell me that I'm wrong with these calculations -
> > but be specific if you find any problems, as I suspect others may glom
> > onto these figures as gospel and I'd hate to have the wrong data in
> > there.
> >
> >http://www.loligo.com/asterisk/misc/amazon-ec2.xls
> >
> > The net of my calculations is that a small instance of 20 users in a
> > standard office environment would cost about $75 per month, which when
> > compared to running a server in-house works out to be (raw cost, not
> > including admin time and not discounting out-of-office bandwidth) only
> > $38.56 more.  Very interesting.
>
> For 20$ or slightly more you can rent a Xen or OpenVZ virtual host which
> will probably do as well.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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And in France it is possible to have a dedicated server with 100 mbit /160
gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real unlimited.
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Re: [asterisk-users] Faxing with asterisk

[Michael]

On Mon, 16 Feb 2009 22:14:49 Fabio Mosti wrote:
> Hi All,
>
> I need to setup asterisk to receive fax.
>
> I'm try Spandsp (opensource) and Attrafax (commercial) both on
> asterisk 1.4.23) but the results are disappointing.
> with spandsp many times the fax arrives cut.
> with Attrafax i have some problem.
>
> Anyone have any idea or solution (Opensource or commercial) to suggest me ?

Unless you have a very stable connection with no contention (like a leased 
line or fibre) to your provider, give up now.

ADSL, cable etc doesn't work properly.

FoIP is a lot worse then VoIP.

Michael.

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Re: [asterisk-users] Faxing with asterisk

[Michael]

> > Anyone have any idea or solution (Opensource or commercial) to suggest me
> > ?
> >
> > Best Regards
> >
> > Try hylafax with IAXmodem. The best results i had it the multitech modems
>
> directly connected to FXS PCI card, you have a nice web interface if you
> wish also (avantfax) You can find some nice install scripts at the elastix
> forums.

Best results are with Hylafax and Multitech serial modems connected directly 
to the PSTN.

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Re: [asterisk-users] Faxing with asterisk

[Grygoriy Dobrovolskyy]

2009/2/16 Michael 

>
> > > Anyone have any idea or solution (Opensource or commercial) to suggest
> me
> > > ?
> > >
> > > Best Regards
> > >
> > > Try hylafax with IAXmodem. The best results i had it the multitech
> modems
> >
> > directly connected to FXS PCI card, you have a nice web interface if you
> > wish also (avantfax) You can find some nice install scripts at the
> elastix
> > forums.
>
> Best results are with Hylafax and Multitech serial modems connected
> directly
> to the PSTN.
>

Well you dont need asterisk then. U think it is nice to have some cdr's for
the incoming faxes.
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Re: [asterisk-users] Faxing with asterisk

[Grygoriy Dobrovolskyy]

2009/2/16 Grygoriy Dobrovolskyy 

>
>
> 2009/2/16 Michael 
>
>>
>> > > Anyone have any idea or solution (Opensource or commercial) to suggest
>> me
>> > > ?
>> > >
>> > > Best Regards
>> > >
>> > > Try hylafax with IAXmodem. The best results i had it the multitech
>> modems
>> >
>> > directly connected to FXS PCI card, you have a nice web interface if you
>> > wish also (avantfax) You can find some nice install scripts at the
>> elastix
>> > forums.
>>
>> Best results are with Hylafax and Multitech serial modems connected
>> directly
>> to the PSTN.
>>
>
> Well you dont need asterisk then. U think it is nice to have some cdr's for
> the incoming faxes.
>
>
misstype "I think"
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Re: [asterisk-users] Faxing with asterisk

[Steve Underwood]

Fabio Mosti wrote:
> Hi All,
>
> I need to setup asterisk to receive fax.
>
> I'm try Spandsp (opensource) and Attrafax (commercial) both on
> asterisk 1.4.23) but the results are disappointing.
> with spandsp many times the fax arrives cut.
> with Attrafax i have some problem.
>
> Anyone have any idea or solution (Opensource or commercial) to suggest me ?
>   
You have two solutions which give trouble, and you assume the problem is 
with them, and not something else in your system. Interesting logic.

You don't indicate the kind of setup you are using. If you are trying to 
FAX over a VoIP connection, its probably that connection making FAXes 
fail. If you are using mISDN, it is probably that making FAXes fail. If 
you are using zaptel to bring in FAXes from the PSTN, spandsp should 
work well.

If you are using the agx addons to interface spandsp to asterisk, they 
have just been updated to work with the latest spandsp. This should be 
more robust in its error recovery than the older versions of spandsp. 
However, with a good connection to the PSTN you shouldn't be hitting 
recovery conditions often enough for that to be too serious.

Regards,
Steve


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Re: [asterisk-users] Faxing with asterisk

[Steve Underwood]

Steve Underwood wrote:
> Fabio Mosti wrote:
>   
>> Hi All,
>>
>> I need to setup asterisk to receive fax.
>>
>> I'm try Spandsp (opensource) and Attrafax (commercial) both on
>> asterisk 1.4.23) but the results are disappointing.
>> with spandsp many times the fax arrives cut.
>> with Attrafax i have some problem.
>>
>> Anyone have any idea or solution (Opensource or commercial) to suggest me ?
>>   
>> 
> You have two solutions which give trouble, and you assume the problem is 
> with them, and not something else in your system. Interesting logic.
>
> You don't indicate the kind of setup you are using. If you are trying to 
> FAX over a VoIP connection, its probably that connection making FAXes 
> fail. If you are using mISDN, it is probably that making FAXes fail. If 
> you are using zaptel to bring in FAXes from the PSTN, spandsp should 
> work well.
>
> If you are using the agx addons to interface spandsp to asterisk, they 
> have just been updated to work with the latest spandsp. This should be 
> more robust in its error recovery than the older versions of spandsp. 
> However, with a good connection to the PSTN you shouldn't be hitting 
> recovery conditions often enough for that to be too serious.
>   
One more thing - check your gains. Modems adapt to varying signal 
levels, and are very insensitive to gain. However, I have received 
numerous wave files for analysis from different people using Asterisk 
where the gain is so crazily high that the signal is in a continuous 
clipping condition. There is no way anything can be reliably decoded 
from that. Voice over these channels must sound dreadful, yet some 
people will swear their system is working fine. :-\

Steve


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[asterisk-users] SpanDSP question for Steve

[Michael]

Hello-

Firstly thanks very much for the work you have put into SpanDSP and the time 
you spend to assist people here :-)

I am currently running SpanDSP 0.0.5 with Call Weaver. Is there any or 
sufficient gain to be had from upgrading SpanDSP?

Michael

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Re: [asterisk-users] Faxing with asterisk

[Fabio Mosti]

2009/2/16 Steve Underwood :

> You don't indicate the kind of setup you are using.

I use asterisk (Spandsp)  with a IAX2 trunk (ethernet connection) to
another asterisk (zap).

client->asterisk (Spandsp)->asterisk (zap)->fax

> Regards,
> Steve

Best Regards,

Fabio

>
>
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Re: [asterisk-users] Faxing with asterisk

[Doug Lytle]

Fabio Mosti wrote:
> I use asterisk (Spandsp)  with a IAX2 trunk (ethernet connection) to
> another asterisk (zap).
>
> client->asterisk (Spandsp)->asterisk (zap)->fax
>   

We have two remote phone systems connected via IAX and 1 fax server at 
the corporate offices with a PRI (ZAP).  The fax server and remote phone 
systems are running HylaFAX+. 

When one of the remotes want to send faxes between facilities, HylaFAX+ 
on the phone system captures the fax locally as a PDF and then send them 
to the remote fax server for delivery.

Works pretty good and no need to fax over IP.

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] OpenSky: Digium Skype gateway?

[Olivier]

2009/2/13 John Todd 

>
> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
>
> > Hi there,
> >
> > is gizmo the first user of the Digium Skype solution, or do they use a
> > different approach/product - any clue?
> >
> > http://www.gizmo5.com/pc/opensky/
> >
> > Philipp
>
>
> I know nothing about their solution that I can say with assurance
> other than it's not the Digium/Skype gateway software.


Any update on this software ?


>
> JT
>
>
> ---
> John Todd   
> email:jt...@digium.com
> Digium, Inc. | Asterisk Open Source Community Director
> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
> direct: +1-256-428-6083 http://www.digium.com/
>
>
>
>
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Re: [asterisk-users] Faxing with asterisk

[Rob Hillis]

Fabio Mosti wrote:
> 2009/2/16 Steve Underwood :
>
>   
>> You don't indicate the kind of setup you are using.
>> 
>
> I use asterisk (Spandsp)  with a IAX2 trunk (ethernet connection) to
> another asterisk (zap).
>
> client->asterisk (Spandsp)->asterisk (zap)->fax

To quote the Mythbusters, "there's your problem".

Fax over IP = forget it unless the connection between your two Asterisk
machines is some form of LAN connection.  This *may* change a little
when the T.38 support in Asterisk includes a gateway mode, which I don't
believe it does yet. (IIRC 1.6 includes much better support for T.38,
but I don't think it includes this kind of gateway yet - anyone care to
correct me?)


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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

[SIP]

Grygoriy Dobrovolskyy wrote:
>
>
> 2009/2/13 Tzafrir Cohen  >
>
> On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
> >
> > I've been involved with getting better data for running Asterisk on
> > the Amazon EC2 cloud computing system.  Here are some calculations
> > I've made on costs based on current published prices on Amazon's
> > system.  Feel free to tell me that I'm wrong with these
> calculations -
> > but be specific if you find any problems, as I suspect others
> may glom
> > onto these figures as gospel and I'd hate to have the wrong data in
> > there.
> >
> >http://www.loligo.com/asterisk/misc/amazon-ec2.xls
> >
> > The net of my calculations is that a small instance of 20 users in a
> > standard office environment would cost about $75 per month,
> which when
> > compared to running a server in-house works out to be (raw cost, not
> > including admin time and not discounting out-of-office
> bandwidth) only
> > $38.56 more.  Very interesting.
>
> For 20$ or slightly more you can rent a Xen or OpenVZ virtual host
> which
> will probably do as well.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> 
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> 
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
>
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>
> And in France it is possible to have a dedicated server with 100 mbit
> /160 gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real
> unlimited.
>
Seriously? Where?  Sign me up!

N.

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Re: [asterisk-users] Faxing with asterisk

[Marc STORCK]

The Attrafax software that was mentioned at the beginning of the thread does 
support Gateway mode.

Regards,

Marc

-Original Message-
Fabio Mosti wrote:
> 2009/2/16 Steve Underwood :
>
>   
>> You don't indicate the kind of setup you are using.
>> 
>
> I use asterisk (Spandsp)  with a IAX2 trunk (ethernet connection) to
> another asterisk (zap).
>
> client->asterisk (Spandsp)->asterisk (zap)->fax

To quote the Mythbusters, "there's your problem".

Fax over IP = forget it unless the connection between your two Asterisk
machines is some form of LAN connection.  This *may* change a little
when the T.38 support in Asterisk includes a gateway mode, which I don't
believe it does yet. (IIRC 1.6 includes much better support for T.38,
but I don't think it includes this kind of gateway yet - anyone care to
correct me?)


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Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)

[Jerry Jones]

>
> Sorry about off-topic, but can you advise the mail client who is  
> able to organise the web mailing list topic as web interface does ?  
> (i mean by blocks/topics) I wold be glad to use something else with  
> the same usability, but dont see any alternative.
> Thank you

Just turn on threading

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Re: [asterisk-users] Michael Graves post

[Michael Graves]

On Mon, 16 Feb 2009 16:05:48 +1300, Matt Riddell wrote:

>On 10/02/2009 5:08 a.m., Michael Graves wrote:
>> I unwittingly started this on Facebook, which I don't user very much.
>> Here's the gist of it.
>> 
>> A Strange Brew: VoIP/Telephony Crossed With Surround Sound
>> 
>> It couldn't be the puritanical kind of approach used in music
>> recording. It would be more a matter of using surround panning to
>> position participants in an synthetic soundfield. I wonder if this has
>> been done to any degree elsewhere?
>> 
>> Stereo is extremely limited in scope. Most of a synthetic stereo image
>> is manipulated using simplistic level based panning, not unlike an old
>> school balance control. It's coarse and two dimensional at best.
>
>Erm - excluding the use of prefade reverb it's actually one dimensional
>- moves left to right - prefade reverb allows you to move backward and
>forward - bringing it to 2 dimensional.
>
>> I'm thinking that UHJ format ambisonic encoding might prove more
>> useful. It allows for accurate, controllable three dimensional
>> positioning while only using the equivalent of a stereo stream.
>
>Surround sound is two dimensional - it just uses the room reverb/delays
>instead of added ones.  I.E. you hear the sound as being in front of you
>rather than to the side of you.

Ah yes. Lets put a little finer point on the terminology.

Stereo is essentially on dimensional.

Surround, as in the common commercial Dolby surround and the like are
two dimensional. These are also sometimes refered to as "planar
surround."

>Three dimensional would require height - which isn't really that useful.

Yes, real three dimensional surround includs a hieght component and is
properly called "periphonic surround."

>The problem is the listening environment - most people don't have
>surround for this - I do like the idea of widening the sound field -
>although it's already doable with phasing and stereo speakers.
>
>I don't think there's even a good stereo conference room.

In all surround encoding schemes backward compatability with the basic
stereo and mono presentation is crucial. The surround scheme should not
damage lesser playback situations.

Effective planar surround is possible from only two channels, allowing
for sources located anywhere in the horizontal plane around the
listener. Ambisonic UHJ format encoding is an example of this. 

OTOH, Dolby et all are not generally good examples. 

Phasing tricks using stereo speakers are good as far as generating an
effect. That is, synthesizing the perception of some image width, but
not accurately reproducing an audible scene. There are many fine
commercial examples of this. I own a old Carver "Sonic Holography"
processor which sounds really nice. Pops a stereo image open
tremendously, although being analog circuitry it's very noisy by
today's standards.

The most common basis of operation of such devices is inter-aural phase
and level differences, often called "head related transfer functions"
(HRTSs)

It's true that the playback circumstances, especially speaker placement
is crucial to surround reproduction. Only 4 speakers are required for
basic surround reproduction, but their placement may be important to
sucessful imaging.

The surround image is likely only going to be optimal in one "sweet
spot" in the listening room. If many people are gathered around a large
table it seems likely that noone will hear the optimal effect.

The question is, will anyone benefit at all? That is, is there any
merit in using existing surround processing techniques to encode
directional cues into conference calling systems?

There's also a question of scale. Does this sort of thing scale down
below full-bore telepresence suites? Or is it just the icing on a $750k
room? Simply a marketing tool?

Michael

--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245




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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

[Grygoriy Dobrovolskyy]

2009/2/16 SIP 

> Grygoriy Dobrovolskyy wrote:
> >
> >
> > 2009/2/13 Tzafrir Cohen  > >
> >
> > On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
> > >
> > > I've been involved with getting better data for running Asterisk on
> > > the Amazon EC2 cloud computing system.  Here are some calculations
> > > I've made on costs based on current published prices on Amazon's
> > > system.  Feel free to tell me that I'm wrong with these
> > calculations -
> > > but be specific if you find any problems, as I suspect others
> > may glom
> > > onto these figures as gospel and I'd hate to have the wrong data in
> > > there.
> > >
> > >http://www.loligo.com/asterisk/misc/amazon-ec2.xls
> > >
> > > The net of my calculations is that a small instance of 20 users in
> a
> > > standard office environment would cost about $75 per month,
> > which when
> > > compared to running a server in-house works out to be (raw cost,
> not
> > > including admin time and not discounting out-of-office
> > bandwidth) only
> > > $38.56 more.  Very interesting.
> >
> > For 20$ or slightly more you can rent a Xen or OpenVZ virtual host
> > which
> > will probably do as well.
> >
> > --
> >   Tzafrir Cohen
> > icq#16849755  
> > jabber:tzafrir.co...@xorcom.com
> > 
> >  >
> > +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> > 
> > http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> > 
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > And in France it is possible to have a dedicated server with 100 mbit
> > /160 gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real
> > unlimited.
> >
> Seriously? Where?  Sign me up!
>
> N.
>
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>

It is not a biz list but i am not owning the company, it is ovh.com click to
"kimsufi"or go directly to http://www.kimsufi.com/ oh it is 19.99 but not a
160 gb but a 250 gb and still unlimited.
have fun.
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[asterisk-users] TeleKaam - Voice Portal for Students and Parents

[Kashif Naeem]

Hello All

We have started a voice portal for Parents and Students. They can listen
Grades, Attendance status and other relevant information over *phone*.
Please read features below and to listen Demo IVR call at *00 92 42 8315427.
*Initially we are deploying it for a School and planning to spread it for
Colleges and Universities. *Your comments and feedback are valuable to us.*

*TeleKaam Features*
**

   - Parent-Teacher meetings can be held by TeleKaam.
   - Parents can get their child result, attendance status in sms.
   - After listening the results parents can talk to respective Teacher.
   - During call hold new courses and facilities offered can be announced to
   parents.

**
*Parents can listen following information over phone:*

a) Children's Results

b) Teacher Comments

c) Attendance status of their child.

*Parents can set different types of alerts. For example:*

a) If my child is absent for more then 2 days then send me sms or auto dial.

b) If teacher has something important to communicate then send me sms or
auto dial.


  Regards,

Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kas...@haditelecom.com
MSN: kashif__na...@hotmail.com
Gmail: meet.kas...@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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[asterisk-users] AstDB wildard searches

[Geoff Lane]

Hi All,

I'm looking for a way to filter the AstDB cidname family to show only
those entries with a specified area code in the Asterisk CLI. If this
were a SQL database it would be something like:
SELECT number, name FROM cidname WHERE number LIKE '1234%'
I've tried "database show cidname 1234*" and substituted "%", "$", "-"
for the wildcard character. I also tried "?" but the Asterisk CLI
wouldn't let me type question marks.

Does AstDB support wildcard characters in this way?

BTW, I know that I can do
 asterisk -rx "database show cidname" | grep 
at the Linux shell prompt, but I'm looking for a way to do this
without leaving the Asterisk CLI.

TIA,

-- 
Geoff


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Re: [asterisk-users] AstDB wildard searches

[Doug Lytle]

Geoff Lane wrote:
>  asterisk -rx "database show cidname" | grep 
> at the Linux shell prompt, but I'm looking for a way to do this
> without leaving the Asterisk CLI.
>   

I don't believe it does.

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] AstDB wildard searches

[Jared Smith]

On Mon, 2009-02-16 at 15:14 +, Geoff Lane wrote:
> I'm looking for a way to filter the AstDB cidname family to show only
> those entries with a specified area code in the Asterisk CLI.

I don't think this is possible with the current AstDB code.

>  If this
> were a SQL database it would be something like:
> SELECT number, name FROM cidname WHERE number LIKE '1234%'
> I've tried "database show cidname 1234*" and substituted "%", "$", "-"
> for the wildcard character. I also tried "?" but the Asterisk CLI
> wouldn't let me type question marks.
> 
> Does AstDB support wildcard characters in this way?

If you have that many items in a database and want to do those types of
filters, why not stick them in a SQL database and use func_odbc to
retrieve them from your SQL database inside the dialplan?


-- 
Jared Smith
Digium, Inc. | Training Manager 




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Re: [asterisk-users] AstDB wildard searches

[Geoff Lane]

On Monday, February 16, 2009, Jared Smith wrote:

> If you have that many items in a database and want to do those types
> of filters, why not stick them in a SQL database and use func_odbc
> to retrieve them from your SQL database inside the dialplan?

Thanks for your suggestion. My Asterisk machine has MySQL, and I might
go down that route at some time. However, my query is part of an
almost academic exercise in which I'm trying to find out what AstDB is
capable of. Unfortunately your excellent book has only enticing hints;
and, although I have shedloads of experience with RDBMSs, this is the
first Berkeley DB I've used.

Thanks again,

-- 
Geoff


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[asterisk-users] command show channels concise

[Jerry Geis]

I am getting a priveldged command error on the manager API.

16-Feb-09 11:51 am asterisk_command() Action: Login
16-Feb-09 11:51 am asterisk_command() Username: XXX
16-Feb-09 11:51 am asterisk_command() Secret: 
16-Feb-09 11:51 am asterisk_command() Events: off
16-Feb-09 11:51 am DEBUG: Response: Success[CR ][LF ]Message: 
Authentication accepted[CR ][LF ][CR ][LF ]
16-Feb-09 11:51 am asterisk_command() Action: Command
16-Feb-09 11:51 am asterisk_command() Command: show channels concise
16-Feb-09 11:51 am DEBUG: Response: Follows[CR ][LF ]Privilege: 
Command[CR ][LF ]

manager.conf has:
[XXX]
secret=
permit=127.0.0.1/255.255.255.0
read = system,call,command,all,agent,user
write = system,call,command,all,agent,user

I thought that was all I needed to run that command?

I am using 1.4.23.

Jerry

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Re: [asterisk-users] AstDB wildard searches

[Jared Smith]

On Mon, 2009-02-16 at 16:30 +, Geoff Lane wrote:
> Thanks for your suggestion. My Asterisk machine has MySQL, and I might
> go down that route at some time. However, my query is part of an
> almost academic exercise in which I'm trying to find out what AstDB is
> capable of. Unfortunately your excellent book has only enticing hints;

Well... I'm glad you liked the book, and yes, it needs more examples of
what AstDB can do.  I'll keep that in mind for the next time we revise
the material.

> and, although I have shedloads of experience with RDBMSs, this is the
> first Berkeley DB I've used.

If you're coming from a traditional relational database mindset, the
AstDB is going to seem very weak.  What I tell people in my training
courses is that the AstDB database is simple way of storing key-value
pairs.  You simply use it to store and organize values, and then
retrieve one value at a time.  Contrast this with relational databases,
where you often run queries that retrieve more than one value at a time,
do joins across tables, order and filter the results, etc.

Hopefully that helps make things a bit more clear.


-- 
Jared Smith
Digium, Inc. | Training Manager 




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Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

[Eric Chamberlain]

On Feb 13, 2009, at 9:59 AM, John Todd wrote:

>
> I've been involved with getting better data for running Asterisk on
> the Amazon EC2 cloud computing system.  Here are some calculations
> I've made on costs based on current published prices on Amazon's
> system.  Feel free to tell me that I'm wrong with these calculations -
> but be specific if you find any problems, as I suspect others may glom
> onto these figures as gospel and I'd hate to have the wrong data in
> there.
>
>   http://www.loligo.com/asterisk/misc/amazon-ec2.xls
>
> The net of my calculations is that a small instance of 20 users in a
> standard office environment would cost about $75 per month, which when
> compared to running a server in-house works out to be (raw cost, not
> including admin time and not discounting out-of-office bandwidth) only
> $38.56 more.  Very interesting.
>

The big advantage I like, is the ability to have identical production  
and development environments, without having to continuously run the  
development environment.

When writing up how to install DAHDI on an Asterisk EC2 instance, I  
went through several instances, I could bring up an instance in  
minutes, use it for 10 or 15 minutes and then throw it away.

I could do something similar with VMware ESX attached to a SAN, but it  
is much more capital intensive, even with leasing.

For a single box, EC2 probably isn't going to be cheaper.  But if you  
have a dynamic environment; need web, asterisk, and database servers;  
scaleable storage; and off-site backup, EC2 starts getting more cost  
effective.

Storing Asterisk realtime data in Amazon's SimpleDB and voicemail in  
S3, would make for a very interesting and scalable solution.

--
Eric Chamberlain, Founder
RF.com - http://RF.com/








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Re: [asterisk-users] AstDB wildard searches

[Geoff Lane]

On Monday, February 16, 2009, Jared Smith wrote:

> Hopefully that helps make things a bit more clear.

It does - many thanks for your help.

-- 
Geoff


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Re: [asterisk-users] command show channels concise

[Mark Michelson]

Jerry Geis wrote:
> I am getting a priveldged command error on the manager API.
> 
> 16-Feb-09 11:51 am asterisk_command() Action: Login
> 16-Feb-09 11:51 am asterisk_command() Username: XXX
> 16-Feb-09 11:51 am asterisk_command() Secret: 
> 16-Feb-09 11:51 am asterisk_command() Events: off
> 16-Feb-09 11:51 am DEBUG: Response: Success[CR ][LF ]Message: 
> Authentication accepted[CR ][LF ][CR ][LF ]
> 16-Feb-09 11:51 am asterisk_command() Action: Command
> 16-Feb-09 11:51 am asterisk_command() Command: show channels concise
> 16-Feb-09 11:51 am DEBUG: Response: Follows[CR ][LF ]Privilege: 
> Command[CR ][LF ]
> 
> manager.conf has:
> [XXX]
> secret=
> permit=127.0.0.1/255.255.255.0
> read = system,call,command,all,agent,user
> write = system,call,command,all,agent,user
> 
> I thought that was all I needed to run that command?
> 
> I am using 1.4.23.
> 
> Jerry

That's not an error message. That is the response given to a Command action 
assuming that a command was provided and the command is not blacklisted.

Mark Michelson

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[asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

I need your help: please help test the gender detection module at 575-613-4392.

I wrote a gender detection module and thought I'd try it out. It only takes a 
second. I've been showing 90%+ accuracy and I want
to make sure it's working correctly. Rain and significant background noise 
seems to throw it off, so I still have a bit of work to do.

Have your friends and significant others call too. Also, let me know if you 
have any need for the module.

Justin Newman
nt_jnewman at yahoo.com


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[asterisk-users] DTMF not completely muted

[Michael Smith]

Hi all,

When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
at the end of the recording.

I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards:
a TE420 w/Octasic and pri_net signalling, and a TE220 w/o Octasic using em_wink
signalling. Calls come in using either PRI or E&M depending on Mitel routing
(you don't want to know...).

Either way, the driver is doing software DTMF detection. (Looking through the
DAHDI driver, hardware DTMF detection is always disabled by default.)

The calls are then sent to another Asterisk instance by SIP with RFC2833 for
DTMF. I dumped the RTP going back and forth between the Asterisks -- the RFC2833
DTMF is being sent properly, but the blips are in the audio stream.

Does anyone have suggestions for preventing the blips, or cleaning them up
afterward? I was thinking I could run a script over all voicemail greetings to
find the first DTMF tone and crop the wav file, but it's not so easy -- the DTMF
blips are way too short to be picked up by spandsp's DTMF detector.

I have some SIP phones and they of course don't have any problems recording
voicemail greetings without blips -- they all use RFC2833 and no real tones are
in the audio stream.

Mike


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[asterisk-users] vmsecret question

[Klaus Darilion]

Hi!

I wonder what is teh meaning if vmsecret is not defined.

Does this mean that the voicebox can be accessed without PIN code? Or 
does it mean that the voicebox can not be accessed at all (of course 
except using the s Parameter of VoicemailMain()) ?

thanks
klaus

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Re: [asterisk-users] DTMF not completely muted

[Kevin P. Fleming]

Michael Smith wrote:

> When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
> and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
> voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
> at the end of the recording.

The blip at the beginning is expected; the DTMF detector won't trigger
until it has seen a long enough continuous tone to recognize it, and
then it will begin muting the audio. The only way around this would be
to artificially delay the audio stream by 20-40ms to accommodate the
muting, but most people would not enjoy the additional latency this
would create.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

This module detects gender and approximate age range. I'm working on getting 
it's accuracy to 80%+ on a consistent basis, after implementing filters to 
remove background noise and other artifacts.

It's designed for a number of things. To start, I have several clients 
(primarily mobile content and servers providers) that want to profile and 
generate demographics of their users for selling advertising. They also want to 
understand their user base. Plus, some customers have found that male and 
female users tend to respond differently to different prompts, flows, etc. This 
helps in designing a system that meets needs of many different types of users.

Of course, there are many other uses and I'm sure people can generate some cool 
ideas.

Let me know how it works when you try the test number at 575-613-4392. Also, 
let me know if you have any interest in the module.

Justin
nt_jnewman at yahoo.com





From: Ron Joffe 
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk 
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

That's an interesting module.

Care to elaborate on what you designed it for ?

Thanks,

Ron



On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> I need your help: please help test the gender detection module at
> 575-613-4392.
>
> I wrote a gender detection module and thought I'd try it out. It only takes
> a second. I've been showing 90%+ accuracy and I want to make sure it's
> working correctly. Rain and significant background noise seems to throw it
> off, so I still have a bit of work to do.
>
> Have your friends and significant others call too. Also, let me know if you
> have any need for the module.
>
> Justin Newman
> nt_jnewman at yahoo.com



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Re: [asterisk-users] DTMF not completely muted

[Michael Smith]

Kevin P. Fleming  digium.com> writes:

> Michael Smith wrote:
> 
> > When the Dahdi driver detects DTMF, it seems it's not muting the first
> > 5-15 ms and sometimes the last 2-10 ms of the DTMF tone.
> 
> The blip at the beginning is expected; the DTMF detector won't trigger
> until it has seen a long enough continuous tone to recognize it, and
> then it will begin muting the audio. The only way around this would be
> to artificially delay the audio stream by 20-40ms to accommodate the
> muting, but most people would not enjoy the additional latency this
> would create.

This makes sense, so I'm looking for a way to post-process the voicemail
greetings because it's kind of ugly to leave the blips in there.

Although it is odd that it doesn't cut the recording as soon as the detector
triggers. If it did, the wav file would end after the first blip, and there
would never be a second blip. It seems like it waits for button-up to stop the
recording.

Mike


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Re: [asterisk-users] Michael Graves post

[Matt Riddell]

On 17/02/2009 3:05 a.m., Michael Graves wrote:
> Phasing tricks using stereo speakers are good as far as generating an
> effect. That is, synthesizing the perception of some image width, but
> not accurately reproducing an audible scene. There are many fine
> commercial examples of this. I own a old Carver "Sonic Holography"
> processor which sounds really nice. Pops a stereo image open
> tremendously, although being analog circuitry it's very noisy by
> today's standards.

Yeah, awesome till you get a stereo->mono mix and then phase
inconsistencies make it mute itself (i.e. left phase inverted from right
then summed).

> The surround image is likely only going to be optimal in one "sweet
> spot" in the listening room. If many people are gathered around a large
> table it seems likely that noone will hear the optimal effect.

Yeah, although they would hear some effect if the speakers were far
enough away.

> The question is, will anyone benefit at all? That is, is there any
> merit in using existing surround processing techniques to encode
> directional cues into conference calling systems?

I just don't know anyone with hardware capable of reproducing it.  Don't
get me wrong, I think it would be awesome, but it needs to be supported
at each endpoint (or as you stated - fall back to a working methodology).

> There's also a question of scale. Does this sort of thing scale down
> below full-bore telepresence suites? Or is it just the icing on a $750k
> room? Simply a marketing tool?

Heh, if you're paying $750k for a room, you should definitely have this
- as well as a full hd telepresence suite.  It would definitely allow
multiple speakers to be more easily distinguished from each other while
talking at the same time.

-- 
Kind Regards,

Matt Riddell
Director
___

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Re: [asterisk-users] [OT] Gmail is broken

[Matt Riddell]

On 16/02/2009 10:23 p.m., Grygoriy Dobrovolskyy wrote:
> Sorry about off-topic, but can you advise the mail client who is able to
> organise the web mailing list topic as web interface does ? (i mean by
> blocks/topics) I wold be glad to use something else with the same usability,
> but dont see any alternative.
> Thank you

Running Thunderbird (shredder) nightlies here without any problems -
some awesome new features (unread folders) but not really on topic for here.

-- 
Kind Regards,

Matt Riddell
Director
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[asterisk-users] How to beep before transfer ...

[Robert Augustyn]

Hi,
When I transfer a call to an extension, the person I call does not have any 
idea when that transfer happened so it is a guessing game.
Is there a way to send a beep to the caller just before transferring the call? 
Preferably by setting something in FreePbx?
 
Sincerely, 
robert 




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Re: [asterisk-users] WiFi SIP phone w/VPN?

[Jay Milk]

Ken D'Ambrosio wrote:
> Hi, all.  My subject line says it all: is there a WiFi SIP phone with VPN
> abilities?  Failing that, a WiFi phone that runs Linux?  I already know
> one phone that does meet my requirements -- the iPhone.  The new software
> comes with a Cisco VPN client, and a SIP client can be had from
> third-party vendors for jailbroken phones.  And, while I'm not averse to
> the idea,
> a) it ain't cheap, and
> b) it's a bit hack.
>
> I've googled my heart out, but haven't found anything else that (I'm sure)
> meets all three requirements.
>
> Thanks!
>
> -Ken
My Nokia E71 has a decent sip-client and vpn capabilities.  I use it as 
an extension all day long (and also as a GSM phone and pda), but I 
haven't used the vpn and sip together, yet.  As an added bonus, it's an 
S60/3 with a well-documented API and an active development community.  
It's still pricey -- ~$300, but there are less expensive wifi/sip 
equipped Nokias out there.

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Re: [asterisk-users] DTMF not completely muted

[Wilton Helm]

The DADHI function is probably intended for more generalized use.  Maybe for 
recording voicemail greetings it should not be used and a different function 
used instead.  There is no reason why it isn't possible to backup in the 
recorded message and erase the blip.  The detection time should approximate a 
known constant, and that much could be removed.  I've certainly seen it done in 
other settings.  That avoids the need for extra delay while still allows 
completely removing the tone.

Wilton
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Re: [asterisk-users] OpenSky: Digium Skype gateway?

[Alejandro Lengua]

What about receiving Skype calls on Gizmo or other SIP device?
Looking into the website I don't see anything regarding that.


On Mon, Feb 16, 2009 at 8:05 AM, Olivier  wrote:
>
>
> 2009/2/13 John Todd 
>>
>> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
>>
>> > Hi there,
>> >
>> > is gizmo the first user of the Digium Skype solution, or do they use a
>> > different approach/product - any clue?
>> >
>> > http://www.gizmo5.com/pc/opensky/
>> >
>> > Philipp
>>
>>
>> I know nothing about their solution that I can say with assurance
>> other than it's not the Digium/Skype gateway software.
>
> Any update on this software ?
>
>>
>>
>> JT
>>
>>
>> ---
>> John Todd   email:jt...@digium.com
>> Digium, Inc. | Asterisk Open Source Community Director
>> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
>> direct: +1-256-428-6083 http://www.digium.com/
>>
>>
>>
>>
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Re: [asterisk-users] DTMF not completely muted

[Michael Smith]

Wilton Helm  compuserve.com> writes:

> The DADHI function is probably intended for more 
> generalized use.  Maybe for recording voicemail greetings it should not be 
> used and a different function used instead.  There is no reason why it 
> isn't possible to backup in the recorded message and erase the blip.  The 
> detection time should approximate a known constant, and that much could be 
> removed.

Yes, that might be the way to go. I'm playing around with a modified
__ast_play_and_record() that stops recording when the button is pressed, not
released. I also have it hacking off the last 150ms of the recording if '#' is
pressed. I'm not sure 150ms is enough, actually -- I think I want to cut off the
noise of the button push / fumbling, and at least with my Linksys ATA there's
sometimes a separate blip up to 300ms before tone detection.

Mike


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[asterisk-users] mp3player() to shuffle playlist

[reza adinata]

Hi all,

I am trying to make a scenario when someone dial *10*, the mp3player()
function would act and play a list of MP3 files.

However, I have no idea how to randomize the function (mpg123 is
capable of shuffling the MP3 files, buat how to implement it in
extensions.conf?)

Perhaps any of you sucessfully made a playlist and shuffling method
for mp3player() in extensions.conf and would like to share it with me.


thanks all for your comments

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[asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)

[Alfred Monticello]

I'm thinking of starting a partyline, where people call in and talk to other 
people. For record keeping and billing purposes, I'd like to go by the callers 
telephone number. 

This method works fine as long as the caller doesn't have callerid blocked, but 
what are my options if they do block their number? I know there must be a way 
to report it, because there is a service provider here in my area that if I 
call and block my number, they are still able to obtain it. I know that when 
dialing a toll-free number, that the number is reported regardless. But what 
about regular non-toll free numbers?

Does anybody have any ideas how I can do this? Are there any providers out 
there that offer this service over PRI or some other method?

Thank you in advance.


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Re: [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)

[Paul Hales]

The old classic is to say something like ' your callerid is blocked,
please get out your credit card'

PaulH


Alfred Monticello wrote:
> I'm thinking of starting a partyline, where people call in and talk to
> other people. For record keeping and billing purposes, I'd like to go
> by the callers telephone number.
>
> This method works fine as long as the caller doesn't have callerid
> blocked, but what are my options if they do block their number? I know
> there must be a way to report it, because there is a service provider
> here in my area that if I call and block my number, they are still
> able to obtain it. I know that when dialing a toll-free number, that
> the number is reported regardless. But what about regular non-toll
> free numbers?
>
> Does anybody have any ideas how I can do this? Are there any providers
> out there that offer this service over PRI or some other method?
>
> Thank you in advance.
>
>
>
> 
>
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[asterisk-users] Optimizing this script for calling Skype users from Asterisk

[Michael Robertson]

I have written this configuration script which uses OpenSky to make Skype
calls directly from Asterisk devices using my companies SIP to Skype
gateway. Users can dial skype_anyskypeusername or manually add names or
extensions which can get mapped to the correct dialing sequence. The right
sequence is usern...@opensky.gizmo5.com but that gets mapped to sipphone
address so I set that up to map directly to the final address.

I need a couple test sites so if anyone wants to test Skype calling on their
Asterisk network please send me email and I'l enable longer calling. Also in
the 2 hard coded examples below  (563 and echo) i want to also reference
gizmo5 and not repeatedly have proxy01.sipphone. Can someone tell me how to
construct the tightest syntax for that? Thanks.

-- MR



[gizmo5]
type=peer ;COPY THIS CONFIG
host=198.65.166.131 ;INTO YOUR sip.conf
fromdomain=proxy01.sipphone.com;THIS WILL
canreinvite=no ;ALLOW ANY
nat=yes ;DEVICE OR
CLIENT
dtmfmode=rfc2833   ;CONNECTED TO YOUR
insecure=very  ;ASTERISK SERVER TO
CALL
qualify=yes;SKYPE USERS
SEVERAL WAYS.
fromuser=YOURSIP;BY DIALING SKYPE NAMES
OR NUMERIC SHORTCUTS
authuser=YOURSIP   ;ENTERED INDIVIDUALLY
BELOW
username=YOURSIP ;OR BY DIALING
skype_skypeusername
secret=YOURPASS;OR THE 333 ALIASES
disallow=all;ENTERED at
my.gizmo5.com
allow=ulaw;
allow=alaw;SEE
gizmo5.com/opensky
allow=ilbc ;FOR MORE INFO

[general]
exten => _1333.,1,Goto(opensky,,1)  ;COPY THIS CONFIG
exten => _333.,1,Goto(opensky,,1);INTO YOUR
exten => _skype[_].,1,Goto(opensky,,1) ;extensions.confexten =>
563,1,Dial(SIP/skype_echo...@proxy01.sipphone.com)
  ;To dial a Skype user by dialing 563 in this example echo123
exten => 
echo,1,Dial(SIP/skype_echo...@proxy01.sipphone.com)
;To dial a Skype name in this example echo will dial echo123

[opensky]
exten => _1.,1,NoOp('opensky dial')
exten => _1.,2,Dial(SIP/${ext...@gizmo5|120|j)
exten => _1.,3,Hangup()

-
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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Gondar Monn]

Looks like my provider is not passing dtmf correctly .. Had a serious
laugh, system kept asking me if I was ready., ended up finding myself
talking to the IVR .



On Mon, Feb 16, 2009 at 11:45 AM, Asterisk Asterisk
wrote:

> This module detects gender and approximate age range. I'm working on
> getting it's accuracy to 80%+ on a consistent basis, after implementing
> filters to remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients
> (primarily mobile content and servers providers) that want to profile and
> generate demographics of their users for selling advertising. They also want
> to understand their user base. Plus, some customers have found that male and
> female users tend to respond differently to different prompts, flows, etc.
> This helps in designing a system that meets needs of many different types of
> users.
>
> Of course, there are many other uses and I'm sure people can generate some
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392.
> Also, let me know if you have any interest in the module.
>
> Justin
> nt_jnewman at yahoo.com
>
> --
> *From:* Ron Joffe 
> *To:* asterisk-users@lists.digium.com
> *Cc:* Asterisk Asterisk 
> *Sent:* Monday, February 16, 2009 11:05:24 AM
> *Subject:* Re: [asterisk-users] Please help test the gender detection
> module at 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> > I need your help: please help test the gender detection module at
> > 575-613-4392.
> >
> > I wrote a gender detection module and thought I'd try it out. It only
> takes
> > a second. I've been showing 90%+ accuracy and I want to make sure it's
> > working correctly. Rain and significant background noise seems to throw
> it
> > off, so I still have a bit of work to do.
> >
> > Have your friends and significant others call too. Also, let me know if
> you
> > have any need for the module.
> >
> > Justin Newman
> > nt_jnewman at yahoo.com
>
>
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[asterisk-users] TeleKaam - Voice Portal for Students and Parents

[Kashif Naeem]

 Hello All

We have started a voice portal for Parents and Students. They can listen
Grades, Attendance status and other relevant information over *phone*.
Please read features below and to listen Demo IVR call at *00 92 42 8315427.
*Initially we are deploying it for a School and planning to spread it for
Colleges and Universities. *Your comments and feedback are valuable to us.*

*TeleKaam Features*
**

   - Parent-Teacher meetings can be held by TeleKaam.
   - Parents can get their child result, attendance status in sms.
   - After listening the results parents can talk to respective Teacher.
   - During call hold new courses and facilities offered can be announced to
   parents.

**
*Parents can listen following information over phone:*

a) Children's Results

b) Teacher Comments

c) Attendance status of their child.

*Parents can set different types of alerts. For example:*

a) If my child is absent for more then 2 days then send me sms or auto dial.

b) If teacher has something important to communicate then send me sms or
auto dial.


  Regards,

Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kas...@haditelecom.com
MSN: kashif__na...@hotmail.com
Gmail: meet.kas...@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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Re: [asterisk-users] Credit Card processing machines

[Andrew Joakimsen]

On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere  wrote:
>
> Anyone have much luck with these on ATA's?  I have a few sites that use
> them succesfully with multi-port Audiocodes boxes, but just connected ten
> machines to Linksys 2102s and they are very flaky.  Using u-law on a 100Mb
> switched network that is barely utilized, then out a T1 on a Sangoma card.
>
> Perhaps there is some tuning on the Linksys or the credit card machine
> itself?  Going to look into reducing the baud rate on the machines, but
> sadly the bank has them password protected and wants to charge a
> "reprogramming fee" :(

They make credit card terminals with Ethernet -- use that instead.

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[asterisk-users] Stress Testing IVR

[Rajkumar S]

Hi,

How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be "programmed" to send out digits after specified
time to simulate users pressing menu items. If it can originate large
number of calls simultaneously then it's great!

Does any one have any recommendations ? Any other method to stress
test an IVR call flow?

with regards,

raj

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[asterisk-users] unistim channel problem

[Ralf Träskman]

Hi


[Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type 
registered for 'USTM'
[Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to 
create channel of type 'USTM' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)

I get this after I restart my asterisk 1.6, it all worked yesterday.
I have the unistim module loaded.
Could it be that I have set keepaliave in unistim.conf to 500, I had to do that 
outerwise my phones would show server unreachable after approx 2 minutes.

What can I do?

/ralf


Ralf Träskman, IT
AdLibris AB, Sveavägen 56C, 111 34 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com 
www.adlibris.com
P Please consider the environment before printing this e-mail

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