Re: [asterisk-users] Using asterisk with DSP chips

2011-11-20 Thread Sazzad 
Is it the wrong place for this question?

On Sat, Nov 19, 2011 at 1:23 AM, Sazzad  wrote:

> Hi,
>
> I have to use asterisk with some dedicated DSP chips, which will do the
> expensive G729 CODEC computing, so that the server processor has minimum
> load. I was informed, I've to use GPAK to implement this. So far I've
> researched, I've found that, Asterisk uses Dahdi to *talk* to external
> devices.
>
> Can you please direct me to references/documentations that clarifies,
>
>1. The over all architecture of asterisk with dedicated DSP chips,
>i.e., how does asterisk integrates with dedicated dsp chips?
>2. Whether embedded scenario is a must for such development or is it
>possible to use asterisk on server and dsp's through some PCI/Ethernet?
>3. What could possibly be the build process in such scenario? Tools
>i.e., cross compilers etc?
>4. How do asterisk know, that when a call arrives it has to use a
>particular DSP chip for Encoding/Decoding?
>
> The GPAK user's manual states,
>
> The physical layer of communication between the host processor and a G.PAK
>> DSP is often dependent upon the hardware design.  This interface can be via
>> a host port interface (HPI), shared memory, a PCI bus, Ethernet or any
>> other fast data pathway.
>
>
> So far I've comprehended from the above quotation, it is possible to use
> asterisk on a server and DSPs on some remote location (say, not built into
> the same board where asterisk is running) . Am I correct?
>
> Thank you very much for patient reading.
>
> --
> Sincerely,
> Sazzad Bin Kamal
>



-- 
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Re: [asterisk-users] Using asterisk with DSP chips

2011-11-20 Thread Doug Lytle 


Sazzad wrote:

Is it the wrong place for this question?


Is it the wrong place for the question?  No. Does anybody that has read 
your message know the answer, probably not.


Doug


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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] Using asterisk with DSP chips

2011-11-20 Thread Sazzad 
I see. thanks anyway.

On Sun, Nov 20, 2011 at 6:20 PM, Doug Lytle  wrote:

>
> Sazzad wrote:
>
>> Is it the wrong place for this question?
>>
>
> Is it the wrong place for the question?  No. Does anybody that has read
> your message know the answer, probably not.
>
> 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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>



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Sazzad Bin Kamal
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Re: [asterisk-users] Using asterisk with DSP chips

2011-11-20 Thread David Backeberg 
On Fri, Nov 18, 2011 at 2:23 PM, Sazzad  wrote:
> Hi,
> I have to use asterisk with some dedicated DSP chips, which will do the
> expensive G729 CODEC computing, so that the server processor has minimum
> load. I was informed, I've to use GPAK to implement this. So far I've

I had never heard of GPAK before your post, so I had to look it up. I
still do not really understand it.

There are at least a few ways to combine asterisk and G.729. And
honestly, I would say your problem is not so much the cpu load, but
the licensing, because G.729 is under patent. You can buy G.729
licenses from Digium, and do G.729 natively in asterisk.

http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC

> Whether embedded scenario is a must for such development or is it possible
> to use asterisk on server and dsp's through some PCI/Ethernet?

It is also possible to have asterisk speak open codecs, like G.711,
and have another ethernet-connected device to the codec, and pay a
different vendor for that licensing.

For example, you can use a Cisco 3945, load it up with DSPs, and setup
the router config to do the transcoding between G.711 and G.729. I've
done this before, and it works, but you have to shell out $10k + for
the router and smartnet and all the DSPS if you are doing as many
simultaneous channels as I'm using.

But I'm still not sure this is your actual question. I realize English
is probably not your first language, however, if you contracted for a
job and you HAVE to use GPAC because somebody else told you that is
the job requirement, I have no idea how to do that.

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[asterisk-users] Deleting AstDB family at start

2011-11-20 Thread Matt Hamilton 

Is it possible to delete the keys belonging to a family in AstDB at Asterisk 
startup? I would like to repopulate it from another source each time Asterisk 
is restarted. 

I know there is a DBdeltree() function. Is there a context that only 
runs once (automatically) at Asterisk startup (so that I can call this 
function)?

Also is AstDB lookup faster than a func_odbc lookup? Is there a faster way to 
perform a lookup in Asterisk; e.g. create a lookup table in memory perhaps?

I'm new to Asterisk...

Thanks,
Matt
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[asterisk-users] check if devices reachable in queue

2011-11-20 Thread Matt Hamilton 


I would like to perform 2 checks on a queue:

1. if the caller stays in the queue for a certain time, I would like to forward 
him to phone A.

2. if the devices/members in the queue are not reachable, I would like to 
forward him to a phone B.

The first one is straight-forward via the timeout.


I'm looking for a fast/practical way of accomplishing the second one. In other 
words, before sending a call to a queue, I would like to see if the 
members/devices in that queue are available/reachable. 


I define the members statically in queue.conf and QUEUE_MEMBER_COUNT gives the 
count of those - doesn't care if  they are available/reachable or not (even if 
phone is unhooked, still counted). 

I should be able to loop through each member and use  ${DEVICE_STATE()}. for every incoming call, isn't this overkill? Any other way?

Thanks a lot,
Matt


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Re: [asterisk-users] Using asterisk with DSP chips

2011-11-20 Thread Eric Wieling 
If you want to use your own DSP transcoder, try asking on asterisk-dev.  If you 
simply want to use a hardware based transcoder Digium and Sangoma have cards.

Sangoma: http://sangoma.com/products/hardware_products/transcoding.html
Digium: http://www.digium.com/en/products/hardware/voice

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg
Sent: Sunday, November 20, 2011 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using asterisk with DSP chips

On Fri, Nov 18, 2011 at 2:23 PM, Sazzad  wrote:
> Hi,
> I have to use asterisk with some dedicated DSP chips, which will do 
> the expensive G729 CODEC computing, so that the server processor has 
> minimum load. I was informed, I've to use GPAK to implement this. So 
> far I've

I had never heard of GPAK before your post, so I had to look it up. I still do 
not really understand it.

There are at least a few ways to combine asterisk and G.729. And honestly, I 
would say your problem is not so much the cpu load, but the licensing, because 
G.729 is under patent. You can buy G.729 licenses from Digium, and do G.729 
natively in asterisk.

http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC

> Whether embedded scenario is a must for such development or is it 
> possible to use asterisk on server and dsp's through some PCI/Ethernet?

It is also possible to have asterisk speak open codecs, like G.711, and have 
another ethernet-connected device to the codec, and pay a different vendor for 
that licensing.

For example, you can use a Cisco 3945, load it up with DSPs, and setup the 
router config to do the transcoding between G.711 and G.729. I've done this 
before, and it works, but you have to shell out $10k + for the router and 
smartnet and all the DSPS if you are doing as many simultaneous channels as I'm 
using.

But I'm still not sure this is your actual question. I realize English is 
probably not your first language, however, if you contracted for a job and you 
HAVE to use GPAC because somebody else told you that is the job requirement, I 
have no idea how to do that.

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Re: [asterisk-users] Using asterisk with DSP chips

2011-11-20 Thread Sazzad 
*@David*
Thanks for your reply. The specification for this job is not clear yet.
Like whether we'll be provided with related DAHDI driver or not. If not, I
have to write one. So I was interested about the 'know how'. Hence I think
my precise question will be:

Is there any open source DAHDI driver for a DSP chip, that Asterisk uses
> for transcoding, so that I can study it ?

*
*
@Eric

Eric, thanks. I think you've got the point. Thanks for those links. I
remember I've compiled Asterisk with all those DAHDI drivers.
Specially, wctc4xxp drivers meant to be used for TC400 series (according to
documentation). Yes, now I think, we are not provided with Sangoma/Digium
cards (which they hopefully have drivers prepared), I've to seek some
wisdom at dev-list for other DSP chips.

-- 
Sincerely,
Sazzad Bin Kamal
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[asterisk-users] Continue AGI after Dial() following caller hang up?

2011-11-20 Thread David Cunningham 
Hello,

We would like to continue a Perl AGI after a Dial() it has done completes
following caller hangup. We would like to do this in the same AGI, and not
using a new AGI from the 'h' extension. It works fine when the called party
hangs up and the 'g' option is used, but not for caller hangup.

Is this possible?

If not a confirmation that this is the case would be very helpful.

Thanks for any advice!

-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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[asterisk-users] How to use password file with Authenticate Application

2011-11-20 Thread virendra bhati 
Hi List,

I want to use text file to get password information with Authenticate
Application. I am using asterisk 1.6.2.11. I made text file at
/tmp/pass.txt with below information.

*pass.txt*

Virendra: 81dc9bdb52d04dc20036dbd8313ed055
Vijay : 9996535e07258a7bbfd8b132435c5962
Virendra Bhati: 7bccfde7714a1ebadf06c5f4cea752c1
*
SIP.conf :-

[2218]*
type=friend
secret=**
callerid="Virendra" <9172341457>
host=dynamic; This device needs to register
nat=yes ; X-Lite is behind a NAT router
canreinvite=no  ; Typically set to NO if behind NAT
disallow=all
dtmfmode=inband
allow=speex
allow=ulaw
allow=alaw
allow=gsm   ; GSM consumes far less bandwidth than ulaw
insecure=invite,port
context=bhati-test
qualify=yes
accountcode=Virendra*

**extensions.conf*

[bhati-test]

exten => 1234,1,Answer()

same => n,NoOp(Welcome to WaveCrest Conferencing Soultion)
same => n,playback(WC)
same => n,Authenticate(/tmp/pass.txt,am,5)
same => n,MeetMe(1234,Mp)
same => n,Hangup()

*CLI output:-

* == Using SIP RTP CoS mark 5
-- Executing [1234@bhati-test:1] Answer("SIP/2218-064f", "") in new
stack
-- Executing [1234@bhati-test:2] NoOp("SIP/2218-064f", "Welcome to
WaveCrest Conferencing Soultion") in new stack
-- Executing [1234@bhati-test:3] Playback("SIP/2218-064f", "WC") in
new stack
--  Playing 'WC.gsm' (language 'en')
-- Executing [1234@bhati-test:4] Authenticate("SIP/2218-064f",
"/tmp/pass.txt,am,5") in new stack
--  Playing 'agent-pass.ulaw' (language 'en')
--  Playing 'auth-incorrect.ulaw' (language 'en')
--  Playing 'auth-incorrect.ulaw' (language 'en')
--  Playing 'vm-goodbye.ulaw' (language 'en')
  == Spawn extension (bhati-test, 1234, 4) exited non-zero on
'SIP/2218-064f'

Please help me where I am wrong
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