Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Hi, anyone know?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 17 May 2010 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adding a context from the console

 

Hi, is it possible to add a context from the console using the dialplan
command?

Thanks

Lee

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Re: [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.

[Motiejus Jakštys]

I doubt codec will change anything, but you can try.

In audacity, Effect -> Change Tempo...

2010/5/19 DHAVAL INDRODIYA :
> hi Motiejus,
>
> Can you give a command for converting it to normal voice , in audacity.
>
> also i tired with more users still problem persists ,
>
> can i try with gsm format , what you say?
>
> regards
> Dhaval
>
> 2010/5/18 Motiejus Jakštys 
>>
>> Hi,
>> The record is not double faster, it's 50% faster (100 seconds original
>> record -> 66.6 seconds recording). Reducing tempo by 33% without
>> losing pitch sort of "fixes" the situation, although adds alot garbage
>> to sound file (you can do this in Audacity).
>> Sample rate 8kHz is OK, changing it to 5280 fixes the tempo, but
>> reduces pitch to unacceptable.
>>
>> Try with more callers in a conference, does it change anything
>> (increased/decreased tempo)?
>>
>> You could also try ConfBridge:
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge
>> or other conference backends (Conference, Konference...)
>> These could solve the problem if Dahdi is broken.
>>
>> On Tue, May 18, 2010 at 11:46 AM, DHAVAL INDRODIYA
>>  wrote:
>> > Hi Motiejus,
>> >
>> > sorry for inconvenience , because asterisk mailing list could not accept
>> > wav
>> > file attachment
>> >
>> > here i am attached a file named test.wav,
>> >
>> > regards
>> > Dhaval
>>
>>
>>
>> 2010/5/18 Motiejus Jakštys :
>> > Please check WAV headers, what is the sample rate of the file? It
>> > should be 8kHz. Does the WAV sound normal when you decrease sample
>> > rate by hand?
>> >
>> > You can just upload one WAV for testing - I'll say what may be wrong
>> > with it.
>> >
>> > On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA
>> >  wrote:
>> >> hello All,
>> >>
>> >> i have one issue with Asterisk Meetme Application
>> >>
>> >> i am recording through Meetme channels through option 'r' and format
>> >> for
>> >> recording a file is 'wav'
>> >>
>> >> lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5.
>> >>
>> >> i have very strange problem of meetme_recording ,
>> >>
>> >> once conference starts recording file having a   recording is 2x faster
>> >> than
>> >> normal recording .
>> >>
>> >> is there any setting to solve it out , my card type is TE410P used E1
>> >> lines
>> >> .
>> >>
>> >> please help me . any help appreciated.
>> >>
>> >> regards
>> >> Dhaval
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> _
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>> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> >>               http://www.asterisk.org/hello
>> >>
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>> >> To UNSUBSCRIBE or update options visit:
>> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>>
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Re: [asterisk-users] Play MusicOnHold and continue with dialplan

[Asterisk]

Cheers. That is exactly what I need (I wonder how come I didn't find out that 
app) :-)

Thanks! Alex

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, May 18, 2010 11:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan

Here's one way
Exten => s,1,noop(dial with moh)
Exten => s,n,dial(tech/1,10,m)
Exten => s,n,WaitExten(10,m)
Exten => s,n,dial(tech/2,10,m)
Exten => s,n,WaitExten(10,m)

The waitexten(10,m) plays musiconhold waiting for a 1 digit extension.  As
long as there's not one in the context, you're good.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Sent: Tuesday, May 18, 2010 4:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan

Thanks, but in my particular case I need to do pause between dials (using
Wait() command). How could I implement MoH also when Wait is in progress (in
single extensions that is)? Is this even possible, or is the only way to
encapsulate the logic in one extension and do Dial(Local/lo...@extension,,m)
in another one?

Regards,
Alex

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, May 18, 2010 4:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan

The simplest way to do this is this:
Exten => s,1,noop(dial with moh)
Exten => s,n,dial(tech/1,10,m)
Exten => s,n,dial(tech/2,10,m)


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Sent: Tuesday, May 18, 2010 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Play MusicOnHold and continue with dialplan

Hi guys,

Is it possible to start playing MusicOnHold to the caller but also continue
with the dialplan in single extension, something like this:

exten => s,1,StartPlayingMoh()
exten => s,n,Wait(10)
exten => s,n,Dial(someone...)
exten => s,n,Wait(10)
exten => s,n,Dial(someone else...)
...

Regards,
Alex


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Re: [asterisk-users] About option U in Dial Ast version 1.6.2

[Vardan]

Hello
as I understand, nobody not used this option?


-- 
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Vardan wrote:
> Has any one used this?
>U(x[^arg[^...]]):
>   x - Name of the subroutine to execute via Gosub
>   arg - Arguments for the Gosub routine
>   Execute via Gosub the routine  for the *called* channel before
>   connecting to the calling channel. Arguments can be specified to
> the Gosub
>   using '^' as a delimiter. The Gosub routine can set the variable ${GO
>   SUB_RESULT} to specify the following actions after the Gosub returns.
>   ${GOSUB_RESULT}:
>   ABORT: Hangup both legs of the call.
>   CONGESTION: Behave as if line congestion was
>   encountered.
>   BUSY: Behave as if a busy signal was encountered.
>   CONTINUE: Hangup the called party and allow the
>   calling party to continue dialplan execution at the next
> priority.
>   GOTO:^^: Transfer the call
>   to the specified priority. Optionally, an extension, or
> extension
>   and priority can be specified.
>   NOTE: You cannot use any additional action post answer options in
>   conjunction with this option. Also, pbx services are not run on the
> peer
>   (called) channel, so you will not be able to set timeouts via the
> TIMEOUT()
>   function in this routine.
>
>
> Thanks


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[asterisk-users] Re-invite from Asterisk Server: Port number changes

[Vinod Parameswaran]

Hello list,

I am trying to test a scenario wherein two clients configured on two diffrent 
boxes try to communicate with each other by means of Asterisk. The softphone on 
both the boxes is zoiper. One of the boxes is Unix, and has the server running 
on it. The other is Windows.

When I make a call between clients (Unix -> Windows), the signaling works fine, 
but I cannot listen to audio on the Windows box. The audio device on the 
Windows box has been tested to be working fine.

Upon analyzing the wireshark SIP logs, I can see that Asterisk sends a 
re-invite in which the RTP port number in SDP is different from that configured 
on zoiper.

I suspect that this is probably the reason for the audio non-availability on 
the Windows box. I have enabled allowdirectmedia as part of my SIP settings.

I have posted the logs at clipnet:

http://cl1p.net/sip.conf (SIP.conf used for this test)
http://cl1p.net/Call_logs_Unix_lo (Wireshark logs captured on the local 
interface on the Unix box)
http://cl1p.net/Call_logs_Unix_eth0 (Wireshark logs captured on the eth0 
interface on the Unix box)
http://cl1p.net/Call_logs_Windows (Wireshark logs captured on the Windows box)

I would appreciate your thoughts.

Thanks
Vin


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Re: [asterisk-users] a2billing DID and Queues

[Tarek Sawah]

the simple way i can see it is the following;let's say you have  did starts 
with 1708
[from-did]exten = _1708XXX,1,Answerexten 
= _1708XXX,n,Queue(SALES,,)exten = h,1,Hangup



--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308









> Date: Tue, 18 May 2010 20:47:12 -0700
> From: toqee...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] a2billing DID and Queues
>
> Hi all,
>
> I have configured asterisk and a2billing.for inbound i have also configured 
> did and its forwarded to sip extensions.
>
> But i want to enable queues with inbound numbers(DID).But i could not find a 
> way to do this in a2billing.
>
>
>
> I want enable that if some did comes to asterisk/a2billing it should be 
> forwarded to queues not sip extensions and
>
> their i want to enable hunting so if one extensions does not receive the call 
> so it should be forwarded to the next
>
>
> extensions.
>
> So please help, Any help will highly appreciated.
>
> Thanks
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
  
_
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Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

[Olivier]

2010/5/19 Gordon Henderson

>

> On Wed, 19 May 2010, Olivier wrote:
>
> > 2010/5/18 Danny Nicholas 
> >
> >>  Dumb question ? wouldn?t it be easier to monitor a web interface than a
> >> phone with 100 lights?
> >>
> > Yes and no : operator already has a Flash Operator Panel on its screen.
> > Information displayed by FOP is richer (you can see who is talking to
> who)
> > but operator feels easier with dedicated buttons for both displaying
> > activity and issuing transfers.
>
> I've deployed a few Grandstream phones with a single button-box. The box
> has 56 additional keys to the 7 on the GXP2000. It can support a 2nd box
> connected in to give 119 BLF buttons/speed-dials.
>
> BLF was flakey in early Grandstreams, but seems to work fine for me in
> these systems - but I know Grandstream gets a bad rap by many these days,
> but it's not too expensive for a trial.
>
> Gordon
>

It's fine to know about Grandstream.

I'm very curious to gather "field proven" experiences as the reason why some
vendors (such as aastra) limit BLF number (too much time to register each
BLF during boot ? ...) while they offer a larger number of buttons, might be
ones that make a trial fail.

Indeed, several IP phones support 100 and more buttons but how many of them
support BLF features successfully ?

If Snom user could also testify, that would be very interesting to know.


>
>
>
>  >
> > I think 100 is the upper limit for both kinds of tools where at a glance,
> > you can see all extensions : I think above a certain user count (120 ?),
> > operator would prefer to specifically query its console to get current
> > specific extensions phone activity.
> >
> >>
> >>  --
> >>
> >> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jose Flores
> >> Galicia
> >> *Sent:* Tuesday, May 18, 2010 2:32 PM
> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> >> *Subject:* Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
> >>
> >>
> >>
> >> Hi
> >>
> >>
> >>
> >> Indeed, limited to only 50 BLF, thats why operator was placed two aastra
> >> phones and set a ring group for both.
> >>
> >>
> >>
> >> Best Regards
> >> Jose Flores Galicia
> >> <>
> >> BriefCode && Code Based Training
> >>
> >>  2010/5/18 Olivier 
> >>
> >>
> >>
> >> On Tue, May 18, 2010 at 11:58 AM, Olivier  wrote:
> >>
> >>  Hi,
> >>
> >> Can you share successful experience with a SIP hardphone supporting 100
> BLF
> >> ?
> >> Which phone would you suggest for that ?
> >>
> >> (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect
> the
> >> state (Idle, Ringing, OnCall) of a local extension.
> >>
> >> Regards
> >>
> >>
> >>   --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>   http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >>
> >> --
> >> Thank you  with regards,
> >> Gopalakrishnan A.N,
> >>
> >>
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>   http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >> --
> >> _
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> >>   http://www.asterisk.org/hello
> >>
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> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
>
> --
> _
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Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

[John Novack]

Olivier wrote:
>
>
> 2010/5/18 Danny Nicholas mailto:da...@debsinc.com>>
>
> Dumb question -- wouldn't it be easier to monitor a web interface
> than a phone with 100 lights?
>
> Yes and no : operator already has a Flash Operator Panel on its screen.
> Information displayed by FOP is richer (you can see who is talking to 
> who) but operator feels easier with dedicated buttons for both 
> displaying activity and issuing transfers.
>
Of course s/he does. A lesson those of us who have been in the telephone 
industry for a quarter century learned a long time ago.
Users want and like buttons and lights, and systems that don't support 
this in an easy to use manner will not succeed.
DSS/BLF consoles are a MUST for any busy operator, who is expected to 
not only answer and switch calls but do many other tasks during the day
A commonly available feature in many/most systems, and not limited to 50 
or even 100 stations.

John Novack

> I think 100 is the upper limit for both kinds of tools where at a 
> glance, you can see all extensions : I think above a certain user 
> count (120 ?), operator would prefer to specifically query its console 
> to get current specific extensions phone activity.
>
> 
>
> *From:* asterisk-users-boun...@lists.digium.com
> 
> [mailto:asterisk-users-boun...@lists.digium.com
> ] *On Behalf Of
> *Jose Flores Galicia
> *Sent:* Tuesday, May 18, 2010 2:32 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] OT - Which SIP hardphone with 100
> BLF ?
>
> Hi
>
> Indeed, limited to only 50 BLF, thats why operator was placed two
> aastra phones and set a ring group for both.
>
> Best Regards
> Jose Flores Galicia
> >>
> BriefCode && Code Based Training
>
> 2010/5/18 Olivier mailto:oza_4...@yahoo.fr>>
>
> On Tue, May 18, 2010 at 11:58 AM, Olivier  > wrote:
>
> Hi,
>
> Can you share successful experience with a SIP hardphone
> supporting 100 BLF ?
> Which phone would you suggest for that ?
>
> (In case that matters, each BLF is supposed to SUBSCRIBE
> to and reflect the state (Idle, Ringing, OnCall) of a
> local extension.
>
> Regards
>
>
> --
> 
> _
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar
> every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Thank you with regards,
> Gopalakrishnan A.N,
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every
> Thurs:
> http://www.asterisk.org/hello
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> asterisk-users mailing list
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>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
> 
>
>
>
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> 14:26:00
>
>   

-- 
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Re: [asterisk-users] Adding a context from the console

[Tilghman Lesher]

On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
> Hi, is it possible to add a context from the console using the dialplan
> command?

Yes, just add an extension to it.  The context will be created as needed.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Many thanks.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
> Hi, is it possible to add a context from the console using the
dialplan
> command?

Yes, just add an extension to it.  The context will be created as
needed.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] DTMF Input from the User

[taimur hasan]

Hello

I am new to Asterisk. I want to know is there any way to get DTMF input from 
the user in the Dialplan. 

Regards
Taimur Hasan
-THQ-  !!!ONE

  
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Re: [asterisk-users] DTMF Input from the User

[Jim Dickenson]

Use read application
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 19, 2010, at 9:42 AM, taimur hasan wrote:

> 
> Hello
> 
> I am new to Asterisk. I want to know is there any way to get DTMF input from 
> the user in the Dialplan. 
> 
> Regards
> Taimur Hasan
> -THQ-  !!!ONE
> 
> 
> 
> Hotmail: Powerful Free email with security by Microsoft. Get it now. -- 
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> 
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Re: [asterisk-users] DTMF Input from the User

[Danny Nicholas]

Read is best for multiple digit DTMF input.  For Single-Digit DTMF, you can
use WaitExten.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Wednesday, May 19, 2010 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF Input from the User

 

Use read application

-- 

Jim Dickenson

mailto:dicken...@cfmc.com

 

CfMC

http://www.cfmc.com/

 

 

 

On May 19, 2010, at 9:42 AM, taimur hasan wrote:






Hello

I am new to Asterisk. I want to know is there any way to get DTMF input from
the user in the Dialplan. 

Regards
Taimur Hasan

-THQ-  !!!ONE





  _  

Hotmail: Powerful Free email with security by Microsoft. Get it now.
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Re: [asterisk-users] About option U in Dial Ast version 1.6.2

[Philipp von Klitzing]

Hi!

> as I understand, nobody not used this option?

Would you like everyone on this list to give you an answer? ;->

If you have a _real_ question about the U option of Dial(), then post it, 
and someone here might or might not have an answer.

Philipp


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Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

[Philipp von Klitzing]

Hi!

> If Snom user could also testify, that would be very interesting to
> know. 

While you can do it, I would not recommend to have two sidecars with a 
snom 370. Especially the boot-up can become very slow if there are lot of 
SUBSCRIBEs to be issued, even if you tweak some of the subscription 
settings of the phone. Looks like the device's CPU cannot handle that 
many subscriptions too well (and I strongly suspect most other vendors 
have the same issue).

About the snom 8xx phones: No experience there, they have a stronger CPU 
and the corresponding sidecar product "snom Vision" is very new (it has 
fewer buttons).

Maybe all this will change when Asterisk supports list subscription in 
some distant future. But until then you will have to do with fewer 
buttons - or use software.

Philipp


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Re: [asterisk-users] DTMF Input from the User

[taimur hasan]

Thanks a lot...

Regards
Taimur Hasan

-THQ-  !!!ONE



From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 May 2010 12:14:15 -0500
Subject: Re: [asterisk-users] DTMF Input from the User




















Read is best for multiple digit DTMF
input.  For Single-Digit DTMF, you can use WaitExten.

 









From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson

Sent: Wednesday, May 19, 2010
12:00 PM

To: Asterisk Users Mailing List -
Non-Commercial Discussion

Subject: Re: [asterisk-users] DTMF
Input from the User



 

Use read application







-- 





Jim Dickenson





mailto:dicken...@cfmc.com





 





CfMC





http://www.cfmc.com/





 





 




 





On May 19, 2010, at 9:42 AM, taimur hasan wrote:













Hello



I am new to Asterisk. I want to know is there any way to get DTMF input from
the user in the Dialplan. 



Regards

Taimur Hasan



-THQ-  !!!ONE















Hotmail: Powerful Free email with security by Microsoft. Get it now. -- 

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Re: [asterisk-users] About option U in Dial Ast version 1.6.2

[Vardan Harutyunyan]

OK I will try to explain my problems, sorry for my English.

So, I want to indicate in dial plan, that the dialed channel is connectted.

For example(extensions.conf):

exten = s,1,Dial(ToSomeOne)
exten = s,n,Hangup

In this example, you can know about that the channel is connected after 
if hangup.
So, if you do this:

exten = s,1,Dial(ToSomeOne,,M(macros))
exten = s,n,Hangup


[macro-macros]
exten = s,1,Noop(Call conected)


After Noop in macros, the dial plan is do nothing.

This is a work.

If I want make this using AEL (Asterisk 1.6.2), the macro is not working.
While how we know, macro in AEL is changed to gosub.
I can continue use the macros, that I have make in extensions.conf, but 
I dont want.
So, in Dial application in asterisk 1.6.2 is now new option U.
It the same, as M, but used gosub meanings.
But he dont work same like M.

Can any body help me to understand how this work.

Thanks


-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

Philipp von Klitzing wrote:
> Hi!
>
>> as I understand, nobody not used this option?
>
> Would you like everyone on this list to give you an answer? ;->
>
> If you have a _real_ question about the U option of Dial(), then post it,
> and someone here might or might not have an answer.
>
> Philipp
>
>


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[asterisk-users] Asterisk and RFC 3261

[Tarek Sawah]

Greetings List,Trying to interconnect with a new provider.. the require 
a compliance with RFC 3261  so knowing less than needed about RFC 
documentations.. i would like to know if Asterisk is actually in compliance 
with RFC 3261 or not.. Can any one help with this?
Regards
--
Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
USA: +1 347 562 2308




  
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[asterisk-users] Cause and cure for "Exceptionally long voice queue length queuing to Local"?

[David Cunningham]

Hello,

We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?

One explanation I've come across is that the peer is congested and
sending RTCP messages asking us to slow the RTP down. Is there any way
we can verify this?

[May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice
queue length queuing to Local/12126412...@asterisk-phone-7e3d;1

Thanks in advance!


-- 
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http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180

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Re: [asterisk-users] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?

[Danny Nicholas]

Sip debug peer?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, May 19, 2010 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cause and cure for "Exceptionally long voice
queuelength queuing to Local"?

Hello,

We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?

One explanation I've come across is that the peer is congested and
sending RTCP messages asking us to slow the RTP down. Is there any way
we can verify this?

[May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice
queue length queuing to Local/12126412...@asterisk-phone-7e3d;1

Thanks in advance!


-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180

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Re: [asterisk-users] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?

[David Cunningham]

What should I expect see if it is the peer asking us to slow down RTP?

Thanks again.


On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas  wrote:
> Sip debug peer?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Cunningham
> Sent: Wednesday, May 19, 2010 3:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Cause and cure for "Exceptionally long voice
> queuelength queuing to Local"?
>
> Hello,
>
> We're seeing lots of warnings like the following, running Asterisk
> 1.6.1.12. Does anyone know the cause or cure?
>
> One explanation I've come across is that the peer is congested and
> sending RTCP messages asking us to slow the RTP down. Is there any way
> we can verify this?
>
> [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice
> queue length queuing to Local/12126412...@asterisk-phone-7e3d;1
>
> Thanks in advance!
>
>
> --
> David Cunningham, Voisonics
> http://voisonics.com/
> US toll-free: +1 888 842 2720
> UK: +44 (0) 20 3298 1642
> Australia: +61 (0) 2 9037 2180
>
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>
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-- 
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http://voisonics.com/
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UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180

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Re: [asterisk-users] Asterisk and RFC 3261

[bruce bruce]

That is the RFC number for SIP. Yes, Asterisk is compliant with RFC. I am
not sure to what degree but I haven't ever faced non-compliance on SIP RFC 3261
ever with any provider.

-Bruce

On Wed, May 19, 2010 at 2:28 PM, Tarek Sawah  wrote:

>
>
> Greetings List,Trying to interconnect with a new provider.. the require
> a compliance with RFC 3261  so knowing less than needed about RFC
> documentations.. i would like to know if Asterisk is actually in compliance
> with RFC 3261 or not.. Can any one help with this?
> Regards
> --
> Tarek Sawah
> Integrated Digital Systems
> CCNA, MCSE, RHCE, VoIP
> USA: +1 347 562 2308
>
>
>
>
>
> _
> The New Busy is not the old busy. Search, chat and e-mail from your inbox.
>
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> --
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>
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[asterisk-users] Sip phone does not call

[ayodele abejide]

Hello group,

 

I have asterisk running on my ubuntu machine, and I have a
peer to peer network with an XP machine, both of the running x-lite client, I 
try
calling either of the soft phone from the other and the response I get is on my
asterisk console is as below: 

 

 

[May 19 19:31:18] NOTICE[1476]: chan_sip.c:20006
handle_request_invite: Call from '1000' to extension '3000' rejected because
extension not found.

 

[May 19 19:31:39] NOTICE[1476]: chan_sip.c:21298
handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000

 

[May 19 19:31:44] NOTICE[1476]: chan_sip.c:20006
handle_request_invite: Call from '1000' to extension '1000' rejected because
extension not found.

 

 

My Diaplan Settings (extensions.conf)

 

[globals]



 

 

[general]



autofallthrough=yes



 

 

[default]



exten => s,1,Verbose(1|Unrouted call handler)



exten => s,n,Answer()



exten => s,n,Wait(1)



exten => s,n,Playback(tt-weasels)



exten => s,n,Hangup()



 

 

[incoming_calls]



 

 

[internal]



exten => 1000,1,Verbose(1|Extension 1000)



exten => 1000,n,Dial(SIP/1000,30)



exten => 1000,n,Hangup()



 

 

exten => 3000,1,Verbose(1|Extension 3000)



exten => 3000,n,Dial(SIP/1000,30)



exten => 3000,n,Hangup()

 

 

Sip Settings (sip.conf)

 

[general]

context=default

bindport=5060

srvlookup=yes

 

[1000]

type=friend

host=dynamic

context=phones

 

[3000]

type=friend

host=dynamic

context=phones

  
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Re: [asterisk-users] Asterisk Cluster

[David Backeberg]

On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime  wrote:
> Hello  Everyone,
>                         I  must deploy an asterisk system that can support
> at least 500 pstn outbound calls.
> It's a challenge as  it's the first time i'm gonna build such a large
> system.
> I want to have your advice on hardware, software and so on . What i have in
> my plan is a cluster of servers with quad PRI cards.
> I will appreciate your advice.

I don't know what you've done for your smaller deployments. Different
people have different opinions.

My personal preference, if this is United States:

DS3 -> Adtran channel bank, break out PRIs
PRIs -> multiple Cisco PRI->SIP appliances. A 3845 maxed out can do 24
PRIs. I recommend using two and splitting load for capacity /
failover.
Using pure SIP between the gateway appliances and at least two
asterisk boxes for flexibility and failover-ability.

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Re: [asterisk-users] Sip phone does not call

[Jim Dickenson]

The two phones belong to context phones and the two extensions are in context 
internal. In context phones you need to include => internal so that context 
phones knows about those extensions. Or put the two extensions in context 
phones and not context internal.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 19, 2010, at 2:05 PM, ayodele abejide wrote:

> Hello group,
>  
> I have asterisk running on my ubuntu machine, and I have a peer to peer 
> network with an XP machine, both of the running x-lite client, I try calling 
> either of the soft phone from the other and the response I get is on my 
> asterisk console is as below:
>  
>  
> [May 19 19:31:18] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call 
> from '1000' to extension '3000' rejected because extension not found.
>  
> [May 19 19:31:39] NOTICE[1476]: chan_sip.c:21298 handle_request_subscribe: 
> Received SIP subscribe for peer without mailbox: 1000
>  
> [May 19 19:31:44] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call 
> from '1000' to extension '1000' rejected because extension not found.
>  
>  
> My Diaplan Settings (extensions.conf)
>  
> [globals]
>  
>  
> [general]
> autofallthrough=yes
>  
>  
> [default]
> exten => s,1,Verbose(1|Unrouted call handler)
> exten => s,n,Answer()
> exten => s,n,Wait(1)
> exten => s,n,Playback(tt-weasels)
> exten => s,n,Hangup()
>  
>  
> [incoming_calls]
>  
>  
> [internal]
> exten => 1000,1,Verbose(1|Extension 1000)
> exten => 1000,n,Dial(SIP/1000,30)
> exten => 1000,n,Hangup()
>  
>  
> exten => 3000,1,Verbose(1|Extension 3000)
> exten => 3000,n,Dial(SIP/1000,30)
> exten => 3000,n,Hangup()
>  
>  
> Sip Settings (sip.conf)
>  
> [general]
> context=default
> bindport=5060
> srvlookup=yes
>  
> [1000]
> type=friend
> host=dynamic
> context=phones
>  
> [3000]
> type=friend
> host=dynamic
> context=phones
> 
> Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. 
> -- 
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Re: [asterisk-users] Asterisk Cluster

[Adolphe Cher-aime]

Thank you David. I was thing about the cisco solution but cost is the  
issue as I will so many DSP to for this amount of calls.

Regards



Adolphe Cher-aime
 From my Iphone

On May 19, 2010, at 4:23 PM, David Backeberg   
wrote:

> On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime  > wrote:
>> Hello  Everyone,
>> I  must deploy an asterisk system that can  
>> support
>> at least 500 pstn outbound calls.
>> It's a challenge as  it's the first time i'm gonna build such a large
>> system.
>> I want to have your advice on hardware, software and so on . What i  
>> have in
>> my plan is a cluster of servers with quad PRI cards.
>> I will appreciate your advice.
>
> I don't know what you've done for your smaller deployments. Different
> people have different opinions.
>
> My personal preference, if this is United States:
>
> DS3 -> Adtran channel bank, break out PRIs
> PRIs -> multiple Cisco PRI->SIP appliances. A 3845 maxed out can do 24
> PRIs. I recommend using two and splitting load for capacity /
> failover.
> Using pure SIP between the gateway appliances and at least two
> asterisk boxes for flexibility and failover-ability.
>
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Re: [asterisk-users] Asterisk Cluster

[Jeff Brower]

Adolphe-

> Thank you David. I was thing about the cisco solution
> but cost is the issue as I will so many DSP to for this
> amount of calls.

If you're not doing G729 or other LBR codec (or encryption, or echo can with 
long tail length, or other high level
requirement for RTP processing) then you should not need a DSP based solution; 
server based processing should be
sufficient.

-Jeff

> Adolphe Cher-aime
>  From my Iphone
>
> On May 19, 2010, at 4:23 PM, David Backeberg 
> wrote:
>
>> On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime > > wrote:
>>> Hello  Everyone,
>>> I  must deploy an asterisk system that can
>>> support
>>> at least 500 pstn outbound calls.
>>> It's a challenge as  it's the first time i'm gonna build such a large
>>> system.
>>> I want to have your advice on hardware, software and so on . What i
>>> have in
>>> my plan is a cluster of servers with quad PRI cards.
>>> I will appreciate your advice.
>>
>> I don't know what you've done for your smaller deployments. Different
>> people have different opinions.
>>
>> My personal preference, if this is United States:
>>
>> DS3 -> Adtran channel bank, break out PRIs
>> PRIs -> multiple Cisco PRI->SIP appliances. A 3845 maxed out can do 24
>> PRIs. I recommend using two and splitting load for capacity /
>> failover.
>> Using pure SIP between the gateway appliances and at least two
>> asterisk boxes for flexibility and failover-ability.


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Re: [asterisk-users] Asterisk Cluster

[Jonathan Thurman]

On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime  wrote:
> Hello  Everyone,
>                         I  must deploy an asterisk system that can support
> at least 500 pstn outbound calls.
> It's a challenge as  it's the first time i'm gonna build such a large
> system.
> I want to have your advice on hardware, software and so on . What i have in
> my plan is a cluster of servers with quad PRI cards.
> I will appreciate your advice.

Your up front cost is going to be a little higher with TDM -> SIP
devices, but your management will be a lot easier.

AudioCodes also has equipment that can support a DS3 connection, or
multiple T1s directly to SIP.

For example, If you are getting 22 T1s then get two AudioCodes Mediant
2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
to 84 T1s or three DS3 or one OC3.

Then leave call management up to Asterisk.  Of course, have redundancy
everywhere you can.

-Jonathan

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Re: [asterisk-users] Asterisk Cluster

[Jeff Brower]

Jonathan-

> On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime  
> wrote:
>> Hello  Everyone,
>>                         I  must deploy an asterisk system that can support
>> at least 500 pstn outbound calls.
>> It's a challenge as  it's the first time i'm gonna build such a large
>> system.
>> I want to have your advice on hardware, software and so on . What i have in
>> my plan is a cluster of servers with quad PRI cards.
>> I will appreciate your advice.
>
> Your up front cost is going to be a little higher with TDM -> SIP
> devices, but your management will be a lot easier.
>
> AudioCodes also has equipment that can support a DS3 connection, or
> multiple T1s directly to SIP.
>
> For example, If you are getting 22 T1s then get two AudioCodes Mediant
> 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
> If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
> to 84 T1s or three DS3 or one OC3.
>
> Then leave call management up to Asterisk.  Of course, have redundancy
> everywhere you can.

In this scenario, does Asterisk still touch every RTP packet?  Or can 'native 
bridge' mode be enabled?

-Jeff


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Re: [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?

[Anthony Messina]

On Tuesday 11 May 2010 01:25:30 pm Tim Nelson wrote:
> I have a handful of Asterisk 1.4.x installations where users dial 'outbound
> calls' to the PSTN even though the destination is on the same Asterisk box
> or on another Asterisk box on the same network. Instead of paying twice
> for the call to go out to the PSTN on one channel and back in on another
> channel, I'd like the ability to lookup the destination number in a MySQL
> database and if found, change the way the call is routed. The call routing
> update could be as simple as issuing a Goto() to change contexts or
> priorities in the current context.

you could use DUNDi for this and avoid external DB and/or AGI.  -a

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Asterisk Cluster

[Michelle Dupuis]

The High Availability HASTerisk (HAAST) product on www.generationd.com is a 
software solution that does automatic failover, etc between multiple asterisk 
machines.  I'm guessing this could be part of an overall solution for you


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Brower 
[jbro...@signalogic.com]
Sent: Wednesday, May 19, 2010 8:02 PM
To: Jonathan Thurman
Cc: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk Cluster

Jonathan-

> On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime  
> wrote:
>> Hello  Everyone,
>> I  must deploy an asterisk system that can support
>> at least 500 pstn outbound calls.
>> It's a challenge as  it's the first time i'm gonna build such a large
>> system.
>> I want to have your advice on hardware, software and so on . What i have in
>> my plan is a cluster of servers with quad PRI cards.
>> I will appreciate your advice.
>
> Your up front cost is going to be a little higher with TDM -> SIP
> devices, but your management will be a lot easier.
>
> AudioCodes also has equipment that can support a DS3 connection, or
> multiple T1s directly to SIP.
>
> For example, If you are getting 22 T1s then get two AudioCodes Mediant
> 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
> If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
> to 84 T1s or three DS3 or one OC3.
>
> Then leave call management up to Asterisk.  Of course, have redundancy
> everywhere you can.

In this scenario, does Asterisk still touch every RTP packet?  Or can 'native 
bridge' mode be enabled?

-Jeff


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Re: [asterisk-users] Asterisk Cluster

[Adolphe Cher-aime]

Thank you Jonathan.
I really appreciate

Adolphe Cher-aime
 From my Iphone

On May 19, 2010, at 6:26 PM, Jonathan Thurman  
 wrote:

> On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime  > wrote:
>> Hello  Everyone,
>> I  must deploy an asterisk system that can  
>> support
>> at least 500 pstn outbound calls.
>> It's a challenge as  it's the first time i'm gonna build such a large
>> system.
>> I want to have your advice on hardware, software and so on . What i  
>> have in
>> my plan is a cluster of servers with quad PRI cards.
>> I will appreciate your advice.
>
> Your up front cost is going to be a little higher with TDM -> SIP
> devices, but your management will be a lot easier.
>
> AudioCodes also has equipment that can support a DS3 connection, or
> multiple T1s directly to SIP.
>
> For example, If you are getting 22 T1s then get two AudioCodes Mediant
> 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
> If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
> to 84 T1s or three DS3 or one OC3.
>
> Then leave call management up to Asterisk.  Of course, have redundancy
> everywhere you can.
>
> -Jonathan
>
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Asterisk Cluster

[Adolphe Cher-aime]

Jonathan for redundancy which software do you recomand?

Adolphe Cher-aime
 From my Iphone

On May 19, 2010, at 6:26 PM, Jonathan Thurman  
 wrote:

> On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime  > wrote:
>> Hello  Everyone,
>> I  must deploy an asterisk system that can  
>> support
>> at least 500 pstn outbound calls.
>> It's a challenge as  it's the first time i'm gonna build such a large
>> system.
>> I want to have your advice on hardware, software and so on . What i  
>> have in
>> my plan is a cluster of servers with quad PRI cards.
>> I will appreciate your advice.
>
> Your up front cost is going to be a little higher with TDM -> SIP
> devices, but your management will be a lot easier.
>
> AudioCodes also has equipment that can support a DS3 connection, or
> multiple T1s directly to SIP.
>
> For example, If you are getting 22 T1s then get two AudioCodes Mediant
> 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
> If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
> to 84 T1s or three DS3 or one OC3.
>
> Then leave call management up to Asterisk.  Of course, have redundancy
> everywhere you can.
>
> -Jonathan
>
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Asterisk Cluster

[Adolphe Cher-aime]

Thanks Michelle

Adolphe Cher-aime
 From my Iphone

On May 19, 2010, at 8:02 PM, Michelle Dupuis  wrote:

> The High Availability HASTerisk (HAAST) product on  
> www.generationd.com is a software solution that does automatic  
> failover, etc between multiple asterisk machines.  I'm guessing this  
> could be part of an overall solution for you
>
> 
> From: asterisk-users-boun...@lists.digium.com [asterisk-users- 
> boun...@lists.digium.com] On Behalf Of Jeff Brower  
> [jbro...@signalogic.com]
> Sent: Wednesday, May 19, 2010 8:02 PM
> To: Jonathan Thurman
> Cc: Asterisk Users List
> Subject: Re: [asterisk-users] Asterisk Cluster
>
> Jonathan-
>
>> On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime > > wrote:
>>> Hello  Everyone,
>>>I  must deploy an asterisk system that can  
>>> support
>>> at least 500 pstn outbound calls.
>>> It's a challenge as  it's the first time i'm gonna build such a  
>>> large
>>> system.
>>> I want to have your advice on hardware, software and so on . What  
>>> i have in
>>> my plan is a cluster of servers with quad PRI cards.
>>> I will appreciate your advice.
>>
>> Your up front cost is going to be a little higher with TDM -> SIP
>> devices, but your management will be a lot easier.
>>
>> AudioCodes also has equipment that can support a DS3 connection, or
>> multiple T1s directly to SIP.
>>
>> For example, If you are getting 22 T1s then get two AudioCodes  
>> Mediant
>> 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans.
>> If you are getting a DS3, get a Mediant 3000.  The M3000 supports up
>> to 84 T1s or three DS3 or one OC3.
>>
>> Then leave call management up to Asterisk.  Of course, have  
>> redundancy
>> everywhere you can.
>
> In this scenario, does Asterisk still touch every RTP packet?  Or  
> can 'native bridge' mode be enabled?
>
> -Jeff
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://www.asterisk.org/hello
>
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Re: [asterisk-users] Asterisk Cluster

[Jonathan Thurman]

On Wed, May 19, 2010 at 6:13 PM, Adolphe Cher-aime  wrote:
> Jonathan for redundancy which software do you recomand?

Without knowing exactly what you are trying to do beside having at
least 500 outbound calls, that would be impossible to say.  I mostly
use a home grown HA Linux configuration (CentOS, cman, MySQL, GFS2)
with Asterisk Realtime.  I would use what you know, as long as it
scales to what you need.  If it doesn't, then I would get someone to
help that has a solution.

-Jonathan

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Re: [asterisk-users] file command with alaw file

[Pham Quy]

On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote:
> On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote:
> > hi all,
> > 
> > I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue
> > Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record
> > audio clip with mixmonitor() as alaw file (softphone is also configured
> > with alaw active only). Using file command i can get the following
> > information
> > 
> > 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio,
> > ITU G.711 A-law, mono 8000 Hz
> > 
> > But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5
> > #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i
> > could get the same information with file command. File command
> > recognized alaw file as JPEG image:
> > 
> > 983006584-20100517-123825.alaw: JPEG image data
> > 
> > I guess i may miss something when i setup the new on on Centos 5.5, but
> > u dont know what it is. Anyone have idea about this?
> > 
> > please help.
> > 
> > Thanks in advance.
> > Quyps
> 
> I did check content of two alaw files (using a hex editor) generated
> from two aboves cases. For the one fromo CentOS 5.2, beginning bytes
> look likes : 
> 
> riff1^0.wavefmt@...@...fact.^0.data.^0...
> 
> and the one from CentOS 5.5 
> 
> ..RQVTVXEMBAX
> 
> It seem like the first one have some information about file format,
> which make our convert tool works correctly, and which the second one
> doesnt have.
> 
> Can you explain to me this different, and how can i get the information
> as the first one?
> 
> Thanks in advances,
> Quyps

This question have been asked for a while, I really need some help
here? 

Thanks in advance.
Quyps 


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