[asterisk-users] Where is the Digium DS3 card?

[Matt Darnell]

Any know what Digium hasn't released the DS3 card?

It was supposed to be out a while ago.

-Matt

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
> Any know what Digium hasn't released the DS3 card?
>
> It was supposed to be out a while ago.

There was a fundamental problem with the chipset used, which precluded the
card from being useful.  Specifically, the chipset only permitted the first
255 channels to be addressed (instead of the full 672).  Since that time, and
partly due to this circumstance, Digium no longer announces the release of
cards until they are ready to be shipped, with drivers and all.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Al Baker]

ok - but has a new release date be announced  ?
or
Has Digium officially dropped the product ?

Steve Totaro wrote:
> On Sun, Apr 6, 2008 at 3:47 AM, Tilghman Lesher
> <[EMAIL PROTECTED]> wrote:
>   
>> On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
>>  > Any know what Digium hasn't released the DS3 card?
>>  >
>>  > It was supposed to be out a while ago.
>>
>>  There was a fundamental problem with the chipset used, which precluded the
>>  card from being useful.  Specifically, the chipset only permitted the first
>>  255 channels to be addressed (instead of the full 672).  Since that time, 
>> and
>>  partly due to this circumstance, Digium no longer announces the release of
>>  cards until they are ready to be shipped, with drivers and all.
>>
>>  --
>>  Tilghman
>>
>> 
>
> Sounds prudent.  I liken it to announcing a vehicle that runs on
> water, maybe, one day.
>
> Anyways, who or why would anyone want to put all 28 eggs in one basket?
>
> Thanks,
> Steve Totaro
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk with lumenvox

[Al Baker]

I don't use Lumenvox, yet.
I have a large client who has expressed interest in having "something 
like it" deployed.
For such a "revolutionary" product there is very little chatter in the 
list about it.
I really had hoped to hear from a bunch of people who had deployed it 
and how it worked out for them.

Josué Conti wrote:
> Hi Al, how are you?
> You use Lumenvox? What we think of the performance of the engine?
> Thank you for your attention
>
> Regards!
>
> Josué
>
> 2008/4/5, Al Baker <[EMAIL PROTECTED]>:
>   
>> I had posted earlier asking about folks real world experiences with
>> with Lumenvox, and the thread 'strangely' disappeared after some
>> bloke from down under justed sodded himself over my straight simple
>> questions.
>> Hm- makes you wonder.
>>
>> Josué Conti wrote:
>> 
>>> Hello everyone.
>>> I wish I could continue with the approval of the engine Lumenvox, for
>>> voice recognition, but not a development of acceptable engine,
>>> Please could help me in achieving test?
>>> As I said earlier we have a project that will involve a very large
>>> number of licenses for Voice recognition, but I would count on help
>>> from Lumenvox, for this case.
>>> Could you help me?
>>>
>>> Best Regards
>>>
>>> Josué
>>>
>>> 2008/3/19 Josué Conti <[EMAIL PROTECTED]>:
>>>
>>>   
 Hello everyone, Rodrigo and Philipp Hello, I would like to know how to
  properly configure the engine Lumenvox no asterisk, I am trying to
  dial by vox actually like that the user should dial for receipt of my
  business, is attended by an IVR system with voice recognition that
  allows the user to say who would like to talk and the asterisk foward
  the call.
  Set up the asterisk below, but the system recognizes the voice, but
  does not guide the call, running immediately after a hangup, what is
  wrong with my settings? I can not very material support on the issue,
  could help me?
  I am not really achieving great results in my tests with engine Lumenvox:
  I am trying to test a simple scheduling dialing by voice, where the
  system identify the user by name and system called in your phone
  number, but I am not able, could help me?
  If I did not say any word, the system is static, but if I say any
  Word, even different words grammar.gram (ura.gram) of the system
  Performs the following priorities file extensions.conf, please, can
  You help me?

  Best Regards

  Josué

  Our programming files are configured this way:
  Ipbx: / etc / asterisk # vim lumenvox.conf
  ; LumenVox configuration file
  [General]
  Servers = 127.0.0.1; Speech Engine Servers to use.
  Save_sound_files = no; Set to yes to save sound files for use with Speech 
 Tuner
  [Grammars]
  ura = / etc / asterisk / grammars / ura.gram
  [Default]
  Vad_snr_sensitivity = 50
  Vad_volume_sensitivity = 50
  Vad_eos_delay = 1250
  Vad_wind_back = 750
  End_of_speech_timeout = 15000
  Use_oov_filter = no
  ;;
  ;; 
  Ipbx: / etc / asterisk # vim extensions.conf
  [General]

  [Globals]

  DYNAMIC_FEATURES => # pickupexten hangup atxfer # # blidxfer

  [Default]
  Length => 2000.1, Playback (Ura / instit / instit_casa)
  Length => 1515.1, Playback (Ura / parabens)

  ;;
  ;;
  ; Pilot URA
  Length => 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1)
  Length => 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1)
  Length => 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1)
  Length => 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1)

  ; IVR URA

  ;
  [URA]
  ;
  Length => s, 1, Answer ()
  Length => s, n, Wait (3)
  Length => s, n, NoOp (entry Ura)
  Length => s, n, Set (TRIES = 0)
  ; Length => s, n, ResponseTimeout (10)
  Length => s, n, BackGround (Ura /abertura)
  Length => s, n, Playback (beep)
  ; Length => s, n, BackGround (Ura / abertura1)
  Length => s, n, Goto (lumenvox-test, s, 1)
  [Lumenvox-test]
  Length => s, 1, Answer
  Length => s, n, Wait (1)
  Length => s, n, SpeechCreate ()
  Length => s, n, SpeechActivateGrammar (Ura)
  Length => s, n, SpeechStart ()
  Length => s, n, SpeechBackground (liggol / abertura)
  Length => s, n, SpeechDeactivateGrammar (Ura)
  Length => s, n, Goto (institutional, s, 1 - $ SPEECH_TEXT (0) ())
  [Institutional]
  Length => s, 1, Playback (Ura / instit / instit)
  Length => s, 2, congestion (3)
  Length => s, 3, hangup

  ipbx: / etc / asterisk / grammars 

Re: [asterisk-users] Zaptel data mode not supported?

[Steve Totaro]

On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
>  > You need to have the kernel compiled specially for it to work.
>
>  Are you sure? What exactly is needed?
>  I think you need to rebuild the kernel on Centos, but on Debian this
>  happens to be supported in the default kernel. Didn't get to test that
>  support yet, though.
>

Tzafrir,

I am not sure actually.

Many years ago I was tasked with setting up E1s, one for data and one
for voice.  There was no definitive guide, but putting *many* pieces
together around the web, I came across blog (hotwo back then) and many
other pieces on how to recompile the kernel with the correct options,
they were not on by default.  This was Whitebox or CentOS (RedHat in
other words).

Never tried on Debian.

I am going to try with a Sangoma T1 on Monday, the ./Setup install
script makes it look like it should be "simple".  We shall see.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On 4/6/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
> > Any know what Digium hasn't released the DS3 card?
> >
> > It was supposed to be out a while ago.
>
> There was a fundamental problem with the chipset used, which precluded the
> card from being useful.  Specifically, the chipset only permitted the first
> 255 channels to be addressed (instead of the full 672).  Since that time,
> and
> partly due to this circumstance, Digium no longer announces the release of
> cards until they are ready to be shipped, with drivers and all.
>
> --
> Tilghman

Would that be 254?  Seems like 254 is always the cap.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 3:47 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Sunday 06 April 2008 02:01:49 Matt Darnell wrote:
>  > Any know what Digium hasn't released the DS3 card?
>  >
>  > It was supposed to be out a while ago.
>
>  There was a fundamental problem with the chipset used, which precluded the
>  card from being useful.  Specifically, the chipset only permitted the first
>  255 channels to be addressed (instead of the full 672).  Since that time, and
>  partly due to this circumstance, Digium no longer announces the release of
>  cards until they are ready to be shipped, with drivers and all.
>
>  --
>  Tilghman
>

Sounds prudent.  I liken it to announcing a vehicle that runs on
water, maybe, one day.

Anyways, who or why would anyone want to put all 28 eggs in one basket?

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel data mode not supported?

[Steve Totaro]

Check page 38 of 74.  A real pain.  Hopefully either Tzafrir is
correct with a different distro (Debian)vor Sangoma makes it simple.

Thanks,
Steve Totaro

On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>  > On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
>  >  > You need to have the kernel compiled specially for it to work.
>  >
>  >  Are you sure? What exactly is needed?
>  >  I think you need to rebuild the kernel on Centos, but on Debian this
>  >  happens to be supported in the default kernel. Didn't get to test that
>  >  support yet, though.
>  >
>
>  Tzafrir,
>
>  I am not sure actually.
>
>  Many years ago I was tasked with setting up E1s, one for data and one
>  for voice.  There was no definitive guide, but putting *many* pieces
>  together around the web, I came across blog (hotwo back then) and many
>  other pieces on how to recompile the kernel with the correct options,
>  they were not on by default.  This was Whitebox or CentOS (RedHat in
>  other words).
>
>  Never tried on Debian.
>
>  I am going to try with a Sangoma T1 on Monday, the ./Setup install
>  script makes it look like it should be "simple".  We shall see.
>
>  Thanks,
>  Steve Totaro
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 05:37:03AM -0400, Steve Totaro wrote:
> > There was a fundamental problem with the chipset used, which precluded the
> > card from being useful.  Specifically, the chipset only permitted the first
> > 255 channels to be addressed (instead of the full 672).  Since that time,
> > and
> > partly due to this circumstance, Digium no longer announces the release of
> > cards until they are ready to be shipped, with drivers and all.
> >
> Would that be 254?  Seems like 254 is always the cap.

I would bet cash that it's 256 channels, numbered 0-255.

The limit is 8 bits of address.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk with lumenvox

[Josué Conti]

OK! I'm also in the process of testing the engine Lumenvox, could not
yet approve a solution to the Portuguese language(testing with Spanish
Colombian), but will be testing this week.
Thank you for your attention, when achieve test I return follow up,ok?
Best Regards

Josué

2008/4/6 Al Baker <[EMAIL PROTECTED]>:
> I don't use Lumenvox, yet.
>  I have a large client who has expressed interest in having "something
>  like it" deployed.
>  For such a "revolutionary" product there is very little chatter in the
>  list about it.
>  I really had hoped to hear from a bunch of people who had deployed it
>  and how it worked out for them.
>
>
>
>  Josué Conti wrote:
>  > Hi Al, how are you?
>  > You use Lumenvox? What we think of the performance of the engine?
>  > Thank you for your attention
>  >
>  > Regards!
>  >
>  > Josué
>  >
>  > 2008/4/5, Al Baker <[EMAIL PROTECTED]>:
>  >
>  >> I had posted earlier asking about folks real world experiences with
>  >> with Lumenvox, and the thread 'strangely' disappeared after some
>  >> bloke from down under justed sodded himself over my straight simple
>  >> questions.
>  >> Hm- makes you wonder.
>  >>
>  >> Josué Conti wrote:
>  >>
>  >>> Hello everyone.
>  >>> I wish I could continue with the approval of the engine Lumenvox, for
>  >>> voice recognition, but not a development of acceptable engine,
>  >>> Please could help me in achieving test?
>  >>> As I said earlier we have a project that will involve a very large
>  >>> number of licenses for Voice recognition, but I would count on help
>  >>> from Lumenvox, for this case.
>  >>> Could you help me?
>  >>>
>  >>> Best Regards
>  >>>
>  >>> Josué
>  >>>
>  >>> 2008/3/19 Josué Conti <[EMAIL PROTECTED]>:
>  >>>
>  >>>
>   Hello everyone, Rodrigo and Philipp Hello, I would like to know how to
>    properly configure the engine Lumenvox no asterisk, I am trying to
>    dial by vox actually like that the user should dial for receipt of my
>    business, is attended by an IVR system with voice recognition that
>    allows the user to say who would like to talk and the asterisk foward
>    the call.
>    Set up the asterisk below, but the system recognizes the voice, but
>    does not guide the call, running immediately after a hangup, what is
>    wrong with my settings? I can not very material support on the issue,
>    could help me?
>    I am not really achieving great results in my tests with engine 
> Lumenvox:
>    I am trying to test a simple scheduling dialing by voice, where the
>    system identify the user by name and system called in your phone
>    number, but I am not able, could help me?
>    If I did not say any word, the system is static, but if I say any
>    Word, even different words grammar.gram (ura.gram) of the system
>    Performs the following priorities file extensions.conf, please, can
>    You help me?
>  
>    Best Regards
>  
>    Josué
>  
>    Our programming files are configured this way:
>    Ipbx: / etc / asterisk # vim lumenvox.conf
>    ; LumenVox configuration file
>    [General]
>    Servers = 127.0.0.1; Speech Engine Servers to use.
>    Save_sound_files = no; Set to yes to save sound files for use with 
> Speech Tuner
>    [Grammars]
>    ura = / etc / asterisk / grammars / ura.gram
>    [Default]
>    Vad_snr_sensitivity = 50
>    Vad_volume_sensitivity = 50
>    Vad_eos_delay = 1250
>    Vad_wind_back = 750
>    End_of_speech_timeout = 15000
>    Use_oov_filter = no
>    ;;
>    ;; 
>    Ipbx: / etc / asterisk # vim extensions.conf
>    [General]
>  
>    [Globals]
>  
>    DYNAMIC_FEATURES => # pickupexten hangup atxfer # # blidxfer
>  
>    [Default]
>    Length => 2000.1, Playback (Ura / instit / instit_casa)
>    Length => 1515.1, Playback (Ura / parabens)
>  
>    ;;
>    ;;
>    ; Pilot URA
>    Length => 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1)
>    Length => 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1)
>    Length => 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1)
>    Length => 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1)
>  
>    ; IVR URA
>  
>    ;
>    [URA]
>    ;
>    Length => s, 1, Answer ()
>    Length => s, n, Wait (3)
>    Length => s, n, NoOp (entry Ura)
>    Length => s, n, Set (TRIES = 0)
>    ; Length => s, n, ResponseTimeout (10)
>    Length => s, n, BackGround (Ura /abertura)
>    Length => s, n, Playback (beep)
>    ; Length => s, n, BackGround (U

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 04:48:19 Al Baker wrote:
> ok - but has a new release date be announced  ?
> or
> Has Digium officially dropped the product ?

Once again, Digium does not announce products until they are ready to ship,
drivers included.  Therefore, I cannot say, either way.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cool New Website

[Dovid B]

> "I think they are using a specially programmed version of Asterisk to do 
> this."
>
>
> Don't you mean:
>
> "I am using a specially programmed version of Asterisk to do this."
>
> ?
>
> domain:  dialaway4free.com
> created: 16-Jan-2008
> last-changed:16-Jan-2008
> registration-expiration: 16-Jan-2009
>
> registrant-firstname:Goran
> registrant-lastname: Donev
> registrant-organization: Donev Technology Consulting Inc
>
>
> Also, clean up your grammar and spelling errors on the site if you want
> anyone to take it seriously.  It's a good idea, and I hope you go far
> with it, but geez, that site looks like it was written by an
> over-caffeinated 12 year old.
>
> Brooks R. Bridges
> Telecommunications Manager
> Ifbyphone, Inc.
> Phone: (847) 983-3000
> Fax: (847) 676-6553
> [EMAIL PROTECTED]
> http://www.ifbyphone.com
>
>

Let me know when to start humming "And another one bites the dust.." 



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel data mode not supported?

[Steve Totaro]

Sorry,

I cannot find the link to the actual Digium link but here are examples
from the wiki:
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration

http://www.voip-info.org/wiki-Asterisk+Data+Configuration

Tomorrow, I will see if data T1 on a Sangoma card is much more simple.
 If I find the Digium PDF I will post it.

Thanks,
Steve Totaro

On Sun, Apr 6, 2008 at 6:12 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> Check page 38 of 74.  A real pain.  Hopefully either Tzafrir is
>  correct with a different distro (Debian)vor Sangoma makes it simple.
>
>  Thanks,
>  Steve Totaro
>
>
>
>  On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro
>  <[EMAIL PROTECTED]> wrote:
>  > On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>  >  > On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
>  >  >  > You need to have the kernel compiled specially for it to work.
>  >  >
>  >  >  Are you sure? What exactly is needed?
>  >  >  I think you need to rebuild the kernel on Centos, but on Debian this
>  >  >  happens to be supported in the default kernel. Didn't get to test that
>  >  >  support yet, though.
>  >  >
>  >
>  >  Tzafrir,
>  >
>  >  I am not sure actually.
>  >
>  >  Many years ago I was tasked with setting up E1s, one for data and one
>  >  for voice.  There was no definitive guide, but putting *many* pieces
>  >  together around the web, I came across blog (hotwo back then) and many
>  >  other pieces on how to recompile the kernel with the correct options,
>  >  they were not on by default.  This was Whitebox or CentOS (RedHat in
>  >  other words).
>  >
>  >  Never tried on Debian.
>  >
>  >  I am going to try with a Sangoma T1 on Monday, the ./Setup install
>  >  script makes it look like it should be "simple".  We shall see.
>  >
>  >  Thanks,
>  >  Steve Totaro
>  >
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel data mode not supported?

[Steve Totaro]

Sorry for all the replies, I found the Digium PDF on Data mode.

http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf

Good luck getting them to support it though ;)

I will post my Sangoma results tomorrow.

Thanks,
Steve Totaro

On Sun, Apr 6, 2008 at 10:49 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> Sorry,
>
>  I cannot find the link to the actual Digium link but here are examples
>  from the wiki:
>  http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
>
>  http://www.voip-info.org/wiki-Asterisk+Data+Configuration
>
>  Tomorrow, I will see if data T1 on a Sangoma card is much more simple.
>   If I find the Digium PDF I will post it.
>
>  Thanks,
>  Steve Totaro
>
>  On Sun, Apr 6, 2008 at 6:12 AM, Steve Totaro
>
>
> <[EMAIL PROTECTED]> wrote:
>  > Check page 38 of 74.  A real pain.  Hopefully either Tzafrir is
>  >  correct with a different distro (Debian)vor Sangoma makes it simple.
>  >
>  >  Thanks,
>  >  Steve Totaro
>  >
>  >
>  >
>  >  On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro
>  >  <[EMAIL PROTECTED]> wrote:
>  >  > On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen <[EMAIL PROTECTED]> 
> wrote:
>  >  >  > On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
>  >  >  >  > You need to have the kernel compiled specially for it to work.
>  >  >  >
>  >  >  >  Are you sure? What exactly is needed?
>  >  >  >  I think you need to rebuild the kernel on Centos, but on Debian this
>  >  >  >  happens to be supported in the default kernel. Didn't get to test 
> that
>  >  >  >  support yet, though.
>  >  >  >
>  >  >
>  >  >  Tzafrir,
>  >  >
>  >  >  I am not sure actually.
>  >  >
>  >  >  Many years ago I was tasked with setting up E1s, one for data and one
>  >  >  for voice.  There was no definitive guide, but putting *many* pieces
>  >  >  together around the web, I came across blog (hotwo back then) and many
>  >  >  other pieces on how to recompile the kernel with the correct options,
>  >  >  they were not on by default.  This was Whitebox or CentOS (RedHat in
>  >  >  other words).
>  >  >
>  >  >  Never tried on Debian.
>  >  >
>  >  >  I am going to try with a Sangoma T1 on Monday, the ./Setup install
>  >  >  script makes it look like it should be "simple".  We shall see.
>  >  >
>  >  >  Thanks,
>  >  >  Steve Totaro
>  >  >
>  >
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 09:44:30AM -0500, Tilghman Lesher wrote:
> On Sunday 06 April 2008 04:48:19 Al Baker wrote:
> > ok - but has a new release date be announced  ?
> > or
> > Has Digium officially dropped the product ?
> 
> Once again, Digium does not announce products until they are ready to ship,
> drivers included.  Therefore, I cannot say, either way.

Well, then.  The answer to the first question is "no, a new release date has
not been announced."

And the answer to the second one is orthogonal to your assertion, Tilghman.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Michael Cargile]

Another reason I am sure that Digium has not released a DS3 TDM card is the
fact that asterisk currently cannot handle that many channels. I am speaking
from experience on this. We have build before a predictive dialer with 16
PRIs. In order to do this and not have audio quality issues we had to use an
8 core Intel Xeon server with 16 gigs of ram, a 6 drive RAID 10, and two
octal echo canceling Sangoma cards. This also required numerous OS tweaks
and dial plan optimizations. The amount of time spend on this was not worth
the final product.

This is not to say that Asterisk will not be able to support this in the
future. In the 1.6 tree, they have change a number of core data structures
and the type of locking used around them which should allow far more
channels to pass through Asterisk with much lower load. I would not even
attempt this though till somewhere around the 1.6.5 release so that the vast
majority of the bugs can be worked out.

In the mean time, if someone really needs to handle that many channels I
would suggest purchasing a DS3 to T1 mux and pass the T1s onto mutliple
Asterisk servers setup in a cluster. In the end you will end up spending far
less money and time setting the system up. I also saw recently at a trade
show a DS3 to SIP converter which might also lower the cost as you would not
need T1 cards. The only issue is that they are a some what new technology
where as DS3 to T1 muxes have been around for some years now and can be
found on ebay for around 700 dollars.

Michael Cargile
Director of Consulting
The Vicidial Group
www.vicidial.com


On Sun, Apr 6, 2008 at 9:55 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:

> On Sun, Apr 06, 2008 at 05:37:03AM -0400, Steve Totaro wrote:
> > > There was a fundamental problem with the chipset used, which precluded
> the
> > > card from being useful.  Specifically, the chipset only permitted the
> first
> > > 255 channels to be addressed (instead of the full 672).  Since that
> time,
> > > and
> > > partly due to this circumstance, Digium no longer announces the
> release of
> > > cards until they are ready to be shipped, with drivers and all.
> > >
> > Would that be 254?  Seems like 254 is always the cap.
>
> I would bet cash that it's 256 channels, numbered 0-255.
>
> The limit is 8 bits of address.
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth   Baylink
> [EMAIL PROTECTED]
> Designer The Things I Think   RFC
> 2100
> Ashworth & Associates http://baylink.pitas.com '87
> e24
> St Petersburg FL USA  http://photo.imageinc.us +1 727 647
> 1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
>   -- (Joseph Stalin)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Sunday 06 April 2008 04:48:19 Al Baker wrote:
>  > ok - but has a new release date be announced  ?
>  > or
>  > Has Digium officially dropped the product ?
>
>  Once again, Digium does not announce products until they are ready to ship,
>  drivers included.  Therefore, I cannot say, either way.
>
>  --
>  Tilghman
>

Was this policy put in place after announcing the DS3 dud?

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

Should have read, "I cannot recommend the Atran MX2800 M13 *enough*"

Two controller cards, two power supplies, battery backup, this is a
nice little box.

It will probably be the most solid piece of equipment in your data center.

Thanks,
Steve Tototaro

On Sun, Apr 6, 2008 at 11:12 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> I cannot recommend the Adtran MX2800 M13, it has redundant everything
>  and is very easy to setup and not very expensive either.
>
>  Thanks,
>  Steve Totaro
>
>
>
>  On Sun, Apr 6, 2008 at 11:04 AM, Michael Cargile <[EMAIL PROTECTED]> wrote:
>  > Another reason I am sure that Digium has not released a DS3 TDM card is the
>  > fact that asterisk currently cannot handle that many channels. I am 
> speaking
>  > from experience on this. We have build before a predictive dialer with 16
>  > PRIs. In order to do this and not have audio quality issues we had to use 
> an
>  > 8 core Intel Xeon server with 16 gigs of ram, a 6 drive RAID 10, and two
>  > octal echo canceling Sangoma cards. This also required numerous OS tweaks
>  > and dial plan optimizations. The amount of time spend on this was not worth
>  > the final product.
>  >
>  > This is not to say that Asterisk will not be able to support this in the
>  > future. In the 1.6 tree, they have change a number of core data structures
>  > and the type of locking used around them which should allow far more
>  > channels to pass through Asterisk with much lower load. I would not even
>  > attempt this though till somewhere around the 1.6.5 release so that the 
> vast
>  > majority of the bugs can be worked out.
>  >
>  > In the mean time, if someone really needs to handle that many channels I
>  > would suggest purchasing a DS3 to T1 mux and pass the T1s onto mutliple
>  > Asterisk servers setup in a cluster. In the end you will end up spending 
> far
>  > less money and time setting the system up. I also saw recently at a trade
>  > show a DS3 to SIP converter which might also lower the cost as you would 
> not
>  > need T1 cards. The only issue is that they are a some what new technology
>  > where as DS3 to T1 muxes have been around for some years now and can be
>  > found on ebay for around 700 dollars.
>  >
>  > Michael Cargile
>  > Director of Consulting
>  > The Vicidial Group
>  > www.vicidial.com
>  >
>  >
>  >
>  >
>  > On Sun, Apr 6, 2008 at 9:55 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>  >
>  > > On Sun, Apr 06, 2008 at 05:37:03AM -0400, Steve Totaro wrote:
>  > > > > There was a fundamental problem with the chipset used, which 
> precluded
>  > the
>  > > > > card from being useful.  Specifically, the chipset only permitted the
>  > first
>  > > > > 255 channels to be addressed (instead of the full 672).  Since that
>  > time,
>  > > > > and
>  > > > > partly due to this circumstance, Digium no longer announces the
>  > release of
>  > > > > cards until they are ready to be shipped, with drivers and all.
>  > > > >
>  > > > Would that be 254?  Seems like 254 is always the cap.
>  > >
>  > > I would bet cash that it's 256 channels, numbered 0-255.
>  > >
>  > > The limit is 8 bits of address.
>  > >
>  > > Cheers,
>  > > -- jra
>  > > --
>  > > Jay R. Ashworth   Baylink
>  > [EMAIL PROTECTED]
>  > > Designer The Things I Think   RFC
>  > 2100
>  > > Ashworth & Associates http://baylink.pitas.com 
> '87
>  > e24
>  > > St Petersburg FL USA  http://photo.imageinc.us +1 727 647
>  > 1274
>  > >
>  > > Those who cast the vote decide nothing.
>  > > Those who count the vote decide everything.
>  > >   -- (Joseph Stalin)
>  > >
>  > > ___
>  > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  > >
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 11:12:33AM -0400, Steve Totaro wrote:
> I cannot recommend the Adtran MX2800 M13, it has redundant everything
> and is very easy to setup and not very expensive either.

We'll assume you meant that you can't recommend them enough.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

I cannot recommend the Adtran MX2800 M13, it has redundant everything
and is very easy to setup and not very expensive either.

Thanks,
Steve Totaro

On Sun, Apr 6, 2008 at 11:04 AM, Michael Cargile <[EMAIL PROTECTED]> wrote:
> Another reason I am sure that Digium has not released a DS3 TDM card is the
> fact that asterisk currently cannot handle that many channels. I am speaking
> from experience on this. We have build before a predictive dialer with 16
> PRIs. In order to do this and not have audio quality issues we had to use an
> 8 core Intel Xeon server with 16 gigs of ram, a 6 drive RAID 10, and two
> octal echo canceling Sangoma cards. This also required numerous OS tweaks
> and dial plan optimizations. The amount of time spend on this was not worth
> the final product.
>
> This is not to say that Asterisk will not be able to support this in the
> future. In the 1.6 tree, they have change a number of core data structures
> and the type of locking used around them which should allow far more
> channels to pass through Asterisk with much lower load. I would not even
> attempt this though till somewhere around the 1.6.5 release so that the vast
> majority of the bugs can be worked out.
>
> In the mean time, if someone really needs to handle that many channels I
> would suggest purchasing a DS3 to T1 mux and pass the T1s onto mutliple
> Asterisk servers setup in a cluster. In the end you will end up spending far
> less money and time setting the system up. I also saw recently at a trade
> show a DS3 to SIP converter which might also lower the cost as you would not
> need T1 cards. The only issue is that they are a some what new technology
> where as DS3 to T1 muxes have been around for some years now and can be
> found on ebay for around 700 dollars.
>
> Michael Cargile
> Director of Consulting
> The Vicidial Group
> www.vicidial.com
>
>
>
>
> On Sun, Apr 6, 2008 at 9:55 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>
> > On Sun, Apr 06, 2008 at 05:37:03AM -0400, Steve Totaro wrote:
> > > > There was a fundamental problem with the chipset used, which precluded
> the
> > > > card from being useful.  Specifically, the chipset only permitted the
> first
> > > > 255 channels to be addressed (instead of the full 672).  Since that
> time,
> > > > and
> > > > partly due to this circumstance, Digium no longer announces the
> release of
> > > > cards until they are ready to be shipped, with drivers and all.
> > > >
> > > Would that be 254?  Seems like 254 is always the cap.
> >
> > I would bet cash that it's 256 channels, numbered 0-255.
> >
> > The limit is 8 bits of address.
> >
> > Cheers,
> > -- jra
> > --
> > Jay R. Ashworth   Baylink
> [EMAIL PROTECTED]
> > Designer The Things I Think   RFC
> 2100
> > Ashworth & Associates http://baylink.pitas.com '87
> e24
> > St Petersburg FL USA  http://photo.imageinc.us +1 727 647
> 1274
> >
> > Those who cast the vote decide nothing.
> > Those who count the vote decide everything.
> >   -- (Joseph Stalin)
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Help, problems with calls sent from nextone gateway

[JoezSweet]

Hi all,

I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway.

Call audio is fine and all seems well but after 15 to 20 sec the call
drops

Most of them are dropped while setting up after 5 - 10 sec
This fails much more often then it is successful

Anyone have a clue on this?
Please fine trace below
Thanks
Joez

Trace :-

Using INVITE request as basis request - [EMAIL PROTECTED]
Found peer 'enswitch-local'
Found RTP audio format 18
Peer audio RTP is at port 82.197.XXX.XXX:20476
Found audio description format G729 for ID 18
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/ 
video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0  
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 82.197.XXX.XXX:20476
Looking for 00556181138037 in from-internal (domain 87.247.224.11)
list_route: hop: 

<--- Transmitting (NAT) to 87.247.224.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  
87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 
5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
Record-Route: 
From: ;tag=3416305095-406953
To: 00556181138037 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Integrics Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: 
Content-Length: 0


<>
Audio is at 87.247.XXX.YYZ port 15364
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
From: "asterisk" ;tag=as1f4953ef
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Integrics Enswitch
Max-Forwards: 70
Date: Fri, 04 Apr 2008 13:31:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2597 2597 IN IP4 87.247.XXX.YYZ
s=session
c=IN IP4 87.247.XXX.YYZ
t=0 0
m=audio 15364 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
 -- Called [EMAIL PROTECTED]
asterisk2*CLI>
<--- SIP read from 216.19.ZZZ.ZZZ:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6
From: "asterisk" ;tag=as1f4953ef
Call-ID: [EMAIL PROTECTED]
To: ;tag=040431081453123850101510433
Contact: 
Content-Length: 0


<--- SIP read from 87.247.XXX.YYY:5060 --->
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Record-Route: 
Max-Forwards: 69
To: 00556181138037 
From: ;tag=3416305095-406953
Call-ID: [EMAIL PROTECTED]
CSeq: 1 CANCEL
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,  
REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0
Via: SIP/2.0/UDP 82.197.XYZ.XYZ: 
5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
Contact: 
Content-Length: 0
X-Enswitch-Source: 82.197.XYZ.XYZ:5060
X-Enswitch-External: yes

Sending to 87.247.XXX.YYY : 5060 (NAT)
<--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP  
87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 
5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
From: ;tag=3416305095-406953
To: 00556181138037 ;tag=as6ec74197
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Integrics Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<--- Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP  
87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 
5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
Record-Route: 
From: ;tag=3416305095-406953
To: 00556181138037 ;tag=as6ec74197
Call-ID: [EMAIL PROTECTED]
CSeq: 1 CANCEL
User-Agent: Integrics Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: 
Content-Length: 0


<--- SIP read from 87.247.XXX.YYY:5060 --->
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0
From: ;tag=3416305095-406953
Call-ID: [EMAIL PROTECTED]
To: 00556181138037 ;tag=as6ec74197
CSeq: 1 ACK
User-Agent: Enswitch SIP proxy
Content-Length: 0


<->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '[EMAIL PROTECTED] 
' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
From: "asterisk" ;tag=as1f4953ef
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Integrics Enswitch
Max-Forwards: 70
Content-Length: 0


<--- SIP read from 216.19

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 10:13:43 Steve Totaro wrote:
> On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher wrote:
> > On Sunday 06 April 2008 04:48:19 Al Baker wrote:
> >  > ok - but has a new release date be announced  ?
> >  > or
> >  > Has Digium officially dropped the product ?
> >
> >  Once again, Digium does not announce products until they are ready to
> > ship, drivers included.  Therefore, I cannot say, either way.
>
> Was this policy put in place after announcing the DS3 dud?

The policy was put into place for a number of reasons, the main one being that
it's generally a good idea and is therefore pretty industry standard.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help, problems with calls sent from nextone gateway

[Steve Totaro]

On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <[EMAIL PROTECTED]> wrote:
> Hi all,
>
>  I'm having problems with calls dropping after 15 - 20 seconds from a
>  particular provider. The are using a NexTone gateway.
>
>  Call audio is fine and all seems well but after 15 to 20 sec the call
>  drops
>
>  Most of them are dropped while setting up after 5 - 10 sec
>  This fails much more often then it is successful
>
>  Anyone have a clue on this?
>  Please fine trace below
>  Thanks
>  Joez
>
>  Trace :-
>
>  Using INVITE request as basis request - [EMAIL PROTECTED]
>  Found peer 'enswitch-local'
>  Found RTP audio format 18
>  Peer audio RTP is at port 82.197.XXX.XXX:20476
>  Found audio description format G729 for ID 18
>  Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/
>  video=0x0 (nothing), combined - 0x100 (g729)
>  Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
>  (nothing), combined - 0x0 (nothing)
>  Peer audio RTP is at port 82.197.XXX.XXX:20476
>  Looking for 00556181138037 in from-internal (domain 87.247.224.11)
>  list_route: hop: 
>
>  <--- Transmitting (NAT) to 87.247.224.5:5060 --->
>  SIP/2.0 100 Trying
>  Via: SIP/2.0/UDP
>  87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
>  Via: SIP/2.0/UDP 82.197..XYZ.XYZ:
>  5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>  Record-Route: 
>  From: ;tag=3416305095-406953
>  To: 00556181138037 
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 1 INVITE
>  User-Agent: Integrics Enswitch
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Contact: 
>  Content-Length: 0
>
>
>  <>
>  Audio is at 87.247.XXX.YYZ port 15364
>  Adding codec 0x100 (g729) to SDP
>  Adding non-codec 0x1 (telephone-event) to SDP
>  Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
>  INVITE sip:[EMAIL PROTECTED] SIP/2.0
>  Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
>  From: "asterisk" ;tag=as1f4953ef
>  To: 
>  Contact: 
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 102 INVITE
>  User-Agent: Integrics Enswitch
>  Max-Forwards: 70
>  Date: Fri, 04 Apr 2008 13:31:55 GMT
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Type: application/sdp
>  Content-Length: 263
>
>  v=0
>  o=root 2597 2597 IN IP4 87.247.XXX.YYZ
>  s=session
>  c=IN IP4 87.247.XXX.YYZ
>  t=0 0
>  m=audio 15364 RTP/AVP 18 101
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=no
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-16
>  a=silenceSupp:off - - - -
>  a=ptime:20
>  a=sendrecv
>
>  ---
>  -- Called [EMAIL PROTECTED]
>  asterisk2*CLI>
>  <--- SIP read from 216.19.ZZZ.ZZZ:5060 --->
>  SIP/2.0 100 Trying
>  CSeq: 102 INVITE
>  Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6
>  From: "asterisk" ;tag=as1f4953ef
>  Call-ID: [EMAIL PROTECTED]
>  To: ;tag=040431081453123850101510433
>  Contact: 
>  Content-Length: 0
>
>
>  <--- SIP read from 87.247.XXX.YYY:5060 --->
>  CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
>  Record-Route: 
>  Max-Forwards: 69
>  To: 00556181138037 
>  From: ;tag=3416305095-406953
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 1 CANCEL
>  Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,
>  REFER, SUBSCRIBE, PRACK, UPDATE
>  Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0
>  Via: SIP/2.0/UDP 82.197.XYZ.XYZ:
>  5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>  Contact: 
>  Content-Length: 0
>  X-Enswitch-Source: 82.197.XYZ.XYZ:5060
>  X-Enswitch-External: yes
>
>  Sending to 87.247.XXX.YYY : 5060 (NAT)
>  <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
>  SIP/2.0 487 Request Terminated
>  Via: SIP/2.0/UDP
>  87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
>  Via: SIP/2.0/UDP 82.197..XYZ.XYZ:
>  5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>  From: ;tag=3416305095-406953
>  To: 00556181138037 ;tag=as6ec74197
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 1 INVITE
>  User-Agent: Integrics Enswitch
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Content-Length: 0
>
>
>  <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
>  SIP/2.0 200 OK
>  Via: SIP/2.0/UDP
>  87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
>  Via: SIP/2.0/UDP 82.197..XYZ.XYZ:
>  5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>  Record-Route: 
>  From: ;tag=3416305095-406953
>  To: 00556181138037 ;tag=as6ec74197
>  Call-ID: [EMAIL PROTECTED]
>  CSeq: 1 CANCEL
>  User-Agent: Integrics Enswitch
>  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>  Supported: replaces
>  Contact: 
>  Content-Length: 0
>
>
>  <--- SIP read from 87.247.XXX.YYY:5060 --->
>  ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
>  Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0
>  From: ;tag=3416305095-406953
>  Call-ID: [EMAIL PROTECTED]
>  To: 00556181138037 ;tag=as6ec74197
>  CSeq: 1 ACK
>  User-Agent: Enswitch S

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 11:58 AM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
> On Sunday 06 April 2008 10:13:43 Steve Totaro wrote:
>
> > On Sun, Apr 6, 2008 at 10:44 AM, Tilghman Lesher wrote:
>  > > On Sunday 06 April 2008 04:48:19 Al Baker wrote:
>  > >  > ok - but has a new release date be announced  ?
>  > >  > or
>  > >  > Has Digium officially dropped the product ?
>  > >
>  > >  Once again, Digium does not announce products until they are ready to
>  > > ship, drivers included.  Therefore, I cannot say, either way.
>  >
>
> > Was this policy put in place after announcing the DS3 dud?
>
>  The policy was put into place for a number of reasons, the main one being 
> that
>  it's generally a good idea and is therefore pretty industry standard.
>
>
>
>  --
>  Tilghman
>

Ah crud, I about to announce my perpetual motion machine!  Guess I have to wait.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Michael Cargile]

 On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen <
[EMAIL PROTECTED]> wrote:

> We've already connected ~600 analog extensions to Asterisk and we were
> far from reaching the bottlenecks. We have used a machine that is hardly
> a top-of-the line server (a dual-core Dell machine), with some 20
> Astribank 32 FXS-s connected to it.
>


There is big difference between a TDM card and a USB device. Also were you
actively using all ~600 extensions at once?

The system we were building was a predictive dialer. Not only were all
channels engaged at one time but we were actively establishing and
disconnecting channels at a rate of over 15 per second.

Anyone who is looking for a DS3 TDM card is probably not looking to use it
to hookup channel banks with it. They are looking to increase their line
capacity, thus increasing their number of concurrent channels.

Michael Cargile
Director of Consulting
The Vicidial Group
www.vicidial.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Tzafrir Cohen]

On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
>  On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen <
> [EMAIL PROTECTED]> wrote:

(Off-list, and not expecting an on-list reply)

> 
> > We've already connected ~600 analog extensions to Asterisk and we were
> > far from reaching the bottlenecks. We have used a machine that is hardly
> > a top-of-the line server (a dual-core Dell machine), with some 20
> > Astribank 32 FXS-s connected to it.
> >

(And nicely enough, the point of the message was left unquoted: that 672 
channels are certainly possible, if there's a will.)

> 
> 
> There is big difference between a TDM card and a USB device. 

Our device is both a USB device and a TDM one. Analog vs. digital (or
extension vs. trunk) may be relevant. As for PCI vs. USB - they keep
telling us USB makes things more complicated :-)

> Also were you
> actively using all ~600 extensions at once?

No. In those limited tests we made we got to ~200 FXS channels running.

> 
> The system we were building was a predictive dialer. Not only were all
> channels engaged at one time but we were actively establishing and
> disconnecting channels at a rate of over 15 per second.
> 
> Anyone who is looking for a DS3 TDM card is probably not looking to use it
> to hookup channel banks with it. They are looking to increase their line
> capacity, thus increasing their number of concurrent channels.

While I don't have your experince, I would still speculate that even for
trunks the average capacity is not ther same load as that of a
predictive dialer.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 10:58:56AM -0500, Tilghman Lesher wrote:
> > Was this policy put in place after announcing the DS3 dud?
> 
> The policy was put into place for a number of reasons, the main one being that
> it's generally a good idea and is therefore pretty industry standard.

Two words: Osborne Two.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
> On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
> >  On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen <
> > [EMAIL PROTECTED]> wrote:
> 
> (Off-list, and not expecting an on-list reply)

Ooh, geee: Why Reply-To Munging is Considered Harmful.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help, problems with calls sent from nextone gateway

[JoezSweet]

Hi there,

We get witheld caller cli from client and no cli output, but i dont  
think it's the problem.

We had a test with about 200 calls and we got an ACD of about 30 sec,
while from another client with asterisk, for the same route we get  
about 3 min ACD.
Beside that we get calls dropped after few sec from invite.

So we're thinking on a compatibility problem between nextone and  
asterisk,
or kind of codec stuff we've Digium G729 licensed.

Client configuration is 1 IP for signalling 1 for media and sending G. 
729

Any idea on what can be the problem?

Thanks
Giovanni


Il giorno 06/apr/08, alle ore 18:04, Steve Totaro ha scritto:
> On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <[EMAIL PROTECTED]>  
> wrote:
>> Hi all,
>>
>> I'm having problems with calls dropping after 15 - 20 seconds from a
>> particular provider. The are using a NexTone gateway.
>>
>> Call audio is fine and all seems well but after 15 to 20 sec the call
>> drops
>>
>> Most of them are dropped while setting up after 5 - 10 sec
>> This fails much more often then it is successful
>>
>> Anyone have a clue on this?
>> Please fine trace below
>> Thanks
>> Joez
>>
>> Trace :-
>>
>> Using INVITE request as basis request - [EMAIL PROTECTED]
>> Found peer 'enswitch-local'
>> Found RTP audio format 18
>> Peer audio RTP is at port 82.197.XXX.XXX:20476
>> Found audio description format G729 for ID 18
>> Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/
>> video=0x0 (nothing), combined - 0x100 (g729)
>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
>> (nothing), combined - 0x0 (nothing)
>> Peer audio RTP is at port 82.197.XXX.XXX:20476
>> Looking for 00556181138037 in from-internal (domain 87.247.224.11)
>> list_route: hop: 
>>
>> <--- Transmitting (NAT) to 87.247.224.5:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
>> Via: SIP/2.0/UDP 82.197..XYZ.XYZ:
>> 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>> Record-Route: 
>> From: ;tag=3416305095-406953
>> To: 00556181138037 
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 INVITE
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: 
>> Content-Length: 0
>>
>>
>> <>
>> Audio is at 87.247.XXX.YYZ port 15364
>> Adding codec 0x100 (g729) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
>> INVITE sip:[EMAIL PROTECTED] SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
>> From: "asterisk" ;tag=as1f4953ef
>> To: 
>> Contact: 
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE
>> User-Agent: Integrics Enswitch
>> Max-Forwards: 70
>> Date: Fri, 04 Apr 2008 13:31:55 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Content-Type: application/sdp
>> Content-Length: 263
>>
>> v=0
>> o=root 2597 2597 IN IP4 87.247.XXX.YYZ
>> s=session
>> c=IN IP4 87.247.XXX.YYZ
>> t=0 0
>> m=audio 15364 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> ---
>> -- Called [EMAIL PROTECTED]
>> asterisk2*CLI>
>> <--- SIP read from 216.19.ZZZ.ZZZ:5060 --->
>> SIP/2.0 100 Trying
>> CSeq: 102 INVITE
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6
>> From: "asterisk" ;tag=as1f4953ef
>> Call-ID: [EMAIL PROTECTED]
>> To: ;tag=040431081453123850101510433
>> Contact: 
>> Content-Length: 0
>>
>>
>> <--- SIP read from 87.247.XXX.YYY:5060 --->
>> CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
>> Record-Route: 
>> Max-Forwards: 69
>> To: 00556181138037 
>> From: ;tag=3416305095-406953
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 CANCEL
>> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,
>> REFER, SUBSCRIBE, PRACK, UPDATE
>> Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0
>> Via: SIP/2.0/UDP 82.197.XYZ.XYZ:
>> 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>> Contact: 
>> Content-Length: 0
>> X-Enswitch-Source: 82.197.XYZ.XYZ:5060
>> X-Enswitch-External: yes
>>
>> Sending to 87.247.XXX.YYY : 5060 (NAT)
>> <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
>> SIP/2.0 487 Request Terminated
>> Via: SIP/2.0/UDP
>> 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY
>> Via: SIP/2.0/UDP 82.197..XYZ.XYZ:
>> 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc
>> From: ;tag=3416305095-406953
>> To: 00556181138037 > [EMAIL PROTECTED]>;tag=as6ec74197
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 1 INVITE
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Content-Length: 0
>>
>>
>> <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP
>> 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.Y

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 11:45:51 Jay R. Ashworth wrote:
> On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
> > On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
> > >  On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir
> > > Cohen < [EMAIL PROTECTED]> wrote:
> >
> > (Off-list, and not expecting an on-list reply)
>
> Ooh, geee: Why Reply-To Munging is Considered Harmful.  :-)

Oh, please, don't start THAT flame war.  People who consider Reply-To
munging harmful have obviously failed to read the ENTIRE rfc (especially
the part where it specifically says the use of the  Reply-To header is for use
on listservs).  :-P

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
> >  On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir Cohen <
> > [EMAIL PROTECTED]> wrote:
>
> (Off-list, and not expecting an on-list reply)
>
> >
> > > We've already connected ~600 analog extensions to Asterisk and we were
> > > far from reaching the bottlenecks. We have used a machine that is hardly
> > > a top-of-the line server (a dual-core Dell machine), with some 20
> > > Astribank 32 FXS-s connected to it.
> > >
>
> (And nicely enough, the point of the message was left unquoted: that 672
> channels are certainly possible, if there's a will.)
>
> >
> >
> > There is big difference between a TDM card and a USB device.
>
> Our device is both a USB device and a TDM one. Analog vs. digital (or
> extension vs. trunk) may be relevant. As for PCI vs. USB - they keep
> telling us USB makes things more complicated :-)
>
> > Also were you
> > actively using all ~600 extensions at once?
>
> No. In those limited tests we made we got to ~200 FXS channels running.
>
> >
> > The system we were building was a predictive dialer. Not only were all
> > channels engaged at one time but we were actively establishing and
> > disconnecting channels at a rate of over 15 per second.
> >
> > Anyone who is looking for a DS3 TDM card is probably not looking to use it
> > to hookup channel banks with it. They are looking to increase their line
> > capacity, thus increasing their number of concurrent channels.
>
> While I don't have your experince, I would still speculate that even for
> trunks the average capacity is not ther same load as that of a
> predictive dialer.
>

I used an Adtran MX2800 to break out 28 T1s (PRI D chan on fourth T1
using NFAS) per quad port server.

As for call setup, hearsay, is that Asterisk is very slow or not
capable of handling high volume call setups.

CallWeaver, forks, or whatever you want to call them are many times
better at setting up SIP calls per second than Asterisk.

I do not have the numbers, I only did inbound, but maybe someone could
elaborate on actual figures.

Thanks,
Steve totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032

[Ruben Zamora]

Hi,
I have a same problem, last week i was working with TE120 with a little 
echo in some call,  I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no 
more echo in my call.

But know i have de same probelm with my incoming audio stream gets 
clipped / dropped when you speak.

Thanks
Ruben

Lex Lethol escribió:
> Hi,
>
> I've used all kinds of digium cards without troubles.  My last
> installation is using a TDM2400p with VPMADT032 echo cancel module and
> after a week of use we noticed that any incoming audio stream gets
> clipped / dropped when you speak or when ambient noise is high.  The
> call basically feels as in a half-duplex channel, but only to the
> person behind our asterisk.  I found a quick way to recreate by
> placing a call using zapata channel, someplace that has an audio
> stream (ie. music on hold from another pbx).  When one talks into the
> phone, one can notice the incoming audio getting muted until you stop
> talking.
>
> First I thought it had to do with polycom configuration although we
> use the same setup for all installations (VAD, etc), but the same
> happens with other sip phones and after more tests I can only recreate
> this using the TDM2400p's FXO trunks.  I have an older TDM2400p with
> no VPMADT032 in production (without this problem), this leads me to
> believe there maybe something wrong with VPMADT032 module or with my
> card in particular.
>
> Today I rebuilt everything from scratch using latest asterisk 1.2
> release, rechecked with the TDM2400p manual zapata configs just to
> make sure I wasn't missing something.  As the manual suggests, I am
> just using echocancel=yes and this should set 128 default value for
> the card.  In the general zapata options there we have
> echocancelwhenbridged=yes.  I have played with all yes/no combinations
> without luck.
>
> Interrupts and timing stuff are OK, we have good incoming and outgoing
> audio quality (as long as its not at the same time).
>
> Anyone else using this card showing the same problems?
>
> Any zaptel/asterisk gurus wanna take a shot at this?
>
> Thanks in advance for your feedback/comments.
>
> Lex
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Michael Cargile]

Sorry about that not used to gmail. :-)

Michael Cargile
Director of Consulting
The Vicidial Group
www.vicidial.com


On Sun, Apr 6, 2008 at 12:45 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:

> On Sun, Apr 06, 2008 at 07:38:06PM +0300, Tzafrir Cohen wrote:
> > On Sun, Apr 06, 2008 at 12:12:17PM -0400, Michael Cargile wrote:
> > >  On Sun, Apr 6, 2008 at 11:27 AM, Tzafrir
> Cohen <
> > > [EMAIL PROTECTED]> wrote:
> >
> > (Off-list, and not expecting an on-list reply)
>
> Ooh, geee: Why Reply-To Munging is Considered Harmful.  :-)
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth   Baylink
> [EMAIL PROTECTED]
> Designer The Things I Think   RFC
> 2100
> Ashworth & Associates http://baylink.pitas.com '87
> e24
> St Petersburg FL USA  http://photo.imageinc.us +1 727 647
> 1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
>   -- (Joseph Stalin)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Michael Cargile]

On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:

> While I don't have your experince, I would still speculate that even for
> trunks the average capacity is not ther same load as that of a
> predictive dialer.
>

This is very true.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] UK POTS - Is there a better card than TDM400P available ?

[Matt Brown]

Hi,

I am feeling very frustrated with the Digium TDM400P, I have 3 x FXS  
1x FXO modules and I have tried various things and different versions  
of Asterisk and Zaptel to no avail.

Clearly there are issues with this card, so I am wondering - is there  
a card out there that does the following without the inherent problems  
of the TDM400 ?

I.e a card that can reliably do:

UK CID
Distinctive Ring Detection.

Any pointers would be great.

Thanks.

( or is there an Asterisk version and Zaptel version in the 1.4 branch  
that fixes these issues ? )

Matt Brown





___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel data mode not supported?

[Alex Kauffmann]

Steve Totaro wrote:
> Sorry for all the replies, I found the Digium PDF on Data mode.
>
> http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
>
> Good luck getting them to support it though ;)
>
> I will post my Sangoma results tomorrow.
>
> Thanks,
> Steve Totaro
>
> On Sun, Apr 6, 2008 at 10:49 AM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
>   
>> Sorry,
>>
>>  I cannot find the link to the actual Digium link but here are examples
>>  from the wiki:
>>  http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
>>
>>  http://www.voip-info.org/wiki-Asterisk+Data+Configuration
>>
>>  Tomorrow, I will see if data T1 on a Sangoma card is much more simple.
>>   If I find the Digium PDF I will post it.
>>
>>  Thanks,
>>  Steve Totaro
>>
>>  On Sun, Apr 6, 2008 at 6:12 AM, Steve Totaro
>>
>>
>> <[EMAIL PROTECTED]> wrote:
>>  > Check page 38 of 74.  A real pain.  Hopefully either Tzafrir is
>>  >  correct with a different distro (Debian)vor Sangoma makes it simple.
>>  >
>>  >  Thanks,
>>  >  Steve Totaro
>>  >
>>  >
>>  >
>>  >  On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro
>>  >  <[EMAIL PROTECTED]> wrote:
>>  >  > On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen <[EMAIL PROTECTED]> 
>> wrote:
>>  >  >  > On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote:
>>  >  >  >  > You need to have the kernel compiled specially for it to work.
>>  >  >  >
>>  >  >  >  Are you sure? What exactly is needed?
>>  >  >  >  I think you need to rebuild the kernel on Centos, but on Debian 
>> this
>>  >  >  >  happens to be supported in the default kernel. Didn't get to test 
>> that
>>  >  >  >  support yet, though.
>>  >  >  >
>>  >  >
>>  >  >  Tzafrir,
>>  >  >
>>  >  >  I am not sure actually.
>>  >  >
>>  >  >  Many years ago I was tasked with setting up E1s, one for data and one
>>  >  >  for voice.  There was no definitive guide, but putting *many* pieces
>>  >  >  together around the web, I came across blog (hotwo back then) and many
>>  >  >  other pieces on how to recompile the kernel with the correct options,
>>  >  >  they were not on by default.  This was Whitebox or CentOS (RedHat in
>>  >  >  other words).
>>  >  >
>>  >  >  Never tried on Debian.
>>  >  >
>>  >  >  I am going to try with a Sangoma T1 on Monday, the ./Setup install
>>  >  >  script makes it look like it should be "simple".  We shall see.
>>  >  >
>>  >  >  Thanks,
>>  >  >  Steve Totaro
>>  >  >
>>  >
>>
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   
Thank you for the replies.  It was my understanding that rebuilding the 
kernel was necessary in 2.4 but everything needed was already included 
in 2.6 series.  My bad I guess.  I was trying to find a use for old 
cards I have, but If i'm going to have to use a Sangoma, I'll just use 
Vyatta which supports them out of the box.  Too bad that the agreement 
between Vyatta and Digium has not resulted in them supporting Digium 
cards yet.

Alex



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 1:53 PM, Michael Cargile <[EMAIL PROTECTED]> wrote:
>
>
> On Sun, Apr 6, 2008 at 12:38 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
> wrote:
> > While I don't have your experince, I would still speculate that even for
> > trunks the average capacity is not ther same load as that of a
> > predictive dialer.
> >
>
> This is very true.
>

The setup and teardown are the devils in the machine.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] OT Good Price $.099 - Voip/ Skype PC Handset UP609 "Regular $29.99"

[Steve Totaro]

http://www.surpluscomputers.com/store/main.aspx?p=ItemDetail&item=CES11532

Ground shipping $9 to MD.  Never used so I cannot comment on quality
and Linux is not listed as compatible but I suppose as long as your
audio jacks work, then the handset will too.

Unfortunately, shipping goes up linearly with units ordered.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Benny Amorsen]

Tilghman Lesher <[EMAIL PROTECTED]> writes:

> Oh, please, don't start THAT flame war.  People who consider Reply-To
> munging harmful have obviously failed to read the ENTIRE rfc (especially
> the part where it specifically says the use of the  Reply-To header is for use
> on listservs).  :-P

It's harmful whether an RFC says it is or not.


/Benny



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Andrew Kohlsmith (lists)]

On April 6, 2008 11:12:33 am Steve Totaro wrote:
> I cannot recommend the Adtran MX2800 M13, it has redundant everything
> and is very easy to setup and not very expensive either.

Agreed; I've set these up and they are rock effing solid. We did have a shelf 
controller die and without the backup shelf controller, we had our only DS3 
down for several days... the replacement shelf controller was lost by UPS... 
talk about a train wreck learning experience.  :-)

Ahh, the good old days...

-A.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen <[EMAIL PROTECTED]> wrote:
> Tilghman Lesher <[EMAIL PROTECTED]> writes:
>
> > Oh, please, don't start THAT flame war.  People who consider Reply-To
> > munging harmful have obviously failed to read the ENTIRE rfc (especially
> > the part where it specifically says the use of the  Reply-To header is for 
> > use
> > on listservs).  :-P
>
> It's harmful whether an RFC says it is or not.
>
>
> /Benny
>

Is RFC 3389 harmful?

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need help with Cisco 7960

[Steve Totaro]

You probably have to unlock it first.  Google or voip-info.org is your friend.

On Sun, Apr 6, 2008 at 5:02 PM, Christian <[EMAIL PROTECTED]> wrote:
> Hello all,
> I need some help with my Cisco 7960 enabling TFTP. Does anyone know what 
> numbers to press in the menu? Or can I enable this through telnet?
> Many thanks,
> Christian
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Need help with Cisco 7960

[Christian]

Hello all,
I need some help with my Cisco 7960 enabling TFTP. Does anyone know what 
numbers to press in the menu? Or can I enable this through telnet?
Many thanks,
Christian


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need help with Cisco 7960

[Christian]

Hello,
I know how to unlock the phone and what the password is.
I am asking this kind of question because i am visually impaired and cannot see 
the screen.
many thanks,
Christian


On 2008-04-06 at 17:05 Steve Totaro wrote:

>You probably have to unlock it first.  Google or voip-info.org is your
>friend.
>
>On Sun, Apr 6, 2008 at 5:02 PM, Christian <[EMAIL PROTECTED]> wrote:
>> Hello all,
>> I need some help with my Cisco 7960 enabling TFTP. Does anyone know what
>numbers to press in the menu? Or can I enable this through telnet?
>> Many thanks,
>> Christian
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>___
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Jay R. Ashworth]

On Sun, Apr 06, 2008 at 05:02:53PM -0400, Steve Totaro wrote:
> http://www.metasystema.net/essays/reply-to.mhtml
> 
> Wildefires are harmful to forests and aniaml life but are essentially
> a good thing in the balance and cycles of nature.  It all depends on
> your viewpoint.

Yes, I've seen that, and most of its arguments are specious, at best.
They amount to "I am too stupid to find a mail user agent with List
Reply, and too lazy to switch to it".

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need help with Cisco 7960

[Steve Totaro]

In that case, I guess I would ask somone with better sight to help me
out, uless they have braille on the buttons.

Thanks,
Steve Totaro

On Sun, Apr 6, 2008 at 5:09 PM, Christian <[EMAIL PROTECTED]> wrote:
> Hello,
> I know how to unlock the phone and what the password is.
> I am asking this kind of question because i am visually impaired and cannot 
> see the screen.
> many thanks,
> Christian
>
>
>
> On 2008-04-06 at 17:05 Steve Totaro wrote:
>
> >You probably have to unlock it first.  Google or voip-info.org is your
> >friend.
> >
> >On Sun, Apr 6, 2008 at 5:02 PM, Christian <[EMAIL PROTECTED]> wrote:
> >> Hello all,
> >> I need some help with my Cisco 7960 enabling TFTP. Does anyone know what
> >numbers to press in the menu? Or can I enable this through telnet?
> >> Many thanks,
> >> Christian
> >>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen <[EMAIL PROTECTED]> wrote:
> Tilghman Lesher <[EMAIL PROTECTED]> writes:
>
> > Oh, please, don't start THAT flame war.  People who consider Reply-To
> > munging harmful have obviously failed to read the ENTIRE rfc (especially
> > the part where it specifically says the use of the  Reply-To header is for 
> > use
> > on listservs).  :-P
>
> It's harmful whether an RFC says it is or not.
>
>
> /Benny
>

http://www.metasystema.net/essays/reply-to.mhtml

Wildefires are harmful to forests and aniaml life but are essentially
a good thing in the balance and cycles of nature.  It all depends on
your viewpoint.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 15:56:22 Steve Totaro wrote:
> On Sun, Apr 6, 2008 at 4:28 PM, Benny Amorsen <[EMAIL PROTECTED]> 
wrote:
> > Tilghman Lesher <[EMAIL PROTECTED]> writes:
> > > Oh, please, don't start THAT flame war.  People who consider Reply-To
> > > munging harmful have obviously failed to read the ENTIRE rfc
> > > (especially the part where it specifically says the use of the 
> > > Reply-To header is for use on listservs).  :-P
> >
> > It's harmful whether an RFC says it is or not.
> >
> >
> > /Benny
>
> Is RFC 3389 harmful?

RFC 3261 is *definitely* harmful.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Tilghman Lesher]

On Sunday 06 April 2008 16:22:58 Jay R. Ashworth wrote:
> On Sun, Apr 06, 2008 at 05:02:53PM -0400, Steve Totaro wrote:
> > http://www.metasystema.net/essays/reply-to.mhtml
> >
> > Wildefires are harmful to forests and aniaml life but are essentially
> > a good thing in the balance and cycles of nature.  It all depends on
> > your viewpoint.
>
> Yes, I've seen that, and most of its arguments are specious, at best.
> They amount to "I am too stupid to find a mail user agent with List
> Reply, and too lazy to switch to it".

And the arguments on the other side come down to "I'm using an ISP
which can't correctly configure their mailserver, and I'm too lazy to set one
up myself." and "I'm too lazy to check the headers when I send out a reply."

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Newbie Polycom: Headset Suggestion for IP601

[Lee, John (Sydney)]

Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
Thanks.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Zaptel data mode not supported?

[Andreas van dem Helge]

On Sun, Apr 6, 2008 at 1:52 PM, Alex Kauffmann <[EMAIL PROTECTED]> wrote:
>  Thank you for the replies.  It was my understanding that rebuilding the
>  kernel was necessary in 2.4 but everything needed was already included
>  in 2.6 series.

*HEAVILY* dependent on the distribution!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problems with Unicall and TE122B

[Ruben Zamora]

Hi

By the past 2 months i have install a server with Asterisk with a E1 in 
Axtel(Mexico). The call presented a Echo in
randoms call,  inside and outside.I decide migrate for the last 
version Asterisk 18.1,Zaptel 1.49 and Unicall and instalaled
a new Digim Card TE122 B (Echo cancelation).   The echo goes away.   But 
i stat gaving probelms with  my incoming audio
 stream gets clipped / dropped when they speak.

I dont know if i need to move a parameter in unicall.conf, or in another 
file.

Thanks

Ruben

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?

[James Williamson]

Snap,

Well, after trying to buying a TDM400P and then getting persuaded to buy 
a TDM410P
because they no longer sell the 400 model I'd say I'm not impressed. It 
took three 2.6
kernel builds (zaptel 1.4 won't even build with the latest kernel 
release) before I finally got the
kernel to build and recognise the device. Can't get callerid to work, 
have followed the instructions
to the letter, does it really have to be this hard? Must say I'm tempted 
to just send the card
back and forget about it and just use Cisco CallManager. Digium's 
support to be fair have been
responsive but unable to help.

James
> Hi,
>
> I am feeling very frustrated with the Digium TDM400P, I have 3 x FXS  
> 1x FXO modules and I have tried various things and different versions  
> of Asterisk and Zaptel to no avail.
>
> Clearly there are issues with this card, so I am wondering - is there  
> a card out there that does the following without the inherent problems  
> of the TDM400 ?
>
> I.e a card that can reliably do:
>
> UK CID
> Distinctive Ring Detection.
>
> Any pointers would be great.
>
> Thanks.
>
> ( or is there an Asterisk version and Zaptel version in the 1.4 branch  
> that fixes these issues ? )
>
> Matt Brown
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Touch Recording

[Lee, John (Sydney)]

I had this problem before...the following appeared in a previous post...
For some reasons, the "*" and "1" must be pressed pretty quickly together on 
the Polycom phone before it can be transmitted successfully to Asterisk.
Does anyone know if that can be tuned?
Sure... go to features.conf, and change the value of the featuredigittimeout 
option.
Hope this helps.


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, 
David C
Sent: Monday, 7 April 2008 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] One Touch Recording

Hi All,

For some reason One Touch Recoding does not work over ZAP but it does work when 
I call another extension.  Both Dial commands have the W option for the calling 
party to enable recording.

Does anyone know why it works internally but not over ZAP.  I have a TE110P 
card on an E1 connection. 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Touch Recording

[Klaverstyn, David C]

Thanks for that.  I have the timeout set to 3000 ms and I have been pressing 
the *1 within 500 ms so I don't think it is related to that.  As I can do it 
over SIP but not ZAP does not make much sense to me.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John 
(Sydney)
Sent: Monday, 7 April 2008 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] One Touch Recording

I had this problem before...the following appeared in a previous post...
For some reasons, the "*" and "1" must be pressed pretty quickly together on 
the Polycom phone before it can be transmitted successfully to Asterisk.
Does anyone know if that can be tuned?
Sure... go to features.conf, and change the value of the featuredigittimeout 
option.
Hope this helps.


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, 
David C
Sent: Monday, 7 April 2008 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] One Touch Recording

Hi All,

For some reason One Touch Recoding does not work over ZAP but it does work when 
I call another extension.  Both Dial commands have the W option for the calling 
party to enable recording.

Does anyone know why it works internally but not over ZAP.  I have a TE110P 
card on an E1 connection. 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] One Touch Recording

[Klaverstyn, David C]

Hi All,

 

For some reason One Touch Recoding does not work over ZAP but it does
work when I call another extension.  Both Dial commands have the W
option for the calling party to enable recording.

 

Does anyone know why it works internally but not over ZAP.  I have a
TE110P card on an E1 connection. 

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Alex Balashov]

Michael Cargile wrote:

> Another reason I am sure that Digium has not released a DS3 TDM card is 
> the fact that asterisk currently cannot handle that many channels. I am 
> speaking from experience on this. We have build before a predictive 
> dialer with 16 PRIs. In order to do this and not have audio quality 
> issues we had to use an 8 core Intel Xeon server with 16 gigs of ram, a 
> 6 drive RAID 10, and two octal echo canceling Sangoma cards. This also 
> required numerous OS tweaks and dial plan optimizations. The amount of 
> time spend on this was not worth the final product.

I hope some optimisations were made to the ViciDIAL code as well, where 
surely the greatest efficiency gains are to be reaped.

> In the mean time, if someone really needs to handle that many channels I 
> would suggest purchasing a DS3 to T1 mux and pass the T1s onto mutliple 
> Asterisk servers setup in a cluster. In the end you will end up spending 
> far less money and time setting the system up. I also saw recently at a 
> trade show a DS3 to SIP converter which might also lower the cost as you 
> would not need T1 cards. The only issue is that they are a some what new 
> technology where as DS3 to T1 muxes have been around for some years now 
> and can be found on ebay for around 700 dollars.

Good ISDN gateways with DS3 interfaces that can output a whole DS3 of 
VoIP channels are hard to come by.  A Cisco AS5400 claims to, and has 
the DSP density for it, but between its processing power and its TDM bus 
cannot handle much more than about half of that.  Of the Cisco media 
gateway line, as AS5850 is probably your best bet.

A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay.  Apex 
too, perhaps.  Haven't tried to see how much it can handle when TDM->RTP 
translation is required.

The cluster idea you suggest is probably most economical, but it sure 
does require a lot of redundant T1 interfaces.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Touch Recording

[Steve Totaro]

On Mon, Apr 7, 2008 at 1:49 AM, Klaverstyn, David C
<[EMAIL PROTECTED]> wrote:
> Thanks for that.  I have the timeout set to 3000 ms and I have been pressing 
> the *1 within 500 ms so I don't think it is related to that.  As I can do it 
> over SIP but not ZAP does not make much sense to me.
>
>
>  -Original Message-
>  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John 
> (Sydney)
>  Sent: Monday, 7 April 2008 3:38 PM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>
> Subject: Re: [asterisk-users] One Touch Recording
>
>  I had this problem before...the following appeared in a previous post...
>  For some reasons, the "*" and "1" must be pressed pretty quickly together on 
> the Polycom phone before it can be transmitted successfully to Asterisk.
>  Does anyone know if that can be tuned?
>  Sure... go to features.conf, and change the value of the featuredigittimeout 
> option.
>  Hope this helps.
>
>  
>  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, 
> David C
>  Sent: Monday, 7 April 2008 3:28 PM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  Subject: [asterisk-users] One Touch Recording
>
>  Hi All,
>
>  For some reason One Touch Recoding does not work over ZAP but it does work 
> when I call another extension.  Both Dial commands have the W option for the 
> calling party to enable recording.
>
>  Does anyone know why it works internally but not over ZAP.  I have a TE110P 
> card on an E1 connection.
>
>

This page should help.  http://www.voip-info.org/wiki/view/Asterisk+DTMF

Try changing the length of the DTMF and also try changing the format
such as info, rfc2933, inband, auto.  See if any of those help.  Do
them one at a time and test so if it gets fixed, you know what the
exact problem was.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Steve Totaro]

On Mon, Apr 7, 2008 at 2:01 AM, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Michael Cargile wrote:
>
>  > Another reason I am sure that Digium has not released a DS3 TDM card is
>  > the fact that asterisk currently cannot handle that many channels. I am
>  > speaking from experience on this. We have build before a predictive
>  > dialer with 16 PRIs. In order to do this and not have audio quality
>  > issues we had to use an 8 core Intel Xeon server with 16 gigs of ram, a
>  > 6 drive RAID 10, and two octal echo canceling Sangoma cards. This also
>  > required numerous OS tweaks and dial plan optimizations. The amount of
>  > time spend on this was not worth the final product.
>
>  I hope some optimisations were made to the ViciDIAL code as well, where
>  surely the greatest efficiency gains are to be reaped.
>
>
>  > In the mean time, if someone really needs to handle that many channels I
>  > would suggest purchasing a DS3 to T1 mux and pass the T1s onto mutliple
>  > Asterisk servers setup in a cluster. In the end you will end up spending
>  > far less money and time setting the system up. I also saw recently at a
>  > trade show a DS3 to SIP converter which might also lower the cost as you
>  > would not need T1 cards. The only issue is that they are a some what new
>  > technology where as DS3 to T1 muxes have been around for some years now
>  > and can be found on ebay for around 700 dollars.
>
>  Good ISDN gateways with DS3 interfaces that can output a whole DS3 of
>  VoIP channels are hard to come by.  A Cisco AS5400 claims to, and has
>  the DSP density for it, but between its processing power and its TDM bus
>  cannot handle much more than about half of that.  Of the Cisco media
>  gateway line, as AS5850 is probably your best bet.
>
>  A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay.  Apex
>  too, perhaps.  Haven't tried to see how much it can handle when TDM->RTP
>  translation is required.
>
>  The cluster idea you suggest is probably most economical, but it sure
>  does require a lot of redundant T1 interfaces.
>
>  --
>  Alex Balashov
>  Evariste Systems
>  Web: http://www.evaristesys.com/
>  Tel: (+1) (678) 954-0670
>  Direct : (+1) (678) 954-0671
>  Mobile : (+1) (706) 338-8599
>

A T3 MUXed into 28 T1 PRIs in one, or a few trunk groups inherently
has redundancy.  If a box dies, the calls are dropped (unless you are
doing reinvite) and any call backs go right to the
Ts that are not in alarm.

Running stripped down Linux OS boxen with quad port T1 cards and four
or five lines in extensions.conf, no unneeded modules or software
loaded, asterisk 1.2, entries for zaptel and zapata, and a couple
entries in sip.conf builds a worry free solution.  Asterisk uptime 2
years, system uptime two years and twenty minutes.

Thanks,
Steve Totaro

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where is the Digium DS3 card?

[Alex Balashov]

Steve Totaro wrote:

> A T3 MUXed into 28 T1 PRIs in one, or a few trunk groups inherently
> has redundancy.  If a box dies, the calls are dropped (unless you are
> doing reinvite) and any call backs go right to the
> Ts that are not in alarm.

True - and if you're simply using CT3 as an economical method of getting 
say, a dozen T1s into a gateway, that is probably an advantage.  But if 
that's the case, it would not be cost-effective to shell out extra money 
for additional PCs with quad T1 cards just to provide failover in the 
event that the primaries fail.  80/20 rule and all that.

The point is that most people that want a DS3 interface really do want 
to pump in a DS3's worth of calls, more or less, in which case they 
really can't afford to have those DS1s going spare just for redundancy's 
sake.  And if you are doing substantially less than a DS3's worth of 
calls, you probably shouldn't be looking at a DS3 interface to begin 
with unless that's just an incredibly lucrative way to get channelised 
PRIs in from a vendor - and with typical the cost of UNE DS3 loops vs 
T1s, that's not necessarily so.

Secondly, an industrial-grade ISDN media gateway designed for telco 
environments (like a Cisco AS, say) isn't going to go down frequently 
enough to merit this kind of concern.  Don't get me wrong, I am the last 
to go on record saying that Cisco voice equipment (or any other) doesn't 
fail from time to time -- ha.  But, again, 80/20 rule.  A PC is much 
more likely to fail within the same MTBF.

So yes, a single gateway handling a DS3 can go down.  But so can an M13 
mux.  You've got single points of failure either way.

If one is in the sort of environment where such high availability really 
is a concern (typically a telco setting), one probably needs to invest 
in a big DACS and redundant, protection-switched DS3 paths (and 
protection line cards for them on the DACS side) as well as redundant 
gateways, or at least redundant DS3 line cards in the chassis.  At that 
point of stringent availability, this discussion becomes a wee bit moot 
because most likely you would not be using Asterisk and PCs in such a 
setting anyway.

> Running stripped down Linux OS boxen with quad port T1 cards and four
> or five lines in extensions.conf, no unneeded modules or software
> loaded, asterisk 1.2, entries for zaptel and zapata, and a couple
> entries in sip.conf builds a worry free solution.  Asterisk uptime 2
> years, system uptime two years and twenty minutes.

Yes, but total cost of ownership goes up because you need someone to do 
all that, and even so, despite the impressive uptime you mention, PCs do 
need a lot more maintenance, upkeep and worry.

With dedicated media gateways, you just plug in, set up and it works.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users