Re: [asterisk-users] Forward DID to another server

[ALEX BALASHOV]

In a manner of speaking.

DHAVAL INDRODIYA wrote:

> is this answer of my question?
> 
> i cant understand?
> regards
> Dhaval
> 
> On Mon, Nov 2, 2009 at 12:25 PM, Alex Balashov 
> mailto:abalas...@evaristesys.com>> wrote:
> 
> Dhaval,
> 
> Why is your name capitalised?
> 
> DHAVAL INDRODIYA wrote:
> 
>  > hello all,
>  >
>  > i have 2 asterisk boxes on that 1 have public IP Address and
> another is
>  > only have local IP address
>  >
>  > now on public IP there are some 7 DID  forwarded , now i want to
> forward
>  > 3 DID out of 7 DID to
>  >
>  > local machine we called server B , I know there are DIal , and Switch
>  > statement in asterisk ,
>  >
>  > but is there any other convenient way to do this. because if call
> ratio
>  > is high then my call legs become
>  >
>  > very long.
>  >
>  > regards
>  > Dhaval
>  >
>  >
>  >
> 
>  >
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>  >
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> 
> 
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
> 
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> 
> 
> 
> 
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-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Forward DID to another server

[DHAVAL INDRODIYA]

is this answer of my question?

i cant understand?
regards
Dhaval

On Mon, Nov 2, 2009 at 12:25 PM, Alex Balashov wrote:

> Dhaval,
>
> Why is your name capitalised?
>
> DHAVAL INDRODIYA wrote:
>
> > hello all,
> >
> > i have 2 asterisk boxes on that 1 have public IP Address and another is
> > only have local IP address
> >
> > now on public IP there are some 7 DID  forwarded , now i want to forward
> > 3 DID out of 7 DID to
> >
> > local machine we called server B , I know there are DIal , and Switch
> > statement in asterisk ,
> >
> > but is there any other convenient way to do this. because if call ratio
> > is high then my call legs become
> >
> > very long.
> >
> > regards
> > Dhaval
> >
> >
> > 
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Async Agi problem

[Robert Bielik]

Ok, now pretty much everything is up 'n running, however when I try to send an 
ANSWER (or any) command to *, it replies with
org.asteriskjava.manager.response.ManagerError "Permission Denied". In 
manager.conf for the *-java client, I have

read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi
write = system,call,agent,user,config,command,reporting,originate

* is 1.6.1.4 and *-java is 1.0.0

Ideas?
TIA
/Rob




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Re: [asterisk-users] Forward DID to another server

[Alex Balashov]

Dhaval,

Why is your name capitalised?

DHAVAL INDRODIYA wrote:

> hello all,
> 
> i have 2 asterisk boxes on that 1 have public IP Address and another is 
> only have local IP address
> 
> now on public IP there are some 7 DID  forwarded , now i want to forward 
> 3 DID out of 7 DID to
> 
> local machine we called server B , I know there are DIal , and Switch 
> statement in asterisk ,
> 
> but is there any other convenient way to do this. because if call ratio 
> is high then my call legs become
> 
> very long.
> 
> regards
> Dhaval
> 
> 
> 
> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users


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

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[asterisk-users] Forward DID to another server

[DHAVAL INDRODIYA]

hello all,

i have 2 asterisk boxes on that 1 have public IP Address and another is only
have local IP address

now on public IP there are some 7 DID  forwarded , now i want to forward 3
DID out of 7 DID to

local machine we called server B , I know there are DIal , and Switch
statement in asterisk ,

but is there any other convenient way to do this. because if call ratio is
high then my call legs become

very long.

regards
Dhaval
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Re: [asterisk-users] IVR

[Samuel Nair]

Try running your asterisk service with the -vvvc option or connect to it 
via the -r option, and then try making a call that would cause it to 
land in the default context, you will see the way asterisk traverses the 
dial plan, this will give you good debug info.

sam!!

Thomas Perron wrote:
> Hi Juan,
> I have this:
>
> [default]
> ;include => stdexten
> include => big10-IVR
> include => cleveland-IVR
> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>
> You recommend I have this:
>
> [default]
> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>
> I tried this and it does not seem to work.
> Other thoughts?
> Where located please?
>
>
>
> 2009/11/1 "Juan E. Rodríguez"  >
>
> As I see here, you do not have to include the big10 context inside
> the default context, as you have an extension defined to reach
> that context and its extention is start extension.
> If the cleveland-IVR is based on the start extension too, the same
> applies.
>
> Besides that, it would work...(maybe not the way you expect... :-) )
>
> Regards,
> Juan
>
> Thomas Perron wrote:
>> Is this going to work:
>>
>> [default]
>> include => stdexten
>> include => big10-IVR
>> include => cleveland-IVR
>> exten => _17035745353,1,Goto(big10-IVR,s,1)
>> exten => _15672528431,1,Goto(cleveland-IVR,s,1)
>>
>>
>> [big10-IVR]
>> exten => s,1,Answer()
>> exten => s,n,Background(dir-welcome)
>> ;exten => s,n,WaitExten(1)
>> ;exten => s,n,Background(astcc-please-enter-your)
>> 
>>
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>
> 
>
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Re: [asterisk-users] IVR

[Thomas Perron]

Hi Juan,
I have this:

[default]
;include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _1703XXX,1,Goto(big10-IVR,s,1)
exten => _1567XXX,1,Goto(cleveland-IVR,s,1)

You recommend I have this:

[default]
exten => _1703XXX,1,Goto(big10-IVR,s,1)
exten => _1567XXX,1,Goto(cleveland-IVR,s,1)

I tried this and it does not seem to work.
Other thoughts?
Where located please?



2009/11/1 "Juan E. Rodríguez" 

>  As I see here, you do not have to include the big10 context inside the
> default context, as you have an extension defined to reach that context and
> its extention is start extension.
> If the cleveland-IVR is based on the start extension too, the same applies.
>
>
> Besides that, it would work...(maybe not the way you expect... :-) )
>
> Regards,
> Juan
>
> Thomas Perron wrote:
>
> Is this going to work:
>
> [default]
> include => stdexten
> include => big10-IVR
> include => cleveland-IVR
> exten => _17035745353,1,Goto(big10-IVR,s,1)
> exten => _15672528431,1,Goto(cleveland-IVR,s,1)
>
>
> [big10-IVR]
> exten => s,1,Answer()
> exten => s,n,Background(dir-welcome)
> ;exten => s,n,WaitExten(1)
> ;exten => s,n,Background(astcc-please-enter-your)
>
> --
>
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>
>
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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[Joseph]

On 11/01/09 13:01, John Novack wrote:

[snip]

>Isn't the IAXy a discontinued product?
>The web page is a 3rd party solution to provision the IAXy, if my memory
>isn't completely shot, and isn't built into the IAXy
>
>John Novack

It looks like it is discontinued.  When did they discontinued it?  It was a 
nice little device, I don't like the internal PC cards they are getting 
obsolete 
as fast as the PC bus is getting obsolete; besides if something goes wrong, how 
long does it take to remove internal card and put it into another PC and 
configure it.

Yes, it was the third party web-page, is gone too. 

One day IAX will be replaced by SIP :-(

-- 
Joseph

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Re: [asterisk-users] IVR

[Juan E. Rodríguez]

As I see here, you do not have to include the big10 context inside the
default context, as you have an extension defined to reach that context
and its extention is start extension.
If the cleveland-IVR is based on the start extension too, the same
applies. 

Besides that, it would work...(maybe not the way you expect... :-) )

Regards,
Juan

Thomas Perron wrote:
Is this going to work:
  
[default]
include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _17035745353,1,Goto(big10-IVR,s,1)
exten => _15672528431,1,Goto(cleveland-IVR,s,1)
  
  
[big10-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
;exten => s,n,WaitExten(1)
;exten => s,n,Background(astcc-please-enter-your)
  

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Re: [asterisk-users] include statements in IVR

[Thomas Perron]

ok
thanks


On Sun, Nov 1, 2009 at 11:16 PM, Peter  wrote:

> Check the channel driver configuration file, or fire up CLI  with max
> verbosity and monitor its output while calling the dialplan
> extensions. CLI is like a good friend that tells you whats going on
> and if there are any errors in you configuration.
>
> Peter
>
> On Nov 2, 2009, at 4:39 AM, Thomas Perron wrote:
>
> > How do I check
> >
> > On 11/1/09, Peter  wrote:
> >> Try removing the include statements from the default context and see
> >> what happens. Also double check to make sure calls are sent to the
> >> default context.
> >>
> >> Peter
> >>
> >> On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
> >>
> >>> I want to match specific contexts to menus.
> >>> If users dial a number (example:  1703444) then start with
> >>> context big10-IVR
> >>> If users dial a number (example:  1567444) then start with
> >>> context cleveland-IVR
> >>> It is not working.  I have played with the include statements and am
> >>> close but no cigar.
> >>>
> >>> Here is a part of my config.  Please send comments.  Thank you
> >>>
> >>>
> >>> [default]
> >>> ;include => stdexten
> >>> include => big10-IVR
> >>> include => cleveland-IVR
> >>> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> >>> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
> >>>
> >>>
> >>> [big10-IVR]
> >>> exten => s,1,Answer()
> >>> exten => s,n,Background(dir-welcome)
> >>> ;exten => s,n,WaitExten(1)
> >>> ;exten => s,n,Background(astcc-please-enter-your)
> >>> ;exten => s,n,Background(zip-code)
> >>> ;exten => s,n,Wait(7)
> >>> exten => s,n,Background(washington-dc)
> >>> ;exten => s,n,Authenticate(,a)
> >>> ;exten => s,n,Background(pin-number-accepted)
> >>> exten => s,n,Playback(queue-thankyou)
> >>> exten => s,n,Background(ginger110109)
> >>>
> >>> [cleveland-IVR]
> >>> exten => s,1,Answer()
> >>> exten => s,n,Background(dir-welcome)
> >>> exten => s,n,WaitExten(1)
> >>> exten => s,n,Background(astcc-please-enter-your)
> >>> exten => s,n,Background(zip-code)
> >>> exten => s,n,Wait(7)
> >>> exten => s,n,Background(washington-dc)
> >>> exten => s,n,Authenticate(,a)
> >>> exten => s,n,Background(pin-number-accepted)
> >>> exten => s,n,Playback(queue-thankyou)
> >>> exten => s,n,Background(ginger110109)
> >>> exten => s,n,Hangup()
> >>>
> >>> ___
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> >>> --
> >>>
> >>> asterisk-users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
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> >>
> >
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Re: [asterisk-users] include statements in IVR

[Peter]

Check the channel driver configuration file, or fire up CLI  with max  
verbosity and monitor its output while calling the dialplan  
extensions. CLI is like a good friend that tells you whats going on  
and if there are any errors in you configuration.

Peter

On Nov 2, 2009, at 4:39 AM, Thomas Perron wrote:

> How do I check
>
> On 11/1/09, Peter  wrote:
>> Try removing the include statements from the default context and see
>> what happens. Also double check to make sure calls are sent to the
>> default context.
>>
>> Peter
>>
>> On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
>>
>>> I want to match specific contexts to menus.
>>> If users dial a number (example:  1703444) then start with
>>> context big10-IVR
>>> If users dial a number (example:  1567444) then start with
>>> context cleveland-IVR
>>> It is not working.  I have played with the include statements and am
>>> close but no cigar.
>>>
>>> Here is a part of my config.  Please send comments.  Thank you
>>>
>>>
>>> [default]
>>> ;include => stdexten
>>> include => big10-IVR
>>> include => cleveland-IVR
>>> exten => _1703XXX,1,Goto(big10-IVR,s,1)
>>> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>>>
>>>
>>> [big10-IVR]
>>> exten => s,1,Answer()
>>> exten => s,n,Background(dir-welcome)
>>> ;exten => s,n,WaitExten(1)
>>> ;exten => s,n,Background(astcc-please-enter-your)
>>> ;exten => s,n,Background(zip-code)
>>> ;exten => s,n,Wait(7)
>>> exten => s,n,Background(washington-dc)
>>> ;exten => s,n,Authenticate(,a)
>>> ;exten => s,n,Background(pin-number-accepted)
>>> exten => s,n,Playback(queue-thankyou)
>>> exten => s,n,Background(ginger110109)
>>>
>>> [cleveland-IVR]
>>> exten => s,1,Answer()
>>> exten => s,n,Background(dir-welcome)
>>> exten => s,n,WaitExten(1)
>>> exten => s,n,Background(astcc-please-enter-your)
>>> exten => s,n,Background(zip-code)
>>> exten => s,n,Wait(7)
>>> exten => s,n,Background(washington-dc)
>>> exten => s,n,Authenticate(,a)
>>> exten => s,n,Background(pin-number-accepted)
>>> exten => s,n,Playback(queue-thankyou)
>>> exten => s,n,Background(ginger110109)
>>> exten => s,n,Hangup()
>>>
>>> ___
>>> -- 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
>>
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] include statements in IVR

[Thomas Perron]

How do I check

On 11/1/09, Peter  wrote:
> Try removing the include statements from the default context and see
> what happens. Also double check to make sure calls are sent to the
> default context.
>
> Peter
>
> On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
>
>> I want to match specific contexts to menus.
>> If users dial a number (example:  1703444) then start with
>> context big10-IVR
>> If users dial a number (example:  1567444) then start with
>> context cleveland-IVR
>> It is not working.  I have played with the include statements and am
>> close but no cigar.
>>
>> Here is a part of my config.  Please send comments.  Thank you
>>
>>
>> [default]
>> ;include => stdexten
>> include => big10-IVR
>> include => cleveland-IVR
>> exten => _1703XXX,1,Goto(big10-IVR,s,1)
>> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>>
>>
>> [big10-IVR]
>> exten => s,1,Answer()
>> exten => s,n,Background(dir-welcome)
>> ;exten => s,n,WaitExten(1)
>> ;exten => s,n,Background(astcc-please-enter-your)
>> ;exten => s,n,Background(zip-code)
>> ;exten => s,n,Wait(7)
>> exten => s,n,Background(washington-dc)
>> ;exten => s,n,Authenticate(,a)
>> ;exten => s,n,Background(pin-number-accepted)
>> exten => s,n,Playback(queue-thankyou)
>> exten => s,n,Background(ginger110109)
>>
>> [cleveland-IVR]
>> exten => s,1,Answer()
>> exten => s,n,Background(dir-welcome)
>> exten => s,n,WaitExten(1)
>> exten => s,n,Background(astcc-please-enter-your)
>> exten => s,n,Background(zip-code)
>> exten => s,n,Wait(7)
>> exten => s,n,Background(washington-dc)
>> exten => s,n,Authenticate(,a)
>> exten => s,n,Background(pin-number-accepted)
>> exten => s,n,Playback(queue-thankyou)
>> exten => s,n,Background(ginger110109)
>> exten => s,n,Hangup()
>>
>> ___
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] include statements in IVR

[Peter]

Try removing the include statements from the default context and see  
what happens. Also double check to make sure calls are sent to the  
default context.

Peter

On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:

> I want to match specific contexts to menus.
> If users dial a number (example:  1703444) then start with  
> context big10-IVR
> If users dial a number (example:  1567444) then start with  
> context cleveland-IVR
> It is not working.  I have played with the include statements and am  
> close but no cigar.
>
> Here is a part of my config.  Please send comments.  Thank you
>
>
> [default]
> ;include => stdexten
> include => big10-IVR
> include => cleveland-IVR
> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>
>
> [big10-IVR]
> exten => s,1,Answer()
> exten => s,n,Background(dir-welcome)
> ;exten => s,n,WaitExten(1)
> ;exten => s,n,Background(astcc-please-enter-your)
> ;exten => s,n,Background(zip-code)
> ;exten => s,n,Wait(7)
> exten => s,n,Background(washington-dc)
> ;exten => s,n,Authenticate(,a)
> ;exten => s,n,Background(pin-number-accepted)
> exten => s,n,Playback(queue-thankyou)
> exten => s,n,Background(ginger110109)
>
> [cleveland-IVR]
> exten => s,1,Answer()
> exten => s,n,Background(dir-welcome)
> exten => s,n,WaitExten(1)
> exten => s,n,Background(astcc-please-enter-your)
> exten => s,n,Background(zip-code)
> exten => s,n,Wait(7)
> exten => s,n,Background(washington-dc)
> exten => s,n,Authenticate(,a)
> exten => s,n,Background(pin-number-accepted)
> exten => s,n,Playback(queue-thankyou)
> exten => s,n,Background(ginger110109)
> exten => s,n,Hangup()
>
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[asterisk-users] include statements in IVR

[Thomas Perron]

I want to match specific contexts to menus.
If users dial a number (example:  1703444) then start with context
big10-IVR
If users dial a number (example:  1567444) then start with context
cleveland-IVR
It is not working.  I have played with the include statements and am close
but no cigar.

Here is a part of my config.  Please send comments.  Thank you


[default]
;include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _1703XXX,1,Goto(big10-IVR,s,1)
exten => _1567XXX,1,Goto(cleveland-IVR,s,1)


[big10-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
;exten => s,n,WaitExten(1)
;exten => s,n,Background(astcc-please-enter-your)
;exten => s,n,Background(zip-code)
;exten => s,n,Wait(7)
exten => s,n,Background(washington-dc)
;exten => s,n,Authenticate(,a)
;exten => s,n,Background(pin-number-accepted)
exten => s,n,Playback(queue-thankyou)
exten => s,n,Background(ginger110109)

[cleveland-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
exten => s,n,WaitExten(1)
exten => s,n,Background(astcc-please-enter-your)
exten => s,n,Background(zip-code)
exten => s,n,Wait(7)
exten => s,n,Background(washington-dc)
exten => s,n,Authenticate(,a)
exten => s,n,Background(pin-number-accepted)
exten => s,n,Playback(queue-thankyou)
exten => s,n,Background(ginger110109)
exten => s,n,Hangup()
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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[John Novack]

Joseph wrote:
> On 10/30/09 12:55, Vincent wrote:
>   
>> Hello
>>
>> Since SIP/RTP is a pain to use with road warriors who need to connect
>> 
> >from any location over the Internet, I'd like to get them some IAX
>   
>> phones instead.
>>
>> For those of you using this protocol instead of SIP, what would you
>> recommend as IAX hardphones and Windows (and ideally Mac) softphones?
>> 
>
> How about Digium iaxy adapter, I've used it in the past, it register to your 
> asterisk as soon as you plug it to any network (borrow any hotels phone, plug 
> it 
> into the iaxy adapter) and you have your solution.  
> The is a web-page that will allow you to provision the adapter over the 
> Internet if you have to (don't have the link).
>
>   
Isn't the IAXy a discontinued product?
The web page is a 3rd party solution to provision the IAXy, if my memory 
isn't completely shot, and isn't built into the IAXy

John Novack

> 
>
>
>
> Checked by AVG - www.avg.com 
> Version: 8.5.423 / Virus Database: 270.14.39/2470 - Release Date: 10/30/09 
> 15:18:00
>
>   

-- 
Dog is my co-pilot


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[asterisk-users] IVR

[Thomas Perron]

Is this going to work:

[default]
include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _17035745353,1,Goto(big10-IVR,s,1)
exten => _15672528431,1,Goto(cleveland-IVR,s,1)


[big10-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
;exten => s,n,WaitExten(1)
;exten => s,n,Background(astcc-please-enter-your)
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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[Joseph]

On 11/01/09 11:04, Alexander Lopez wrote:
>What version of the IAXy are you running the ones that I have do not
>have a web interface and require IAXprov to provision?

[snip]

The one I have is using standard Linux command line for provisioning but one 
time I run onto a web-page that offered iaxy provisioning, though I can not 
find 
it anymore.
I know there is Windows XP and 2000 provisioning software. 

-- 
Joseph

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Re: [asterisk-users] pattern matching DID

[Thomas Perron]

Does anyone know of a low price SIP termination service to Nepal?  For
VoIP calling card solutIon

On 11/1/09, Aggio Alberto  wrote:
> Hi,
> it's quite straightforward: you can do your dialplan like this (default is
> the default context answered when inbound calls happen) - remember the
> underscores! -
>
>
> [default]
>
> exten => _1703,1,Goto(place-IVR,s,1)
>
> exten => _1567 ,1,Goto(place-other,s,1)
>
>
>
> [place-IVR]
>
> exten => s,1,Answer
>
> exten => s,2,Background(menu-file)
>
> exten => 1,1,Goto(submenu,1)
>
> exten => 2,1,Goto(submenu,2)
>
>  (...)
>
>
>
>
>
> [place-other]
>
> exten => s,1,Answer
>
> exten => s,n,...
>
> (...)
>
> exten => s,n,Hangup
>
> If you want to jump into a specific part of context, you should put a label
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
>
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
> Sent: domenica 1 novembre 2009 21.46
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] pattern matching DID
>
> I have two DID numbers.
> I want callers who dial 1 703  to get placed in a specific part of
> IVR
> I want other callers who dial 1 567  to get placed in a different
> area.
> How do I do this please?
>
>

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Re: [asterisk-users] pattern matching DID

[Thomas Perron]

Where is everyone located?  I am in Virginia, USA

On 11/1/09, Aggio Alberto  wrote:
> Hi,
> it's quite straightforward: you can do your dialplan like this (default is
> the default context answered when inbound calls happen) - remember the
> underscores! -
>
>
> [default]
>
> exten => _1703,1,Goto(place-IVR,s,1)
>
> exten => _1567 ,1,Goto(place-other,s,1)
>
>
>
> [place-IVR]
>
> exten => s,1,Answer
>
> exten => s,2,Background(menu-file)
>
> exten => 1,1,Goto(submenu,1)
>
> exten => 2,1,Goto(submenu,2)
>
>  (...)
>
>
>
>
>
> [place-other]
>
> exten => s,1,Answer
>
> exten => s,n,...
>
> (...)
>
> exten => s,n,Hangup
>
> If you want to jump into a specific part of context, you should put a label
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
>
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
> Sent: domenica 1 novembre 2009 21.46
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] pattern matching DID
>
> I have two DID numbers.
> I want callers who dial 1 703  to get placed in a specific part of
> IVR
> I want other callers who dial 1 567  to get placed in a different
> area.
> How do I do this please?
>
>

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Re: [asterisk-users] pattern matching DID

[Thomas Perron]

Thank you.
I am trying it shortly.  This is a lot of fun.  I am trying to find
places where I can get customers with IVR or anything relating to
Asterisk.  Any ideas?
Cheers
Tom

On 11/1/09, Aggio Alberto  wrote:
> Hi,
> it's quite straightforward: you can do your dialplan like this (default is
> the default context answered when inbound calls happen) - remember the
> underscores! -
>
>
> [default]
>
> exten => _1703,1,Goto(place-IVR,s,1)
>
> exten => _1567 ,1,Goto(place-other,s,1)
>
>
>
> [place-IVR]
>
> exten => s,1,Answer
>
> exten => s,2,Background(menu-file)
>
> exten => 1,1,Goto(submenu,1)
>
> exten => 2,1,Goto(submenu,2)
>
>  (...)
>
>
>
>
>
> [place-other]
>
> exten => s,1,Answer
>
> exten => s,n,...
>
> (...)
>
> exten => s,n,Hangup
>
> If you want to jump into a specific part of context, you should put a label
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
>
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
> Sent: domenica 1 novembre 2009 21.46
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] pattern matching DID
>
> I have two DID numbers.
> I want callers who dial 1 703  to get placed in a specific part of
> IVR
> I want other callers who dial 1 567  to get placed in a different
> area.
> How do I do this please?
>
>

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Re: [asterisk-users] pattern matching DID

[Aggio Alberto]

Hi,
it's quite straightforward: you can do your dialplan like this (default is the 
default context answered when inbound calls happen) - remember the underscores! 
-


[default]

exten => _1703,1,Goto(place-IVR,s,1)

exten => _1567 ,1,Goto(place-other,s,1)



[place-IVR]

exten => s,1,Answer

exten => s,2,Background(menu-file)

exten => 1,1,Goto(submenu,1)

exten => 2,1,Goto(submenu,2)

 (...)





[place-other]

exten => s,1,Answer

exten => s,n,...

(...)

exten => s,n,Hangup

If you want to jump into a specific part of context, you should put a label 
near the 'n' priority where you want to jump to (eg. exten => 
s,n(jumphere),) then specify that label into Goto() 
application.

Cheers,
//Al.




From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: domenica 1 novembre 2009 21.46
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pattern matching DID

I have two DID numbers.
I want callers who dial 1 703  to get placed in a specific part of IVR
I want other callers who dial 1 567  to get placed in a different area.
How do I do this please?

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[asterisk-users] Dialstatus

[Joseph]

I can not seem to get dial status to work,
in sip.conf I have: qualify=yes

simple plan:
exten => 51,1,Dial(SIP/11,20,r)
exten => 51,n,Goto(s-${DIALSTATUS},1)
exten => s-Busy,1,Hangup()
exten => s-Answer,1,Macro(atb)

I'm dialing from exten.11 to exten.11 so I get busy signal and the channel 
should hangup but it doesn't

How to print/display dialstatus? I'm using ATA

-- 
#Joseph

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[asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension

[covici]

Hi.  When I dial a Dahdi extension using asterisk 1.6.0, and there is no
answer, the extension hangs up, but the dial status is busy instead of
no answer.  How do I get this to work -- do I need to update dahdi?  The
card is an X400p using its FXS module.

Thanks in advance for any ideas on this.

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

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

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[asterisk-users] pattern matching DID

[Thomas Perron]

I have two DID numbers.
I want callers who dial 1 703  to get placed in a specific part of
IVR
I want other callers who dial 1 567  to get placed in a different
area.
How do I do this please?
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[asterisk-users] PSTN Lines and AEX808B

[Torintino T]

I
have a setup of Asterisk 1.2.28 and Digium AEX808B (8 FXOs), I connected the
PSTN lines to the Digium card, everything is working fine but the issue is that
when I make calls through the PSTN lines, some of calls get out successfully and
the other give me a different dummy long ring back tone for about 12 seconds
until it hangs up.


I
have taken the server with the card to another location in the same city, the
same PSTN Company, but different exchange, everything worked fine, as well I 
tested
it with a Prima Cell, and worked fine,


I
want to find the issue to try to sort it out.


Any
help will be highly appreciated.
The following is  zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en


context=from-pstn
;signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
busydetect=yes
busycount=3
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=8.0
txgain=2.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

group=1

;Include AMP configs
#include zapata_additional.conf

Thanks

  
_
Windows Live: Keep your friends up to date with what you do online.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010___
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Re: [asterisk-users] Tutorial for SIP user

[Thomas Perron]

I am having the same issue.
Please assist.


On Sun, Nov 1, 2009 at 1:27 PM, giancarlo lombardo
wrote:

> Dear all,
> I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
> trouble, I see on XLITE console:
>
> Registration Error: 503 - Service unavailable.
> Someone have a tutorial or a step by step description how to do that ?
>
> Thanks in advance
>
> --
> Giancarlo Lombardo
>
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Re: [asterisk-users] Tutorial for SIP user

[Farooq Hussain]

Dear Giancarlo,

On which OS your are installing XLITE. If you are trying to connect XLITE
using Winodws XP please make a entry in your firewall. I think that would
solve your problem

On Sun, Nov 1, 2009 at 10:27 AM, giancarlo lombardo  wrote:

> Dear all,
> I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
> trouble, I see on XLITE console:
>
> Registration Error: 503 - Service unavailable.
> Someone have a tutorial or a step by step description how to do that ?
>
> Thanks in advance
>
> --
> Giancarlo Lombardo
>
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-- 
Thanks

Farooq Hussain
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[asterisk-users] Execute the specified macro for the called channel AFTER connecting to the calling channel.

[Joseph]

How to tell Dial to execute Macro AFTER connecting to the channel?

Dial macro definition:
Execute the specified macro for the called channel before connecting to the 
calling channel.

and I want to execute the macro after connecting (channels answers).

-- 
Joseph

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[asterisk-users] Tutorial for SIP user

[giancarlo lombardo]

Dear all,
I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
trouble, I see on XLITE console:

Registration Error: 503 - Service unavailable.
Someone have a tutorial or a step by step description how to do that ?

Thanks in advance

-- 
Giancarlo Lombardo
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Re: [asterisk-users] Error in MeetMe modules ?

[Samuel Nair]

You need to have the dadhi_dummy driver loaded, or have a digium (or 
similar) card plugged in. Meetme needs a timing source. dahdi_dummy is 
used as the timing source in case you dont have a card.

sam!!

Phibee Network Operation Center wrote:
> Hi
>
> when i use MeetMe, i have this errors:
> app_meetme.c: Unable to open pseudo device
> Where is the problems ?
>
>
> i have too warning and error into my logs:
>
> [Nov  1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open 
> pseudo channel for timing...  Sound may be choppy.
> [Nov  1 07:26:17] WARNING[18544] config.c: Realtime mapping for 
> 'iaxpeers' found to engine 'mysql', but the engine is not available
> [Nov  1 07:26:17] ERROR[18544] codec_dahdi.c: Failed to open 
> /dev/dahdi/transcode: No such file or directory
>
> What is the process for resolv this problems ?
>
> "Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine 
> is not available", i don't understand because
> realtime Sip+Extension works and cdr too
>
> Thanks for your help
>
>
>
>
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>   


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Re: [asterisk-users] usage of manager events to create custom reports

[Steve Edwards]

On Sun, 1 Nov 2009, nik600 wrote:

> due to some custom requirements we are planning to use the manager 
> events for creating some custom reports.
>
> So the call-flow in the events listener will be:
>
> 1) new event detected
> 2) check if the event has an Uniqueid information
> 3) push the event into a stack reserved for Uniqueid
> 4) if the event if Hangup write the information of the stack reserved
> fro Uniqueid and then free memory
>
> I'm planning to write this in php, i think that this code is very
> light to be run even after a lot of events because i free memory after
> the conclusion of each call.

When I think "high load" I don't think PHP, I think C.

Freeing the memory after each call may be exactly the wrong approach. 
Consider if you keep 1,000 bytes of info for each call and can handle 
1,000 simultaneous calls that's only 1MB. If you allocate this array 
statically you have no memory allocation issues and eliminate a lot of 
executing code.

> Then (on a separate server) there will be a re-processing of the file
> extracting all the information required from a call.

Why not write a row to a database (on that separate server) at the end of 
each call and eliminate this step. You will save a bunch of disk I/O on 
you Asterisk system and eliminate a whole bunch of race and failure 
conditions.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Originate with Local channel to any app-only extension hangs up immediately?

[eric weaver]

In 1.4.26, I'm experiencing a case where a manager- (or CLI-) originated
call where the channel is Local and the extension primarily runs an app
(e.g. Playback)  immediately gets up.  The extension may be rung but is
cancelled.

Anybody seen this?  Know what to do about it?
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[asterisk-users] Exchange 2007 and Voicemail with Imap Storage

[Ryan Wyse]

Hey has anyone gotten Exchange 2007 working with Imap Storage?  I know  
that Exchange 2007 didn't support the usual delegated Imap access  
until Rollup 4, but it should work now right?  I have it working fine  
with Exchange 2003.  When I do DomainName\Username\mailboxalias, I get  
authentication failures on Exchange 2007 with a username of "Username 
\mailboxalias".  If I try the alternate login of "usern...@domain/ 
mailboxalias" I get an invalid remote specification from the c-client  
library.  I'm using imap2007e.  The strange thing is that both of  
those logins appear to work fine when using the php imap_open command,  
which is also built against the c-client libraries, though they appear  
to be built against the older 2001 version.

Thanks,

Ryan Wyse

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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[Alexander Lopez]

What version of the IAXy are you running the ones that I have do not
have a web interface and require IAXprov to provision?

=> -Original Message-
=> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
=> boun...@lists.digium.com] On Behalf Of Joseph
=> Sent: Saturday, October 31, 2009 3:07 PM
=> To: Asterisk Users Mailing List - Non-Commercial Discussion
=> Subject: Re: [asterisk-users] [IAX] Recommended soft- and hardphones?
=> 
=> On 10/30/09 12:55, Vincent wrote:
=> >Hello
=> >
=> >Since SIP/RTP is a pain to use with road warriors who need to
connect
=> >from any location over the Internet, I'd like to get them some IAX
=> >phones instead.
=> >
=> >For those of you using this protocol instead of SIP, what would you
=> >recommend as IAX hardphones and Windows (and ideally Mac)
softphones?
=> 
=> How about Digium iaxy adapter, I've used it in the past, it register
=> to your asterisk as soon as you plug it to any network (borrow any
=> hotels phone, plug it
=> into the iaxy adapter) and you have your solution.
=> The is a web-page that will allow you to provision the adapter over
=> the Internet if you have to (don't have the link).
=> 
=> --
=> Joseph
=> 
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=> 
=> --
=> This message was scanned by Connected.Net and is believed to be
clean.
=> Click here to report this message as spam.
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=> 


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[asterisk-users] Skyp SIP? - what is free for a home *

[hbk]

Hi,

I get confused about all solutions for Skype!

I want to connect as simple as possible out home *  to be able to at
least answer Skype calls.
Now I use a PC USB box and a FXO, works ok both call directions but uses
a PC.

Any good and free idea ?

Thank you!
HB

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[asterisk-users] usage of manager events to create custom reports

[nik600]

Dear all

due to some custom requirements we are planning to use the manager
events for creating some custom reports.

I've enabled cdr_manager, then in manager.conf i've enabled
timestampevents = yes and in queue.conf eventmemberstatus = yes.

I know that these settings can generate a lot of manager events but
i'm planning to have a very simple application on the Asterisk server
that keep all that events from the manager socket and put them into a
separate file for each call.

To decide when to write the call file i'm planning to wait for the
Hangup event).

So the call-flow in the events listener will be:

1) new event detected
2) check if the event has an Uniqueid information
3) push the event into a stack reserved for Uniqueid
4) if the event if Hangup write the information of the stack reserved
fro Uniqueid and then free memory

I'm planning to write this in php, i think that this code is very
light to be run even after a lot of events because i free memory after
the conclusion of each call.

Then (on a separate server) there will be a re-processing of the file
extracting all the information required from a call.

I'm writing to you just to know:

- what do you think about this kind of approach
- if someone else has done something similar and wants to share his experience
- how much is affordable the events generation excpecially in system
with a high load

Thanks to all for any contribute.

Hi

-- 
/*/
nik600
http://www.kumbe.it

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[asterisk-users] Error in MeetMe modules ?

[Phibee Network Operation Center]

Hi

when i use MeetMe, i have this errors:
app_meetme.c: Unable to open pseudo device
Where is the problems ?


i have too warning and error into my logs:

[Nov  1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open 
pseudo channel for timing...  Sound may be choppy.
[Nov  1 07:26:17] WARNING[18544] config.c: Realtime mapping for 
'iaxpeers' found to engine 'mysql', but the engine is not available
[Nov  1 07:26:17] ERROR[18544] codec_dahdi.c: Failed to open 
/dev/dahdi/transcode: No such file or directory

What is the process for resolv this problems ?

"Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine 
is not available", i don't understand because
realtime Sip+Extension works and cdr too

Thanks for your help




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Re: [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?

[Phibee Network Operation Center]

Hi

ok i have understand ;=)

bye


Phibee Network Operation Center a écrit :
> Hi
>
> actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
>
> I read on the wiki:
>
> ===
> Database Config
> put the following in res_mysql.conf
>
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = myuser
> dbpass = mypass
> dbport = 3306
>
> Values in sip.conf or iax.conf like in older versions of * are no longer 
> used.
>
>
> Database Table
> Lets create the table we need:
>
> NOTE: You can use any table name you wish, just make sure the table name 
> matches what you have the family name bound to.
>
> ===
>
>
> But i don't see where i put the Table Name ? (if i don't want use 
> sip_buddies)
>
> and he have a sample of Table Structure, can i add a new champs for my 
> personnal
> software without problems ?
>
> Thanks
> jerome
>
>
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>
>   


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Re: [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?

[Samuel Nair]

You can set it up in extconfig.conf:

iaxusers => mysql,asterisk,iaxusers
iaxpeers => mysql,asterisk,iaxusers
sipusers => mysql,asterisk,sipusers
sippeers => mysql,asterisk,sipusers
voicemail => mysql,asterisk,voicemail
extensions => mysql,asterisk,extensions
queues => mysql,asterisk,queues
queue_members => mysql,asterisk,queue_members
meetme => mysql,asterisk,meetme

sam!!

Phibee Network Operation Center wrote:
> Hi
>
> actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
>
> I read on the wiki:
>
> ===
> Database Config
> put the following in res_mysql.conf
>
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = myuser
> dbpass = mypass
> dbport = 3306
>
> Values in sip.conf or iax.conf like in older versions of * are no longer 
> used.
>
>
> Database Table
> Lets create the table we need:
>
> NOTE: You can use any table name you wish, just make sure the table name 
> matches what you have the family name bound to.
>
> ===
>
>
> But i don't see where i put the Table Name ? (if i don't want use 
> sip_buddies)
>
> and he have a sample of Table Structure, can i add a new champs for my 
> personnal
> software without problems ?
>
> Thanks
> jerome
>
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   


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Re: [asterisk-users] [IAX] Recommended soft- and hardphones?

[sdcha...@gmail.com]

Vincent,

You can get hardphones from http://www.atcom.cn/

I've been using a few since long and its pretty reliable. Go for the
AT620's. They are not that costly.

Softphone with Zoiper in the free version doesnt support G723 or G729
codecs. If you need them try IAXLITE.

Just google IAXLITE, you can find it.

Cheers.

2009/10/31 Joseph :
> On 10/30/09 12:55, Vincent wrote:
>>Hello
>>
>>Since SIP/RTP is a pain to use with road warriors who need to connect
>>from any location over the Internet, I'd like to get them some IAX
>>phones instead.
>>
>>For those of you using this protocol instead of SIP, what would you
>>recommend as IAX hardphones and Windows (and ideally Mac) softphones?
>
> How about Digium iaxy adapter, I've used it in the past, it register to your 
> asterisk as soon as you plug it to any network (borrow any hotels phone, plug 
> it
> into the iaxy adapter) and you have your solution.
> The is a web-page that will allow you to provision the adapter over the 
> Internet if you have to (don't have the link).
>
> --
> Joseph
>
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-- 
Sunil Charly

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