Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g

2007-10-03 Thread Walt Joyce
For another tone frequency for the outside dialtone, try putting this
value "[EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL 
PROTECTED];*(.4/0/1),10(*/0/2+3)" in the Outside 
Dialtone field. It will give you a slight pause followed by a different
dialtone frequency. On a Linksys/Siprua 941, that would be at the top
of the Regional page.

However, you won't hear any secondary dialtone unless you put a comma
after EVERY initial '9' in the dialplan string for each line in use.
On a 941, that would be at the bottom of the Ext 1 and Ext 2 pages of 
the web interface. I suggest the dialplan string of:
(*xx|[1-7]xx|9,[3469]11|98|99|9,[2-9]xx|9,11|9,[2-9]xx|9,1[2-9]xx[2-9]xx|9,011xxx.)

- Walt Joyce


Eric "ManxPower" Wieling wrote:
> I can't help you with that.  I only wanted to point out that ignoreopat 
> is not what you need.
> 
> On Polycom SIP phones you continue dialtone by placing a , in the 
> phone's dialplan.  SIP phones have their own internal dialplan that is 
> not part of Asterisk's dialplan.  You would have to check the docs for 
> your phone.  Not all SIP phones can continue dialtone.
> 
> bilal ghayyad wrote:
> 
>>I need to select a line from the Zap group channel
>>using the SIP Phone (not FXO and not FXS ports).
>>
>>ignorepat does not work?
>>
>>Also, what is the method to let the second dial tone
>>has another tone frequency?
>>
>>Regards
>>Bilal
>>
>>
>>No, ignorepat is for FXS ports (FXS ports use FXO
>>signaling).  Also, 
>>ignorepat does not apply to SIP phones, because SIP
>>phones provide
>> their 
>>own dialtone, not a dialtone provided by Asterisk.
>>
>>Al lists wrote:
>>
>>>Correction, on FXO port not FXS,
>>>second, read his email first:
>>>"Also, how it will be possible to assign an
>>
>>dedicated
>>
>>>line (connected to FXO) to an
>>>button on the Polycom IP Phone or Broadtel IP Phone,
>>>so if user select that button
>>>then he will be sure that his outside call will be
>>
>>via
>>
>>>that specific line."
>>>Just assign a key on your phone to dial that
>>
>>extension, and you will
>> have
>>
>>>dial tone on selected line,
>>>then as a traditional PBX you can send any digits to
>>
>>your provider.
>>
>>>On 10/1/07, Eric ManxPower Wieling <[EMAIL PROTECTED]>
>>
>>wrote:
>>
>>>>ignorepat continues dialtone after a leading digit
>>
>>has been dialed
>> on
>>
>>>>FXS ports.  How does ignorepat help this guy?
>>>>
>>>>Al lists wrote:
>>>>
>>>>>ignorpat is your friend
>>>>>
>>>>>On 9/30/07, Tzafrir Cohen
>>
>><[EMAIL PROTECTED]> wrote:
>>
>>>>>>On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
>>
>>ghayyad wrote:
>>
>>>>>>>Dear List;
>>>>>>>
>>>>>>>How can I place a call via Zap/g1 (group) but
>>
>>need to
>>
>>>>>>>determine the line (FXO port)
>>>>>>>that will go via it?
>>>>>>
>>>>>>Simply don't use groups. Use channels directly.
>>
>>To dial via the
>>
>>>>specific
>>>>
>>>>>>Zaptel channel NN, use Zap/NN
>>>>>>
>>>>>>Am I missing anything?
> 
> 
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Re: [asterisk-users] SIP Panel?

2007-10-02 Thread Walt Joyce
Matt -

I'd like the sourcecode for the SIP panel.

- Walt

Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Terry Giufre-Sweetser wrote:
> 
>>Dear List,
>>
>>Has anyone found or written a status panel application, windows or 
>>linux, that uses SIP notifies and subscriptions, to gather the status of 
>>SIP extensions from Asterisk?
>>
>>And displsy nicely on a GUI?
> 
> 
> I wrote a program a while ago - don't know if it will still work:
> 
> http://www.sineapps.com/sinepeers.php
> 
> Let me know if you want the sourcecode, it's probably buried somewhere
> in my svn repository.
> 
> - --
> Kind Regards,
> 
> Matt Riddell
> Director
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Re: [asterisk-users] SIP Panel?

2007-09-26 Thread Walt Joyce
Yes, I have. It is not difficult. I use the Asterisk Manager interface.
Is there a particular question?

- Walt

Terry Giufre-Sweetser wrote:
> Dear List,
> 
> Has anyone found or written a status panel application, windows or 
> linux, that uses SIP notifies and subscriptions, to gather the status of 
> SIP extensions from Asterisk?
> 
> And displsy nicely on a GUI?
> 

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