date:20070928

Re: [asterisk-users] 3-way calling

2007-09-28 Thread Rilawich Ango

Do u mean meetme?  It is total different from my case.
In meetme, everybody need to know and dial the conference room number
to get into the conference room.  In my case, party A,B,C may not know
the conference number.  A only knows B numbers and B only knows C
numbers.

On 9/28/07, Pamela Weis [EMAIL PROTECTED] wrote:
 it is probably not what you are looking for.
 but simply use a conference room of asterisk for those 1 line phones.

 pamela

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Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

2007-09-28 Thread Olivier

Strange !
We successfully used SuperMicro boards without any IRQ problems.

What is SuperMicro's reply, concerning this IRQ problems ?
They sure have interest to solve this or at least explain why it can't be
done.

Regards
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Re: [asterisk-users] IAX configuration

2007-09-28 Thread yonoko molomo

Hi,
i think that is not the point.
the call works, what is not working is the IAX config.
somehow i need to put manually all users of the foreign asterisk
(user, password...).

if i put type=friend, it does not work in any case.
if i put type=peer it works only if i define the users of the foreign
asterisk and also an entry for the foreign server

that should not be the normal behaviour, i guess.

any ideas?
thanks



2007/9/27, Mojo with Horan  Company, LLC [EMAIL PROTECTED]:
 when using variables, use ${variablename} instead of $(variablename) --
 (squiggly braces instead of parentheses) -- I'm not sure parentheses are
 allowed.

 yonoko molomo wrote:
  Now I update the extensions.conf file accordingly.
  exten = clientA_Number,1,Dial(sip/$(exten),10)

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[asterisk-users] Conference call today at 12:30 PM EDT

2007-09-28 Thread randulo

Hey folks,

Here's your chance to report in about Astricon, ask or answer general
asterisk questions, talk about your asterisk-related (or voip-related)
projects, sites, work, anything. We interested and listening. We have
a great core group on these conferences, even though Indiana is
disproportionately represented for some reason :)

This conference is NOT limited to developers or gurus, anyone
interested in VOIP and asterisk is welcome to join anytime.

Let's talk! http://www.VoipUsersConference.org

You don't have to register now, you can call in on any phone (or via asterisk):

Call (724) 444-7444
Enter 22622# then 1# or your PIN # if you have one.

Asterisk instructions for a painless dialplan experience are here:

http://www.voipusersconference.org/asterisktalkshoecallinsetup.htm

Last but not least, Windows and Mac users can use the built in SIP
client from Talkshoe.com to call in with a single click. Batteries not
included.

rr

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[asterisk-users] Non-USASCII chars in sip.conf?

2007-09-28 Thread Per Jessen

This must have been asked before, but googling didn't help much. 
How do I define a callerid that contains non-USASCII characters? E.g. ä,
ö, ü, å, ø, æ etc. ? 


/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] call relation in call transfer

2007-09-28 Thread Rilawich Ango

Thanks.
Actually, I want to have some information about the call transfer just
like to queue_log in queue.
According to your message, there is no such mechanism to associate the
call in call transfer.  How about any variable that I can identify the
call which is made by call transfer?
As I know there is a variable ${BLINDTRANSFER} that will fill in a
value in blind transfer.  However, I can't find any variable that will
fill in a attended transfer.  Anyone can advise?


On 9/28/07, Alex Balashov [EMAIL PROTECTED] wrote:

 On Fri, 28 Sep 2007, Rilawich Ango wrote:

  In CDR, I found that there are 3 records after doing call transfer.
  However, 3 of them are individual record that is very difficult to
  identify they are related to call transfer.  My question is how to
  identify the call with a clear flow, from CDR or by other means, is a
  call transfer.

Do they have a common criterion?  If they do not have a common
 criterion, it is probably not logically possible to associate them.
 Asterisk is a back-to-back user agent, so it builds out distinct legs for
 every call with unique Call-IDs and dialogue tags.  This makes it hard to
 meaningfully associate call flows like this inherently, unless you do
 state tracking in the software to make this possible.

This has been an ongoing topic of discussion periodically on the
 Asterisk Developers' List (asterisk-dev).  It seems there is considerable
 interest in reworking the CDR engine to account for this type of situation
 more meaningfully.  You may wish to search the list archives for greater
 insight into what core developers are thinking, or to join the list and
 add your two cents to what you want to see from it.  You're definitely not
 the first person to run into this or regard it as a serious impediment. :)

 Cheers,

 --
 Alex Balashov

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Re: [asterisk-users] What's the deal with ATAcomm?

2007-09-28 Thread Doug

At 15:37 9/27/2007, Jerry Jones wrote:
I will miss them. It was nice having a local company with a few
Polycoms in stock most of the time. A month or so ago we needed some
quick and were unable to contact them, either through their toll free
or local numbers. I swung by their office last week and nocticed it
was vacant.

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