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Oracle’s Plans for Java Unveiled at JavaOne

Oracle had lots of Java announcements at this year’s JavaOne. So far the plans include: ‘The availability of an early access version of JDK 7 for the Mac OS, plans to “bridge the gap” between Java ME and Java SE, an approach to modularizing Java SE 8 that will rely on the Jigsaw platform, a new project that aims to use HTML5 to bring Java to Apple’s iOS platform, the availability of JavaFX 2.0, a pending proposal to open source that technology, gearing up Java EE for the cloud, and a delay in the release of Java 8.'”

Source: Slashdot.org

Questions on Dahdi

I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found it
is not running, Instead it runs when service starts but I can’t find its
status when I type in service dahdi status.

I am using Asterisk 1.8.7 on centos 5.7 32 bit.


call pickup

Am 05.10.2011 20:42, schrieb Marek Cervenka:
> hello,
> is there some way to notify people in the same pickup group about call
> from caller to callee?
> i.e. i have call from 111 to 222
> there are 222,333,444 in the same pickup group
> 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup
> the call with *8
> siemens have this on their sip openstage phones. how they do this?

You can have that with subscriptions/hints, for example Snom phones
can display not only a call to one of the peers but also the caller and

This works jaw to cheek with BLF (busy lamp field) which allows to monitor
other extensions’ status (in_use, ringing…).

Of course you can be member of a pickup group without “monitoring” the
status of any of the peers, and you can monitor a peer’s status without
being in the same pickup group (although not pickup the call then,
obviously :-)



Hello list

i have one question related to meetme,i have to providers with the first one
i put the number with 9 digit 520XXXXXX and all works without issue, with
the second i put just the last 3 numbers 500 with meetme there is nothing

but when i put the last 3 numbers like below i can call my sip without any
problem, could you please inform me if the issue is related to my provider
of the issue come from asterisk

exten => 500,1,Dial(SIP/228, 30)


first provider
exten => 520XXXXXX,1,Answer
exten => 520XXXXXX,n,Wait(4)
exten => 520XXXXXX,n,Meetme
second provider

exten => 500,1,Answer
exten => 500,n,Wait(4)
exten => 500,n,Meetme

there is no meetme with this one


conf =>1234,5678

thanks and regards

Passive wait in dialplan?

Hello, everyone

Here part of my dialplan context

same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
same => n,Wait(0.2)
same => n,GoTo(COMMAND_SWITCH)
same => n,NoOp(— NOT REACHED —)

UserEvent sends blah-event via AMI to high-level UI, user makes
decision and issues some command via Action:SetVar, then dialplan
continues to work.

The problem is, in dialplan there is an active wait loop, i.e. waiting
mechanism which rapidly checks some var(consuming processor resources
and flooding logs). Is it possible to create passive waiting loop
within current abilities of Asterisk 1.8?