Oracle's Plans for Java Unveiled at JavaOne

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"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…

Java News 4 years ago 0 Answers

Questions on Dahdi

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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. CK

Asterisk Users 4 years ago 1 Answer

call pickup

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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…

Asterisk Users 4 years ago 1 Answer

meetme

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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) extensions.conf first provider
exten => 520XXXXXX,1,Answer

Asterisk Users 4 years ago 0 Answers

Passive wait in dialplan?

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Hello, everyone Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2 [blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same => n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
same => n,GoToIf($["${COMMAND}"="${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

Asterisk Users 4 years ago 0 Answers