Red Hat’s OpenShift Adds Full Java Lifecycle Offering

Report
Question

Red Hat’s OpenShift platform as a service offering has been in public beta for a while now. It offers a fairly simple way for people to jumpstart “cloud” development efforts by abstracting out all the messy business of setting up application and database servers. Instead, you simply publish your source code to OpenShift, and their platform does the rest. Supported languages are those used heavily by nimble, agile startup types: PHP, Python, Ruby. Interestingly, OpenShift also supports Java. That’s not a language that many people associate with cloud solutions. Today, Red Hat is announcing that they’re improving their support…

Java News 3.7 years ago 0 Answers

More than one route to a destination

Report
Question

Hi, I have a setup with 5 remote offices, each having a Asterisk PBX.
I then have a central office, also with an Asterisk PBX.
The remote offices have 2 links to the central office, a large link,
and a smaller, but more reliable link.
Unfortunately, using IAX is not an option for me.
Can I use 2 SIP Trunks from each remote offices to the central site
and permit 2 simultaneous calls across the SIP trunk that passes over
the smaller line, and permit 10 simultaneous calls across the larger
link?

Asterisk Users 3.7 years ago 2 Answers

Forcing a CODEC

Report
Question

Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all
other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX
trunks, I was hoping that it would be possible to specify alaw and
ulaw as the first two CODEC choices for the SIP phones, as well as in
their sip.conf configurations, but that I could use the IAX trunks
(with bandwidth=high) to force the phones to…

Asterisk Users 3.7 years ago 4 Answers

Multiple SIP endpoint registrations

Report
Question

Hi guys,
I want to ask if its possible to make calls using one SIP account,
The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center,
so the iPhone app will be using the same SIP account for all users
lets say for example:
iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center)
Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone…

Asterisk Users 3.7 years ago 0 Answers

Goto Queue, does not work, it should play message or any thing

Report
Question

Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten => 5631040,1,Playback(WelcomeMessage)
exten => 5631040,2,Goto(OrangeCMG,s,1)
So how I can overcome this? Regards
Bilal

Asterisk Users 3.7 years ago 3 Answers