* You are viewing the archive for November 15th, 2011

Red Hat’s OpenShift Adds Full Java Lifecycle Offering

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 of Java on OpenShift with support for “full Java lifecycle for developers”.

Read More: http://techcrunch.com

More than one route to a destination

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?
I also wish to have priorities, so that more important calls are sent
over the smaller link (but more reliable) and the larger link used for
less important calls.
Can you do this priority based on the user ID of the caller?

Another question:
If a user with a SIP client starts off in remote office1, and then
moves to remote office4, can then keep the same phone number?

Kind Regards

James

Forcing a CODEC

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 use their third CODEC
choice, g722, because that would be the only CODEC specified for the
IAX trunks (following disallow=all).

Unfortunately, that doesn’t work. Although the Asterisk console
reports that g722 is being used, when I listen to the connection it’s
obvious that a G.711 CODEC is being used. Curiously, the reverse does
work: if g722 is specified as the first CODEC of choice for the
phones, it is possible to use the IAX trunks to force them to use
alaw/ulaw instead.

Is a solution to this problem?

I’m using Debian squeeze with Asterisk 1.6.2.9.

Cheers,

Jaap

Multiple SIP endpoint registrations

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 users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)?
Regards
Faraj Khasib

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

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

Calling an independent gateway from asterisk

Amar,

In general, gateways don’t register. They are simply defined as a peer and calls are routed to them in the dialplan. When I do this I usually use the local channel to get to the
dialing contexts.

Get in touch if you need a more detailed example

Bruce Ferrell

On 11/14/2011 10:01 PM, Amar Akshat wrote:
> Hello,
> I have a testing scenario at hand. I want to make a call from Asterisk
> CLI or AMI to an external network gateway. Is this possible.
> Let me explain the use case.
>
> Asterisk server (say 192.168.5.10) has few registered endpoints or softphone.
> Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060),
> listening for SIP invites, but this gateway is not registered with
> Asterisk,
>
> can I send out SIP invites (call) to this external gateway, without
> having to register on Asterisk.
>