Setting up a PBX for Israel<->US communication

[Tzafrir Cohen]

On Thu, Feb 26, 2009 at 03:21:54PM +1100, Amos Shapira wrote:
> 2009/2/26 Ori Berger <linux-il at orib.net>:
> > Some information that may be useful if anyone is still interested:
> 
> Thanks for the info.
> 
> >
> > - I can recommend grnvoip's termination service: They have good routes, good
> > rates, competent technical support. They do not officially support IAX2
> > termination (only SIP and H323), but they will provide it if asked
> > (supposedly; I'm using SIP termination). I heard great things about voipjet,
> > but apparently they now actively require you to be a non-person entity
> > (read, company) to join.
> 
> I bought a $US50 credit with grnvoip and they sent me the exact
> Asterisk configuration for my account when I asked. They also
> initiated contact when they saw that I haven't used my credit within a
> few days and offered support. That gave me an impression of good
> service(TM).
> 
> >
> > - The cheap setup described by Arik is perfect for call _routing_ so long as
> > the asterisk server is only there for routing, and can "step out" of the
> > communication chain once call routing is finished. Otherwise, at least with
> > a Xen setup on vpslink, the CPU slice is not regular enough to provide
> > acceptable quality, even for things like a voicemail app. (Everything works,
> > but sound is occasionally choppy). OpenVZ might be better; Lylix.net might
> > be better; I only have experience with Xen, and it's NOT good enough.
> 
> I setup a Xen guest with CentOS 5.2 and Asterisk 1.4.23(?) at VPSLink
> and it's generally pretty accessible from where I am.
> 
> Why do you think OpenVZ might be better? Does it have a known
> advantage in Asterisk hosting?
> 
> Googl'ing for "asterisk hosting provider" Lylix.net indeed comes up
> near the top and seem to be asterisk-centric but their cheapest plan
> of AsteriskNow is $35/month. No competition for the $8/month from
> VPSLink.
> 
> >
> > - In order to enable Asterisk to step out of call routing (and network
> > routing), the DID mapping protocol and the termination protocol must be the
> > same -- either both should be IAX2 (when using VoipJet) or both should be
> > SIP (when using grnvoip). Otherwise, asterisk will need to remain "on the
> > line" to do protocol translation.
> >
> > - Asterisk rocks! It takes a little effort to configure, and looks weird at
> > first (at least to my originally telephony-uninitiated self), but in most
> > cases, there's a good reason for the way it needs to be configured. I think
> > it's worthwhile to try to understand why Asterisk is built the way it is,
> > rather than just look for an easy to configure GUI.
> 
> I'm still struggling with Asterisk configuration. At some stage a
> worker of mine got the sample echo test to work from my workplace but
> that exhausted his Asterisk knowledge and I wasn't able to repeat that
> test from home.
> 
> I also tried to follow the Asterisk book
> (http://www.the-asterisk-book.com/unstable/) and didn't manage to do
> any dialing through even with the simplest, first configuration
> example (http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html#asterisk-konfigurieren)
> 
> But I'm actually stuck at a more basic stage - I can't get incoming
> audio on any of the software SIP clients I tried on my Ubuntu (8.10,
> i386). I tried Twinkle (recommended here for its better logging),
> Ekiga and Gizmo.

Let's start with something simpler: a call between the phone and
Asterisk itself: an echo test, playback, voicemail extension, or
whatever.

-- 
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