Setting up a PBX for Israel<->US communication

[Amos Shapira]

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.

The only instance where I got incoming Audio was with Ekiga calling to
sip:500 at ekiga.net (the ekiga echo test service) which allowed me to
hear myself at about 4 second delay. Even their callback service
(sip:520 at ekiga.net) kept saying that my SIP client rejected their
callback.

I use an ADSL2+ with D-Link DSL-G604T modem/router. I also have a
Sipura ATA-3000 connected to my ISP's VoIP (SIP) service with no
problem (and no port forwarding required in the modem). I can't
configure the ATA to use other SIP servers because it supports
registration to only one SIP server at a time and I don't want to
loose access to the ISP's VoIP service.

What SIP software do others use here that works for them?
Does it support STUN? Ekiga's config droid says I need to use STUN (so
I use stun.ekiga.net).
Can anyone help me troubleshoot the audio problems?

Ultimately, I'd like to connect Nokia E71 via SIP to the asterisk
server, but that's another complexity I'll wait with until I get
Asterisk working with a software SIP client.

Thanks,

--Amos