[Asterisk-Users] Asterisk <-> Gizmo

[Leif Neland]

---- Original Message ----
From: <steve at daviesfam.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com> Sent: Wednesday, December 21, 2005
9:14 AM Subject: Re: [Asterisk-Users] Asterisk <-> Skype
anywhere/anyhow?

> On Tue, 20 Dec 2005, AR Tarzi wrote:
>
>> could you please tell how it interfaces with Asterisk? Could I
>> receive calls into Asterisk? send calls out?
>> I've just downloaded it and am searching (unsuccessfully) for these
>> on Gizmo's site/software.
>
> Gizmo isn't just a soft phone.  Like Skype, its a service.  Unlike
> Skype, though, the service is open to the rest of the SIP world.
>
> So - to call your Asterisk system from Gizmo, simply tell Gizmo to
> dial extension at asterisk.server.host.name.  To call Gizmo from
> Asterisk, simply tell it to dial "SIP/gizmophonenumber at sipphone.com"
>
It 'sort of works'.

I can call from gizmo to my *, but the url for incoming is 
SIP/gizmophonenumber at proxy01.sipphone.com

DTMF from gizmo does not work

If gizmo is dialing into the queue, gizmo doesn't notice the prompts from * 
(which I can see in the *log), but keeps playing ringtones. But when the 
phone is answered, gizmo knows. and the connection is made.

(The queue works as expected, when I call from eg my cellphone to *)

So an Answer() is needed before queue().

Leif