[Freeswitch-users] mod_dingaling and google voice

[Avi Marcus]

Or get a gizmo5 account and then you can just do SIP forwarding to FS,
without the pstn loop. You can find them around for a fee bucks. I have a
few, not sure if I'm going to use them...
-Avi
On Mar 1, 2011 4:59 AM, "envelopes envelopes" <sunwood360 at gmail.com> wrote:
> I don't use GV for direct inbound, because it is not working very well.
>
> - Get an account from sipgate.com;
> - let GV forward inbound to your sipgate PSTN #
> - and your FS register as a sipgate client.
>
>
>
> On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman <
> freeswitch at cartissolutions.com> wrote:
>
>> What FS build are you running? I'd like to compare notes with you on your
>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do
>> you have that set up?
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>
>> never mind. add this line <action application="ring_ready"/> and restart
FS
>>
>> fixed the issue.
>>
>>
>>
>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <
>> sunwood360 at gmail.com> wrote:
>>
>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>> ringback or not sure whether the other party has answered the call.
>>> is there any config variable to set up so that I will be notified if my
>>> call is answered?
>>>
>>> thanks!
>>>
>>>
>>
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