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@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@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@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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