I am taking on a new project. I already started this a week ago and It’s still really a work in progress.
So,from my title you may know that I am trying out Freeswitch. I had Asterisk for so long I can’t even remember now. I mainly use my Asterisk setup for google voice and I don’t like that their implementation breaks a lot. I really don’t know if its Asterisk’s fault but I am just willing to try out something new ,hopefully this will be a whole lot better.
A couple of days ago ,I already figured out how to get incoming and outgoing GV(google voice) calls. Now all I need is to make a few adjustments to the dialplan,if I figure this out.*crosses fingers* The dialplan for me I think is much easier to understand than asterisk ,I just need a refresher on REGEX because all the logic depends on it.
Browsing through the Freeswitch wiki ,I found that refresher on REGEX.Thanks a lot to the Freeswitch community,I feel like I am . . .home. *winks*
For several months or is it a year already ,I can’t remember honestly, Sipbroker’s access to Gizmo/Sipphone was borked.Today when I tried it I was amazed that it worked!
This is awesome coz I a was using this to call the RTP-300 ATA that I sent to the Philippines. I was really bummed that It quit working coz it was really easy to just call Sipbroker’s PSTN gateway and dial the Gizmo/Sipphone number that I setup.
Now I can call again using my Cellphone wherever I am!!! Awesome really!
On a side note,I notice that Sipbroker has a Philippine flag on their access code. Could it be that their servers are located in the Philippines? Interesting….