I am running Freeswitch from git and lately I notice that it’s not working properly. Something changed that I could not put my finger on it.
So this is my setup:
- Freeswitch from git – I almost always update this when I can. I don’t remember when but I one of the update I did broke the thing
- Dingaling/GoogleVoice – I don’t know if this is FS or a google voice problem. Maybe google made some changes and screwed up freeswitch
- Debian Sid – updated all the time
- PFsense – latest and greatest router .I am doing NAT through this box. I might have made some changes that might have affected my FS . Need to investigate this.
What I have done so far:
- I had set stun on jingle_profile and I still hear no audio , DTMF works however
- Setting stun off disables DTMF and still no audio
- Turned off Call Screening/Presentation
- I am reverting to version 1.2.stable today. Before I was tracking Master.
Reverting to stable worked!
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….