Search found 93 matches
- Tue Oct 29, 2013 5:42 am
- Forum: Technical Support
- Topic: GoogleVoice DialPlan not dialing out
- Replies: 6
- Views: 6753
Re: GoogleVoice DialPlan not dialing out
Same problem here since early today.
- Wed Aug 28, 2013 7:40 pm
- Forum: Report a bug
- Topic: VPN issues
- Replies: 2
- Views: 2030
Re: VPN issues
I know SS does not involve in audio path, that's all the reason I use it in the first place! For the B to A call, I have to use the callback function: sys.Callback("A", "B") then all fine. If I use: sys.Dial("A") then no inbound audio on B. There must be a bug somewhere in SS to cause such a differe...
- Wed Aug 28, 2013 5:21 am
- Forum: Report a bug
- Topic: VPN issues
- Replies: 2
- Views: 2030
VPN issues
I have two end points registered to SS: A: public IP, so no NAT involved. B: through a VPN tunnel. When A call out to B: no problem at all. When B call out to A: A can hear B but B cannot hear A. This is the major problem and seems to be all SS's fault. I've tried [ma=false] etc. When B places a Goo...
- Sun Sep 02, 2012 7:54 am
- Forum: Report a bug
- Topic: No inbound audio with Nimbuzz and Fring
- Replies: 2
- Views: 2362
Re: No inbound audio with Nimbuzz and Fring
Well, I set up a FreeSWITCH just to test it for nimBuzz, no NAT problem. I'm sure FS is not doing media transcoding nor proxying. I enabled "inbound-bypass-media" as described here: http://wiki.freeswitch.org/wiki/Bypass_Media and here: http://devblog.brahmancreations.com/content/freeswitch-g729-pas...
- Thu Aug 30, 2012 6:41 am
- Forum: Report a bug
- Topic: No inbound audio with Nimbuzz and Fring
- Replies: 2
- Views: 2362
No inbound audio with Nimbuzz and Fring
There must be some NAT handling issues between SS and Nimbuzz and Fring etc. No inbound audio when NAT is involved. If I let them connect directly to the sip provider such as sipgate, i.e., phone -> nimbuzz -> sipgate, no problem. Once SS is in the middle, i.e., phone -> Nimbuzz -> SS -> Sipgate, th...
- Fri Aug 17, 2012 9:01 pm
- Forum: Feature Requests
- Topic: CrowdCall dialing?
- Replies: 2
- Views: 3587
Re: CrowdCall dialing?
Somebody will (if not yet) post the dialing mechanism, I think, all you do is to implement it into SS.
- Thu Aug 16, 2012 11:52 pm
- Forum: Feature Requests
- Topic: CrowdCall dialing?
- Replies: 2
- Views: 3587
CrowdCall dialing?
One of the two reasons that attracted me to SS is: GoogleVoice dialing (the other is no audio touch, still unique). Crowdcall is similar to GV, but free calls to many other countries such as India,China, France, Germany, etc. You install an app on your smartphone, give a # for it to call back, then ...
- Tue Aug 16, 2011 12:52 am
- Forum: Feature Requests
- Topic: May i use gtalk directly in sipsorcery?
- Replies: 4
- Views: 3010
Re: May i use gtalk directly in sipsorcery?
The major problem of SS service is not about features but reliability. Other such services all use the well established software - asterisk etc. Aaron need more people to beta test this software. Running an old version of SS on my computer is easy, but to update and keep up with what Aaron uses on h...
- Wed Aug 03, 2011 9:23 pm
- Forum: News
- Topic: Special offer for Beta users until 31 July 2011
- Replies: 8
- Views: 14256
Re: Special offer for Beta users until 31 July 2011
I'm a light user too, the free account is well enough to handle my in and out calls with voipdiscount, sipgate and ipkall.
- Tue Aug 02, 2011 1:32 pm
- Forum: Feedback
- Topic: CSipSimple for Android
- Replies: 5
- Views: 13293
Re: CSipSimple for Android
Grooveip or gtalk uses g711 exclusively, not reliable on 3g.