The release notes mentions:
- GV with SIP2SIS (PC-based SIP to Skype transcoding) re-packetization modification
can someone explain what this is?
i use siptosis on PC along with obi(SIP trunk) but not sure where GV comes into picture.
Calls bridged from Google Voice to SipToSis had audio problems with the default SipToSis settings of:
audio_frame_size=240,240,240,160
It was necessary to change SipToSis to:
audio_frame_size=160,160,240,160
Brief testing indicates this issue has been resolved and changing audio_frame_sizes is no longer necessary.
Where do I set the audio_frame_size=160,160,240,160 ?
Quote from: Salvatore on December 04, 2011, 09:53:31 AM
Where do I set the audio_frame_size=160,160,240,160 ?
In siptosis.cfg, but you don't need to change it if you update to 1.3.0(2651).
I upgraded to 1.3.0(2651) but I am still having troubles bridging calls.
Incoming calls come in via SIP. I am able to bridge incoming calls to:
a. ph (connected analog phone line) - no voice at all.
b. SP1 (dial out using SIP) - two way voice.
c. li (connected line to telco/pbx) - no voice.
with 1.2.x all of this is working fine. Is it possible to get the firmware image for 1.2 so we can downgrade? Is there any other alternate solution?