On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling k...@ken-ton.com wrote:
Setting the codec negotiation to scrooge resolved my problems w/
CallCentric.
I'd bet that'd do it for him as well.
*Lessons Learned by me:*
1.) Listen to Brian.
2.) When in doubt, refer to rule 1.
Can I get
Setting the codec negotiation to scrooge resolved my problems w/ CallCentric.
I'd bet that'd do it for him as well.
Lessons Learned by me:
1.) Listen to Brian.
2.) When in doubt, refer to rule 1.
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0
On Dec 23, 2009, at 11:57 AM,
That usually means they are saying 30 but sending 10 which is broken.. you
can't say hey i'm sending 30 and then send 10... find a new provider or beat
them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP
They don't operate their own voip gateways, just run an SBC in front of a
bunch of other providers. So someone they are reselling is using Sonus
gear. I use them to originate to some destinations but in the US I avoid
them due to the sonus stuff that pops up on certain routes.
On Wed, Dec 23,
If I only care about outbound audio, is there a way to force the audio
packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there
still this same issue?
--matt
On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker r...@rupa.com wrote:
They don't operate their own voip gateways, just
You can disable auto-adjust in the sip profile., but that might just
make it worse, no warranty:
param name=rtp-autofix-timing value=false /
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 23-Dec-09, at 11:41 AM, Matthew
You might also have to set the codec negotiation to scrooge
/b
On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote:
You can disable auto-adjust in the sip profile., but that might just make it
worse, no warranty:
param name=rtp-autofix-timing value=false /
Mathieu Rene
Avant-Garde
