[Asterisk-Users] Intervivo sip.conf?

[Mark Turner]

Hi Dave,

On Sun, 17 Oct 2004, David Croft wrote:
> 
> I have tried your config and variations on it but have the same problems.

Sorry to hear that you're still having problems.  If you email me your
sip.conf and extensions.conf then I'd be happy to take a look.

> Placing a call out using intervivo, regardless of dtmfmode setting, DTMF 
> tones are not recognised by the recipient. The same applies to receiving 
> dtmf digits.

I did mention that I never got around to making DTMF work from my home
Asterisk server, but it will be possible.  My guess is that there is
a mis-match between the DTMF mode settings at either end, i.e. in your
config and in our server config.  We have a (hidden by default) config
option on your control panel that allows you to specify the DTMF mode
manually, which should allow us to fix this for you.

> Also, unless I set insecure=very (which I shouldn't need to), I get 
> these messages when someone tries to call in:
> 
> Oct 16 18:08:21 NOTICE[7175]: chan_sip.c:7162 handle_request: Failed to 
> authenticate user "xxx" <sip:xxx at 217.168.22.129>;tag=as30592e8c
> 
> where xxx is the number they're calling from. They get a busy signal.
> 
> Any ideas?

I'm sure we'll sort it once I've seen your config files.

Cheers,

Mark.

p.s. If you're not keen on emailing your config files to my home address
(why should you believe that I really work for InterViVo) then feel
free to email them to support at intervivo.net instead and I'll grab them
from there.