On Fri, 1 Oct 2004, David Croft wrote:>
> Anyone have a working sip.conf for Intervivo? (with bidirectional audio,
> dtmf and authentication!)
I use....
register =>
0845NNNNNNN:PASSWORD:0845NNNNNNN@sip.intervivo.net/YOURINTERNALEXTENSIONNUMBER
externip = EXTERNALADDRESSOFHOMENATFIREWALL
nat = yes
And....
[ivv]
type=friend
secret=PASSWORD
username=0845NNNNNNN
host=sip.intervivo.net
fromuser=0845NNNNNNN
externip = EXTERNALADDRESSOFHOMENATFIREWALL
nat=yes
canreinvite=no
reinvite=no
notransfer=yes
qualify=yes
I *think* you can get away with not having some of the NAT stuff now,
but I'm not 100% sure and daren't try changing it from afar in case it
breaks our home phone system and my wife wouldn't be impressed. :)
In extensions.conf I have....
[macro-ivv]
exten => s,1,Dial(SIP/${ARG1}@ivv)
And....
[pstn-via-ivv]
exten => _0[1-9].,1,Macro(ivv,${EXTEN})
I *don't* have DTMF working at home at the moment 'cos I'm routing
all
calls via a Pheenet EL400 (allows me to integrate my two PSTN lines and
my two Dect bases with the VOIP world) and I haven't figured out how to
tell the EL400 to pass DTMF in a compatible way yet.
My home extensions.conf is a bit of a mess at the moment with lots of
stuff in there to route to other VOIP networks instead of using the free
gateways via InterViVo, so I'd rather not show too much more of what I
have until I've tidied it up. I also implement parallel ring on the home
* server rather than using the same functionality via the control panel.
Lots of tidying needed. :(
BTW, we're about to add a new feature on your VOIP control panel on our
website which will allow you to choose what codec we use when sending
calls to you, handy if you'd prefer to force ilbc to keep the bandwidth
usage down.
BTW2, I'm the CTO at InterViVo and it was me and my team that built
and manage our VOIP service. I'd be more than happy to help you get up
and running with Asterisk but please email via this list rather than to
me personally so that my colleagues will see it if I'm not around.
Cheers,
Mark.