[Asterisk-Users] Intervivo sip.conf?

[David Croft]

Thanks Mark,

I have tried your config and variations on it but have the same problems.

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.

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?

David


Mark Turner wrote:
> 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 at 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.
> 
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