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:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED]/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/[EMAIL PROTECTED])
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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