Me too.

Voip: Callcentric.com
OS2008 Diablo - network error
OS2008 out of box - connected fine, but I only got one way communication (no incoming sound, outgoing sound worked ok).

Tested with softphone on my pc from same network and everything works fine.

Disabled "Discover Public IP" and I can connect now, BUT something is very wrong. I get ringing when I dial, but no answer therefore call is not going out properly. At least with the previous OS I could make a call even if I couldn't hear anyone talking!

I suspect that callcentric requires codec settings that are not supported; here is what I need to set somehow (but I cannot find settings for these):

Silence Suppression NO or OFF
G.723 Rate 6.3 kbps
Codec Orders Place codec's that are available in the following order (from 1 highest priority, to 12 lowest priority):
1. G.729
2. G.723 6.3k
3. GSM
4. iLBC
5. Speex 8k
6. G.728
7. G.726
8. Speex 16k
9. Speex 32k
10. G.722
11. G.711 mu-law
12. G.711 A-law


Other Settings:

STUN Server NO or OFF
Subscribe for Message Waiting Indicator (MWI) NO or OFF
Send DTMF RTP (RFC 2833)
DTMF Payload Type 101

Well, any solution for this jet? I'm too would like to get my 1und1-sip account to work, but only get the authentication error. my second account (sipgate.de) works fine. btw. I'm on latest diablo, 4.2008.30-2

Have you tried disabling the "discover public address" option in the advanced configuration of your SIP account?

Yes, tried with that enabled and disabled, no difference

Not sure if I am having the same problem , anyway....

I am having trouble with incoming calls to rtcomm using gizmo5 as my service provider.
When a call comes in the device lights up, the rtcomm voice application automatically opens and the cpu spikes as you would expect , but the answer dialogue doesn't pop-up and the caller is told I am offline.

I have tested it with my voxalot number and that works perfectly.

Outgoing calls work fine.

Incoming and outgoing calls work fine if I use the gizmo5 softphone instead of rtcomm.

anyone else got this problem ?

update - I am now forwarding my gizmo5 number to voxalot and the incoming calls work fine that way ?! what a strange bug

this is what i've discovered so far:
-i've got the problem of failed login on diablo only behind a nat (quite obvious), the same settings (discover public address enabled) works fine on a plain connection, no drop after 30 secs
-on chinhook everything works fine, but if i disable the discover public address the call will drop after about 30 secs
-on diablo the login works only with discover public address disabled but this make the incoming calls drop after about 30 secs

i'm waiting to come back from holidays to set up the test environment to sniff the packets

Random thought here... not sure if it's relevant or not. Under Chinook, SIP authentication would fail after installing support for .ogg files -- something to do with the codecs. I haven't tried this under Diablo yet, but I'll try to give it a shot this weekend.

Callcentric works for me. The trick seems to be to disable STUN, and set an outgoing proxy. Perhaps STUN is broken in diablo.

In detail, my rtcomm settings are:

Username: <my number> @ callcentric.com
Password: <my password>
Use for PSTN calls enabled

With non-default advanced settings:

Transport: UDP
Outgoing proxy: callcentric.com
Discover public address: off

I don't know anything about 1und1, but perhaps something similar would work. I haven't really tried to get Gizmo to work either.

Hi all,
I'm with 1und1 in Germany and playing with my n800 running newest Diablo (30-2) I'm experiencing exactly the same problem as Efelon but only from behind a NAT. The same SIP account runs fine from a direct GPRS/EDGE connection with a public IP-number assigned.

I tried to trace the packets on my local router and I can confirm that the error msg is 403 Keine RFC1918-IPs erlaubt
I also can see no connection attempt at all to stun.1und1.de in the course of the registering negotiations, independently on the STUN settings.

cheers,
/burbie

Looks like you 1und1 users are out of luck. It appears that the NIT Telepathy client only uses stun for the data connection, not for SIP. I believe that this is the usual approach; there is no need for stun in order to talk to a server (although the keep-alive is important). But the 1und1 server doesn't seem to accept NATed clients.

So your best option is probably to get a Voxalot account, and let it talk to the 1und1 server. This may be worthwhile anyway. I really like Voxalot; it's wonderful for getting around the various limitations of different SIP providers. It's free if you only want out-going calls, $15 (US) per year to handle one SIP registration, and $25 per year to handle up to five SIP registrations.

Cheers,
Lon