asterisk users - Feb 2007 - peer-to-peer RTP trouble in SIP

Hey,

We have asterisk 1.2.4 (old I know) with a couple of snom
phones, a couple of grandstream phones and around 65 philips
dect stations.
Now the problem:
All calls do peer to peer RTP except the calls from dect
station to dect station.
snom to dect or dect to snom do peer to peer.
So the sip config looks fine because all the 'static
deskphones' honor the REINVITE and start talking to
eachother.
Our supplier told us they dont send SDP with the INVITE. Can
this be the problem causing dect to dect calls to always use
asterisk in the RTP path ?

-- 

Michiel van Baak
michiel@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called
users?"

23 feb 2007 kl. 09.52 skrev Michiel van Baak:
> Hey,
>
> We have asterisk 1.2.4 (old I know) with a couple of snom
> phones, a couple of grandstream phones and around 65 philips
> dect stations.
> Now the problem:
> All calls do peer to peer RTP except the calls from dect
> station to dect station.
> snom to dect or dect to snom do peer to peer.
> So the sip config looks fine because all the 'static
> deskphones' honor the REINVITE and start talking to
> eachother.
> Our supplier told us they dont send SDP with the INVITE. Can
> this be the problem causing dect to dect calls to always use
> asterisk in the RTP path ?
If they do not send SDP with the INVITE there will be no
media at all in the call. Very simple.

Can be that they do not support re-invites. If so, you should
see an error message in the SIP communication.

/O