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<DIV> hi,<BR> <BR> I have the following configuration:<BR> <BR>
Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public
IP)<BR> <BR> i can register fine and call ringing is working as good. The
problem is =<BR> i cant hear audio both ways and i get this
error:<BR> <BR> WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP
Read error:<BR> Resource temporarily unavailable<BR> <BR> my sip.conf
file is as follows:<BR> <BR> [general]<BR> port =3D
5060
; Port to bind to<BR> bindaddr =3D
0.0.0.0
; Address to bind to<BR> ;externip =3D
200.201.202.203 ; Address that we're going to put in
=<BR> SIP<BR> messages if we're behind a
NAT<BR> tos=3Dlowdelay<BR> disallow=3Dall
; Disallow all
codecs<BR> allow=3Dulaw
; Allow codecs in order of preference<BR> <BR>
dtmfmode=3Dinfo<BR> <BR>
[grandstream1]<BR> type=3Dfriend<BR> host=3Ddynamic<BR> secret=3Dmysecret<BR> context=3Doutgoing<BR> nat=3Dyes<BR> reinvite=3Dno<BR> canreinvite=3Dno<BR> qualify=3D2000<BR> <BR>
has anyone done this before?<BR> <BR> chandra</DIV></BODY></HTML>