Make sure that udp packets can get from the server back to the grandstream.
At 12:40 14/01/04, you wrote:
hi,
I have the following configuration:
Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP)
i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error:
WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable
my sip.conf file is as follows:
[general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall ; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference
dtmfmode=3Dinfo
[grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000
has anyone done this before?
chandra
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