[Asterisk-Users] grandstream asterisk configuration

[Chandra]

 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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