[Asterisk-Users] grandstream asterisk configuration

[bam]

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