[Asterisk-Users] grandstream asterisk configuration

[Chandra]

i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to
grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i
have also opened all 5060, 5000-5008 ports in my firewall configuration.
grandstream uses 5004 port for rtp.

what am i missing here? please tell me.

chandra

----- Original Message -----
From: "bam" <bam at cqm.co.uk>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, January 14, 2004 6:42 PM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration


>
> 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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