[Asterisk-Users] grandstream asterisk configuration
Walt Reed
asterisk at linuxguy.com
Wed Jan 14 07:50:10 MST 2004
You may need to run Ethereal to make sure packets are REALLY getting
through.
On Wed, Jan 14, 2004 at 07:35:48PM +0545, Chandra said:
> 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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>
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