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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040114/9a155824/attachment.htm
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
Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: "Chandra" <chandra@digital.com.np> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: asterisk-users@lists.digium.com 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
> 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 unavailableChandra, I have this _exact_ same problem and it's the Netgear router corrupting the UDP checksums in the RTP packets. Specifically, the checksums come out of the phone unset and the router is setting them to incorrect values. Netgear has not yet responded to my support requets. Ethereal will confirm if you're getting the same thing. Swap out the Netgear with a Linksys or other router and I bet it works.