Draytek 2800VG Asterisk Problems

Asterisk Asterisk Support
1

Hi folks,

I have been using asterisk with voiptalk and a 3com router which was working well except no QoS. I have now changed to a Draytek 2800VG but cannot get it to work properly.

The asterisk box is on a Lan (192.168.1.5), on the 3com router port 5060 and 4569 were forwarded to the asterisk ip address. The same settings on the 2800VG result in incoming calls via sip not arriving at the asterisk box at all, outgoing calls can be setup but there is no audio.

Can anyone suggest what I need to do differently on the 2800?

Thanks,
Andy.

2

to make asterisk work behind a NAT you need to do a few things.

  1. Set a reasonable port range in RTP.conf. 100-200 ports are fine, you don’t need 10,000. Use high numbers, 10000-10200 works good.

  2. figure out your external IP. in sip.conf, set externip=YOURexternalIP and localnet= to reflect your network. You might also want to set NAT=yes. If you don’t do this you will get one-way or no audio.

  3. Now on your router, forward udp ports 5060 and whatever range you set in step 1 to your asterisk box. If you do not have the RTP ports forwarded you will get one-way or no audio.

  4. configure your QOS. if it works by IP or MAC, that’s great. If it goes by service or port, remember to prioritize RTP port range. It’s the stuff that goes on the 10k range ports that makes the difference.

hope that helps!

3

Thanks for the tips but still no joy, here is my sip.conf that works fine with the 3Com router:

[voiptalk]
username=username
type=friend
secret=xxxxxx
nat=yes
maxexpirey=180
insecure=very
host=voiptalk.org
fromuser=username
fromdomain=voiptalk.org
dtmfmode=rcf2833
disallow=all
defaultexpirey=160
context=from-pstn
canreinvite=no
authuser=username
allow=g729

The Draytek router has TCP and UDP 5060-5070 forwared to the * machine (192.168.1.5) also ports 8766 to 35000 TCP and UDP, still no joy. I have also tried DMZ to 192.168.1.5 - still the same.

I tried adding

externip=my public ip
localnet=192.168.1.5

Still the same, don’t know what else to try, do I have a router problem or an asterisk config problem, if I put the 3Com router back on it works fine.

Thanks,
Andy.

4

[quote=“woodpecker”]I tried adding

externip=my public ip
localnet=192.168.1.5
[/quote]

are you adding this to the [general] section ??

5

It wasn’t in general but I just tried it in general and its still the same, general now is:-

[general]

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
externip=81.xxx.xxx.xxx
localnet=192.168.1.5

disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf

Any other suggestions??

Thanks,
Andy.

6

It’s working!

Without changing anything else the phone just started ringing and it seems to be working, can someone just confirm the ports that are actually required to be open, I’ve turned off the DMZ now.

Port 5060, does this require UDP and TCP?

Ports 10000-20000 UDP and TCP?

How many high ports do I need open for a maximum of around 10 callers in queues?

Thanks,
Andy.

7

its ALL udp. you just need 5060 and your RTP range.

each call will take up 1-2 ports, I recommend forwarding 100 or so just in case.

also with localnet= you must also put in your subnet mask ie

localnet=192.168.1.1/255.255.255.0