Home Network/Office Server

I will try to make this as short as possible. I have an Asterisk server at work, which is on a private network behind a snapgear firewall. I have configured port forwarding on the snapgear and I know it works.

I know it works because at home, I can plug up my VoIP phone directly to my cable modem and I can dial out; everything works fine. However, when I plug the phone up to my Netgear WGR614 router, I can get dialtone and it seems to dial numbers but I never hear any ringing on the phone. Also, I can’t dial the number from another phone.

I have my sip.conf setup as:
[home]
type=friend
secret=secret
qualify=yes
nat=yes
port=5060
callwaiting=yes
host=dynamic
canreinvite=no
context=office

My extension.conf for this line is setup as:
exten => 0905,1,Answer
exten => 0905,2,Ringing
exten => 0905,3,Wait(2)
exten => 0905,4,Set(CALLERID(name)=0905)
exten => 0905,5,Dial(SIP/home,20)
exten => 0905,6,Queue(test)
exten => 0905,7,Hangup

When the phone is setup behind my Netgear router and I try to dial my number, I see this in Asterisk:
Unable to create channel of type SIP (no route to destination)

However, I do see that my phone is now reachable. I have tried changing the nat=yes to no and I didn’t get dialtone. I added the port=5060 line to see if it would help and it didn’t. I have configured port forwarding on my Netgear, and even set my VoIP phoen as the default DMZ. It still won’t work correctly.

Hi

Its because when connected to the modem you are not behind NAT at your end just * is but when connected to the router both you and the router are behind NAT, and then th ebest option is to use a VPN

Ian

[quote=“ianplain”]Hi

Its because when connected to the modem you are not behind NAT at your end just * is but when connected to the router both you and the router are behind NAT, and then th ebest option is to use a VPN

Ian[/quote]

Thanks. I have decided to get a VoIP phone with LAN and WAN ports. I am going to stick it outside the router, which should solve the problem.

I was really just looking to see if there were any extension.conf or sip.conf settings that could resolve this.