voip.ms

[Maude]

This might be helpful but it's a bit over my head... Can someone have a look ?

http://www.linuxjournal.com/article/9399?page=0,0

edit: my SIP Passthrough is enabled in WAN in the GUI but somehow I don't see anything related to 5060 in iptables-save ???

 

[RMerlin]

@Kronyx and @RMerlin ,

I was thinkering with iptables and inserted those rules in the FILTER section:

admin@Sarah:/tmp/home/root# iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A INPUT -p udp -m udp --dport 5080 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A OUTPUT -p udp -m udp --dport 5060 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A OUTPUT -p udp -m udp --dport 5080 -j ACCEPT

It didn't workout though but I think I might be onto something...

That won't help. INPUT and OUTPUT chains are for traffic targeting the router itself, not your LAN devices.

 

[Maude]

That won't help. INPUT and OUTPUT chains are for traffic targeting the router itself, not your LAN devices.

Those this makes a bit more sense ?

-A PREROUTING -d 70.55.199.224/32 -p udp -m udp --dport 5060 -j DNAT --to-destination 192.168.0.85:5060
-A PREROUTING -d 70.55.199.224/32 -p udp -m udp --dport 10000:10100 -j DNAT --to-destination 192.168.0.85


.85 is the Obihai box

 

[RMerlin]

Those this makes a bit more sense ?

-A PREROUTING -d 70.55.199.224/32 -p udp -m udp --dport 5060 -j DNAT --to-destination 192.168.0.85:5060
-A PREROUTING -d 70.55.199.224/32 -p udp -m udp --dport 10000:10100 -j DNAT --to-destination 192.168.0.85


.85 is the Obihai box

Not really, -d means "destination IP". If the idea is to secure your ATA, then it should rather use -s.

 

[Maude]

Well, no... The idea is to let the out traffic through since I can call and receive calls but can't be heard even though I hear people. It's the same case as Kronyx...

The darn thing is that the exact same setup worked flawlessly with EBox.... No changes to Asus router at all.

So the problem must be on the Bell side of the equation...

 

[RMerlin]

Well, no... The idea is to let the out traffic through since I can call and receive calls but can't be heard even though I hear people. It's the same case as Kronyx...

The darn thing is that the exact same setup worked flawlessly with EBox.... No changes to Asus router at all.

So the problem must be on the Bell side of the equation...

As I was mentioning, no firewall rules should be required for SIP to work properly with Asuswrt.

One potential difference in your case is probably that with EBox your modem was bridged, while with Bell you are most likely in a double nat setup (unless you use the funky 2wire "super DMZ mode", which seems to "kinda" work".)

Personally I'm with Teksavvy's vCable, so my modem is fully bridged.

 

[tijaune]

I do see STUN right in that first screenshot that you posted.

that Stun radio box is not selected (that capture is from FXF port tab). There is another field for Stun server in another tab which is blank on my setting.

oh and I have Thomson modem in automatic ip in wan section, how do you determine if it is bridge?

 

[Maude]

Hem.... If I might say it this way M. Petit Jaune, are with Bell per chance ?

 

[Kronyx]

Let me know Maude if you find something, you are my last resort ! hehe

Merci !

 

[Kronyx]

Well, no... The idea is to let the out traffic through since I can call and receive calls but can't be heard even though I hear people. It's the same case as Kronyx...

The darn thing is that the exact same setup worked flawlessly with EBox.... No changes to Asus router at all.

So the problem must be on the Bell side of the equation...

I don't think the problem is Bell, I have configured my ATA with PPPOE passthrough so it gets an IP from Bell and it work flawlessly, but I don't like to let it directly on internet.

 

[RMerlin]

that Stun radio box is not selected (that capture is from FXF port tab). There is another field for Stun server in another tab which is blank on my setting.

oh and I have Thomson modem in automatic ip in wan section, how do you determine if it is bridge?

DCM425, DCM475 or DCM476? All of these are pure modems, they have no router functionality so they will always be bridged.

I use a DCM475.

 

[tijaune]

ah I'm also using DCM475 and sorry Maude, I'm with EBox.

It does happen some occurences over the past year, sometime the caller can't hear our hello. But it was so rare I did not put any troubleshooting effort.

Thanks for the informative linux journal read!

 

[Kronyx]

You got it fixed tijaune ?

 

[Maude]

Bon,

To recap: I had the exact same router setup working flawlessly with EBOX as for SIP Passthrough. BUT I didn't had a PPPoE or VLAN 35 with EBOX.

Then, this morning I tried this test: I reconnected the ONT to the Bell Hh2000 and the Asus to the HH2000 from Bell WITHOUT the VLAN 35 option in LAN/IPTV set because that's how it connects to the HH2000. The setup worked and phone was working both ways albeit slowly.

I printed the iptables-save output to the external drive.

I then reconnected the ONT to the Asus, once again setting up the LAN/IPTV Internet to VLAN 35, effectively bypassing the HH2000 completely. The phone worked only one way, going out.

I also printed the iptables-save output to the external drive.

I'll post the outputs to the thread tonight as I'm at the office right now.

But the conclusion is clear: either loose 10% of speed and have the phone working or continue debugging. I think I'll continue debugging

 

[Marz888]

I think you guys should try STUN. It just works.... It's STUNning.

 

[Kronyx]

hahaha Marz888, i'll try it tonight.

 

[Maude]

Already do, no effect...

Here are the iptables-save I mentionned this morning and a diff between them...

Can someone have a look ?

TO_HH2000: http://maude-boivin.com/Geo/to_HH2000.txt
TO_ONT: http://maude-boivin.com/Geo/to_ONT.txt
Diff: http://maude-boivin.com/Geo/diff.xt

 

[Kronyx]

I've just put the ATA in the HH200 (wich I use for the TV only) and everything work, so there is something with the ASUS on PPPOE with the ATA on LAN ...

 

[Maude]

@Kronyx , I tought you had the Asus connected to the ONT... That's the setup that doesn't work for me. When I connect the Asus WAN port into the HH2000 and ONT to the HH2000 with PPPoE on the Asus W/O VLAN 35, the setup works but I loose 10% on download speed...

Oh well, I'll probably set it up that way tonight but I would have preferred to find the solution...

 

[tijaune]

I noticed on voip.ms wiki page for Cisco SPA112, RTP port is 10000. On my Grandstream HT701, it is 5004 currently (default value).