Connect SIP phone to Trixbox from Internet.

ltb76
Posts: 4
Member Since:
2009-05-05

Hi,

I have a small issue with my new Trixbox installation.

The server is located in the DMZ.
I have created a SIP trunk to my Danish provider (Musimi) and a “Ring All” ring group. Which my incoming route ends in.

The phones located on my LAN can connect to the Trixbox. The internal (LAN) IP phones can call each other, they can call out using the Musimi trunk. And I can receive external calls through the Musimi trunk on my internal IP phones.

Now I would like to connect an external (WAN) IP phone to the Trixbox.
The external phone registers at the Trixbox, and if I make a call to an internal IP phone the call is received but there is no sound.

I have opened the following ports in my firewall:

5060-5090UDP External IP --> Trixbox
10000-10200UDP External IP --> Trixbox
5060TCP External IP --> Trixbox

I have changed /etc/asterisk/rtp.conf to:
rtpstart=10000
rtpend=10200

Since the firewall is using NAT between WAN and DMZ I have tried the following in /etc/asterisk/sip_nat.conf:
externip=87.72.xxx.yyy
localnet=172.16.30.0/255.255.255.0
localnet=172.16.10.0/255.255.255.0

When I add this, the SIP trunk and the internal IP phones stops to work. So I’ve removed these entrys again.

I do not see rejected packages in the firewall between the external IP phone and the Trixbox.

WAN subnet: 87.72.xxx.yyy
LAN subnet: 172.16.10.0/255.255.255.0
DMZ subnet: 172.16.30.0/ 255.255.255.0

I’m using the latest Trixbox 2.6.2 (and I have installed all updates)

Any help much appreciated,



b14ck
Posts: 773
Member Since:
2009-03-03
You need to forward ports

whoops

--

Randall Degges
Lead Developer, RCI Telecommunications
projectb14ck - http://projectb14ck.org/ - Weblog



b14ck
Posts: 773
Member Since:
2009-03-03
If your server is on DMZ you

If your server is on DMZ you do not need the sip_nat.conf file. You can make it blank. In regards to the external phone not having audio, have you tried modifying rtp.conf and setting the values back to their defaults? (10,000->20,000)? I would try that first, and see if you can get it to work. If not, then let us know and we can continue working on it. It's usually best to have the defaults working before trying other things.

--

Randall Degges
Lead Developer, RCI Telecommunications
projectb14ck - http://projectb14ck.org/ - Weblog



ltb76
Posts: 4
Member Since:
2009-05-05
Now I changed

Now I changed /etc/asterisk/rtp.conf back to
rtpstart=10000
rtpend=20000

changed the firewall forward to include port 10000-20000UDP
and rebooted the Trixbox.

But there is no change.

sip_nat.conf is back to default aswell.

I've been "tailing" /var/log/asterisk/full for debugging - is that the correct log file to concentrate on? Os should I focus on a different log file?



SkykingOH
Posts: 9678
Member Since:
2007-12-17
You need to put your entries

You need to put your entries back in sip_nat.conf

The only reason this would not work is if your firewall is messing with the media.

During a call with no media you can check the rtp debug from the Asterisk CLI.

You can also enable sip debugging to the full log.

--

Scott

aka "Skyking"



nttranbao
Posts: 189
Member Since:
2008-02-16
you should have

you should have "nat=yes", and can use sip_general_custom.conf instead of sip_nat.conf

I am using with sip_general_custom.conf, and it works great.

;----------------------------------------
nat=yes
externip=87.72.xxx.yyy
localnet=172.16.30.0/255.255.255.0
localnet=172.16.10.0/255.255.255.0

--

----------------------
IT/VOIP consultancy, VOIP eStore, Support Forum
Bao Nguyen IT Co., Ltd.
http://www.baonguyen.vn
WE MAKE IT



ltb76
Posts: 4
Member Since:
2009-05-05
THANK YOU ìt works...

I've now added

nat=yes
externip=87.72.xxx.yyy
localnet=172.16.30.0/255.255.255.0
localnet=172.16.10.0/255.255.255.0

to sip_general_custom.conf and it works :)

Funny but the same settings does not work in sip_nat.conf

Thanks for all the suggestions.



Tofuboi
Posts: 1
Member Since:
2007-06-20
Sorry to hijack this thread

Sorry to hijack this thread but it is relevant to my issue.

I am experiencing the same issue basically.

I have one ADSL link out via a NetGear DG834 router. Port 5060 is forwarded to my Trixbox which sits on 192.168.1.140.

It's a completly vanilla setup, no mods or anything. network based calls work fine, i can register from outside on the internet no worries, dial a call, no audio in or out. I've tried everything on here (adjusting the externip= and what not to my details to no avail. I hope someone can help me. :) Thanks! :D



ltb76
Posts: 4
Member Since:
2009-05-05
Have you forwarded port

Have you forwarded port 10000-20000UDP aswell?



jacus
Posts: 1
Member Since:
2009-07-02
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jstraten
Posts: 165
Member Since:
2006-08-16
I just completed my 2.8

I just completed my 2.8 installation and I encounter the same problem.

I can make calls internally and from my local network to external numbers just fine, but I don't have audio on incoming calls from an external SIP trunk. Oddly enough they can hear me...

I use a dedicated IP address behind a Smoothwall. Nothing relevant (except 80 and 8090) is blocked.

Did anybody else resolve this problem? Any suggestions?

Thanks,
Jens



john.t
Posts: 1
Member Since:
2009-08-30
re:

If your server is on DMZ you do not need the sip_nat.conf file. You can make it blank. In regards to the external phone not having audio, have you tried modifying rtp.conf and setting the values back to their defaults? (10,000->20,000)?

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patriot1sk
Posts: 2
Member Since:
2009-11-16
I can't register to server

I have the the same problem, my server is located in the DMZ, and when i try to connect, ping or register nothing. I try to change the sip_nat.conf :
nat=yes
externip=87.72.xxx.yyy
localnet=172.16.30.0/255.255.255.0
localnet=172.16.10.0/255.255.255.0
Can anybody help me with this.Thx



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