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Asterisk: port forward sip protocol to asterisk server

Posted on 2007-11-13
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Last Modified: 2013-11-12
i need to port forward sip protocol to asterisk server so that remot phones can get connected. i have a linksys router
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Question by:jonnydollar
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13 Comments
 
LVL 12

Expert Comment

by:bhnmi
ID: 20276448
Forward Port 5060 (if it is on the default sip port) To the IP address of the trix box
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Author Comment

by:jonnydollar
ID: 20276680
i have done a range of ports 5060-5080 to the asterisk server i think and i am still unable to register
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LVL 36

Expert Comment

by:grblades
ID: 20278405
1) Forward tcp & udp port 5060 to the asterisk server.

2) Forward udp ports 10000-20000 to the asterisk server and check /etc/asterisk/rtp.conf to make sure asterisk is using this port range for rtp.
If you done do this then you will get one way audio.

3) Edit /etc/asterisk/sip.conf and define your local network IP range and the external IP address of the asterisk box (which will be the public IP address of the netgear router).
This is required so that asterisk puts the correct IP address in the SIP packets.

4) In sip.conf for the phones connecting from the internet put 'NAT=yes'. This helps if the phone itself is also behind a NAT device.
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LVL 2

Expert Comment

by:PCMAC
ID: 20279453
I know you're trying to do this with a SIP protocol. Its quite tricky Have you considered IAX?

I've used this site:
http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

But with little success. My best solution was to create a VPN tunnel between two sites. This involved no hacking of the Asterisk box, just the firewalls.
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Author Comment

by:jonnydollar
ID: 20279514
grblades:

I used dyndns to reach my network because my public ip is a dynamic one from the provider. will this still be successful? you mentioned defining the local network ip range and the public ip in the sip.conf file. Can you give me an example?
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LVL 36

Accepted Solution

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grblades earned 500 total points
ID: 20279599
If your local network has a IP range like 192.168.1.x then you would add :-
localnet=192.168.1.0/255.255.255.0

As you use dyndns then you would specify your external hostname instead of its IP address. For example :-
externhost=foo.dyndns.net

As PCMAC said if you are using software phones then the IAX2 protocol might be better as it works much better with firewalls. You cant really use it for hardware phones as there are few available. If you are using a software client then a good IAX2 bases one (which we use) is called DIAX. There is also the additional benefit that it works from hotels etc... which might otherwise block SIP.
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Author Comment

by:jonnydollar
ID: 20279703
i use a combination of soft phones and ip phones. i have one grandstream budgetone bt-101 sip phone and on my computers i use eyeBeam. Is it still reccommended i use IAX?
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Author Comment

by:jonnydollar
ID: 20279712
eyebeam is on a laptop that i would like to register to the remote asterisk server btw
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LVL 36

Expert Comment

by:grblades
ID: 20280336
If eybeam is working for you now then by all means carry on using it.

You may find that it does not work from some hotels etc...
If you find this is often the case then you can switch to DIAX and it will improve the chance of you being able to use it.
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LVL 2

Expert Comment

by:PCMAC
ID: 20280371
Yes, Eyebeam is what I use in hotels setup using IAX2. Works great!
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Author Comment

by:jonnydollar
ID: 20281465
it worked i can register but i cant get any audio coming back from the other end. any suggestions
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LVL 36

Expert Comment

by:grblades
ID: 20282065
Can you post the contents of your rtp.conf file.
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LVL 36

Expert Comment

by:grblades
ID: 20282086
I assume the audio problem is when using SIP?

If you wanted to try IAX2 you will just need to open UDP port 4569 and then make the relevant addition to the iax configuration file and the dialplan (to call the iax extension). You wont have any one way audio problems with IAX2.
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