asterisk stopping sip calls after 20seconds

ma_hiro

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Mar 9, 2010
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#1
Hello, Sirs:

This one is really giving me a headache. I have been using asterisk for quite some time and testing it.

I set up a * server from behind a NAT and forwarded all ports to the router which is a public IP. I have tried X-Lite and Eye-Beam soft phones and created 2 of them from another network on the internet. These 2 softphones are also behind a NAT router, however, I also forwarded the needed ports.

One time I can make the calls thru, but most of the time its not. It seems the problem appear and disappear for whatever reasons. Connectivity is always there since I can browse internet when problem appears.

Even when sip phones call a feature code (like *43 or echo test, call fails after 20sec).

My versions are: asterisk 1.4.29.1 freepbx 2.7.0.0

Calls within the LAN are no probs.

I know it must be a NAT issue, but it seems there is not much discussion on this particular problem of 20sec disconnect.

Can you please help a newbie like me?

Thanks,
ma-hiro
 

ma_hiro

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Mar 9, 2010
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#2
Hello Sirs:

I think this has been fixed already.

First time I hide an asterisk box behind a Nat.

Solution is externip/localnet settings seems to fix it.

Case closed.
 

danardf

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Dec 3, 2007
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#3
Hi And Welcome to our Forum Elastix.

The RTP config is important.
Remember that you must forwarding the RTP ports (10000 to 20000) or if you want least range ports. for example: 10000-10050
Look at rtp.conf

Regards
 

ma_hiro

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#4
Hello, again:

The problem of calls dropping after 20 sec of talking appears again.

to illustrate:

1. (*)server in 2-lan card: 1st nic is public IP and 2nd nic is private lan
2. ATA FXO terminates to pstn, and ATA wan is same subnet of (*)LAN;
3. xlite/eye-beam is somewhere on the net registering to (*) and getting registered

x-lite users are able to call pstn on 1-stage dialling and failing after 20 sec.

Config are:
Router 1:
port forwarded to sip: 5060-5070; 10000-20000 (I tried up to 60000 but probs still)

Router 2:
port forwarded to xlite PC: 5060-5070; 10000-20000

(*) version is 1.29.1/freepbx 2.0

I am wondering why in some test calls seems to connec beyond 20 sec and also failling without doing anything.

any thoughts?

Thank you!
ma_hiro
 

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