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Asterisk with 2 SIP Linksys Phone adapters, NAT problems

 
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IronTool
planetFreshman


Joined: 02 Dec 2005
Posts: 1

PostPosted: Fri Dec 02, 2005 3:36 pm    Post subject: Asterisk with 2 SIP Linksys Phone adapters, NAT problems Reply with quote

Hi,

First of all, congratz for a great resource forum.

Iīve installed Asterisk with 2 Linksys PAP2-EU Phone Adapters. In my lan iīve installed them correctly and i call from one to the other one with no problems.

Iīve moved one of the linksys to another lan with his own router and internet connection. Now I canīt make a call, because in one direction the audio is not arriving.

I think there is some problem with RTP port forwarding. This is my setup

Lan1: Asterisk server and Linksys Phone1. Router Us Robotics 9003
53 UDP->Linksys Phone 1
69 UDP-> Linksys Phone 1
5060 TCP-UDP->Asterisk
5061 TCP-UDP-> Linksys Phone 1
10000-10050 UDP->Asterisk (RTP ports configured in rtp.conf)
16384-16482 UDP-> Linksys Phone 1 (RTP ports as configured in phone)

Lan2: Linksys Phone1. Router Linksys AG241
53 UDP->Linksys Phone 2
69 UDP-> Linksys Phone 2
5060 TCP-UDP-> Linksys Phone 2
16384-16482 UDP-> Linksys Phone 2 (RTP ports as configured in phone)

Sip.conf:
[200]
username= Phone 1
type=friend
secret=1234
record_out=On-Demand
record_in=On-Demand
qualify=yes
port=5061
nat=no
host=dynamic
dtmfmode=rfc2833
context=SIP
canreinvite=no
callerid="Phone1" <200>

[201]
username= Phone2
type=friend
secret=12345
record_out=On-Demand
record_in=On-Demand
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
context=SIP
canreinvite=no
callerid=" Phone2" < 201>

With this config when I call from phone 1 to phone 2, they can hear me but I canīt hear them. If I uncheck 10000-10050 UDPport forwarding, I can hear them but they canīt hear me.

I know that there a lot of problems with Nat translations and SIP RTP ports, but there is a way to make it work???

Thanks in advance.

Note: sorry for my English, itīs not my primary language.
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planetWayne
Site Admin


Joined: 30 Jan 2003
Posts: 280

PostPosted: Mon Dec 12, 2005 11:08 pm    Post subject: Reply with quote

Hiya!
Mmmm you got me. All my stuff is on my local lan so I don't have that problem, saying that I do have SIP and IAX ports forwarded to my Asterisk box

For IAX I have TCP and UDP on port 4569.
and for SIP - TCP 5060, UDP 8000 - 8012

These are used so I can make outbound (and accept inbound) calls via my Asterisk box.

One more thing you could try is to get both sites connected with a VPN. Then you shouldn't need to worry about what ports to open up as the two lans would be 'joined'.

HTH.
Wayne.
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