IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it . You will have to keep asterisk server in static ip or do port forwarding to accept connections from outside .
<br>OR maybe i didnt understand senario properly here . Is it like your "Server with SIP application (public_address)" responds to sip calls made by any program ( like sjphone pc-pc sip ) . If thats case then asterisk should be able to call it like any other program or maybe theres nat scenario playing bad here :-/ . Can you port forward from firewall to asterisk server ??
<br><br> <br><br><div><span class="gmail_quote">On 13/11/06, <b class="gmail_sendername">nik600</b> <<a href="mailto:nik600@gmail.com">nik600@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
On 11/12/06, nik600 <<a href="mailto:nik600@gmail.com">nik600@gmail.com</a>> wrote:<br>> On 11/12/06, Vicky <<a href="mailto:vicky.r@gmail.com">vicky.r@gmail.com</a>> wrote:<br>> > Yep make the server with dynamic ip register to server with static ip ( sip
<br>> > or iax both will do but in sip keep nat=yes while making extension )<br>> ><br>> the problem is that the server with dynamic ip can't register on the<br>> other server!<br>><br>> This is the situation:
<br>><br>><br>> Server with SIP application (public_address)<br>> |<br>> |<br>> - - - Internet<br>> |<br>> |<br>> Firewall (NAT)<br>> |<br>> |<br>> Server Asterisk (private ip:<a href="http://192.168.100.249/public">
192.168.100.249/public</a> ip:public_address_2)<br>> |<br>> Analogic Board<br>> |<br>> Telecom<br>><br>> I want to make a call from Server Asterisk to the server with SIP Application.<br>> The SIP Application can't register to Server Asterisk (because the
<br>> application can't do it, i know, it isn't a good thing....but this is<br>> the application)<br>> When The SIP Application receives a SIP call it responds (because a<br>> dummy SIP user is autoregistered on hisself)
<br>><br>> So i only have to make a call to SIP/user@public_address<br>><br>> I've also tried to setup an asterisk server on my laptop, and make a call to<br>> SIP/user@public_address from the public_address network. It works!
<br>><br>> I only have to setup the Asterisk server in production to make a SIP<br>> call throw the NAT but without any SIP user registered on it.<br>><br>> Can i do that?<br>><br>> Many thanks to all
<br>><br>maybe you need some other information?<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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