1. Username and password are manditory in the XML - you don't have to put anything that makes sense, just fill it in with your DID or something and keep register=false and those details will never do anything.<br><br>2. bridge would be the proper tool, you might have to mess around with proxy media or make sure proxy_media is off to ensure the data is coming from you directly and not negotiated between providers.<br>
<br><div class="gmail_quote">On Tue, Aug 30, 2011 at 3:57 PM, Dan Lan <span dir="ltr"><<a href="mailto:danlanweb@gmail.com">danlanweb@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>Hi, </div>
<div> </div>
<div>I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB</div>
<div>Both SIP trunks are using IP authentication, no need for username and password.</div>
<div> </div>
<div>for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in to acl.conf.xml </div>
<div>
<div><span style="font-family:Arial"><span><node type="allow" cidr="xxx.xxx.xxx.xxx/32"/></span></span></div>
<div><span style="font-family:Arial"><span></span></span> </div>
<div><span style="font-family:Arial"><span>I understand the incoming call will go to the public context, so I think I need to do something here. </span></span></div>
<div><span style="font-family:Arial"><span></span></span> </div>
<div><span style="font-family:Arial"><span>I dont know what to do next.</span></span></div>
<div><span style="font-family:Arial"><span>1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but sofia require me to have the username and password for the trunk. I dont know where to add the SIP_TrunkB in freeswitch, since the provider of SIP_TrunkB only need to recoginize my FS IP address.</span></span></div>
<div><span style="font-family:Arial"><span>2. After I establish SIP_TrunkB, how should I do on public dialplan to route the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or "bridge", could I make a dialplan that can route all the call from IP address of A to IP address of B?</span></span></div>
<div><span style="font-family:Arial"><span></span></span> </div>
<div><span style="font-family:Arial"><span>Sorry for the newbie question, I try to look up on wiki but only got partial information for me.</span></span></div>
<div><span style="font-family:Arial"><span></span></span> </div>
<div><span style="font-family:Arial"><span>Any help and any directions or hints are appreciated</span></span></div>
<div><span style="font-family:Arial"><span></span></span> </div><font color="#888888">
<div><span style="font-family:Arial"><span>Dan Lan</span></span></div></font></div>
<br><br>
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<br></blockquote></div><br>