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. 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. On Tue, Aug 30, 2011 at 3:57 PM, Dan Lan <danlanweb@gmail.com> wrote:
Hi,
I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB Both SIP trunks are using IP authentication, no need for username and password.
for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in to acl.conf.xml <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
I understand the incoming call will go to the public context, so I think I need to do something here.
I dont know what to do next. 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. 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?
Sorry for the newbie question, I try to look up on wiki but only got partial information for me.
Any help and any directions or hints are appreciated
Dan Lan
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