[Freeswitch-users] Connecting Call Between Two SIP Trunks

[Brad Mina]

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 at 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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