PROBLEM:- I have configured the SIP settings with a new SIP provider but the call is not getting established when dialing onto the audio conference bridge, Audio conference bridge provider claims that they have received your call.
REASON:- Your Adobe connect server/ Flash Media Gateway(FMG) is deployed behind NAT and its domain /IP is not routable on internet, When you dial into the audio bridge through SIP , SIP header carries the local/internal IP address of FMG/Connect server and remote SIP provider does not have any idea how to reach to the local/internal IP address.
SOLUTION :- There are two ways to fix this
1. Adding Global Address tag in sip.xml file.
Add/Replace the below tag on sip.xml file on profileID sipGateway
<globalAddress>your-connect-domain</globalAddress>
your-connect-domain could be a domain name , IP address which should be routable on internet.
Example
<Profile>
<profileID>sipGateway</profileID>
<globalAddress>your-connect-domain</globalAddress>
<userName>101</userName>
<password>101</password>
<displayName>sipGateway</displayName>
<registrarAddress>10.10.10.10</registrarAddress>
<doRegister>true</doRegister>
<defaulthost>10.10.10.10</defaulthost>
<hostPort>0</hostPort>
<context>sipGatewayContext</context>
<supportedCodecs>
<codecID>G711u</codecID>
<codecID>speex</codecID>
</supportedCodecs>
</Profile>
2. Deploy FMG on another server/box having a public domain/ IP access.
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