I’m able to receive incoming calls but unable to make outbound call. I checked the “run firewall test” in the SIPSTATION account access panel and the status comes back as fail. I thought that a firewall problem would be on the inbound side but not on outbound. Is this a configuration issue with PBX in a flash setup?
Thanks for the quick reply. I have tried several outgoing calls to various numbers and the X-Lite soft phone does show “Call established” and I hear a voice that says "please wait while I connect your call and background music.
When I dial a bad number I do get the beep tone and voice message “the person you are calling is unavailable, please try again”
So, I’m connected but it sounds like I’m on hold with the background music at the number that I have dialed. I call my cell phone number but it doesn’t ring.
I have read all docs that appear to be related but have not found a solution. Any ideas on what I may be missing?
The firewall test is a fairly simple test.
The test works on most straight forward network configurations but it is possible to fail in some cases even though you are properly configured. It tries to indicate that when it fails.
The purpose of the test is to check if you have your RTP ports (usually 10000-20000) properly forwarded. It does this by finding an unused RTP port within the range of currently configured ports (as configured in rtp.conf) and then sending an http request to a server which turns around and send a packet back to the address the http request came from that looks like a single RTP UDP packet (in ulaw format).
After sending the http request, it then listens for that reply and makes sure it came back properly.
There are at least a couple of things that can make this test fail even if you are configured correctly. First, if you have multiple WAN, depending on your configuration, the packet may be sent back to the WAN that you don’t use for your SIP traffic and fail. Another situation is if you have an SIP aware firewall that sees this packet correctly as a rogue packet and blocks it.
So … although the test probably works on 99+% of FreePBX installations, it is not perfect. If you know you have your proper RTP and SIP ports forwarded to your PBX, then you should not worry about this test.