What are my Failover Options?
Our Failover Service allows you to provide us with alternative delivery phone numbers and/or Backup PBXs in the event that we can't deliver a call to you.
If you do not have failover service set up, we try to send your calls to you from trunk1.freepbx.com. If you are not registered, or the attempt otherwise fails to reach your PBX, we then try sending the call to your from trunk2.freepbx.com. If that doesn't work, we give up.
Failover can prevent this from happening. If you set up an alternative phone number and/or backup PBX as a failover option, this will give us another way to deliver your call.
I have set up your service and am getting one-way or no audio. What do I do?
This is usually the result of not having your ports forwarded. Please see our wiki "Configuring your PBX or device with SIPStation Service," which addresses port forwarding. If you previously had service working and this has recently started to occur, check your externip/externhost settings if using Asterisk. (The Asterisk SIP Settings module controls these if using recent versions of FreePBX.)
Can you help me with internal configurations of my phone system?
Our support is limited to helping you get configured with our SIPStation SIP service. Because of the complexities and differences between customer environments, we are not able to help you configure your internal networking, firewalls, and other related items. FreePBX.org has excellent paid support available when you need an engineer to get onto your internal system and help you with most aspects of your FreePBX/Asterisk-based PBX, phones, and networking. Details can be found at the Support and Professional Services Page.
I can make outbound calls but inbound calls are failing.
You must register with us to receive inbound calls. We deliver the inbound calls as 10-digit DIDs. If you are receiving the specific message: "The number you have dialed is not in service, please check the number and try again," then the call is getting delivered to you, but you probably don't have a proper inbound route set up for the DID. This message is generated by Asterisk. When that occurs, it is the ss-noservice recording.
I get an error that says "noserver" when trying to use the FreePBX SIPSTATION module.
This error is often the result of some very aggressive firewalls with content filtering or equivalent. The Sonicwall is a common one that gives this problem, and you may see it creating problems for Module Admin online access in FreePBX as well. (If this is the case, the content filtering will have to be disabled, or else the PBX or our server target will have to be configured to bypass the filtering.) This error means that the module did not get a response from our servers when trying to obtain your account information. If you can get to your account in the SIPStation Store, then the server is available and the problem is on your side. Other causes could be DNS issues not resolving the hostname. We have had reports from Trixbox/Fonality users that rebooting their system resolved the issue, which could be the result of a first-time installation that did not complete.
I am having trouble using SIPStation behind a PFSense firewall.
This advice may be helpful:
NAT -> Outbound -> Manual Outbound NAT rule generation (AON)
Interface-Source-Source Port-Destination-Destination Port-NAT Address-NAT Port-Static Port
WAN 192.168.0.0/24 * * 500 WAN address * YES
WAN 192.168.0.0/24 * * * WAN address * YES
WAN 127.0.0.0/8 * * * WAN address * YES
—> Check the box that turns off PF scrubbing.
—> Reset the states.
How do I take a packet capture for analysis?
This advice may helpful:
tcpdump -s 3000 -w ~/freepbx-sip.pcap port 5060 or portrange 10000-35000
Initiate the above before an errant call and stop it with a ctrl-c, then download the file to a PC for analysis with Wireshark.
My question is not answered here. How can I contact support?
SIPStation support is available to registered users at https://support.sangoma.com