I am in need of some guidance. I’m having issues with voice audio for ring groups.
I created a ring group with two extensions. I configured sipstation so that when the main number is called it routes to the ring group. Both phones ring at the same time, but when they’re answered there is complete silence. The caller cannot be heard and the caller cannot hear us.
Has anyone else had a similar issue?
Sounds like you have a firewall issue with your trunking. Do you have ports 10000-20000 open for the RTP ports from SIPStation?
No sir, we’ve not opened up RTP ports on the firewall. The reason it’s not setup at this moment is because of our current setup.
- We have an MPLS network with CenturyLink. They also manage the cloud firewall. Behind the firewall we have three sites each with their own pbx/SipStation. I’ve attempted before to port forward both port 5060 and RTP ports to our main site freepbx, but I was unsuccessful getting it to work. I’ve talked with support from CenturyLink and SipStation and they were in agreement they’ve never seen a setup like this so they told me to assign a different outside IP address to each pbx and then port forward that way. However I’ve tried that, but SipStation only sees the CenturyLink oustide IP on the CenturyLink firewall and I’m unsure how to tell it to see the external IP I’ve assigned the pbx even after configuring SIP settings.
SIPStation can force itself out a interface. its just a SIP trunk in Asterisk and Asterisk just sends the traffic to your kernel. Its all in how you setup your Linux networking and routing tables in Linux. Dual WAN interfaces are tough and require you to look into static routing.
We were talking about this extensively last week. Look for “Dynamic Host” in the history and see what you can find.