I’m running FreePBX FreePBX 14.0.13.26 / Asterisk 13.32.0. All modules are up to date. I’m having a problem with audio not working with an incoming call from a SIPSTATION DID. However the audio works perfectly with a call from a VoIP.ms DID.
Note that I have a pfSense firewall in front of the LAN. With the VoIP.ms DID I do not have to open up any ports on the pfSense firewall, and the audio works with outgoing calls as well as calls to the VoIP.ms DID. To get the SIPSTATION External Connectivity–>Check Connectivity to pass the SIPSTATION Firewall Status test, I have to open up UDP ports on the firewall.
For example, let’s say the SIPSTATION DID is 604-222-1001 and the VoIP.ms DID is 604-222-1002, and I set up an Inbound Route with the destination for the 604-222-1001 SIPSTATION DID to be extension 1001, and an Inbound Route with the destination for the VoIP.ms DID to be 1002. Note that extensions 1001 and 1002 are on the same phone, only different lines.
When I make a call through the PSTN (say from my mobile phone) to 604-222-1001 I get no audio but the call from my mobile phone to 604-222-1002 works perfectly.
According to the Configuring your PBX or device with SIPStation Service wiki at https://wiki.freepbx.org/display/ST/Configuring+your+PBX+or+device+with+SIPStation+Service
We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk-based installs. If using newer versions of FreePBX, port 5160 is the default port for ChanSIP so that may be the port you need to forward. Check Asterisk SIP Settings for the bind port of ChanSIP. It may be possible to get your service working without port forwarding, but optimal service will be obtained with the above mentioned ports. You can lock down port UDP/5060 or UDP/5160 depending on bind port of ChanSIP to the trunk1.freepbx.com and trunk2.freepbx.com FQDNs for additional security, but please note, we do from time to time change the IP addresses associated with these FQDNs. Therefore it is best to use the FQDN and not an IP Address. You cannot lock down UDP/10000-20000 to any specific IP address, since the media of a phone call can come from hundreds of different IP addresses.
I have opened up port 5060 in the pfSense firewall and locked it down to trunk1.freepbx.com and trunk2.freepbx.com.
Here are the registration strings (credentials changed) in sip Settings–>Incoming on trunks for both SIPSTATION and VoIP.ms:
YksAbcef1234:[email protected]
YksAbcef1234:[email protected]
12345_user:[email protected]:5080
12345_user:[email protected]:5080
Reports–>Asterisk Info–>Registries–>Chan_Sip Registry shows:
Host dnsmgr Username Refresh State Reg.Time
trunk1.freepbx.com:5060 Y YksAbcef1234 105 Registered Fri, 27 Mar 2020 07:39:04
vancouver1.voip.ms:5080 Y 12345_user 105 Registered Fri, 27 Mar 2020 07:39:02
vancouver2.voip.ms:5080 Y 12345_user 105 Registered Fri, 27 Mar 2020 07:39:02
trunk2.freepbx.com:5060 Y YksAbcef1234 105 Registered Fri, 27 Mar 2020 07:39:04
4 SIP registrations.
However, I am still not able to receive audio when dialing the SIPSTATION DID from PSTN, and I am receiving audio when dialing the VoIP.ms DID.