I have a FreePBX system with a Grandstream GXW4104 using 2 analogue lines (connect from traditional PBX with extension 8495 and 8496). The Grandstream is registered and it works perfectly on outgoing calls.
On incoming calls it receives 1 call from FXO1 and transfers it to Ring group but it doesn’t accept any more calls on the other lines( FXO2,FXO3 and FXO4). The caller gets a ringing tone but no incoming call show on the sip phone. I can make outbound call normally in both 2 line.
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
901 (Unspecified) D No No A 0 UNKNOWN
902 (Unspecified) D No No A 0 UNKNOWN
906 (Unspecified) D No No A 0 UNKNOWN
GXWT1 10.18.30.194 Yes Yes 5060 OK (2 ms)
GXWT11 10.18.30.194 Yes Yes 5060 OK (2 ms)
gxw4104 10.18.30.194 Yes Yes 5060 Unmonitored
6 sip peers [Monitored: 2 online, 3 offline Unmonitored: 1 online, 0 offline]
I have a small server that running on Elastix and Zap16 voip card. I connect analogue line from traditonal PAPX to Elastix. But that server is very old and i need to build a new once. I select a Freepbx and Grandstream GXW4104.
I follow this guide, install freePBX 11. Create chan_sip trunk, and still get only 1 incoming call forward to Voip. Can i confirm that “unconditional call forward to voip” on the gateway is PSTN number ?
Are you expecting a second call on the same pstn line to roll over to a different line if one of your pstn lines is being used and somebody calls that same line?
For that to happen, your pstn provider must configure your lines into a hunt group.
Aside from that, you need to check that the unconditional call forward to voip is correctly set, that parameter is used to send the calls to freepbx. If you have an inbound route, you can use the number 7777, which by default is the feature code to simulate an incoming call.
Firstly, my english is not good, so sorry about that. Let me explain you the topology
Traditional FreePBX have 4 line analog with different extension (8491,8492,8493,8494) connect to GXW4104 and FreePBX. We set 4 extension in 1 one ring group 8999 in Panasonic PAPBX, when user call to 8999, it will forward to sip phone in FreePBX.
I guess something wrong with antivirus on my pc that using at company, because the Status off Sip Registration on GXW4104 is “No” but i try at home with the same setting on FreePBX and GXW4104 is " Yes". I will test it again when i back home and let you know. I will do manual FreePBX and GXW4104 when it work
Finally, It worked after install on real server ( Super Micro ). I guess McAfee anti virus on my pc block the connection between FreePBX (VM) and GXW 4104. All of incoming call and outbound call work well. But I need to create both Sip Trunk and pj chan sip trunk. At unconditional Forward to voip, User ID is set Ring group extension (ch1-4:600)