So here’s my sob story since I’m a bit lost at this point.
Issue : Any calls that go through SPA3102 only have audio from the outside, audio from internal phones is not getting through SPA3102 to the external line. It also sounds like whatever the internal phone user says is echo’d back to them on the phone… The external caller hears nothing but silence.
Obviously sounds like data isn’t getting to SPA3102 or it is not routing it back out, but not sure why…
Story
I have a local internet provider that I’m getting 3 in 1 service with : Internet, Cable TV, ‘Internet Phone’. The cable modem provided has one of those “normal” phone analog plugs. (Which really behind the scenes converts to voip somewhere on there end)
Since I wanted to retain my internet phone with them, but I wanted to implement a voip system in my house, I have done the following (though I might reconsider this).
Phones
Cisco 7960 Phones - 2x (DHCP)
Cisco 7940 Phones - 3x (DHCP)
3CX Soft Phone for my iPhone - 1x (DHCP)
Switch
All network devices patch in through my Cisco 3500XL PoE switch
Router
ASUS RT-N56U
PBX/Vmail/DHCP/TFTP
AsteriskNow box running TFTP and DHCP daemons.
ATA
SPA3102
Basic Call Flow would be like …
Internal to Internal
Cisco/3CX – Switch – PBX – Switch – Cisco/3CX
Internal to External
Cisco/3CX – Switch – PBX – Switch – SPA3102 – “Analog Phone port” (FXS?)
External to Internal
“Analog Phone port” (FXS?) on cable modem – SPA3102 – Switch – PBX – Switch – Cisco/3CX
When I orginally put everythin together months ago, I had some initial problems with ATA and it had to do with having to enable the “lan” and “wan” sides of the
device even though I did not need them.
After enabling that, everything has worked fine, for months.
The only recent changes were that we received a new cable modem a couple weeks ago; however, it worked fine for those past couple weeks with the new modem, nor can I seem to understand how it would matter that much…
Dedicated external IP address changed a couple weeks ago as well, but it worked fine regardless. I did today, during troubleshooting, update the one spot I could see the old address referenced.
Things I have checked :
I have checked that all devices can “see” each other through ping and tracert confirms the devices are not hopping through any routers or gateways so I don’t think this some type of port blocking type issue.
For entertainment value, I have rebooted all devices “just in case” with no change.
Checking the FreePBX log reveals an occasional error:
[Aug 18 15:23:51] VERBOSE[5126] netsock.c: == Using SIP RTP CoS mark 5
[Aug 18 15:23:51] WARNING[5126] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Aug 18 15:23:51] VERBOSE[5126] netsock.c: == Using SIP RTP TOS bits 184
Researching this appears to indicate there is an extension that isn’t reachable which is reasonable as I have extensions defined that are not active at the moment. I DO NOT recall seeing this when I first setup the system so I’m not 100% sold on that.
I’m hoping someone may have experienced this and can give me the 30 second answer; otherwise, I think my next step will be to bust out a packet sniffer to see if the packets are being sent (and to where)
If anyone wants to help directly, I could setup a webex session, etc.
This is my first PBX build so I’m definitely open to assistance.
Thanks!