Trying to use OBI202 as a Proxy to GV from Freeswitch
ubergoober:
I'm attempting to adapt the instructions for letting an IP phone proxy thru the OBi202 device to make outgoing calls via Google Voice. Unsuccessfully.
I've seen other instructions for freepbx and the two variations for the 100/110 devices.
Here's how I'm configured:
OBi202 in Bridge Mode, connected to router via the LAN port.
ITSP Profile D/General -> Signaling Protocol: SIP
ITSP Profile D/SIP -> ProxyServer: IP Address of FreeSwitch Server
ITSP Profile D/SIP -> ProxyServerPort: 5080
ITSP Profile D/SIP -> Registrar Server: blank
ITSP Profile D/SIP -> OutboundProxyPort: 5080
Voice Services/SP4 -> X-ServProvProfile: D
Voice Services/SP4 -> X_InboundCallRoute: {17935>(Msp1):sp1}
Voice Services/SP4 -> X_RegistrarEnalbe: blank
Voice Services/SP4 -> X_UserAgentPort: 5080
Voice Services/SP4 -> X_Proxy: checked
Voice Services/SP4 -> AuthUserName: 17935
Voice Services/SP4 -> AuthPassword: matches Freeswitch Gateway Password
Voice Services/SP4 -> URI: 17935@IP Address of Freeswitch Server
My gateway does register, but only after putting the number of the obi device@pnn.obihai.com in the realm field of my freeswitch gateway.
When I offer a call to the OBI, SNGREP shows an invite, 401 unauthorized, ack, invite (sdp), 100 trying and finishes up with 486 Busy Here.
RTP shows it's attempting PCMU/8000, so I don't expect a codec issue.
Wondering what else is required for the OBI to want to route this call via SPi.
I'm sure somebody smarter than me must have done this before. Thanks in advance of any help.
Tom
restamp:
I've got several OBis serving as ITSPs for FreePBX Asterisk servers, and am in the process of documenting what I've done, but it's going to take a few days since I'm tied up with other things at the moment. I'm not familiar with FreeSwitch, but a couple comments:
I presume your FreeSwitch is external to your local LAN and that it has a static IP address. OTOH, the OBi is behind a NAT on the LAN, right? Here's the catch: If the OBis external IP is a dynamic one, which is most likely the case, *it* has to register with server and not the (more traditional) other way around.
Why do you need a URI?
That said, it looks like you've got the two talking and SIP almost working. I presume SP1 is GV and SP4 is SIP. What's with the {17935>(Msp1):sp1}? Just for grins, what happens if you just set X_InboundCallRoute to simply {sp1}, uncheck X_Proxy, and get rid of the URI? If that works, set the GV X_InboundCallRoute to {sp4(17935)} (or whatever number you have assigned to your GV trunk) and see if you can call in.
I hope to get my notes to the point where I can put them online by next week. They are based on Asterisk, but if you haven't gotten things up and running by then, they may help give you some ideas.
Good luck.
ubergoober:
Restamp. Good advice. Your suggests have resulted in a successful call.
The Freeswitch server is in the same network here at home behind the firewall. The URI is something I tried based on a confluence page on the freeswitch website referencing obi1x0 devices. It turns out to be a red herring. I'll report back to this thread if my experience is in any way unreliable.
Thank you!
Tom
ubergoober:
I spoke too soon. The phone selected a line I had not expected. My calls fail with a 480 temporarily unavailable.
restamp:
If your freeswitch and OBi are on the same LAN, there is really no need for Registration and/or Authentication. Hard code the IPs and rely on them for authentication, it's really much simpler. Give the OBi a static IP and follow the instructions here:
https://www.dslreports.com/forum/r31954649-
(Also check X_SpoofCallerID in the SIP ITSP Profile.)
This is the stuff I'm cleaning up to document, but this worked for me as is. I have no idea how to translate the Peer Details to FreeSwitch-speak, but you probably do. Also make sure the Freeswitch offers ulaw (and opus, too, if it has it).
Good luck.
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