To clarify, I have been testing the connection by attempting calls to and from my mobile phone. If I use my mobile to call the number linked to the VOIP account that the SPA112 connects to, the call is received fine and the call works perfectly; I can hear voice at both ends fine.
If I attempt to call the mobile from the phone behind the SPA112 I immediately get a reorder tone as soon as the number is dialed by the SPA112 (i.e. immediately if it matches a number in the dial plan followed by S0 or after the short wait period otherwise etc.)
On the other hand, if I use a softphone (I've tried MicroSIP and Blink) I can place calls to the same mobile through the same VOIP account from the same LAN behind the same NAT as the SPA112, and the calls are received and bidirectional voice comms works just fine.
I'm not sure if I can make calls internal to my LAN with this setup... I only have the one VOIP account, and have no SIP server on the LAN (if that's the right terminology, I'm new to this).
Unfortunately I'm not sure what you're refering to when you talk about ACL and COR. The topology is very straight forward: the SPA112 (along with a couple of PCs) on a LAN behind a NAT router with an internet connection on the other side. The SPA112 is configured as a DMZ, though I don't believed this had any impact for better or worse. In no test for either SPA112 or softphone have I forwarded any ports or changed any router configuration at all (with the exception of the DMZ).
So the question is why the softphone can place calls fine, but the SPA112 can only receive calls. The dial plan is an obvious answer, but I'm fairly certain I've ruled this out.
If someone can tell me how to get more detailed logging out of the SPA112 that would be very useful as well.
no direct recommendation as to your issue, but I would be sure you're running the latest firmware. We just installed one of these units at our office for use with a fax machine, and had nothing but problems until updating the firmware. Since then it has been smooth sailing.
Thanks to your comment about syslog I've gotten much better logging happening by setting my PCs ip as the "debug server" and monitoring it with wireshark.
Comparing the SIP logging to a successful call from a softphone (monitored using wireshark again) it would appear that my problem is that INVITES are being placed to "XXXXXXXX@my.sip,proxy.com" as opposed to "XXXXXXXX@my.sip.domain.com" as it were i.e. my proxy server is being used as the SIP domain, which is incorrect for my VOIP provider.
To resolve the problem I had to configure the "proxy" as the SIP domain, and the "outbound proxy" as the actual proxy and always use the outbound proxy as I could not discover a way to specify the SIP domain anywhere in the configuration.
Seems like a bizarre solution but my setup is now working fine.
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