I want to connect some analog telephones to FreePBX using an ADIT 600 with a CMG Router. According to most of the documentation online, the CMG Router only work with MGCP. I think that info is dated, because the CMG Router I got from eBay has software version 4.01 (released 2006) that supports SIP, and I believe PJSIP at that.
I managed to get the phones talking to FreePBX. The *43 loopback test works and I can leave voice messages for extensions. I just can’t make any of the phones ring. They immediately go to voicemail. I’ve futzed around quite a bit with FreePBX and the ADIT, but I haven’t figured out why. DND is not enabled on any of the lines. I did notice under Reports → Asterisk Info in the PJSIP peers section, the extensions seem to have found the IP of the ADIT router, but they all show as unavailable.
I appreciate any help, thanks! Even if you could point me to a page in the manual where I should start reading, I would very much appreciate that.
You go channel unavailable before the INVITE could be sent. That means one of:
PJSIP/002 is not defined;
It has failed to register;
It has registered, but the connectivity test, using OPTIONS, is failing.
Querying the endpoint status, with the CLI, should distinguish between these cases (I think it is “pjsip show endpoint pjsip/002”).
In case 3, it could be that there really is no connectivity, in which case you need to check addresses and trace the request and response to where one gets lost. or it could be that the CMG’s handling of bad requests is broken (they should still produce a response, which is all that is needed). In the latter case, you should disable the qualify test.
I ran pjsip show endpoint 002 from asterisk cli on the web interface and got a big old block of text. I think this is the important part:
Endpoint: 002/002 Unavailable 0 of inf
InAuth: 002-auth/002
Aor: 002 1
Contact: 002/sip:[email protected]:5060 9e277741ad Unavail nan
I went into extension 002 and set the qualify frequency to 0, and tried again. This time I got ringback tone, but the other phone did not ring. This is what the command brought back:
Endpoint: 002/002 Not in use 0 of inf
InAuth: 002-auth/002
Aor: 002 1
Contact: 002/sip:[email protected]:5060 9e277741ad NonQual nan
That suggests that the there isn’t a complete round trip path; you really don’t have connectivity. You’d need to use the CLI command “pjsip set logger on” to check requests were being sent to the right address and had correct Via and Contact addresses, and if so, you would need to trace them through the network, to see where they got lost.
I ran the port monitor in the ADIT, and I could see the invites coming in, and it seemed to ignore them. It also seems to ignore the options / qualify messages as well. I think FreePBX tries to ring the phone at line 394. I’ll have to see if I can find any helpful documentation for the CMG Router. It might be that I need a special feature key to make SIP work properly. Thanks for the help! I’ll be sure to give an update if I make any progress.
The main log on the ADIT, as well as the log in the CMG router didn’t have anything related to the calls. There was an alarm that came in, but then cleared once the CMG registered with FreePBX, but that’s as detailed as it gets. I ran the port monitor again, and saved that here:
I also ran print config for the router in slot 6 and for the overall unit:
It looks like the ADIT supports syslog, so when I get the chance, I’d like to setup a server to catch all that data and see if I get anything more detailed.