@stephendt wrote:
Hey guys, been tearing my hair out over this one. I originally gave this a go a couple of months ago with no success, so I decided to swap over my ATA today and give it another go. I originally tried with an SPA3102, and am now trying with an SPA3000 and am getting the same results, so I'm thinking it's more to do with the asterisk configuration more than anything.
I currently have a PSTN line that cannot get ported to a SIP provider for various reasons and still receives a number of inbound calls. It's never used for outbound calls so I don't have an inbound route (we use the new SIP phones for that), but I'd like these calls to ring on our various IP phones.
I've followed this guide to the letter: http://wiki.freepbx.org/pages/viewpage.action?pageId=55476525. I'd post screenshots but it'd be a waste of time, I literally have the same configuration - I've triple checked. I get the feeling this guide is incomplete or out of date, because it just isn't working on my FreePBX 13.0.188.8 server.
Essentially what is happening is when I dial my PSTN number, my desired softphone doesn't ring.
I had a quick look at the asterisk console output when I try to dial in and I'm getting this:
[2016-10-08 17:14:48] NOTICE[5578]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"pstn" <sip:XXXXXXXXXX@192.168.1.47>' failed for '192.168.1.25:5062' (callid: 29189091-56f96099@192.168.1.25) - No matching endpoint found [2016-10-08 17:14:48] NOTICE[5578]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"pstn" <sip:XXXXXXXXXX@192.168.1.47>' failed for '192.168.1.25:5062' (callid: 29189091-56f96099@192.168.1.25) - No matching endpoint found [2016-10-08 17:14:48] NOTICE[5578]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"pstn" <sip:XXXXXXXXXX@192.168.1.47>' failed for '192.168.1.25:5062' (callid: 29189091-56f96099@192.168.1.25) - Failed to authenticate [2016-10-08 17:14:48] NOTICE[5578]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"pstn" <sip:XXXXXXXXXX@192.168.1.47>' failed for '192.168.1.25:5062' (callid: 29189091-56f96099@192.168.1.25) - No matching endpoint found [2016-10-08 17:14:48] NOTICE[5578]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"pstn" <sip:XXXXXXXXXX@192.168.1.47>' failed for '192.168.1.25:5062' (callid: 29189091-56f96099@192.168.1.25) - Failed to authenticateXXXXXXXX = my mobile, 192.168.1.47 = FreePBX13, 192.168.1.25 = my ATA.
It's almost as if the inbound route has no idea where to send it - even though I've specifically told it to route to a particular extension. The extension works fine otherwise. The destination extension is using pjsip whilst the trunk is using chan_sip, but I don't think this would be a problem.
If anyone could give me some clues that would be really appreciated.
Posts: 1
Participants: 1