Can't receive inbound calls

I am getting my INVITE messages from my provider but I can’t seem to point the DID anywhere.

My provider uses one IP for outbound calls which I have successfully setup. They also provide another IP for inbound calls. I have that inbound IP listed on the same outbound truck under the “Incoming Settings” under user details.

chan_sip.c: Call from ‘VI_in’ (xx.xxx.xxx.xx:5060) to extension ‘1520xxxxxxx’ rejected because extension not found in context ‘default’.
[2015-07-20 16:53:33] VERBOSE[8492][C-00000010] netsock2.c: == Using SIP RTP TOS bits 184
[2015-07-20 16:53:33] VERBOSE[8492][C-00000010] netsock2.c: == Using SIP RTP CoS mark 5
[2015-07-20 16:53:33] NOTICE[8492][C-00000010] chan_sip.c: Call from ‘VI_in’ (xx.xxx.xxx.xx:5060) to extension ‘19036382948’ rejected because extension not found in context ‘default’.

How do I troubleshoot this further?

Thanks!

Reject because extensions not found in context default

:slight_smile:

FreePBX do not use default extension, use from-pstn or from-trunk.

Where would I add the extension? Can I resubmit it or add it manually to the context default file? How do I get my extension to accept an inbound call?

Thanks!

Generally you should use from-pstn as the context for your inbound trunk in FreePBX , did you read the wiki?

The TRUNK accepts the call and sends it to the endpoint you should have defined for that DID, that endpoint might be an extension.

Thanks for the direction Dicko,

I am in and have been reading wiki like crazy. I swear I can’t find anything anywhere (even on Google) about the “Incoming Settings” which I am not sure about what it does or how it handles incoming calls from another host setting.

I guess I need endpoint manager module :slight_smile:

And I swear you can

http://wiki.freepbx.org/display/F2/Trunk+Sample+Configurations

and

http://wiki.freepbx.org/display/HTGS/Configuring+your+PBX

(I would have to say that the second link is a little egocentrically written as it only covers SIPStation trunking, but the first link will fill in that gap)

For configuring your endpoints you might well need an endpoint manager, but really all you need to do is setup the registration server, extension and password on the phone.

I still have my face buried in the wiki - the same links you sent me…thanks for the info.

I still cannot find any detail on, specifically, the incoming settings on the trunk page. I searched to find from another site, adding the provider host info and FPBX connection parameters in the Incoming Settings USER Details might help with my incoming call ‘trunk’ but that is all the info I have on it. Apparently, the “Peer Details” and “User Details” can contain the same information with the exception of the provider host or IP to designate the outgoing and incoming settings/call.

It looks like the “from-pstn” did the trick as I had “default” in the context.

Dicko, what did you see that said that to you. I am trying to learn quickly, effectively and efficiently. I have been at this for a long while. I enjoy troubleshooting and finding out for myself, when I hit a wall, I have to ask…I hate to ask because I know the answer is usually out there somewhere. I am very good at finding informational nuggets. Your help is much appreciated .

SIPstation interests me but I need to work with what I have for the moment. I enjoy a challenge that has no corporate pressure. Endpoint looks like it will suit me well and ease things a bit but I will have to wait before I invest more bux into this. I want to figure what modules will help me best or which ones I may need first. A bit of a learning curve, I will get there.

I now have some codec issue on a Grandstream BT101 but my UVP (Ubiquiti) kicks butt.

All the best!

By the way, there is no way I could have gone this far without the wiki. It is my FPBX bible, extremely helpful!

It is often less confusing to build two trunks to your provider one called provider-in and one called provider-out, instead of confusing yourself with a single bi-directional trunk send calls through your outbound route(s) to provider-out and land your DID’s from provider-in on appropriate endpoints, IWFM, for general background to how SIP works, many find

http://www.voip-info.org

gives a good solid background for “tips and tricks”.