Help with sip register please

FreePBX Endpoints
1

I am new to FreePBX. I just installed it yesterday and I signed up for a free 20 day sip trunk test through sipstation.
I have a custom made soft sip client on 192.168.3.150 and my freePBX setup is on 192.168.3.177
Given the following errors, can someone please give me some direction how to solve this? This same client software does work using the asterisk server that I have at work so it must be something with my setup. Note I do not support sip tls yet.

[2021-10-25 14:54:30] NOTICE[25359] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘<sip : 3aIgIdig6bRv @ 192.168.3.177>’ failed for ‘192.168.3.150 : 5060’ (callid: d519448-0-13c4-65014-446a2-711191fc-446a2) - No matching endpoint found

[2021-10-25 14:54:30] NOTICE[25359] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘<sip : 3aIgIdig6bRv @ 192.168.3.177>’ failed for ‘192.168.3.150 : 5060’ (callid: d519448-0-13c4-65014-446a2-711191fc-446a2) - Failed to authenticate

2

FYI, I added some strategic spaces so the forum wouldn’t think I was adding links. :grinning:

3

It is probably much easier to try and register numeric sip endpoints as FreePBX is after all a PBX so not well prepared for 3aIgIdig6bRv, you will need a predefined matching extension and authority of course

4

Man, I am so green. I tried to figure out what you meant but I can’t. So from my understanding, from: should be in the format of <sip : username @ serverIp>. Perhaps I am missing an entire step. All I did was sign up and install the sipstation. That part looks like it is running. It could also be a bug in the softphone I am using. However I think it is just probably a configuration issue because the softphone works using my work’s asterisk server.

5

Based on this, your softphone is configured to use 3aIgIdig6bRv as your extension/username for registration. It was mentioned that it may be easier to make your extension numbers exactly that - numbers only. If 3aIgIdig6bRv isn’t actually your extension number, then the softphone is configured incorrectly. So far, this doesn’t appear to have anything to do with SIPStation yet, only registering an extension to your system

6

Start with something simple and work up from there:

  1. Create an extension in FreePBX, e.g. 101, and program a popular softphone to use it (MicroSIP, Linphone, Zoiper, etc.)
  2. Confirm that the softphone registers to Asterisk ok.
  3. Confirm that you can call *43 (echo test), hear the prompt and hear your voice.
  4. Set up a second extension 102 and another device and confirm that you can call between them.
  5. Set up SIPStation, create inbound and outbound routes.
  6. Confirm that you can make and receive external calls.
  7. Set up a third extension 103 for your custom soft phone.
  8. Confirm that your custom device registers to Asterisk.
  9. Now you can test calls to and from your custom device.
1 Like
7

That’s awesome. Thank you! I was able to create an extension and use it to dial out so I am getting somewhere! I just have to figure out DID now.

8

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