Newbie Help Please - SPA3102, POTS / PSTN and Freepbx

Hi folks

I’m really struggling with the most basic bits of the SPA3102 setup, would you be kind enough to give me a few pointers?



Happy to provide more info if it helps - just can’t get the pots / pstn to register with the trunk

There are many threads in this forum and elsewhere about SPA3102, the almost identical SPA3000, and similar devices (HT503, HT813, OBi110, OBi212).

If you can’t find your trouble, please post:
Is device on same LAN as PBX?
Using pjsip or chan_sip?
Also using the Phone (FXS) port on the SPA?

At the Asterisk command prompt, type
pjsip set logger on
sip set debug on
then reboot the SPA so it attempts to register. Report what, if anything, appears in the Asterisk log on a failed registration. Also post trunk settings and all non-default PSTN Line settings.

Hi Stewart

Thanks so much for coming back so promptly.

I have spent the last three days trying to get the pstn to register - was also up to 2:30am last night. It is with a heavy heart and only after having googled myself near blind that I have to admit defeat and seek support.

I will set the logger running now - but while I do that here’s my situation - thanks again for your support

My setup is:


I want to:

  • Use only the PSTN/POTS for inbound and outbound calls
  • Use only one single analogue phone as one single extension
  • Inbound and outbound CID to register on FREEPBX

On the SPA3102:

  • LINE1 on port 5160 is registered
  • PSTN on port 5062 will not register and this is my big problem

Questions I have:

  • What are the absolute basic settings to get a FXO registration
  • Does the PSTN line need to be registered if I want inbound and outbound CID to work
  • Should I use PJSIP or SIP
  • Do I need to set anything up on the SIP page of the 3102
  • I know I have to setup inbound and outbound routes ultimately - but do I need to set them up just to get the POTS to register

To help I will post shots of my SPA3102 and Freepbx setup - this is just my test environment, I will change all passwords, numbers and IP’s once I move site to install :wink:






Hi Stewart - can you PM me - is that possible?

From a quick look, on the PSTN Line page, change Proxy to
and in the trunk PEER Details, delete the line

If it still won’t register, turn on sip debug per earlier post, and report what gets logged on a failed attempt.

I don’t know whether there is a technical problem with the forum or you started multiple threads with the same title, but please keep all your posts about this topic) in this thread, by using the Reply button.

A SIP packet (like any UDP or TCP packet) has a source IP address and port number, and a destination IP and port. To be received and properly processed, the destination IP/port must match the IP of the remote device and the port it is ‘listening’ on. On the SPA, the listen ports are set by the SIP Port parameters for Line 1 and PSTN Line. They must be different, but can be anything, because the PBX will learn those values when the device registers. The default values of 5060 and 5062 are fine.

The port specified by Proxy (the number after the colon) must match the listen port of the PBX. If you haven’t changed them, a recent FreePBX has pjsip listening on port 5060 and chan_sip listening on port 5160. So, if both the extension and the trunk for the SPA are using chan_sip, then the Proxy value for both Line 1 and PSTN Line should be
and the PBX will distinguish the registrations by their different source port numbers and user IDs.

If you had set up the Line 1 extension with pjsip and it’s working fine, there is no reason to change that – leave the Proxy setting for Line 1 alone.

Hi Stewart

Apologies for the messy posts yesterday - for some reason I wasn’t allowed to reply only post new messages.

Am pleased to say I finally managed to get both the line and the pstn registered - thank you so much for your help :slight_smile:

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.