New User, 1st post, 1st time use

Hello everyone.

1st post up on here and also the 1st time I have ever ventured into the world of PBX, so please be gentle :slight_smile:

Will start off with a bit of background as to why I started to use FreePBX and what my final intentions are.

In my current office we have two analogue phone lines linked in from the main site telex, but the problem is is needs to be in 5 different places (3 desks and 2 labs). We do have a LAN in here we can piggy back onto so my thought and final aim was to have a VOIP ssytem set up on the LAN that then utilises the two analogue lines to interface us to the rest of the world (When we want to that is ;-))

The setup will be a Cisco 2621 with a 2 port FXO-EU card fitted > Netgear 8 Port PoE Switch > Rasp Pi with FreePBX connected to one of the non powered ports > IP Phones connected to the PoE ports.

Am wanting this to be able to allow for ringing between the extensions that we set up and also so that we can receive incoming calls from the analogue lines (via Cisco and FXO) and also dial out on to the lines too.

Phojnes wise, at present, I only have a Cisco 7942G that was inherited from a previous job. I have sucessfully managed to get the SIP firmware onto this and also appear to be able to change the config using the relevant xml file for the handset.

My main issues are that I cannot for the life of me seem to get the phone to register onto the system.

As this is my first foray into this, do not want to try and do too much at first, I want to be able to get the handset registered and be able to make internal calls. The FXO part will come a bit later on when I have got used to the system.

Is there somehting fundamental that I am missing in my noobness here? Have created an extension and that seems to go fine, but am unsure as to what parameters I need to change in the phones xml config so as to get it to register (main one is I cannot find the place to put in the ‘secret’ part)

Apologies for slight Wall of Text on my first post and also the fact it may be blindingly obvious to sort out, but am pulling out what little hair I have left here.

Many thanks in advance to any and all comments, suggestions and ideas and am looking forward to learning more and contributing more to the community here

Thought I would put an update on as to how things are going.

Probably very noob errors, but as I mentioned, is all new to me.

Had a Polycom 501 turn up to use to, so could make 2 way calls when I resolved my issues.

My main problem was pointing the phones at teh wron ip address for TFTP server. I did a rebuild on the distro I have and then updated online and also downloaded the End Point info for the phoines I will be using.

After reconnecting and changing the TFTP server address for each phone they are now registering and can make 2 way calls, have on hold musak and voicemail appears to be functioning too.

Next step is to configure the Cisco router and then after that it is the daunting stage of interfacing the PBX to the router so that we are able to ‘dial out’

Another week passes and another update.

Now have FreePBX up and runnig a treat for internal calls and also have the Cisco configured to provide DHCP and NTP.

VIC FXO installed into the router and recognised and configurable.

Next steps as follows

1 - Configure FXO card
2 - Have PSTN calls forwarded to Rasp Pi and FreePBX
3 - Have calls sent from Rasp Pi and FreePBX via Cisco to PSTN
4 - Work out why the 7942G and 7940 are not picking up correct time and date when Polycom 501 is (may be template config issues here)

…and only a bit closer to what I am trying to achieve.

Have given up using the Cisco as the FXO interface as this was proving very problematic and am now using a SPA3000, which is offering it’s own selection of issues.

Seem to take one step forward and several steps back on this one.

Have sucessfully set-up the SIP trunk between the SPA and FreePBX but when ever I dial in, the selected extension (or ring group in my case with all extensions in) rings fine, but am unable to answer the call. The moment you pick up the handset the phone appears to drop the call, then it starts ringing again.

Am still unable to make any outside calls onto our on-site exchange but feel this may be a dial plan issue, which I have no idea of resolving.

For my situation I need 3 digit numbers kept on the FreePBX, 4 digit numbers passed as they are to the FXO-PSTN line and also need to keep the functionality where we are able to dial out of our onsite exchange to national numbers. At present this is achieved by prefixing the national number with 0.

I know it is a long shot, but is anyone able to offer some assistance on these final issues I am having?

Cheers in hopeful advance