I am working on my Linksys bringing in a line. Into Linksys and be answered by freepbx. I searched the topic and found one but it was not as clear as the information I received here. I did understand that the registration should be done on the Linksys side not freepbx. I guess I need to use both of the ports for this operation correct?
I have the sip running for nextiva, On Linksys PAP2 line 2 Im bringing a pots line into freepbx. I have followed all protacals Im registered to freepbx by extention which seems that I need to make a sip channel like before. When I call the number I have enabled the pjsip set logger on it does say IP/2.0 400 Missing Contact header. Could you please help me with this?
The Linksys PAP2 using NOTIFY by default instead of OPTIONS for the keep-alive. Asterisk doesn’t accept NOTIFYs from devices like this for anything. You need to change the keep-alive method in the PAP2 to OPTIONS.
Better yet, go buy a HT-802 and not deal with this ancient POS.
im not sure where to find notify and options. its working fine using my sip with nextiva
This thread has gone completely astray. A PAP2T (or HT802) has two FXS (foreign exchange station) ports, each designed to connect to an analog phone, or a device with the same interface (cordless phone base, legacy alarm system, fax machine, etc.)
It cannot connect to a POTS (analog) line or a similar analog line from a cable MTA, fiber ONT, etc. It simply can’t work, because the hardware is not compatible.
To connect Asterisk to an analog line, you can use a device with one or more FXO (foreign exchange office) ports: Current products include Grandstream HT813 and Obihai OBi212. Older devices, often available used at low cost, include Grandstream HT503, Obihai OBi110, Cisco/Linksys SPA3102 and SPA3000.
I agree now that I’m learning. I purchased a Grandstream Hybrid ATA with FXS and FXO Ports (HT813) that will be here tomorrow. I thought that was the best route to go. Have you messed with the cellular devices gsm gateways that allow the same concept? I did not realize that the Linksys only has FXS which is only for a telephone and need FXO to bring in a line. I’m learning so much. Thank you Stewart1.
How was all this setup with Nextiva?
I just received a Grandstream HT813 that has both of the phone ports. I have them both plugged in but have some questions. Do I need to send the registration from the Grandstream or from FreePBX?
Is Nextiva still your voice provider in all this?
No I had to do something else because there system was horiable. So im bringing in a line from the outside. I was not sure how the registration should be setup.
For the FXO port, see
The FXS can be set as a normal extension.
I guess I should have asked you before is there any newer hardware out there that is better?
Who is the provider now? Someone putting out an analog line will be few and far between these days.
Also why wouldn’t you just get a sip trunk for the PBX? A single analog line is worthless.
Well if its anything like Nexteva, it’s worthless. A bad experience. They first gave me numbers that were not able to be assigned because they were ported out by other users, then the tech support did not understand sip trunks. I knew more than they did, and Im just learning…This is a project so I would like to keep my spending to a limit. I would save money over time. I did port my numbers I use for my nonprofit back to Ooma Office for now then later when the funds are available, I will do more.
Im going to send the log because I’m having issues with FX0. The FXS is registered.
OK so Ooma Office and Netiva are the same type of service. You are using a cloud PBX service with Ooma Office. There’s not a single piece of it that is analog on their side. Does the FreePBX have an FXS or FXO card in it? Why is it even using FXS or FXO?
If you’re going to use FreePBX then neither Nextiva or Ooma Office is right for this because they provide the same features that you’re trying to use in FreePBX. As well, you’ve introduced FXS/FXO analog between Ooma and FreePBX which none if it is necessary or needed.
I can’t see any reason there needs to be a HT813 between FreePBX and these services. I can’t see the need for using FXS/FXO in FreePBX. Find a real SIP trunk provider and use them. Or at least figure out how to use your one line account at Ooma to connect as a SIP trunk to the PBX.
You’re just going to keep making this more work than it needs to be trying to solve problems that don’t need to exist and you created.
I am only using Ooma Office temporarily. I have the phone line in FXO and the analog phone in the other port. Has anyone had a chance to look at my log? I just got home from back surgery and trying to figure this out.
Where does FreePBX fit into all this?
I have freepbx coming in as well. I have tried many configurations with grandstream and freepbx but im seeming to have issues with setting up a sip trunk.I dont need but one line since this line will be an time and temperature line.