How to setup FPL Elastix?

[4020315]

Greetings,

I've been a paid subscriber (aka, paid $50 for the config file) since June 2009, happily using my ATA. Now the new year brings out the daring soul in me

I've managed to setup Elastix on a spare PC!!!

The system is up, internal extensions are working, but FPL is not registering.

I've heard of some having similar challenges, while others got it working. How did you do it? To start, I need details on what goes in the PEER and USER sections.

Seeing that existing settings on the forums are only ATA based, I'm planning on compiling all the tricks for future Elastix/Asterisk/Trixbox users.

4020315

[tbrummell]

I use Elastix 1.6/Asterisk 1.4 myself.

First, you need to change your useragent, FPL will not route calls to an "asterisk" UA. I set mine as "LinksysPAP2" and it works fine. I have added the line "useragent=LinksysPAP2" to the sip_general_custom.conf file.

Once that is done, your trunk needs to have the following:

In the PEER details:
context=from-trunk
host=voip.freephoneline.ca
qualify=yes
dtmfmode=rfc2833
username=16135551212 ;whatever your FPL number is goes here...
secret=YourSecretPass
type=friend
insecure=invite
disallow=all
allow=G729&ulaw

User Context and User Details are blank.

The register string is as follows:
16135551212:YourSecretPass@voip.freephoneline.ca/16135551212

This setup has been working for me since Sept. 2010. That should get the trunk up and registered. After that you need to create an Inbound Route so the call is set to an answer point of some type. Personally I use Ring Groups.

[4020315]

Thanks for the setting tbrummell. Unfortunately, both my FPL lines (yes, I have 2 paid lines) are still not registering. I still have my ATA connected, until I get the Elastix working. In that, I have turned off "enabled line 2", thinking ATA registration might be causing an issue.

Both lines are still setup in Elastix (line 1 is still enabled in ATA), yet none are registered. I have used the settings you defined above in PEER details of the trunk, and user context/user details are blank. Register string was as you mentioned above.

And the /etc/Asterisk/sip_general_custom.conf includes the useragent line.

What else could it be?

[4020315]

well...I had assumed that the most fundamental part was auto configured. The "network" settings were not defined, and hence my box had no outside connection. All is well now, and I'm connected!

Thanks for your assistance with the PEER settings.

[tbrummell]

well...I had assumed that the most fundamental part was auto configured. The "network" settings were not defined, and hence my box had no outside connection. All is well now, and I'm connected!

Thanks for your assistance with the PEER settings.

I was going to ask you to double check that. Glad you got it sorted. Have fun with Elastix!

[tbrummell]

both my FPL lines (yes, I have 2 paid lines) are still not registering.

It becomes an addiction once you start running a pbx. Even though mine is at home, on top of the wife's FPL number, I have 2 Ottawa DID's, 1 Campbellford area DID and a toll free number coming in to my system, all from Voip.ms. My outbound long distance is handled by voip.ms & CallWithUs. The flexibility of running your own pbx is awesome.

[Jake]

It becomes an addiction once you start running a pbx.

I would have to agree with that! I have one Ottawa FPL number for my main number, another FPL number I use for my mobile phone to get free calling in and out through it, a US number (free via WhistlePhone), a Southampton (in the UK, free from coms) number for my Mum to call local to her, and a Lymington (again in the UK, free from an Italian company) for my Dad to call local to him. All these DIDs come into the PBX then onto the house phones, and calls get routed out via FPL, WhistlePhone or VoipCheap depending on which number I dial.

I tell my wife I have done all this to save money... but I know differently

[tbrummell]

It becomes an addiction once you start running a pbx.

I tell my wife I have done all this to save money... but I know differently

I told her the same, and her being in to accounting, I needed to show her ROI. Turns out, all the hardware (ip phones and an ATA) will be "paid for" in 11 months of *not* paying Distributel for their service. Since Sept I've spent $5.89 on long distance. A far cry from what we used to pay to Distributel.

[Jake]

I told her the same, and her being in to accounting, I needed to show her ROI. Turns out, all the hardware (ip phones and an ATA) will be "paid for" in 11 months of *not* paying Distributel for their service. Since Sept I've spent $5.89 on long distance. A far cry from what we used to pay to Distributel.

I used the phones we had, and connected the ATA into the house wiring. I since bought the FPL DSL package which got my the TG784 router, so that got me effectively another ATA with 2 more phone ports. We ditched Bell the November before last, so I know we have easily paid for the equipment. I buy a 10 Euro topup on voipcheap about once a year. That gives me 90 days free calling to many places, then the UK, US, and Canada works out about a penny a minute which does us fine. (** Admin feel free to delete this if you feel it is a conflict of interest).

The thing I like the best is when people ask where the number is from, because it is not a 284 or 283 (Smiths Falls) number, and I tell them that it is a number that I don't have to pay a monthly bill for. I like that feeling, and knowing I don't have to give Bell $40 odd a month for less service and calling areas than I get for free now

[pcdunlop]

...another FPL number I use for my mobile phone to get free calling in and out through it...

Can you provide more specifics on how do you setup the FPL with your mobile? I believe you call your FPL number from your mobile and then get a dial tone so that you can call through FPL? How do you forward your incoming calls to your mobile? I would appreciate your help on this.

Thanks!

[Jake]

...another FPL number I use for my mobile phone to get free calling in and out through it...

Can you provide more specifics on how do you setup the FPL with your mobile? I believe you call your FPL number from your mobile and then get a dial tone so that you can call through FPL? How do you forward your incoming calls to your mobile? I would appreciate your help on this.

Thanks!

Well I have it set up through a Freeswitch PBX running on a machine here.

There are two parts to this in the dial plans. I also have one number free calling set up on my mobile, which is the FPL number I use for this.

One dial plan is set up to recognise calls to the FPL number FROM my mobile number. When it sees me calling in it diverts the call to the DISA script I have set up and allows me to call out.
The other dial plan is set up to recogise calls to the FPL number NOT from my mobile number (IE everyone else). This dial plan diverts the call to my mobile using the FPL number.

Because of the way Freeswitch works is it stops when it find a match on the dial plan, so because the first one matches when I call in it doesn't look any more, otherwise when I call in it would just call me back.

There are two downsides to this. One, all the calls come in to my mobile as my FPL number, so I don't really know who is calling (not a biggie for me). The other is people have to text to one number, and call another. Both these downsides are outweighed by the fact I only pay $7 a month for free incoming and outgoing calls to my mobile.

[tbrummell]

All of the above can be done with Elastix as well, but the upside is the callers CID is passed to my mobile.

[curriegrad2004]

Because of the way Freeswitch works is it stops when it find a match on the dial plan, so because the first one matches when I call in it doesn't look any more, otherwise when I call in it would just call me back.

You might want to read through the dialplan documentation on FS's wiki carefully... I've had that same problem before, but once I read through the documentation carefully, I fixed that problem ages ago...Getting regex working is a huge hint here.

[Jake]

Because of the way Freeswitch works is it stops when it find a match on the dial plan, so because the first one matches when I call in it doesn't look any more, otherwise when I call in it would just call me back.

You might want to read through the dialplan documentation on FS's wiki carefully... I've had that same problem before, but once I read through the documentation carefully, I fixed that problem ages ago...Getting regex working is a huge hint here.

Maybe you mis-understood what I put, I don't have a problem. I was saying that it stops [looking for other dial plans to match against] when it finds a dial plan match. My dial plans work very well for what I need right now. I know I am no expert on this, but I have managed to cobble it together to do what I want.

[curriegrad2004]

If you have break set to "never" would solve this problem of FreeSwitch stopping on a first successful match in the dialplan. Setting this to never will tell FS to continue searching the DP regardless of a successful match or not

The people who regularly read the users mailing list of FreeSwitch can give you assistance on getting your dialplans to work if you really want to know more on how the dialplans work.

[4020315]

well...I had assumed that the most fundamental part was auto configured. The "network" settings were not defined, and hence my box had no outside connection. All is well now, and I'm connected!

Thanks for your assistance with the PEER settings.

I was going to ask you to double check that. Glad you got it sorted. Have fun with Elastix!

Now I'm having other issues. Both FPL lines are working, but incoming calls stop terminating to the elastix box every 24 hours, and i get FPL vm. outgoing is fine. A quick check on the FPL portal says "online". I have already configured my "useragent" to be LinksysPAP (which is what shows up on FPL portal).

I had gotten incoming calls working yesterday by removing the "dial rules" (now blank). Not sure why the registration worked, but I can't change blank to blank again

Any other suggestions?

I've already given my box a static IP, and even changed the lease expiry for DHCP (just in case). What else could it be?

[tbrummell]

With the "qualify=yes" option set, it should keep your line registered all the time.

Next time you can't get calls in, log in to the Asterisk CLI and place a test call. Capture the output, if any, and paste it here. If you get no output in the CLI then that means the call is in no way getting to your box. At that point I'd do a sip show peers and see what the status of FPL says.

[Jake]

One thing I did in my freeswitch setup is to add in a parameter to ping the SIP providers, like this example http://wiki.freeswitch.org/wiki/SIP_Pro ... ions_Alive (Keeping Connections Alive). Now I know you are not using freeswitch, but something similar with Elastix might help you if it exists.

[curriegrad2004]

One thing I did in my freeswitch setup is to add in a parameter to ping the SIP providers, like this example http://wiki.freeswitch.org/wiki/SIP_Pro ... ions_Alive (Keeping Connections Alive). Now I know you are not using freeswitch, but something similar with Elastix might help you if it exists.

I never had to configure keep-alives in FreeSwitch. Are you using it behind the NAT or are you actually loading FreeSwitch on the router like I am doing right now?

[tbrummell]

One thing I did in my freeswitch setup is to add in a parameter to ping the SIP providers, like this example http://wiki.freeswitch.org/wiki/SIP_Pro ... ions_Alive (Keeping Connections Alive). Now I know you are not using freeswitch, but something similar with Elastix might help you if it exists.

That's what "qualify=yes" does in Asterisk. It sends an options command every 60 seconds. This keeps the NAT port open at all times, it also keeps the connection up with the SIP provider as they respond to the options command.

[Jake]

That's what "qualify=yes" does in Asterisk. It sends an options command every 60 seconds. This keeps the NAT port open at all times, it also keeps the connection up with the SIP provider as they respond to the options command.

Ahh OK. I thought I would throw it out there just in case.

[mikexmas]

Works great thanks for the help. I just hope FPL is fine with it I am only using it at home to learn how to setup Elastix not to abuse the great service they are providing.

As a side note remember to turn off the voice mail setting on FPL to except inbound routes on your Elastix or FPL will grab the call every-time and send it to there voice mail.

[tbrummell]

As long as your calling patterns don't indicate you are using it as a business line, they are fine with it.

I actually use the FPL vmail, not the onboard Asterisk vmail. That way if my connection is down for any reason my calls still go to voicemail, and I am sent an email with the vmail attached. Not that I'd get said email if my Internet connection was down, since I run my own email server, but you get the idea.

[ben0it27]

I've setup recently a pc with Elastix on a phoneline that was used a HT-502, the setup look fine (everything is up) but when the computer is connected to the internet and someone try to call the #, its automaticly goes to the voice mail of freephoneline, and i see nothing in the asterisk logs nor the status. I've setup the trunk with my uisername/password (the one in the freephoneline summary SIP setting page).

So when someone call it goes directly to voicemail, when i disable the trunk its goes to the ht-502 phone with no problem.

I would like to know what i forget ? i don't see anything in the logs that could help me out.


(Router) --------- (ADSL Modem)
| \ \
| \ \
| \ \
Elastix HT-502 PC

Thanks in advance for any tips

[dibsmft]

Have you forwarded ports to the computer running Elastics?