Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

[jondecker76]

Looking for others that wish to connect LMCE to their analog phone line using a Sipura 3000. It would be nice to document this process so that in the future we can automate the setup.

Goal:
Connect an analog phone line to LMCE to take advantage of some of Asterisks and LMCE's features, such as:

• Forwarding calls depending on status (Asleep, away, at home, etc..)
• Priority caller features
• Integration with security system (call neighbors, your cell phone, police, fire dept. on different security events)
• Answering Machine features / mailbox features
• Dialing from orbiters (phone book, speed dial etc)
• Recording and archiving of conversations
• Hold features and on hold music
• pausing media upon receiving a phone call

As you can see, for those of us that don't want to use VOIP, there are still many benefits of hooking your analog phone system up to LMCE.

The idea is to have the internal phone line treated as 1 extension (all hard wire pstn phones ring together). Beyond that, normal VOIP phones can be used as additional internal lines, which will interface the Analog line through Asterisk.

Another nice feature with this approach and with the Sipura 3000 is that upon a power outtage or failure of the core, the FXO and FXS ports are jumpered automatically, linking your internal phone line to the analog line, allowing the phones to still be used.

Anyone want to put their heads together with me to get this set up? My Sipura 3000 is dieing to be configured and interacting with LMCE!

[bulek]

Looking for others that wish to connect LMCE to their analog phone line using a Sipura 3000. It would be nice to document this process so that in the future we can automate the setup.

Goal:
Connect an analog phone line to LMCE to take advantage of some of Asterisks and LMCE's features, such as:

• Forwarding calls depending on status (Asleep, away, at home, etc..)
• Priority caller features
• Integration with security system (call neighbors, your cell phone, police, fire dept. on different security events)
• Answering Machine features / mailbox features
• Dialing from orbiters (phone book, speed dial etc)
• Recording and archiving of conversations
• Hold features and on hold music
• pausing media upon receiving a phone call

As you can see, for those of us that don't want to use VOIP, there are still many benefits of hooking your analog phone system up to LMCE.

The idea is to have the internal phone line treated as 1 extension (all hard wire pstn phones ring together). Beyond that, normal VOIP phones can be used as additional internal lines, which will interface the Analog line through Asterisk.

Another nice feature with this approach and with the Sipura 3000 is that upon a power outtage or failure of the core, the FXO and FXS ports are jumpered automatically, linking your internal phone line to the analog line, allowing the phones to still be used.

Anyone want to put their heads together with me to get this set up? My Sipura 3000 is dieing to be configured and interacting with LMCE!

Hi,

I'm using Sipura 3000 for connecting analog doorphone to my system... Configuration is I guess pretty similar, so I can probably help with some details...

Sipura is nice device, but has a lot of config options and I haven't configured it since ages when it started to work. I'm only changin sip credentials when I integrate it in new system...

Let's start discussion and wiki page on this one....

Update: I did my first and the only configuration with some web configuration utility, that let you only fill subset of needed information and it prepared configuration file for sipura. After that, I only did minor tweaks to the configuration. I can't find it anymore, but maybe this will be of some help :
http://www.aussievoip.com/wiki/index.php?page_id=119. I can also make snapshots of my setup and send it to you over PM.



regards,

Bulek.

[golgoj4]

Hey guys,

I have a spa-3102 that I use with my system. I wrote a wiki on it http://wiki.linuxmce.org/index.php/Linksys_spa-3102. Hopefully this helps.
Also, the articles I found relevant on setting it up are located at the bottom of that page.

Hope this helps,
Golgoj4

[jondecker76]

thanks a lot - i'll be checking this out during the week!

[jondecker76]

I started working on this a bit. Following your wiki instructions, I got to this step:

If you want the phone attached to your SPA-3102 (Line 1) to ring when people call, you should go to Wizard -> Devices -> Phone Lines, and click Settings for the appropriate phone line(s). There you can modify the ringing behaviour of your new phone.

If I go to Wizard->Devices->Phone Lines, there are no lines listed. There is a drop box to add a new line, but they are all VOIP specific. What shoudl I do here?

thanks,

jon

[Zaerc]

Correct me if I'm wrong, but I think you need to finish the steps under "Configuring FreePBX" first for the lines to show up there. 

By the way, nice wiki page golgoj4!

[jondecker76]

I added the trunk in FreePBX. At the top of the page for my trunk, it has an error

WARNING: This trunk is not used by any routes!This trunk will not be able to be used for outbound calls until a route is setup that uses it. Click on Outbound Routes to setup routing.

Messing around with this is showing me how far behind this area of LMCE is (as far as configuration and ease of use goes) I'm really hoping to get this figured out so the process can be automated

I've started reading the book "Asterisk: The Future Telephony" - I have a feeling like i'm going to have to fully understand Asterisk before I can ever get it configured correctly.

[golgoj4]

I started working on this a bit. Following your wiki instructions, I got to this step:

If you want the phone attached to your SPA-3102 (Line 1) to ring when people call, you should go to Wizard -> Devices -> Phone Lines, and click Settings for the appropriate phone line(s). There you can modify the ringing behaviour of your new phone.

If I go to Wizard->Devices->Phone Lines, there are no lines listed. There is a drop box to add a new line, but they are all VOIP specific. What shoudl I do here?

thanks,

jon

This is something I have not yet figured out how to do. Currently, the setup is only for voip.  I had to do all my customization in freepbx / linksys admin page to get the call into lmce. I have no idea just yet on how to make the configuration options for pstn appear. My uninformed self tells me it involves figuring out a couple things

-base configuration similar to the voip setup only more stripped down as there isnt any authentication with pstn lines.
-pnp some how to auto configure a pstn line when supported ata is connected.

thats basically where I left off a while ago. But to be honest, im still trying to understand fully why it works and how free pbx works in relation to the webadmin.

[jondecker76]

ok. well at least you have a start on this. I'm going to take some time to read through this book and get a better understanding of Asterisk, then disect some scripts and get an idea of how LMCE interacts with Asterisk. Maybe then the answer will pop out at us..

[bulek]

Hi,

I did mess with Asterisk & LMCE integration some time ago and remember few things... The last state of this problem was that Aaron said, that they're seeking for Asterisk guru to fix that Asterisk & FreePBX integration under LMCE, but I guess nothing changed from there. We're using quite ancient versions of FreePBX and Asterisk, cause some problems were fixed in there and compilation/installation script is now the nightmare, only Asterisk guru can solve....

Writing out of memory:
I'm dealing with Asterisk installation under LMCE in this way :
- add everything that is possible to add via web-admin (at least phone devices), then some perl scripts will be run that will put those devices also in freepbx or Asterisk config files (look in /etc/asterisk/).
- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...

I have setup with three VOIP providers, doorphone on Sipura interface and many routes etc...
My logic is usually setup in such manner, that I declare two dummy users (doorphone, housephone) and do all call routing features through them (I guess doing it for each family members is too much work, while we all use common phones). The incoming call logic goes in this way: I just reroute calls to doorphone or housephone user and then use all call routing features you can setup via web-admin under LMCE. There you can determine behaviour based on housemode, incoming caller etc...

I can help you with questions regarding that... Maybe we should start wiki page about using Asterisk under LMCE....

Regards,

Bulek.

[maybeoneday]

hi all,
I'd love to have this as a feature  (tech abilities a bit limited as are funds),   but would this,

   http://www.pctradestore.com/code/ui/main/product_info.aspx?prdid=ATA&catid=7&heading=VOIP%20SIP%20/%20PSTN%20analogue%20ATA%20Adapter%20(%201*%20FXS,%20%201%20*%20FXO%20&%201%20RJ45%20WAN%20)

substitute for  the sipura,( it's £25 as opposed to £150      )

Any comments/advice regarding spec /shortcomings /potential pitfalls greatly appreciated,

regards,
Ian

[Marie.O]

- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...

Bulek,

what kind of things can't be done in the lmce interface? And would you mind to maybe add those things to the web admin interface of lmce?

rgds
Oliver

[jondecker76]

maybeoneday:
It has an FXO and an FXS port, so technically it shoudl work *if* you know how to configure it correctly (in its own admin interface - should'nt be much different on the asterisk end). However, with the cheap price, one would have to imagine that the quality will leave a lot to be desired.


Can anybody that is currently using a VOIP provider such as broadvoice post a screenshot of their Phone Lines admin page?

[bulek]

- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...

Bulek,

what kind of things can't be done in the LinuxMCE interface? And would you mind to maybe add those things to the web admin interface of LinuxMCE?

rgds
Oliver

hmm, a lot of things... You cannot do inbound, outbound routes, trunks etc... Basically you can only add phones, some predetermined VOIP providers and of course quite nice routing settings via web-admin...
I think it would be really complex task to duplicate FreePBX under web-admin and probably pointless too, cause majority of users should have enough existing web-admin interface...

What we would really need right now in this area is to update Asterisk and FreePBX to more current versions for 8.10, but that's a complex task cause current setup has some hacks in it.... and only some experienced Asterisk guru can do this....

Regards,

Bulek.

[golgoj4]

hi all,
I'd love to have this as a feature  (tech abilities a bit limited as are funds),   but would this,

   http://www.pctradestore.com/code/ui/main/product_info.aspx?prdid=ATA&catid=7&heading=VOIP%20SIP%20/%20PSTN%20analogue%20ATA%20Adapter%20(%201*%20FXS,%20%201%20*%20FXO%20&%201%20RJ45%20WAN%20)

substitute for  the sipura,( it's £25 as opposed to £150      )

Any comments/advice regarding spec /shortcomings /potential pitfalls greatly appreciated,

regards,
Ian

I realize the linksys is more expensive, but I found that it has so many features that I dont even use them all. Key in those features was its ability to act as pstn trunk. Now, I dont know that much about phone systems, but one of the reasons I went with the linksys is that is that it was highly recommended on many forums I visited because of its flexibility. So what im sayin in essence is try to find out as much as possible about the device your looking at. google the product name and model number as its a good way to get hits on issues and more information in general from people who took the plunge already. My internet at home is a bit wonky but I will grab some screenies of the linksys admin page as well as what I have going on in free pbx so that you can see what I mean. The main benefit of my spa-3102 is there were guides to help me understand most of it to the extent I could get the system running.

So check out the support out there now for it as well as how configurable it is. From what I read at a minimum it should be able to connect to asterisk, but im not sure about it acting as the pstn bridge TO lmce without further information.

hth
golgo