Author Topic: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)  (Read 16314 times)

jondecker76

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Just another update...

Everything is 100% working and set up. All LMCE standards have been used, and if you compared LMCE interaction and operation of using the pstn line/spa3000 and a VOIP account, you would not know the difference. Absolutely everything is 100% tested and working.. Conferencing, paging through the house, forwarding to other extensions, calling extensions, sending and receiving calls, routing to voicemail, priority callers, you name it! I will be posting instructions on how to set this up soon, but I want to do a few things first:
- standardize naming conventions on a few things. This way, once I automate the process, what you create manually through instructions will be the same as it will be when auto setup is implemented
- Test just a couple more things out (mainly interaction with Security  -  calling your cell phone on a security breach and the like.
- Watch a movie with my wife before she kills me


All I can say is that this is really really really fun, and really helps to complete a full automated house!

After this is all finished, i'm going to implement support for the spa2000 (2 FXS ports, for adding additional hard wired analog lines). I'm preferring this over the Digium cards because it doesn't hog up PCI slots, and only has to connect to the router on the internal network.

maybeoneday

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jondecker,

you're amazing,

I've just (04:10 gmt) finished skimming the o'reilly book, and now I've got a problem,



1) go to bed
2)unpack and install the spa3102 that came  earlier  (actually yesterday, i didn't realise I'd been reading so long)

 ;D

just one befuddled thought, if a dect setup was connected rather than single phone could the dial plan include divert/conference and other capabilities of dect system?,

congrats,
regards,
Ian

Edit...bluetooth phones as handsets for md's?
« Last Edit: January 16, 2009, 05:26:42 am by maybeoneday »

jondecker76

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Ian,

I doubt you could ring separate extensions in a Dect setup (my phones in my house also have only one base station, which all "drone" phones connect to). For this reason, I decided to treat my entire internal landline as 1 phone, and I'm pretty happy with that.

jondecker76

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #33 on: January 16, 2009, 08:33:09 pm »
I have finished a wiki page detailing how to manually set this up. The more people we can get testing this, and the more we can verify everything works as it should, the sooner I can add this as a plug-and-play device for LinuxMCE. Please, lets test EVERYTHING out, I'd like to have this working perfectly and 100% supported for 0810.

Here is the wiki page:
http://wiki.linuxmce.com/index.php/Sipura/Linksys_spa3000_pstn_interface


Problems I know of so far:
1) Upon power outtage, or network failure/core crash, the FXO and FXS lines are bridged so you can still use the phones. Dialing out works fine (just unplug the power and/or network cable from your spa3000 to see!). However, incoming calls in this failsafe mode only ring once. (Update: now fixed)

2) Incoming calls don't directly ring the house phone extension. All orbiters alert of the call, and the call can be directed to go to the house line, but I would have expected that the houseline would ring and could be picked up on a call with no orbiter interaction at all. (UPDATE: Found the problem - this will work once I add the web admin code to install the phone line - this is now fixed)


Please list any other problems you can find. If you know of a way to fix any of the problems we find, try it out and report back here if you are successful.

Thanks, and enjoy!
« Last Edit: January 23, 2009, 05:22:29 am by jondecker76 »

maybeoneday

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #34 on: January 16, 2009, 08:57:58 pm »
GREAT WORK  Jondecker,

& brilliant wikki,

will be installing / testing tommorrow, will feedback asap,

many thanks
Ian

maybeoneday

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #35 on: January 18, 2009, 02:08:35 pm »
Hi jondecker,

been trying your setup with no joy :(
mainly , I think 'cos I'm  in UK, & using spa3102

FOR UK USERS......www.aoakley.com/articles/2008-01-08.php       has the relevant settings (untested by me as yet,imminent) and from there I'll try again with freepbx as per your wiki.

NB  pay close attention to first part concerning cables  and testing phones 


regards,
Ian

jondecker76

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #36 on: January 18, 2009, 05:02:51 pm »
the only settings you are going to have to dial in  should be on the spa3102 - everything else should be the same. Most of the settings should be under the "Regional" tab of the spa3000/3102 - these are the tones and signals the sipura looks for on the line to know whether to answer, hang up, etc.


Found out from someone who called today that music on hold works - though I didn't successfully navigate answering a call on call waiting and returning to the original caller. Will try again later

freymann

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #37 on: January 20, 2009, 06:36:01 pm »
Hi Jon.

I just installed the SPA3000 with mixed success.

I ran a network cable out of the basement office and through a bunch of floor joists to reach the demarc point of the phone line by the fuse panel. I then wired in a couple phone boxes so I could connect a phone cable from the demarc point to the box feeding the house phones (to keep things "normal") but that also lets me connect the SPA3000 in between so I can 'test' and put things back to normal when done.

First, the "House Line" doesn't seem to work at all. If I try to make a call I get a long pause and then a busy signal. Incoming calls appear on the orbiter, but if I route it to the "House Line" no inside phones ring. If I route it to voice mail, I can hear the prompt to leave a message, and when I do and hang up, the message never appears.

After creating the "dummy" phone line, when I went back into the FreePBX admin and looked at the Incoming Route for the boardvoice line, and Custom App option, it contained "custom-linuxmce,103,1". So I went back to the Incoming Route for the House Line, and changed its custom app line to match as instructed. (so it used 103,1 instead of 102,1)

I didn't see anything else in the instructions that varied. I reloaded the router before testing incoming/outgoing calls. When I went back into FreePBX I noticed the "Apply Configuration Changes" bar was there again so I clicked it and continued with the reload, tested again, same results.

I do see status boxes on the top left of the orbiter saying call lost or something in rather odd times (while I'm speaking to the voice mail for instance?)

I do have one major issue...

We use Bell HSE (ADSL) for our internet and when I connect the SPA3000 and then hang the house phones off it, we lose our internet. I guess I'll need to purchase a splitter and run a filtered line into the SPA and leave the jack on the SPA, which would feed the house, empty, feeding the house from the other side of the (unfiltered) splitter at the demarc point.

In this setup, as far as I can tell, the only things I'd gain are:

1) on the orbiters... I'd be able to see the incoming phone number displayed.  Can it display the name of the caller along with the phone number?

2) the answering machine function should work

Would it be able to make outgoing calls (in the case of security alarm messages) set up this way?

The other way of doing the connection would mean relocating equipment and running even more network cable throughout the house, or perhaps running a dedicated phone wire to the upstairs dining-room to the ADSL modem and separating that from the rest of the house feed.

On a more general note, when this is set up in the manner you've described, does somebody have to route the call on an orbiter each time a call comes in? I'd like it to fall through to the house line and ring normally.

We have an existing answering machine and I already know the better half is not interesting in trying to get her messages from LMCE. I believe there is a setting somewhere to disable LMCE from taking messages?



jondecker76

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #38 on: January 20, 2009, 07:16:39 pm »
Freymann

I may have an error somewhere in the instructions I posted in the wiki. Can you get on IRC today, maybe we can find the problem by comparing.

As far as disabling the answering machine (IVR), in the web admin there is a field to adjust a timer for it to kick on. The instructions on the web admin say to set it to 0 to disable it, but this didn't work for me. I set it extremely high so it doesn't kick on at all (90 seconds) - our home answering machine picks it up well before then. (the horrible robotic voice IVR sucks and needs a complete overhaul in my opinion)

Yes, you can set incoming calls to ring on any combination of your media directors and your house line.


While on the subject - here are some things I have learned so far using pstn with Asterisk
- ECHO!  Echo is a major problem. This is unavoidable and due to the facts that both signals transmit over a single pair of wires and the fact that Asterisk introduces some latency. Asterisk does have software echo cancellation - however, it is a part of the zapata driver and only works with PCI cards! You can get reasonable performance from the spa3000 if you match the impedance and gains well, and set  the jitter buffers to low (to reduce latency). This takes some tweaking however.

- Digium sells a very good software echo canceler that can be compiled with zapata drivers ($10 license). Again, this only applies to pci cards that use the zapata driver.

- Digium sells cards with hardware echo canceling - this is probably the best option for call quality, but expect to spend at least $500 for one FXO and one FXS port (ouch!)

- I may try the cheap x100p cards that sell really cheap on ebay (about $30). They use the zapata driver and can make use of the software echo cancellation (both the freeware ones and the commercial one from digium).

- I haven't yet been able to change the on-hold music. I'll be playing with this more as I get free time

freymann

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #39 on: January 20, 2009, 09:28:10 pm »
I may have an error somewhere in the instructions I posted in the wiki. Can you get on IRC today, maybe we can find the problem by comparing.

I just had some free time but the chat link at the top of the forums does nothing for me?

Quote
As far as disabling the answering machine (IVR), in the web admin there is a field to adjust a timer for it to kick on. The instructions on the web admin say to set it to 0 to disable it, but this didn't work for me. I set it extremely high so it doesn't kick on at all (90 seconds) - our home answering machine picks it up well before then.

Ok, I think I found that.

Quote
Yes, you can set incoming calls to ring on any combination of your media directors and your house line.

Awh, this is in pluto-admin... Wizard > Devices > Phone Lines.

I see they have a Local Number Length. It was 7 but we require 10. Changed it.

From this same screen, look for the boardvoice line (my only one) and click on Settings under Actions on the far right and check off "House Phone" for all the scenarios required.

I put a splitter on the phone demarc and have one half feeding the house phones and the other half just feeding the spa3000, and this seems to be working OK.

I still don't end up with any messages if I say send to answering machine on the orbiter though.

EDIT: Actually, in pluto-admin, Telecom > General Voicemail, they appear here, but I can't listen to them on my ubuntu workstation (talks about installing something, then you say sure, then it says nothing installed). On XP in MSIE, I get a mini player but no sound. Can you not send a call to a specific user's voice mail?
« Last Edit: January 20, 2009, 09:53:45 pm by freymann »

freymann

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #40 on: January 21, 2009, 04:47:17 pm »
Hi Jon.

An update on my setup...

I can make outside calls from any orbiter, dialing 9 first...

I can answer a call from any orbiter. I can hear the caller but they don't hear me. I had a mic connected to my MD but most likely I gotta fiddle with a setting somewhere, someplace to activate it?

If I send a call to voicemail, it goes to the general voicemail account which I can only access via pluto-admin. If I click on any of the two user accounts nothing happens. This is weird.

I can see in the future, where having LMCE answer the phone and allow users to select 1 for me or 2 for the wife beneficial... especially if we can then have it forward those messages to our cell phones and/or email boxes.

I finally managed to figure out how to get into IRC and will hang out there a bit today.

bulek

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #41 on: January 21, 2009, 11:10:52 pm »
Hi,

you can also try to call *43 (loop to Asterisk and back)...

Regards,

Bulek.
Thanks in advance,

regards,

Bulek.

jondecker76

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #42 on: January 22, 2009, 08:58:11 pm »
just updated the wiki with another diagram of how the connection should go if you use DSL/ADSL for your internet access

freymann

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #43 on: January 22, 2009, 09:02:12 pm »
Here's a few things I discovered today.

Logged into LMCE as user "Gerry"
Orbiter shows an incoming call. I send to User "Gerry"
Voice Message is recorded, and on the orbiter, it now shows 1 message waiting for me.

If, when the oribiter shows the incoming call, and I click on "Leslie" instead of "Gerry" (in which I assume it would sent to Leslie's voice mail) it still goes to "Gerry"'s voicemail.

Did another test.

Logged into LMCE as user "Leslie"
Orbiter shows an incoming call. I send to User "Leslie"
Voice Message is recorded, and on the orbiter, it now shows 1 message waiting for Leslie (and 1 for Gerry, as above).

To play back voice mail --

At a MD with sound... Make sure correct user is logged into MCE. Click on Telecom and the logged in user's name.

Now you get a display allowing you to change your user status, and below is a list of voice mails, called:

New message 1

New message 2

New Message 3

etc.

Click to the right of the "New message 1" in the unused green area and it should play back that message, over and over. To stop it from playing back, go back to the main screen, Green Button (or F7) and then OFF. The audio system is playing back your messages.

You have to keep going back to Telecom > User > click on the green space, main menu, off to cycle through voicemails.

If I'm logged in as "Gerry" but click on Telecom > Leslie it doesnt' show me her list of voicemails, it shows me mine.

You can log into the Pluto web admin to hear your messages through your web browser... click on Telecomm > My Voicemail then click the play button beside your message. On ubuntu, yesterday, when I tried this, it talked about installing a plugin, which appeared to have failed, and I never did hear anything. Today, now that the computer has been rebooted? It's working. Also seems to work fine under MSIE and WinXP.

To be able to click on Gerry and have it sent to voicemail, I configured Telecom > Call Routing (from the top of the pluto web admin) > Gerry User Mode At Home > Normal Caller to be set to 'Go To voicemail'.


freymann

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #44 on: January 22, 2009, 09:06:46 pm »
Jon mentioned one could press *98 to get to voicemail over a phone connected to the Phone jack on the sipura.

That doesn't work under my setup, I just get a busy signal.

If I dial *43, same thing, just a busy signal.

In fact, I can't make outgoing phone calls from a phone connected to the Phone jack on the sipura. Just get a busy signal. Dialing 9 then the number doesn't make any difference.

I can make a call from an orbiter just fine (dialing 9 first), and I can hear the person at the other end, but they can't hear me.

That's probably all I'll have time to experiment with today.