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

[jondecker76]

Freymann

I should be on IRC tomorrow from around 1300 until 1500 EST. It sounds like something may be wrong with the Line1 setup tab on the spa3000. I'll try to catch you on there tomorrow and we should be able to get that last part figured out. Though if you get a chance, I think I just spotted a mistake in my wiki instructions. In the spa3000/Line1 setup page, for password, instead of l m c e as a password,  put the extension number as the password (so that Display Name, UserID and Password all read the same number like 206 for example). I believe that this will fix your internal line problem.


Regarding some of the other abnormalities you mentioned, I'm sure that there are some bugs in the telecom system - after you are all set up, lets continue to track them down and put Trac tickets in on them so they get fixed. While I am realizing that the spa3000 may not be the pest way to interface the pstn (because of echo problems) - it will allow us to fully test the system, track down bugs and get things fixed. I will soon be getting some PCI pstn cards to try to add support for - in the end these may be the best choice because of echo cancelation.  Thanks a lot for your effort in getting this sorted out!

thanks

Jon

[freymann]

Freymann

I should be on IRC tomorrow from around 1300 until 1500 EST. It sounds like something may be wrong with the Line1 setup tab on the spa3000. I'll try to catch you on there tomorrow and we should be able to get that last part figured out.

Hi Jon. I reviewed the instructions again for Line 1 and all looks well, so maybe some of the defaults are different in sipura? I will drop by IRC tomorrow afternoon and hook up with you. I just connect a wireless phone there which lets me go back to the office and play (oops, I mean, work!)

[jondecker76]

i just edited my post 2 posts up... I think I found your Line1 problem (as outlined in my last post above)

[freymann]

i just edited my post 2 posts up... I think I found your Line1 problem (as outlined in my last post above)

 Yep, changing the password to match the other two fields fixed the problem of not being able to use *98, etc. I now have my own voice on my voice mailbox. Very kewl!

[jondecker76]

great - you should be able to dial an outside line now as well from your analog phones - your setup should be 100% now. I've updated the wiki page to correct the error on settig the password. Time to set up your routing and do some experimenting - if we can get some bug reports in in time maybe we can have the Telecom functioning better for 0810

Please post any bugs you have found so that we can discuss them and verify that it truely is a bug and not operating as intended. I already have a few bug reports i when it comes to changing on hold music.

[freymann]

great - you should be able to dial an outside line now as well from your analog phones - your setup should be 100% now. Time to set up your routing and do some experimenting - if we can get some bug reports in in time maybe we can have the Telecom functioning better for 0810

Yes, may be able to experiment more tomorrow, not sure.

I do have one question which I'm still not clear on.

Connected the way you have illustrated.... when an incoming calls comes in, if nobody is at an orbiter to see/handle it, what happens? Does it just fall through to the built-in answering machine? Do the phones on the House Line ring? or is this dependant on the house mode and user modes? Basically, I'd like [specifically the wife demands] our house phones act just like they do now with no requirement to push buttons on the LMCE screen(s).

[jondecker76]

make sure your phone line settings have your House Line checked for at least the Unarmed security mode (I would also do this for Entertaining and Armed at home). This will ensure that your analog phones always ring when you are home. It wouldn't hurt if House Line were checked for All security modes. This will ensure that it will ring regardless if you are home or not. This way the call can be answered as normal with no Orbiter screen interaction what so ever. The only difference should be between dialing 9 for an outside line(which I am going to experiment to get rid of having to dial a 9 - i think it is just tweaking the dialplan settings in FreePBX->Trunk)

On the other hand, if you uncheck House Line from Phone_Lines settings, then it will only ring on the orbiters (if they are checked).. The from the orbiters you can send the call to House Line, after which the analog phones will ring. This is a cool setup for the Entertaining security mode so you aren't disturbed while entertaining guests.

As you can see, there are a million ways to set this up and pretty much anything you can think of should be possible. The pain is setting it up the first time, but after that it should work pretty well

[freymann]

Hi Jon.

Haven't had much free time lately to experiment with the sipura, but what I've noticed is:

-yes, the echo is there on my end, not sure if it's there at the caller's end too?
-it rings the house line and the orbiters just fine once I adjusted my call preferences and reloaded router. Very kewl.
-the "call display" on the orbiter shows me the phone number, but not the name? Does that pop up the # only?
-if I'm logged in as me, and I notice an incoming call that is for the wife, I can't direct the caller to her voicemail box, it only lets me send it to my voicemail box or the 'general' voicemail box
-the only access to the 'general' voicemail box is through pluto-admin web page or picking up a phone and *98 into mailbox 100.
-there's an option when viewing My Voicemail to activate the forward to my email address... this doesn't work
-Phone Book works fine and is quite handy
-outgoing calls from the orbiters work. How would one go about utilizing a microphone attached to the computer to be able to talk?
-playback of voicemail through orbiter on the tv works fine, although sometimes you have to click on the message line twice before it will start playing. Not a big thing
-after the orbiter disappears on an incoming call (and you did nothing with LMCE), the menu (as if you pressed F7) stays on the screen for a very long time. I have to grab the remote and clear the menu.
-I was able to customize my recorded message from a handset quite easily.
-I was able to retrieve my voicemail messages quite easily from a handset.

I found this page in the wiki too:

LinuxMCE telecom features
http://wiki.linuxmce.org/index.php/LinuxMCE_telecom_features

which may be useful.

We don't use the built-in answering machine just yet. The better half insists she doesn't want to have to flip on the tv to get her messages, she doesn't use the pluto-admin web page at all, and she doesn't want all the house phones hung off the sipura because she doesn't like the echo or having to dial 9 before a call which makes dialing *98 impossible.

In my testing, I've just grabbed a portable phone from the basement and connected that to the sipura and used that for the day. As I've mentioned to you, I split the phone line at the demarc point to feed the sipura and the rest of the house separately. This pleases the wife tremendously as nothing changed internally except call display on the orbiters and an alternate voice mail system.

Next time I'm able to play I will see if I can get freepbx to forward messages to my cell number. I don't think we have call forward on the house line so that would be one way to keep in touch with home calls while away.

This is really good stuff and your instructions regarding the setup were easy to follow and most helpful.

[dlewis]

Nice job... thanks for the update.

On a separate but equal front, I have an x100p card on it's way to my abode... I hope to get that going and draft a write-up. An issue that I foresee is with zaptel (now Dahdi). Dahdi has replaced zaptel with some decent updates. In order to upgrade to Dahdi, you need to re-compile asterisk (which as we know, will be a pain). Therefore, I plan to try the x100p card with an upgraded version of Zaptel (1.4.11). Now, you might be thinking: "1.4.11 is not the newest version...". You are correct. However, since the x100p does not come with hardware echo cancellation, I am going to use the Open Source Line Echo Canceller (OSLEC) to remedy the echo that will probably occur; the latest patch for zaptel as it pertains to OSLEC, is for zaptel version 1.4.11. I've installed the new zaptel and I've also installed OSLEC. Zaptel seemed to give weird errors, but I believe this is due to not having the card installed... Once I get the card (in about 10 business days), then I will report back my findings...

[golgoj4]

Freymann,

I know you can do this with voip calls. Maybe some of the freepbx configs for voip forwarding would be useful?

just thoughts

[dlewis]

Freymann,

I know you can do this with voip calls. Maybe some of the freepbx configs for voip forwarding would be useful?

just thoughts

golgoj4, do you still get an echo with the spa-3102?

[los93sol]

Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed.  I had to create a blank /etc/zaptel.conf then do sudo apt-get install zaptel.  I found that not creating that file ahead of time the install would fail out.  From there it was just a matter of configuring my trunk, inbound, and outbound routes in the FreePBX config.  I was going to do a wiki writeup on this, but at the time I did this nobody on the forums showed interest in pstn lines and lmce.  I for one would prefer to keep my normal pstn line and use on of the providers like voipcheap to cover long distance since they are generally free for 300 minutes/month.  I don't make that many long distance calls so for me this would be a nice alternative.

[dlewis]

Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed.  I had to create a blank /etc/zaptel.conf then do sudo apt-get install zaptel.  I found that not creating that file ahead of time the install would fail out.  From there it was just a matter of configuring my trunk, inbound, and outbound routes in the FreePBX config.  I was going to do a wiki writeup on this, but at the time I did this nobody on the forums showed interest in pstn lines and LinuxMCE.  I for one would prefer to keep my normal pstn line and use on of the providers like voipcheap to cover long distance since they are generally free for 300 minutes/month.  I don't make that many long distance calls so for me this would be a nice alternative.

How was the echo with the x100p? Did you have to install oslec? If so, anything special beyond normal install procedures?

-Derrick

[freymann]

Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed.

I'm a tinkerer and wanted to see what asterisk was all about. I hope to start using the voicemail boxes a little more in the future.

To prepare on testing the forward voicemail to email setting, today I did this on the core:

sudo apt-get install postifx

Afterwards, I edited /etc/postfix/main.cf and put my ISP's outgoing mail server (SMTP) name beside "relayhost = "

and then reloaded postfix with:

sudo /etc/init.d/postfix reload

I still haven't had a chance to call in, direct my call to my voicemail and see what happens, but I don't see why it wouldn't now.

Jon: Perhaps this is something you may want to add to your sipura wiki page?

[los93sol]

One other thing, this is slightly off-topic, but I've been trying to connect a sip device to my asterisk box from another network and so far have been unsuccessful, I cannot even see any communication happen in my asterisk cli.  Has anyone been able to get this type of setup working, the same device connects fine on my local lan.  I tried setting up firewall rules in LMCE's webadmin, but they don't seem to be working.  I know I need 5060 open and by default rtp uses 10000-20000 even ports only, and that all of these ports should be udp.  If someone could shed some light on how to set these rules up in the webadmin, perhaps I'm something very silly the wrong way.

dlewis:  echo is an issue on my sip devices, I need to investigate this more thoroughly, I can say Hi into the phone and I will hear it repeat almost a full second later.  On the other end of the calls they do not hear any echo at all and report that voice quality is excellent.  For now I've been ignoring the echo I hear, but it really is quite distracting when carrying on a lengthy conversation.