hwittenb
join:2003-12-20
 ·Future Nine Corpor..
·callwithus
·Callcentric
1 edit | reply to pigjuice
Re: Using VOIP to extend PSTN service over Ethernet.
said by pigjuice  :I could live with a single extension for now. Could you please post how you have it set up? I have the 3102 and 2002 already installed with static IP addresses, I just need to make it dial and ring. On the PSTN line, unless you want it to "ring-thru" to the handset attached to the 3102 first, I would have a separate analog phone attached to the pstn line and also have the pstn line attached to the FXO (Line) port of the 3102.
On the PSTN tab on the SPA you need to setup the pstn-to-voip gateway so that the call coming in to the SPA will be ring the extension at the same time it is ringing the phone directly attached to the pstn line. If you want it to "ring thru" to the phone attached to the 3102, this will have to be done first with this configuration.
The port numbers used in the example are arbitrary. Any numbers can be used,just keep them straight as to which is which.
SPA3102 PSTN Tab
Sip Port: 5096
Line Enable: YES
NAT Mapping Enable: NO
PSTN To Voip Gateway Enable: YES
PSTN Ring Thru to Line 1: NO
PSTN Caller Auth Method: NO
Off Hook While Calling Voip: NO
PSTN Caller Default DP: 6
Edit: Register NO
Edit: Make Call Without Reg YES
Edit: Send Call Without Reg YES
Dial Plan 6:
...where userid is the userid on the SPA2002 line 1 tab
...where 192.168.1.102 is the ip address of the SPA2002
...where 5300 is the sip port number of the SPA2002 Line 1 tab
PSTN Answer Delay: 1
Voip-to-PSTN Gateway Enable: YES
Voip Caller Auth Method: HTTP Digest
One Stage Dialing: YES
Voip Caller Default DP: 1
Dial Plan 1: (xx.)
VoIP Users and Passwords (HTTP Authentication)
Voip User Auth ID: userid
Voip User 1 DP: 1
Voip User 1 Password: password
Voip Answer Delay: 0
SPA2002 Line 1 Tab
Line Enable: YES
NAT Mapping Enable: NO
Sip Port: 5300 (same number used in the Dial Plan on the 3102)
Proxy: 192.168.1.100:5096
...where 192.168.1.100 is the ip address of the 3102
...where 5096 is the Sip Port of the PSTN tab on the 3102
UserID: userid
...where this is the same user id used on the PSTN Tab under VoIP User 1 Auth ID
Password: password
...where this is the same password used on the PSTN Tab under VoIP User 1 Password
Edit: Register NO
Edit: Make Call Without Reg YES
Edit: Send Call Without Reg YES
Dial Plan (xx.)
With this method of configuration, using HTTP Authorization you get the dial tone on the SPA2002 and the digits dialed go out on the pstn. There is another way to do this configuration where you get the dial tone from the SPA3102. I like this way better.
Edit: If different routers are involved then you may need to use NAT Mapping Enable, you will need to forward some ports, specifically the SIP ports and possibly the RTP ports. |
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pigjuice
join:2005-11-29
Le Roy, WV | It isn't working with these settings, but I'm still going through it. I'll try to wipe the boxes once more. Right now, I have no dial-tone on the 2002.
Both boxes are on the same subnet and can see each other IP wise. |
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hwittenb
join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric
| Login to your boxes as admin/advanced, save the configurations to your hard drive, make a zip file of them using File...SendTo and post the zip file of the configurations.
If you set Make Call without Reg YES you should get a dial tone period if the box is powered, the line is enabled. |
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pigjuice
join:2005-11-29
Le Roy, WV
| I reset both boxes and reentered the settings and it works. I have dial tone and dial out capability on the 2002. Some echo, but nothing serious.
Thanks for your help, this indeed was a simple procedure.
The next step would be adding an extension to the 3102 and having it ring at the same time and dial out the same as the 2002 and setting up some rudimentary intercom between the extension. |
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Fisamo
Premium
join:2004-02-20
Apex, NC
·VOIPo
·AT&T CallVantage
| At the risk of complicating your life, you could consider getting a surplus PC and putting Trixbox (at your site with the PSTN connection) on it. In that case, you can have as many extensions as your bandwidth will allow (keeping in mind, of course, that you would not want to dedicate all of your bandwidth to phone extensions). Also, keep in mind that there is a learning curve. You could probably get a Trixbox set up reasonably quickly with rudimentary settings, but if you decide to play with the bells and whistles, you'll get a headache quickly.  You can use the SPA to get your PSTN line into the Trixbox, or you can get an FXO card (e.g. TDB01B from Digium, if you want brand name--you can't get a "real" X100P anymore--all that's available are knock-off clones with hit-or-miss reliability, from what I've read). |
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pigjuice
join:2005-11-29
Le Roy, WV
| I am considering a solution with Aterix or Trixbox at some point in the future when I get more lines and learn more about this stuff.
My line is working fine the way hwittenb suggested. I have some small issue that I can tolerate: My caller id information isn't forwarding from the PSTN to the extension, which I'm sure it's something that can be set up. I'm also getting echo of my own voice and a strong fade on the other party when both are talking. |
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hwittenb
join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric
| said by pigjuice :My caller id information isn't forwarding from the PSTN to the extension, which I'm sure it's something that can be set up. Maybe the PSTN Answer Delay setting is not long enough to let the SPA decode the incoming caller id. Try increasing the delay a second at a time until you decide that isn't the problem. I'm away from home, but I believe there is also setting to use the incoming PSTN caller id on the voip call.
I'm also getting echo of my own voice and a strong fade on the other party when both are talking.
Probably something to do with the gain settings on the SPA. |
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pigjuice
join:2005-11-29
Le Roy, WV
| I caught another minor problem. If I end the call before the other party does (like leaving a voice message and hanging up when you're done talking) the call doesn't disconnect. After leaving a message, the phone stays off the hook and never hangs up properly. |
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hwittenb
join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric
| said by pigjuice :I caught another minor problem. If I end the call before the other party does (like leaving a voice message and hanging up when you're done talking) the call doesn't disconnect. After leaving a message, the phone stays off the hook and never hangs up properly. Usually the problem is detecting that the pstn line has disconnected. The 3102 has a lot of options for that. Disconnection by the remote SPA is done by a SIP "Bye" signal and the 3102 will acknowledge the BYE and release the call on the PSTN line. I just looked at a trace on my equipment and that is how it works. I'm not sure what would be happening on your setup. You would have to run a trace on the 3102 to figure it out.
It is possible to setup the 3102 so that it will disconnect the call after any "long silence" period that you specify. This might be something to try if you don't mind getting calls arbitrarily disconnected when there is silence for a long period.
You can read about these subjects in the SPA3000 ATA User's Guide that you can download from www.sipura.com/support
I see you use WildBlue for your internet service. Have you found it possible to use voip at all over that type of service? The propagation delays are quite long for voip and the general answer is "no you can't use it" but I'm curious as to whether you have tried it. |
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pigjuice
join:2005-11-29
Le Roy, WV
| I have tried voip in a limited fashion and so far it didn't work. It would connect and immediately drop, or there would be too much jitter to understand anything. I mean to try it more in the future and I am also planing to add second sat line in a few months (starband or unasat) and I will report on how it works. My PSTN line is my last wire to the world I would love to cut it if possible. |
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Fisamo
Premium
join:2004-02-20
Apex, NC
·VOIPo
·AT&T CallVantage
| If there's an entry in your Regional tab for "CPC duration" make sure it's set to 0.75 - 1.0 (unit is seconds). I doubt this is the source of your problem for the cases where you place the call, hang up, and the call is not properly handled, but it would help for incoming calls. |
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DonPedro
join:2005-11-18
dominican re
| reply to pigjuice
I do not understand, I had starband (100 up/500 down) and it worked very well pC to Pc .not comfortable but acceptable PC to landline.
Now I have the new plan (250 up/1000 down)it works fine both and much better with pc to landline.
Sure it is not first digital quality but very acceptable.
Why people have problem : a non accurate dish alignment?
P. |
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