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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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| Posted:
May 14, 2007 - 08:17 PM |
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Hello,
I have 2 SPA-3102 units and I am trying to configure them so I can make p2p calls via a private LAN and have analog phones connected to both of them sharing a single POTS (for incoming and outgoing calls) line connnected to only one of them.
Hardware and connection I curently have:
1) Two spa-3102 each with an analog phone connected to them
2) One of the spa-3102 is connected to a POTS phone line
3) Both spa-3102s are connected together via a private wifi LAN
What I would like to be able to use this for:
1) make p2p calls between the two analog phones connected to each spa-3102 kind of like an intercom
2) incoming calls on the POTS line should ring both analog phones on both spa-3102s
3) Both analog phones should be able to make outgoing calls using the single POTS line connected to one of the spa-3102
Before someone asks, the reason for all this is that each spa-3102 is in a separate building that cannot be wired together but they are linked with a wifi network and I want the two building to use the one POTS line available in only one of the buildings and have the possibility to communicate p2p between the two buildings.
I am new to VOIP and have spent a couple of days searching for info/tutorials on the net but cannot find anything similar to what I want to do. The wizards I found all assume you use a Voip provider in some way.
Any help/pointers tutorial or book references appreciated
Thank You
Patrick Robin |
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dibsmft
Site Admin
Joined: Oct 21, 2005
Posts: 1777
Location: St. John's, Newfoundland and Labrador
Status: Offline
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| Posted:
May 14, 2007 - 08:59 PM |
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Welcome to Voipuser!
If the buidings are close enough for WIFI they are probably close enough to use a three or four unit cordless phone? The DECT phones have a built in intercom facility. This would seem to be a simpler solution. With a 3102 the phones could also be used for Voip. (That is essentially the setup that I have).
You could set up something similar with two 3102 units but you will have to take care to address each phone separately as they are on the same network. On the whole cordless phone solution is simpler. |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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| Posted:
May 14, 2007 - 09:51 PM |
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Thanks for the suggestion but we tried with various phones and they did not work because of line of sight problems. We got wifi working fine with directional antennas outside the buildings. We could not find equivalent solutions in the cordless phone world.
Patrick Robin |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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May 14, 2007 - 09:53 PM |
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You mentioned it is doable with the spa3102.
If you have any pointers/tutorials. I could not find similar setups so far to study.
Thank You
Patrick Robin |
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jcgalvezv
Moderator
Joined: Apr 07, 2005
Posts: 617
Location: San Salvador, El Salvador, Central America
Status: Offline
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| Posted:
May 15, 2007 - 06:14 PM |
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Patrick:
Lets asume 2 devices (SPA3102) A and B. Lets also asume device A's IP address is 192.168.1.50 and device BIn device B's IP address is 192.168.1.51. Finally, lest assume device A is where the POTS line is connected (local device) and Device B is the remote one.
Configure these parameters in device A:
- Line 1 tab, SIP Settings section set SIP port: 5060.
- PSTN Linetab, SIP Settings section set SIP port: 5061.
- In Line 1 and PSTN Line tabs, Proxy and Registration section all fields blank or NO except Make Call Without Reg: YES and Ans Call Without Reg: YES.
- In Line 1 tab, Subscriber Information section set Display Name: <what you want to be displayed when you call>, User ID: 11 and leave other fields empty.
- In PSTN Line tab, Subscriber Information section set Display Name: <what you want to be displayed when you call>, User ID: 12 and leave other fields empty.
- In PSTN Line tab, VoIP-To-PSTN Gateway Setup section set VoIP-To-PSTN Gateway Enable: YES, VoIP Caller Auth Method: NONE, One Stage Dialing: YES, Line 1 VoIP Caller DP: none, VoIP Caller Default DP: none, Line 1 Fallback DP: none.
- In PSTN Line tab, PSTN-To-VoIP Gateway Setup section set PSTN-To-VoIP Gateway Enable: YES, PSTN Caller Auth Method: none, PSTN Ring Thru Line 1: YES, PSTN CID For VoIP CID: YES, PSTN Caller Default DP: 1, Off Hook While Calling VoIP: NO.
- In PSTN Line tab, Dial Plans section set Dial Plan 1: S0<21:@192.168.1.51:5060>.
- In Line 1 tab Dial plan section Dial Plan field set (21S0<:@192.168.1.51:5060>), Enable IP Dialing: YES
Configure these parameters in device B:
- Line 1 tab, SIP Settings section set SIP port: 5060.
- PSTN Linetab, SIP Settings section set SIP port: 5061.
- In Line 1 and PSTN Line tabs, Proxy and Registration section all fields blank or NO except Make Call Without Reg: YES and Ans Call Without Reg: YES.
- In Line 1 tab, Subscriber Information section set Display Name: <what you want to be displayed when you call>, User ID: 21 and leave other fields emply.
- In PSTN Line tab, Subscriber Information section set Display Name: <what you want to be displayed when you call>, User ID: 22 and leave other fields emply.
- In Line 1 tab Dial plan section Dial Plan field set (11S0<:@192.168.1.50:5060>|<#9,:>[*x]x.<:@192.168.1.50:5061>)[/b] and set also Enable IP Dialing: YES
Now, how it works:
- To call device A Line 1 from device B dial 11.
- To call using device A's PSTN Line from device B dial 9# to get a second dial tone, then dial the destination number.
- To call device B Line 1 from device A dial 21.
- Whenever you receive a call in the POTS line connected to device A, phones connected to devices A and B should ring.
Probably, not sure, PSTN to remote device calls won't work since the local device's PSTN line is no registered with any SIP server. Anyway, Give a try.
I have not tried this configuration before, so let me know if this worked for you.
Juan C. |
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dibsmft
Site Admin
Joined: Oct 21, 2005
Posts: 1777
Location: St. John's, Newfoundland and Labrador
Status: Offline
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| Posted:
May 15, 2007 - 07:13 PM |
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Thanks Juan! Saved me some typing...
Before you try anything, make sure that both 3102s are working properly when connected to PSTN and the internet.
Good Luck |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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May 15, 2007 - 09:57 PM |
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Thank you very much Juan,
I made great progress with your instructions.
Doing what you said, everything works except one thing.
When I make a call over the POTS to device A, it rings only once on the device A's phone then I get a different dial tone. it seems device A answers and gives me a dial tone but a higher pitch dial tone. Device B's phone doesn't ring at all
All other features work:
1)I can call device A line 1 from device B line 1 and vice versa using 11 and 21
2)I can make a call from device B via device A's POTS line using #9
Note that yesterday evening after much trials I was able to get a POTS phone call to ring the phone on device B and I could answer it using these instructions: http://www.provu.co.uk/pdf/sipura/spa_b ... pa3000.pdf
But your instruction got me lots further by giving me the p2p feature I wanted (and had no clue how to achieve). Now I only have to make the incoming POTS calls work again. I tried playing with a combination of your instructions and the ones in the pdf above but it still gives the same result.
Any ideas appreciated.
Thank You Juan |
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dibsmft
Site Admin
Joined: Oct 21, 2005
Posts: 1777
Location: St. John's, Newfoundland and Labrador
Status: Offline
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May 15, 2007 - 11:15 PM |
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Good. Nicely done Juan.
I have not got the manual with me but I think you will have to be satisfied with A ringing for a few rings and then transferring the call to B and ringing there. You should be able to do that by going to the PSTN tab and turning on PSTN to voip forwarding
PSTN-To-VoIP Gateway Enable: Yes
Then make a dial plan (above) perhaps
Dial Plan 2: (<:extensionnumber> )
and make that the default dialplan. You may have to adjust the delays a bit. I hope jcgalvezv can make a better suggestion. |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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May 15, 2007 - 11:28 PM |
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Thank You
Assuming you are suggesting doin gthis on device a. It seems to be identical to what Juan gave in his instructions except you write this for the dial plan:
(<:extensionnumber> )
wich I assume translates to (<:192.168.1.51>)
Juan write this: S0<21:@192.168.1.51:5060>
Both cause the phone on device A to ring only once and then the POTS caller gets a dial tone. The phone on remote device B never rings
Thank You
Patrick Robin |
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dibsmft
Site Admin
Joined: Oct 21, 2005
Posts: 1777
Location: St. John's, Newfoundland and Labrador
Status: Offline
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| Posted:
May 15, 2007 - 11:45 PM |
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If I dial my 3102 from PSTN it rings about 5 times and the hands over to voip .... at that stage you want it to call the other extension on voip.
Essentially the same, yes, but you should remove the S0 as I think that passes the call from A to B at once ... you need there to be a delay. (A least that it is the way it seems with my set up). (Thats why I suggested DialPlan 2 so that you can experiment) |
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jcgalvezv
Moderator
Joined: Apr 07, 2005
Posts: 617
Location: San Salvador, El Salvador, Central America
Status: Offline
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| Posted:
May 15, 2007 - 11:57 PM |
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Patrick:
Make sure you have Off Hook While Calling VoIP: No.
Juan C. |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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May 16, 2007 - 02:09 AM |
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>Off Hook While Calling VoIP: No.
It is already.
I tried different variations and found this:
To make incoming POTS incoming calls work again I have to remove the 21 "User id" on device B line1 and remove it from the PSTN default dial plan on device A. Of course doing this disables p2p calling in one direction. The instant I put back "User id" 21, incoming POTS calls cause only one ring and then the caller gets a dial tone (seems the SPA hung up the line)
Thanks
Patrick |
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jcgalvezv
Moderator
Joined: Apr 07, 2005
Posts: 617
Location: San Salvador, El Salvador, Central America
Status: Offline
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| Posted:
May 16, 2007 - 04:23 PM |
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Patrick:
Try doing this in device A:
- In PSTN Line tab, Dial Plans section set Dial Plan 1: S0<:@192.168.1.51:5060>.
- In Line 1 tab Dial plan section Dial Plan: (<21:>S0<:@192.168.1.51:5060>).
and this in device B:
- In Line 1 tab, Subscriber Information section set User ID: <empty>. <empty> means no data in the field.
What we are doing here is removing the user id information.
Give a try and let me know.
Juan C. |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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May 16, 2007 - 04:44 PM |
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Thank You again Juan!
You last suggestion made it all work.
I will keep on playing with it a bit to add "nice to have" features like distinctive ring for the local phones p2p calls compared to external incoming calls and maybe a way to get the phone company "message waiting" dial tone transmitted to the remote device B phone.
Thanks again Juan.
Patrick |
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atelierrobin
Joined: May 14, 2007
Posts: 8
Status: Offline
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| Posted:
May 16, 2007 - 04:48 PM |
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For those attempting this from scratch on the SPA-3102, what caused me lots of head scratching initialy was that I was trying to have this work using strictly the ethernet ports on the SPA-3102. I was under the impression that if the goal is to have the SPA-3102 only communicate together on the local LAN, one should use the ethernet ports to interconnect them to the LAN switch/hub. Well I never got this to work since it seems the SPA-3102 only talk SIP over the "Internet" port. So now I use the Internet ports to connect them to my LAN switch and I do not use the Ethernet ports at all.
Regards,
Patrick Robin |
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