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VoIP2Go

join:2013-12-14

Forward All incoming PSTN calls to a SIP URI

Well, I was sure I could do this with my SPA3102 but having read through the Admin Guide, I'm not sure it's possible. Please correct me if I'm wrong

Basically, I want to forward all incoming calls from the PSTN to a SIP URI without any authentication or other key presses required.

Will my Obi110 allow me to do this if I dig it out from the bottom of my unused gadget box?



Qsig

join:2009-05-18
Kanata, ON

I would do this more at the SIP provider level so you're not hairpinning calls back out from your own internet connection (if they are going back out). I know voip.ms and Callcentric can do this very easily, not sure on other providers.


VoIP2Go

join:2013-12-14

Thanks, but this has to be done without the involvement of a SIP Provider.


hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..
reply to VoIP2Go

You can do this with a SPA3102. You just put the sip uri in the applicable dial plan for the PSTN Line Tab on the SPA3102. Edit: On the SPA3102 you may need to set Enable IP Dialing: Yes on the Line 1 Tab.

You can do this on an OBi110. You just put the sip uri in the InboundCallRoute for the Line Port on the OBi110

I would show examples except the forum strips out some of the characters


garys_2k
Premium
join:2004-05-07
Farmington, MI
Reviews:
·Callcentric
·callwithus

said by hwittenb:

I would show examples except the forum strips out some of the characters

Put the stuff you want to show between (code)...(/code) tags, where you replace the () with square brackets. It will come out like this:
 A bunch of special characters:
~!@#$%^&*()_+`[]:;"'
 


ThaiGuy

join:2008-05-10
Thailand
reply to hwittenb

OK Thanks - I'll have a play with my SPA3102.


hwittenb

join:2003-12-20

SPA3102 example:

 "(S0<:17771234567@in.callcentric.com>) or (S0<:userid@21.22.23.24:5060>)"
 

VoIP2Go

join:2013-12-14

I tried the second string, but even though the call was forwarded to the SIP URI, the hard coded UserID of the SPA3102 is passed in the INVITE as Caller ID. I realized I need to pass on the caller ID from the originating POTS caller. Do you think that might be somehow possible?


hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..

said by VoIP2Go:

I tried the second string, but even though the call was forwarded to the SIP URI, the hard coded UserID of the SPA3102 is passed in the INVITE as Caller ID. I realized I need to pass on the caller ID from the originating POTS caller. Do you think that might be somehow possible?

On the PSTN Line Tab under PSTN-To-VoIP Gateway Setup you need to set "PSTN CID for VoIP CID: Yes" and also you need to set the PSTN Answer Delay to 4 (or maybe 3) seconds, long enough for the SPA3102 to decode the incoming caller id on the analog pstn line.

If it doesn't work right away I would run a sip debug trace to see if the SPA3102 is seeing the incoming caller id. There will be an entry in the trace for the CID if it is decoded.

VoIP2Go

join:2013-12-14

That worked - Thanks. The only problem I have now is that when the PSTN caller hangs up the SPA3102 does not detect anything and the call with the SIP URI at the far end continues indefinitely. I'm actually using a GSM terminal in place of PSTN, so perhaps that has something to do with it. I have GSM terminal 1 calling GSM terminal 2 then SPA3102 connected to GSM terminal 2 forwarding to the SIP URI. I have experimented with a few settings that I thought may affect the ability to detect the disconnection but no luck so far.


hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..

said by VoIP2Go:

That worked - Thanks. The only problem I have now is that when the PSTN caller hangs up the SPA3102 does not detect anything and the call with the SIP URI at the far end continues indefinitely. I'm actually using a GSM terminal in place of PSTN, so perhaps that has something to do with it. I have GSM terminal 1 calling GSM terminal 2 then SPA3102 connected to GSM terminal 2 forwarding to the SIP URI. I have experimented with a few settings that I thought may affect the ability to detect the disconnection but no luck so far.

The SPA3102 has a fallback line silence settings for disconnect detection "Detect PSTN Line Silence: Yes/No, PSTN Silence Threshold:, and Detect VoIP Long Silence Duration: Yes/No". The PSTN Silence Threshold is a sensitivity setting.

It is best, however, if the GSM unit has a setting to give the SPA3102 a CPC signal (loss of voltage) or a disconnect tone when the GSM caller disconnects.

twinclouds

join:2010-06-12
San Diego, CA

1 edit

I tried on my SPA3102 using hwittenb's method. The SIP phone connected to the Sip URI specified would ring. However, the phone connected to the telephone (FXS) port no longer ring. Is there anyway to make both ring? Without enabling the PSTN to VOIP gateway, the phones attached to the telephone (FXS) port will always ring.


hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..

said by twinclouds:

I tried on my SPA3102 using hwittenb's method. The SIP phone connected to the Sip URI specified would ring. However, the phone connected to the telephone (FXS) port no longer ring. Is there anyway to make both ring? Without enabling the PSTN to VOIP gateway, the phones attached to the telephone (FXS) port will always ring.

The SPA3102 inner logic is to first ring the phone attached to the SPA3102 and then go to the PSTN-to-VoIP gateway, i.e. first ring the phone attached to the SPA and then go to the call bridging logic you have setup in the PSTN-To-VoIP gateway. The phone attached to the SPA3102 will ring during the PSTN Answer Delay period assuming you have set the Ring Thru Line 1: Yes setting. The default PSTN Answer Delay setting is 16 seconds. For simultaneous ringing you need to get an OBi adapter.

twinclouds

join:2010-06-12
San Diego, CA

3 edits

I do have an OBi110 and will give it a try. Thanks for your prompt response. I found telephone wiring always introduce some noise. If this approach works well, I can have two sets of cordless phones attach to them and have one PSTN rings both sets.
As for OBi110, it looks like the InboundCallRoute can spesify multiple destinations. I assume {ph} is for the phone. Can I specify multiple SIP URI so it can ring phones attached to more than one ATAs? Can I specify as say: {192.168.1.100},{192.168.1.101},{ph} or different?


hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..

said by twinclouds:

I do have an OBi110 and will give it a try. Thanks for your prompt response. I found telephone wiring always introduce some noise. If this approach works well, I can have two sets of cordless phones attach to them and have one PSTN rings both sets.
As for OBi110, it looks like the InboundCallRoute can spesify multiple destinations. I assume {ph} is for the phone. Can I specify multiple SIP URI so it can ring phones attached to more than one ATAs? Can I specify as say: {192.168.1.100},{192.168.1.101},{ph} or different?

Obi calls this forking the call. I believe you would setup the Line Port Inbound Call Route like this:
{SP2(userid@192.168.1.124:5089),ph}
or for two destinations
{SP2(1234567@192.168.1.124:5089),SP2(177712345678@192.168.1.127:5023),ph}
 
where the sip uri's include the user id and the sip port number in addition to the ip address.

Note: I tested the above calling other phones on my local network. The phones all rang simultaneously. The first phone was on a SPA3102, the other phone on a 2d OBi. The call to the SPA3102 was fine. The phone on the 2d OBi had one-way audio which is a problem you can encounter with OBi adapters and probably you could troubleshoot to find a solution. The OBi adapters, in my opinion, are sometimes very difficult to get working properly with direct ip calling particularily when you are trying to send calls to another address on your local network. It doesn't appear that they really planned to have people using direct ip address calling, they planned to have you use their OBiTalk network.

twinclouds

join:2010-06-12
San Diego, CA

hwittenb: Thank you very much for your detailed instructions and insights. I will try it out to see how it works. Really appreciate your help!