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New Member

SPA3102 : Can I make it to work this way ??


I have a simple analogic telephonic line with a simple analogic phone.

I want to connect the line to "line" connector of my SPA and the telephone to "phone" connector and use it this way:

The phone must be registered to an asterisk pbx on local LAN (I've understood to configure and use the WAN connection anyway)

The incoming calls from telco line will make the phone to ring

The asterisk extensions calls to SPA will make the phone rings as well

The dialed number by the phone will be routed to asterisk or PSTN following a dialplan.

I do NOT need any interaction between PSTN and Asterisk ( no calls from PSTN and asterisk and vice-versa )

Pratically speaking, I want to keep the PSTN---Phone connection transparent with adding the asterisk feature to my phone only

I disabled on  PSTN Line port the "PSTN-to-VoIP" and "VoIP-to-PSTN" features and I set the  "PSTN Ring Thru Line1" feature

I configured Line1 (the phone port) to register as Asterisk pbx extension

I configured Line1 with the following dialplan : (0[1-9]xx.<:@gw0>|3xx.<:@gw0>|2xx) in order to place calls starting with a single 0 or with 3 through the PSTN line and the extension 2XX thyrough asterisk

Incoming calls from PSTN or Asterisk extension are ringing my phone but unfortunately I'm not able to place any call either to PSTN or to Asterisk extensions

Is there any particular setting more to be configured to ????

Thank you for any comment or suggestion



SPA3102 : Can I make it to work this way ??

To place a call from the phone to the pstn line you need to enable the VoIP-to-PSTN gateway (VoIP-to-PSTN Gateway Enable: Yes) and set One Stage Dialing: yes.  Your Line 1 dial plan allowing 0+ and 3+ calls to the pstn line looks OK.  On the PSTN Line Tab the Line 1 VoIP Caller DP will be pointing to a Dial Plan (xx.).  You need to enable the VoIP-to-PSTN gateway because internally the SPA3102 places an internal voip call from the Line 1 phone to the VoIP-to-PSTN gateway.

For the asterisk extension calls your dial plan will allow 3 digit dialing to your asterisk system if the number starts with 2.  You must have some other problem.  What happens when you dial the 3 digit number? 

A sip debug trace will show what the SPA3102 sends to your asterisk system.  To run a sip debug trace you download and install a Syslog program on a local computer, you put the local computer's ip address in the Debug Server field on the SPA3102 System Tab, set the Debug Level to 3 on the System Tab, and on the Line 1 Tab you set the Sip Debug Option to FULL.  You can download a simple Windows Syslog program here: 

New Member

SPA3102 : Can I make it to work this way ??

Thank you for your helpful answer

I thought there was some sort of "analogic bridging" to enable between these two interfaces.

The voip to PSTN enable , instead, made it work !

The dialplan has revealed OK , when I dial 2XX I can call immediately any asterisk extension,

any dial other than 0[1-9]xx. 3xx. and 2xx is refused by the system like needed.

Two questions about dialplans :

Why is suggested 3xx. rather than a simple 3.  ???

If I need two rules with overlapping dials , should I place them in right sequential order ??

i.e.:  numbers starting with 0 via PSTN  but those starting with  0123 via asterisk(voip)  , should I place the more restrictive matching before the other one ??

(in this example (0123xx.|0xx.<:@gw0>) ??

Thank you

New Member

SPA3102 : Can I make it to work this way ??

A note about audio quality:

I noted sometimes an unwanted echo effect during a line1---PSTN call,  it should be inexistent as there is no latency or other network related problem between "inner" voip communication.....

Is there some pararmeter to work around ??

Could it be due to regional setting of PSTN section ??

Thank you

Re: SPA3102 : Can I make it to work this way ??

In a dial plan, the single dot (.) means 0 or more entries of the digit.  If you had just 3. it would mean 3 or 33 or 333 etc. When you enter x it means any character on the phone keypad.

The dial plan logic goes from left to right.  On something like (0123xx.|0xx.<:@gw0>) I would think something that started with 0123xx would go to Line 1 voip, something else that started with 0 would go to the PSTN line.  When in doubt, it is always possible to test your assumptions to see where the call is going using the sip debug trace I mentioned previously.

As for the pstn line echo, first make sure you have the proper impedance setting for the pstn line for your country.  If you still have an echo you could try adjusting the PSTN To SPA gain and SPA to PSTN gain on the PSTN Line Tab.  The default setting is 0.  The allowable range is -15 to 12.

dB of digital gain (or attenuation if negative) to be applied

to the signal sent from the PSTN side to the SPA. The range

is -15 to 12.

dB of digital gain (or attenuation if negative) to be applied

to the signal sent from the SPA to the PSTN side. The range

is -15 to 12.