PSTN & FAX line for PAP2T and SPA3102

AASIFMADNI · ‎07-30-2015

dear Experts,

i am facing problem to configure pap2t and spa3102 devices for PSTN & FAX line.

here is my scenario look into and give me your valuable feed back that what should i do.?

Howard Wittenberg · ‎07-30-2015

You have two problems, the analog phone and the fax machine. You should tackle the analog phone first and get that working.

The fax machine could be much more difficult. You may not be able to make the fax work at all. You will need to be using the G.711 codec. The FAX machine should be set to use the slowest setting possible. The SPA3102 / PAP2T Administration Guide has some suggestions for SPA3102 settings for FAX on page 50. These settings would also be true for the PAP2T. The Admin Guide also discusses the T.38 Fax relay standard. T.38 is not supported by the PAP2T and is only supported by the SPA3102's FXS port, not the SPA3102 FXO (line) port, so that discussion is not relevant in your case.

From your diagram you do not show any router, just fixed ip addresses. For a voip connection you should not be cabled to the internal local network router ("Ethernet" port) of the SPA3102. You need to be connected to the "Internet" port (WAN port) of the SPA3102. This was covered in the previous thread.

If you have tried configuring the units you should post the configurations and describe the test results. If you login to an adapter with a web browser you can save the configuration to your computer's hard drive. With the SPA3102 you need to login to Voice and save and then to Router and save. Make a .zip file of the saved configurations and attach to a posting.

AASIFMADNI · ‎07-31-2015

thanks both of you.

i hear dial tone but when i dial any number dial tone change into busy tone.

below i upload my both devices configurations.

after this configuration when i dial a number to my mobile from pap2t it had fine and i get call from pap2t but when i dial any other number tone change into busy tone.

have a look my configuration and told me that where is problem and mistake...

advance thanks.

Dan Lukes · ‎07-31-2015

You should follow Howard's advices at the first. If it will not help, then ...

Busy tone may have so many causes.

Local dial plan doesn't allow particular number you are dialing.
PAP2T is unable to place call because of network condition/configuration
PAP2T is able to send SIP INVITE to SPA3102 but the cal si rejected by SPA3102

You need to identify cause. TUrn on syslog&debug messages on PAP2T and catch them. Catch SIP packets between PAP2T and SPA3102 as well.

It should reveal information important for further analysis of the issue.

Howard Wittenberg · ‎07-31-2015

You need to login to your SPA3102 again under the "Voice" Tab and save and post the configuration like you did previously under the "Router" tab so we can see how you have configured the SPA3102 PSTN Line Tab.

You do not have a conventional network with a router that directs packets. If I understand correctly your network is point to point between the PAP2T and the SPA3102. There is no connection to the internet.

You have setup the SPA3102 with a Static IP address of 192.168.1.88. In this case the ip address of the PSTN Line Tab would be 192.168.1.88:5061 assuming 5061 is configured as the Sip Port: on the PSTN Line Tab.

On the PAP2T you have Line 1 disabled and Line 2 enabled. I assume you have the analog phone connected to Line 2. On Line 2 you need to change Register: yes to Register: no before you can make a call. You have the dial plan as follows:
(<#9:userid@192.168.1.88:5060>|xx.)

I imagine dialing #9 would call the phone attached to the SPA3102 assuming the Line 1 Tab User ID is userid and the sip port number for the Line 1 Tab is 5060. I can duplicate that when I have two ata's connected with either regular cable or a crossover cable. It should work for you.

With your dial plan, dialing some number other than #9 would send that number to what you have setup as the Proxy, i.e. 192.168.1.88:5061, and since the number does not match the user id of port 5061 the SPA3102 should bridge that call to the PSTN line. I can duplicate that, bridging the call to the PSTN line, when I have the two ATA's attached directly with a regular cable or a cross-over cable. It should work for you.

So, change the Register: Yes setting to Register: No on the PAP2T Line 2 Tab and also post the SPA3102 Voice Tab configuration.

Edit: I believe the Register setting was what was stopping your call from the PAP2T.

AASIFMADNI · ‎07-31-2015

Dear Howard,

i told you earlier that above configuration had uploaded will have work only once and after that when i dial number second time i hear busy tone.

current settings allow me to receive call form pap2t at once..

anyhow i attach configuration file as you said..

AASIFMADNI · ‎07-31-2015

Dear Howard,

here is complete network diagram have a look..

Howard Wittenberg · ‎07-31-2015

So the network is conventional with a router. You have the SPA3102 setup with a static ip address of 192.168.1.88 but show it on the diagram as 192.168.1.3. Any particular reason for that?

AASIFMADNI · ‎07-31-2015

no particular reason for this actually pap2t have 192.168.1.104 ip address that's why i changed spa ip into 192.168.1.103.

now i am uploading all setting one by one in same file.

Howard Wittenberg · ‎08-01-2015

SPA3102 was not attached to PSTN Line Tab when you saved configuration. What is on-hook PSTN Line voltage? You can read this on the SPA3102 INFO Tab. Login the the SPA3102 and read and report the voltage ... PSTN Line Status .. Line Voltage:. This is just to verify one of the default settings.

Here are some changes I would make to the SPA3102 Configuration:

1. Change the User ID: on the PSTN Line Tab to something other than userid, i.e. SPA3102PSTN

2. Under VoIP-To-PSTN Gateway Setup change Line 1 VoIP Caller DP: to 1 (which is (xx.))

3. Under PSTN-To-VoIP Gateway Setup change PSTN Caller Default DP: to 2 (which is (S0<:userid@192.168.1.104:5061))

This will cause an incoming pstn line call to go to the PAP2T

4. Under PSTN-To-VoIP Gateway Setup change PSTN Ring Thru Line 1: to NO

When set to yes it will ring the phone (if any) attached to the SPA3102 before forwarding to the PAP2T

5. Under FXO Timer Values change PSTN Answer Delay to 3 (or 4) seconds (or zero if the CID comes before the 1st ring)

Set to zero the call will immediately forward to the PAP2T. Set to 3 (or 4) seconds you would delay forwarding to the PAP2T if you wish to capture an incoming caller ID. You have PSTN CID For VoIP CID: YES to send this caller id to the PAP2T

6. Under PSTN Disconnect Detection you have Detect PSTN Lone Silence: Yes, PSTN Long Silence Duration: 10 seconds. This is a short period of time. Maybe it should be longer? This setting is used to disconnect the PSTN Line when the SPA3102 does not detect a more conventional disconnect signal such as a CPC signal, disconnect tone, or a polarity reversal.

7. On the PSTN User Tab you have an entry under Cfwd Sel1 Caller: and Cfwd Sel1 Dest:. Erase those entries. The call forwarding is done by the dial plan. Those entries are not PSTN caller ID's in any event.

i told you earlier that above configuration had uploaded will have work only once and after that when i dial number second time i hear busy tone.

When it works once and not the 2d time it is more than likely due to the SPA3102 failure to detect a disconnect signal from the PSTN Line and the PSTN line is still off hook when you try a 2d call. When the conventional signals do not work the SPA3102 has a provision to enable silence for a period of time as a signal to disconnect the call. You have already setup silence on the PSTN Line as a signal, you could also enable silence on the voip side of the call as a signal.

AASIFMADNI · ‎08-06-2015

Dear Howard,

now i am sending you spa and pap2t configuration in PDF format kindly look.

Howard Wittenberg · ‎08-06-2015

The PAP2T configuration looks OK assuming you changed the static ip configuration of the SPA3102 to 192.168.1.103. You previously had it set to another address. By putting the configuration in .pdf format, you did not include the SPA3102 INFO Tab or the Router Tabs which would tell me the SPA3102 ip address and the voltage level of the attached pstn line. When you save the configuration with your web browser you get all the tabs with one save when logged into Voice Tab and one save when logged into the Router Tab.

On the SPA3102 you have the SPA3102 PSTN Caller Default DP set to 1 which is (xx.). This will return a dial tone to the PSTN caller. You should have it set to 2 which is (S0<:userid@192.168.1.104:5061) which will immediately dial the distant PAP2T assuming the PAP2T ip address is 192.168.1.104.

On the SPA3102 you have PSTN Ring Thru Line 1: YES and PSTN Answer Delay: 16 which will attempt to ring the phone (that does not exist on your diagram) attached to the SPA3102 for 16 seconds before accessing the PSTN-to-VoIP Default Caller DP which would ring the distant PAP2T.

On the SPA3102 you have the SPA3102 Line 1 VoIP Caller DP set to 2 which is (S0<:userid@192.168.1.104:5061). This applies to a phone attached to the SPA3102 which you do not show on your diagram so I guess it is not important, however it won't work.

AASIFMADNI · ‎08-19-2015

Dear Harword,

i am so sorry for late reply because of busy schedule.

i am still facing that problem.when i dial any no after 5 seconds i have received busy tone from pap2t.

if i get off RJ11 connector from spa3102 than i hear same busy tone.

please make it easy for me.

thanks

Howard Wittenberg · ‎08-20-2015

It would help to see the current SPA3102 and PAP2T configurations.

Explain what does work and what does not work.

Your problem could be the detection of a disconnect of your attached PSTN Line to the SPA3102.

When your PSTN disconnects the line it must send a disconnect tone that you have configured, or give a CPC signal (voltage drop) or you must do it by detecting silence on the line for a configured period of time. These are configuration settings.

AASIFMADNI · ‎08-20-2015

What should i do???