PSTN line from SPA3102 to PAP2T

[valerio martellotta]

Hello, I need to use pstn line from another building connected to the first one by a radio link. I have a SPA3102 connected to PSTN line and PAP2T in the other building. the configuration I made is the following:

On the SPA3102 PSTN Line Tab:

PSTN-To-VoIP Gateway Enable: yes

PSTN Caller Auth Method: none

PSTN Ring Thru Line 1: no

PSTN CID for VoIP CID: yes

PSTN Caller Default DP: 2

Off Hook While Calling VoIP: no

Dial Plan 2: (S0<:userid@ip_address:port>)

where userid is any userid you have setup on the PAP2T Line configuration, ip_address is the PAP2T ip address, port is the Sip Port configured on the PAP2T Line configuration.

Make Call Without Reg: Yes

PSTN Answer Delay: 3

On the SPA3102 Line Tab:

Enable IP Dialing: Yes

On the PAP2T Line Tab Configuration:

Ans Call Without Reg: Yes

Problem is that I get call from the PAP2T but I'm not able to make it. Can u help understanding my mistake?

thanks

[AASIFMADNI]

Dear Dan Lukes.

I read above all discussion about pap2t and spa3102 i am still confused that which settings is correct because i am facing same valerio martellota problem please help me regarding this that which settings i do 

[Dan Lukes]

Sorry, this discussion has been so wide. And even similar symptoms may not have common matter. Please describe your issue, what you tried, what doesn't work for you. You should create brand new thread for it, IMHO.

[AASIFMADNI]

Dan thanks for suggestion.

i making new thread but i need your help regarding pap2t and spa3102 configuration.. 

[AASIFMADNI]

DAN here is thread link.

https://supportforums.cisco.com/discussion/12570611/pstn-fax-line-pap2t-and-spa3102

[AASIFMADNI]

this is my scenario.kindly help me 

[Howard Wittenberg]

valerio martellotta,

If I understand you correctly you wish

1.  Incoming call on PSTN line attached to the SPA3102 to ring the Phone attached to the PAP2 Line 2

2.  PAP2 dialing to dial on the SPA3102 PSTN line.

Reviewing your attached PAP2 and SPA3102 configurations dated 2/22/2015 you need to make some changes.

PAP2T

For dialing on the SPA3102 you can configure either to receive a dial tone from the SPA3102 at which point you then dial the number, or you can configure for one-stage dialing where the number you dial with the PAP2T is the number that will automatically be dialed on the SPA3102 PSTN line without additional dial entry on your part.  I am configuring for the one-stage dialing where the number you dial on the PAP2T will be dialed on the PSTN line attached to the SPA3102.

PAP2T LINE 2 TAB

Change: Register: Yes to No

Change: Proxy: 192.168.2.88 to 192.168.2.88:5061

This assumes the ip address of the SPA3102 is 192.168.2.88, the SPA3102 PSTN Line Sip Port is 5061

Change Dial Plan to: (xx.)

A dial plan of xx. will allow any numbers to be dialed

SPA3102

Change Register: Yes to No

Change Ring Thru Line 1: Yes to No

Change PSTN Answer Delay: 16 to 3

Assuming 3 seconds is long enough to decode incoming caller id from PSTN call

Change Dial Plan 1: to (xx.)

Change Dial Plan 2: to (S0<:userid@192.168.2.104:5061)

Assuming the ip address of the PAP2T is 192.168.2.104 and userid is the userid configured on the PAP2T Line 2 Tab, and the PAP2T Line2 Sip Port is 5061.  This setting will automatically dial the distant PAP2T.

Your on-hook PSTN Line voltage shows 38v.  You have the Line-In-Use Voltage set to 30v.  This should be OK.  If the on-hook voltage is lower than the Line-In-Use voltage the SPA3102 would not take the like off hook to dial.  This setting is usually made about half way between the on-hook and off-hook voltage.

If you wish to capture a Syslog from the SPA3102 on the System Tab you need to change the Debug Level to 3 and on the PSTN Line Tab you need to set the Sip Debug Option to FULL.

If you wish to capture a Syslog from the PAP2T on the System Tab you need to change the Debug Level to 3 and on the Line 2 Tab you need to set the Sip Debug Option to FULL.  

I would not normally activate to capture a syslog from each device at the same time unless you are using a syslog capture program that identifies the source.

[Dan Lukes]

Busy tone & call state invalid mean the call has been rejected. It doesn't disclose why has been rejected. The reason needs to be discovered. Please follow the advices.

With the information available I can repeat that the dial plan looks very suspicious. You have either very specific goal, or it's just wrong.

You should disclose your intentions - why you have dial plan set to the particular value shown.

Unfortunately, we are not aware about your goal, so we are unable to give you more valuable advice.

Note that I'm still not sure the issue is caused by dial plan misconfiguration. You didn't supplied the required logs.

[Howard Wittenberg]

You are sending the call to the PAP2T to sip port 5061 (192.168.1.31:5061).  This is Line 2 on the PAP2T configuration.  Your .jpg does not indicate that this is the line you have configured.

You said you set the PSTN Answer Delay to 3 seconds.  The .jpg does not include that setting and you should verify that setting.  The call is not initiated until the expiration of the PSTN Answer Delay which has a default setting of 15 seconds.

Otherwise you need to determine the cause of the call failure.  The best method to do that is with the Sip Debug Trace function as suggested. Cisco has a simple pc syslog program that you can use to capture the packets
https://supportforums.cisco.com/document/36921/using-slogsrvexe-utility
You need to run the traces separately.  On the adapter System Tab you need to set the ip address of the computer capturing the trace and set the Debug Level to 3.  On the Line Tab you need to set the Sip Debug Option to FULL.

A better way to communicate the configurations is by using your web browser, save the configuration to your hard drive and then make a .zip file of the saved configuration (single save).