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

[Dan Lukes]

Outgoing call can be refused either at local level, because of Dial_Plan settings or at remote level - SIP request either doesn't reach the destination, or they are rejected by them.

You need to identify the reason.

Turn on syslog&debug and catch them as well as SIP packets sent and received by the device.

It will help you (or us) to analyze the issue.

[valerio martellotta]

thanks for the answer, I would like to use it by dial plan. Attached both of the configurations, Please help me.

[Dan Lukes]

Your Dial_plan look very suspicious. But there is nothing like "universal dial plan". If your unit is used in public phone network, you need to follow national dial plan of the country in question. If your unit is connected to private/corporate phone network, you need to configure dial plan according this network rules.

Unfortunately, you disclosed no details about numbers you are tried to dial nor the kind of voice network the ATA is connected to.

Note 1: I'm not claiming that your current dial plan is causing the issue. I have no informations that allow me to make such kind of conclusion.

Note 2 (off topic): Even your codec settings looks suspicious. In Italy the most common preferred codec is G711a, not G711u. Of course, you need to follow your Telco requirements, so as long as your upstream provider order you to use G711u there is no issue.

[valerio martellotta]

Hello,

those are the full settings of the devices. I make several more attempts but nothing. When I try to make a call from the PAP, I got busy tone and call 1 state: Invalid and Call 1 Tone: Ring Back.

[Howard Wittenberg]

There is no userid configured on the PSTN Line Tab of the SPA3102.  You need to have a userid configured and it needs to match the userid in the dial plan on the PAP2T.

You have the PSTN Line-In-Use Voltage set at 50v.  In this case, if the actual voltage on the PSTN line is lower than 50v the SPA3102 will return a sip error code indicating the service is not available (Sip 503) and you will receive a busy tone.  You can look at the SPA3102 INFO Tab and read your on-hook PSTN Line Voltage.

The default Line-In-Use Voltage setting is 30v.  It is normally set about 1/2 way between the (higher) on-hook voltage and the (lower) off-hook voltage disregarding the polarity.  The SPA3102 uses this setting to determine if the PSTN Line is already in use.  

You are setup for 2-stage dialing, i.e. the SPA3102 should return a dial tone upon receiving the call from the PAP2T.

[valerio martellotta]

Hi, sorry about the delay but I was too busy and I didn't have time to solve the problem. At the moment anyway I've installed another system and I've to fix this problem. I've tried a new configuration and at the moment I can do something different. I can call from the spa3102 and I can hear the dial tone on line 2 of the pap2 but I can't call using it(after that compose the number I get the busy tone). With this configuration anyway only the phone connected on spa3102 ring when I receive a call. I worked with the dial plan and I tried also to install a syslog but I can't see nothing interest and that can help me. 

I changed also the dial plan and I set up the "regional" parameters with the italian ones. 

However my goal is to receive and make call from the pap2 and spa3102 using just the PSTN line. I don't need and I don't want to use, at the moment, any voip account. 

attached my last configurations file. 

thanks again for all 

[Dan Lukes]

E.g. you wish for calls just from Line 2 of PAP2T to SPA3102 and vice versa.

Well. Then set Register to No on both devices. There is no one responding to REGISTER requests. It will not solve your problem, but it will make thinks simpler.

According the issue ...

If I understand correctly, you dialed 3206637253 on Line 2 of PAP2T. It's all I know as you disclosed almost no information about the failing call. As you keep secret even the syslog&debug messages it's hard to give you a valuable advice.

Well. The call path may be broken on PAP2T.

1. If it will not fit Dial Plan the call will be rejected with no further action.

2. Otherwise an INVITE packet will be sent to SPA3102.

Catch SIP messages between PAP2T and SPA3102 and disclose them. It will allow us to distinguish [1] from [2].

[3] If call setup arrive to SPA3102 it may be rejected here. Either because of Dial Plan or because SPA3102 doesn't know where the call should be routed to.

Catch syslog&debug messages and disclose them - they should allow us to verify or reject the [3]. Also, it will reveal where call is routed to unless rejected. Note that SIP messages between PAP2T and SPA3102 should be catched as well if possible.

[4] Unrejected call is router somewhere - either to Line 1 or PSTN line of SPA3102. It may be rejected by peer. Catch syslog&debug messages and disclose them - they help us to identify call progress and abort reason. Note that SIP messages between PAP2T and SPA3102 should be catched as well if possible.

Just as a test you should use numeric phone numbers instead of text strings.

With this configuration anyway only the phone connected on spa3102 ring when I receive a call

"when I receive call" from where ? From PSTN ?

All at all, I'm not sure the SPA3102 can create two legs (line 1 of SPA3102 and PAP2T via SIP) for one incoming calls. SPA3102 is rather VoIP bridge than full featured switch ...

[valerio martellotta]

I have tried your configuration but something is still wrong. 

I try to be more accurate:

1) I want to receive calls from outside callers that compose my PSTN number. In this case I want that my phone connected to spa3102 port "phone" ring and phone connected on "line 2" of pap2t ring too (simultaneously). None of the phones rings with this configuration! Changing "Ring Thru Line 1: No to Yes" make rings the phone connected on spa3102 for just one time. If I try to answer I can hear a fust busy tone.

2) make call from spa3102 with the phone connected on "phone" port OK!!!

3) make call from pap2t with the phone connected on "line 2" port KO!!! (with the configuration suggested I can hear the dial tone but when I compose the number (any) I can hear just silent. Check the file "screenshot" for more info.

Attached also the log file of spa3102 when I call my number and new configuration file

[Howard Wittenberg]

valerio martellotta,

With the SPA3102 you can not ring the phone connected to the SPA3102 and the phone connected to the PAP2T simultaneously. You can ring the phone connected to the SPA3102 for a period of time and then if not answered you can bridge the call to the PAP2T.  For an incoming PSTN Line call you set this up on the PSTN Line Tab with Ring Thru Line 1: Yes to ring the phone attached to the SPA3102 and then set the period of time you wish the ringing to continue to that phone by setting PSTN Answer Delay to the number of seconds you wish the phone attached to the SPA3102 to ring before you bridge the call to the PAP2T.  The default setting of PSTN Answer Delay is 16 seconds.  If you have that setting on the SPA3102 PSTN Line Tab the phone on the SPA3102 will ring for 16 seconds before the phone on the PAP2T will ring.

You make a call from the SPA3102 with the phone connected to the "phone" jack on the SPA3102 on the PSTN Line using the dial plan on the SPA3102 Line 1 Tab.  You could use a dial plan like this on the SPA3102 Line 1 Tab.  (xx.<:@gw0>)

If you also wish to make a call from the SPA3102 to the distant PAP2T with the phone connected on the "phone" port of the SPA3102 you need to add this function to the dial plan on the Line 1 Tab.  You could do this by enhancing the above dial plan with a dial plan like this where you dial #9 to call the PAP2T.
(<#9:userid@192.168.2.104:5061>|xx.<:@gw0>)

If you wish to call the phone attached to the SPA3102 from the phone attached to the PAP2T you could do that from the dial plan on Line 2 Tab of the PAP2T.  A dial plan like this would make that call by dialing #9:

(<#9:userid@192.168.2.88:5060>|xx.)

The syslog you posted is missing some information.  The log is from the SPA3102 and you need to enable Sip Debug Option: FULL on both the Line 1 Tab and the PSTN Line Tab on the SPA3102.

I tested the above on my equipment.  I do not see any error on your configurations regarding the call from your PAP2T calling out the PSTN Line attached to the SPA3102.  If you are still having problems make the changes to Sip Debug Option and post new syslog showing the call.

On the SPA3102 VoIP Caller Default DP: is set to None.  You could have it set to 1, however that should not make any difference.

[valerio martellotta]

Hi Howard Wittenberg,

firstly thanks a million for your help. 

After your last message I can make call from pap2t to spa3102 but I can't hear the caller and the caller can't hear me. 

I don't have a chance to call from spa3102 to pap2t because my PSTN provider asnwer with a registered voice saying that I can't make this kind of call. 

I still don't have the chance to make a call from pap2t. I can hear like the conversation want to start but even in this situation I can't hear nothing and nothing ring anyway. After composing the number I can hear just silent.

I have tryed to catch a log but setting "full" it dosen't work (Syslog server error "Run-time error "13" Type mismatch") . Selecting "1-line" the result is on the attached files. 

One thing more....I have connected the spa3102 only on my local network by "ethernet" port and there is nothing connected on port "internet" port. Is this a problem? Do I have to setup on my router some default route for the traffic between spa3102 and pap2t? 

[Howard Wittenberg · Accepted solution]

One thing more....I have connected the spa3102 only on my local network by "ethernet" port and there is nothing connected on port "internet" port. Is this a problem?

Yes, I would say that generally is a problem.  My opinion is that the SPA3102 was not designed/tested to make calls using its local SPA3102 network and that could be the reason you have audio problems.  You need to be using the "Internet" (WAN) jack and for ip addresses either use DHCP from the router or use a static ip addresses in your main router's subdomain.  

If you change IP addresses, you will need to adjust the ip addresses used in your direct ip calling configuration.   You should use ip addresses that will not change over time if the router is rebooted.  The router job is to route the packets between the ip address used for the SPA3102 and the PAP2T.  You generally do not need to setup a path between the two addresses.

In addition, you are running an older version of the SPA3102 firmware (3.3.6). The current firmware version is 5.2.13 and can be download here:
https://software.cisco.com/download/release.html?mdfid=282414112&softwareid=282463187&release=5.2.13

Make the above change and post some new sip debug traces of problem calls and configurations showing the ip address changes.  The syslog posted show the title of the messages but do not show the detail of the messages (Sip Invite, Sip 200 OK, etc).  The detail will show the addresses passed by the call for the audio.  Perhaps a different syslog program might show the detail.  Cisco has a simple windows pc program that you can download here.

https://supportforums.cisco.com/document/36921/using-slogsrvexe-utility

On the SPA3102 and the PAP2T System tab you have an ip address for both a syslog and a debug server.  You only need an ip address for the debug server.

[valerio martellotta]

Almost done!!! I can make call from pap2t to spa3102 and viceversa (audio ok)

I can call any number from pap2t and from spa3102!!!(audio ok)

I can't receive call to pap2t!!

When I try to call the PSTN line from my mobile the spa3102 start to ring and after 5 second (PSTN Answer Delay setup to 5) it finish to ring and my mobile is like connected but I can hear just a tone like "prompt tone".....if I open the phone connected to pap2t or the phone connected to spa3102 (spa3102 is not ringing anymore) I can here the "Prompt tone" too. I have tried to set "PSTN Ring Thru Line 1:" to no but nothing change.

At the moment I have connected just the "internet" port and nothing at the "ethernet" port. 

thanks again 

[valerio martellotta]

PROBLEM SOLVED!!! PSTN Caller Default DP: set to 2 and not 1.

everything work well at the moment!!!thanks a million for your help

[Dan Lukes]

You should consider rating of valuable Howard's responses. Also consider to use "Correct answer" button. It will help others to found solutions.