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SPA3000 not estabilishing PSTN incoming calls

Alan.Vinicius91
Level 1
Level 1

Good morning everyone!

I'm from Brazil and I currently having many problems to provide a solution for my University's Final Project.

Currently I have the following:
Softphone/Analog Phone <<< >>> Asterisk <<< >>> SPA3000 <<< >>> PSTN line.

I configured and register both PSTN and LINE 1 ports on Asterisk and it's fine in this sense. 
However, everytime that I try calling my softphone/analog phone through PSTN, the SPA3000 routes the call to the softphone/analogphone, it rings, answers, but the call has huge noises and drops after seconds. Sometimes it keeps rings before it drops. 
The strangest thing is that the originator equipment that calls my PSTN number (Cellphone or telelphone), doesn't seem to estabilsh the call, even when the softphone answers it, because the originator keeps receiving the ringing tone.

I ran syslog in the pstn port and when the softphone answered and dropps after seconds, one of the outputs that I received in the log was: "x-asterisk-hangupcausecode=17"

Did anyone have any issues like this?


Thanks!

24 Replies 24

Note the CPC is optional and not well standardized extension of POTS line protocol. Phone company should disclose specification of phone line in question. There may be no CPC defined at all, so voltage dropout should not be considered CPC.

Note that lack of CPC may cause the SPA3000 will not be aware that called phone become on hook so call will be disconnected in full by calling phone on-hook state only.

It's nothing so special - the old relay-based phone switches behave same way - there has been no way to disconnect path from called side.

 

Guys, talking about this CPC and all this stuff made me search for some info.

And I found some related to voltage.
What happens is this:
Hook 0 (V) State:    On    Line Voltage:    0 (V)

This is totally wrong, correct? I should have 48V for instance in line onhook and it shows up 0V.
When I plug a phone instead of the LINE SPA port, I receive dialtone and etc. When I plug the SPA, it always look like this and sometimes my calls to the line port fails.

Hook 0 (V) State:    On    Line Voltage:    0 (V)

This is totally wrong, correct? I should have 48V for instance in line onhook and it shows up 0V.
When I plug a phone instead of the LINE SPA port, I receive dialtone and etc. When I plug the SPA, it always look like this and sometimes my calls to the line port fails. -
 

The SPA doesn't attempt to make a call to the line (FXO) port if the voltage is lower than the "Line In Use" setting (disregarding polarity).

When you plug your telephone company PSTN line into the Line jack on the SPA3000 and then with the PSTN line on-hook you look at the INFO Tab of the SPA3000 you should see Line Voltage of nominally +/- 48v.

If the PSTN line is off-hook the voltage will drop to +/- 6v to 8v or so.  If the voltage shows 0 that generally means the line is not connected.

The voltage comes from the telephone company central office.

If you plug the PSTN line directly into your analog phone and receive dial tone there is a voltage on the line.

 

 

 

Beware that even nominal voltage is network specific. About either 60V or 48V are most common, but even 24V may be seen on some networks. The off-hook state is defined in the term of minimum current flowing thru line.

The line may be shorted with no harm to anything. It will cause voltage drop to 0V (measured at wall socket) while the line still be considered off-hook by CO.

Off-hook voltage is thus matter of line resistance and resistance of phone itself. As long as device have independent power source (e.g. is not line-powered only) it's input resistance can be very low causing off-hook voltage to drop to almost zero.

I don't know the nominal input resistance of SPA3000 in off-hook state. Typical resistances of end devices are few hundreds of ohms (like 200-300).

The "line-is-in-use-logic" is based on line voltage measurement while device is still in high resistance (on-hook) state. If voltage is lower than margin it is considered that there's other device in off-hook state on line causing the voltage drop. It may not work reliably where you are so near to CO. In such case the own resistance of wire will be rather low while impedance of connected device will be still the same causing small drop only thus resulting off-hook voltage will be higher than usual. As maximum line current is limited on CO side as well the things become more complex.

Things like flash support, CID transfers (especially those variants transferring CID in on-hook state), CPC (if supported at all), pulse dialing support, ... make things even more complicated.

It's why some line values are configurable on SPA3000 while other values are considered "valid" in wide range of values. Even in such case particular end device (SPA3000 here) may not be compatible with particular phone network in full.

 

Sorry I'm not offering simple answers to you ;-)

 

... but you got one from Howard already - turn off CPC detection and try again.

 

 

Hi Guys,

I have all inbound calls working properly with a bit the noise, but I ´m able to hear the caller.

Thanks for comments to solve the inbound calls. So, My problem was sip trunk configuration and routing.  

The next step is clean the line, I hope hear the caller without noise. How can I improve that, please?

So, I have changed the following parameters:

- My analog phone has been configured on extension_custom
- I left only once trunk registering PSTN line and line 1
- I have created Inbound call with DID as "My PSTN number" and forward the call to trunk
- I have created a misc application for asterisk understand internal routing
- on the Line 1 Tab: Enable IP Dialing: Yes.
- Off Hook While Calling VoIP: YES
- At trunk I put "from-trunk" to register PSTN line and "from-analog" for line 1.
- In line 1 I have changed all parameters in "port polarity config" to Reverse
- On PSTN line my dial plan is my PSTN did number

 

Regards,

Guys, my problem should be only the SPA.

Right now I have a cable from PSTN (That works, because I tested and plugged many times my telephone to it) connected to the line/FXO port and it shows Line: On Hook and 0 Voltage.

Is there any way that I can test if the FXO port is dead?

Do you know anything about it? Troubleshoot, if firmware upgrade solves or it is not working due to a misconfiguration on the tabs of SPA?

The SPA3000 needs to accurately determine the voltage level on the PSTN line.  If it does not do so the SPA3000 may be defective.

You have already posted results that show the SPA3000 receives ringing from an incoming call and takes action on the call.  The action fails when the SPA3000 believes there is no voltage on the PSTN line and it has received a disconnect signal from the PSTN line.

You have the latest firmware released for the SPA3000.  My copy of the release notes for the firmware say copyright 2007.  Cisco discontinued the product several years ago.

Is there any way that I can test if the FXO port is dead?

Yes.

And No.

Have you equipment to measure loop current as well as tip-to-ring voltage ?

Then measure the values for both seized and unseized state when standard phone is connected (you claimed it works with phone). As we know nothing about your's particular PSTN line we need to establish "baseline".

Unfortunately, we need to establish one more baseline. The 0V announced by WWW UI may be caused by bug in WWW UI code (possibly related to particular configuration and/or firmware) so device works correctly with the exception it show wrong voltage in UI.

But I have SPA3000 nowhere to compare.

I may advise you to reset the SPA3000 to factory default to eliminate the possibility the problem is caused by particular configuration, but regardless of it, it will be hard to make final decision the device is broken or just not configured properly.

 

According the SPA3000 documentation, the PSTN is considered "not connected" if there's no loop current detected or tip-to-ring RMS voltage is below 1V. But line is not considered "not connected" in this particular case, so device seems to be aware that there is no 0V voltage on line. It give you a slight chance it's WWW UI bug only. But don't put so much hope on it.

 

By the way - consider rating of valuable comments. It will help others to found advices.

 

felipecarneiro
Level 1
Level 1

Hello,

I'm trying to configure a SIP trunk between CME and a SPA3000. Calls from the analog phone to VoIP (SIP and SCCP) are working perfectly, but I'm unable to call from the VoIP phones to the PSTN connected on SPA3000.

How have you configured the SPA300 and Asterisk to route calls from VoIP to PSTN?

Thanks in advance!

First paragraph mention SPA3000 connected to CME. Question mention SPA300 connected to Asterisk instead.

Thus I'm confused about the topology.

Also, "I'm unable to call" is rather vague description. It doesn't help to neither understand nor analyze your issue.

 

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