SPA112 registered to SIP provider by no dialtone

[Shaun Haddrill]

Hi Folks,

This is my first time configuring a voip gateway, so I may be missing something simple.

I am trying to configure a SPA112 device to connect to a SIP provider and then plug a regular analogue telephone into the phone port of the ATA device to make phone calls.

Under the voice information page the Registration State is Registered, and there are SIP packets and messages being transferred, so it looks like I have got the SIP connection working correctly.

But when I plug a telephone into either phone port 1 or 2, I hear the same flat tone.

The light to port 2 is solid green, and so I was expecting to hear a dial tone when plugging the phone cable into port 2, but this is not the case.

When I dial numbers from the analogue telephone I hear the beeping of the buttons, but the call is not initiated.

There are so many configurations under the voice settings, are some of the settings specific to the model of telephone? The model telephone I am using is a Gemini IQ330.

If logs and configuration settings would help I can provide those, I have attempted to change as few configurations as is possible to get the SIP connection working.

Many thanks for any assistance that can be provided.

[Leo Laohoo · Accepted solution]

You are from Australia.  

Internode has a very good page (HERE) about how to configure SPA 3000.  Sipura was purchased by Cisco a few years ago.  This fact is relevant because SPA 112 and 122 are both based on the SPA 3000.  

[Shaun Haddrill]

Thank you for the link, it does seem like an in-depth resource for setting the regional tones and dial plans, unfortunately I am stuck at step 7 where it mentions the " PSTN Line tab". Is this an option that exists in the Sipura and not the SPA 112?

[Leo Laohoo]

Is this an option that exists in the Sipura and not the SPA 112?

Look at what the SPA 112 and then compare with what the Sipura has.  Cisco's line of ATAs are based from the Linksys/Sipura range.  So the codings were copied across.  

This is the same process I use to configure my SPA 122.  

[Shaun Haddrill]

Yes I can see the options are almost identical between the Sipura and the SPA 112, but there are some noticeable differences.

I can't find the Disconnect Tone setting and I can't find the PSTN Line section in the options for my SPA 112.

[Shaun Haddrill]

@Leo I understand the difference now, the Sipura has an incoming PSTN line whereas the ATA 112 does not, and therefore does not have a PSTN settings section.

[Dan Lukes]

 the Registration State is Registered

The Registration State of which line is Registered ?

You are trying to use single account for both lines ? It may require specific arrangement or may not be possible at all. Have you two different models of analog phone ? Or you configured single line, but you are trying to use both plugs by single analog phone ?

Please describe the goal ...

When I dial numbers from the analogue telephone I hear the beeping of the buttons, but the call is not initiated.

So, you are hearing flat dial tone. Despite of pressing numbers, the dialing tone is still turned on and no call is initiated. True ? Otherwise, provide correct exact description of behavior you observing, step by step.

[Shaun Haddrill]

Hi Dan,

My goal is to have a single account on 1 line, the reason I currently have two lines configured is because I was trying different settings across both lines to see which would work. While I was testing the different settings I was using different accounts, currently, both lines are registered with two separate accounts.

I haven't tried a different model of analogue phone because I don't have access to a second phone currently, it's definitely something I will need to try next.

So, you are hearing flat dial tone. Despite of pressing numbers, the dialling tone is still turned on and no call is initiated. True ?

True, that's exactly what is happening.

[Dan Lukes]

My goal is to have a single account on 1 line

Thus configure Line 1 only, forget the Line 2, don't plug phone to it. If you configure both lines to the same account, both lines may not work.

Reset SPA112 to factory default, start with Line 1 only. Repeat the test. Let us know the result.

Also, you mentioned no PSTN line. I assume you have no one. Thus, don't search for settings of it.

[Shaun Haddrill]

I've followed your advice, reset to factory settings, and then set the SIP settings and still the same issue occurred.

I then followed the instructions on the Internode website for the Sipura for the second time as suggested by leolaohoo, and there is no change to the dial tone and I'm unable to send/receive calls using an analogue telephone.

I'll try a different telephone tomorrow morning, hopefully that should make a difference.

[Dan Lukes]

Well, the analog phone line is not so standardized interface. It's possible the Gemini IQ330 implementation of POTS interface is incompatible with SPA112 implementation.

You may configure syslog&debug of Voice Application running on SPA112 and catch them.

Or try another phone. It may help us to disclose cause.

[Shaun Haddrill]

I've tried a new telephone, a Panasonic KX-TBG110 Digital cordlress phone, and still the same issue is occurring.

I've attached log files, what log level should I set? currently I have selected to save error logs for both the system and kernel, and I have selected full under the line 1 "SIP Debug Option".

It seems like the phone is connected to the ATA device correctly, if I dial * 4 times I get to the configuration menu.

And looking at the Voice Information page, the ATA device sees if the phone is off the hook or on the hook, see the attached screenshots for more information.

Perhaps it's the SIP forwarding settings? I've currently got the didlogic.net service registered with the device.

[Dan Lukes]

You attached syslog of SPA112's kernel. It have no value for us. The important things are looged by Voice Application syslog. You need to have external server to capture those messages. Read Debug and syslog Messages from SPA1x2 and SPA232D ATA. Set highest log level.

[Shaun Haddrill]

I got outgoing telephone calls working with the original telephone (and the new telephone I purchased) by switching SIP providers.

It must have been a mis-configuration such as codec, or proxy authentication.

All is good now, although I never got outgoing/incoming calls working with the first SIP provider, now I can make voip calls using the current analogue telephones in my office.

The next step is to get the ATA device working with the PABX system!

Regards,

[Leo Laohoo]

The next step is to get the ATA device working with the PABX system!

Thanks for the update, Shaun & glad to see you've got it finally working. 

Think about upgrading PABX to an IP-based.  Look into Asterisk.