Calling out on line port not working from SP

[rjdyck]

I am trying to decide if if I have a bad obi110. This is a new unit. I have some experience with a SPA3000 so this really is not very new for me. I want to be able to call in on an SP and get PSTN dialtone.
This works - SP1->AA->LI
This works Phone->LI
This doesn't work - Sp1->LI
No fancy digit maps. The SP1 inbound route is simply LI
I am connecting to the PSTN alright. I hear a little bit of line hum for a few seconds and then a recording saying "the number you have dialed can not be completed as dialed".
I tried (<@:>):li to eliminate any digits but it didn't help.

Any suggestions?
One other unrelated thing, The Obi can not detect an offhook  on the phone line. I played with the voltage settings to no avail.

[drgeoff]

I'd guess that message is coming from your PSTN service provider.

Log in to the OBi's local web interface.  Click on Status, then on Call History and see what digits are being dialled on the LINE port.

Normally an InboundCallRoute would send CallerID as DTMF tones to the called terminal.  Could that be what your PSTN service provider is receiving and rejecting?

[rjdyck]

The call history shows no digits dialed.
SP1 inbound ringing
line1 outbound connected

A PSTN line never receives caller ID it only sends it and caller ID is sent as FSK not DTMF.
When the line receives a "call for service" ( a loop ) it attaches a DTMF receiver to the line. The receiver is not capable of reacting to FSK even if it was sent down the line. However the DTMF receiver is capable of receiving legacy dial pulses. My guess is that there is pulse on the line ( loop, break, loop ) and the receiver interprets as a digit 1 ( if it was a single pulse ). A single digit other than 0 does not match any dial plan and it times out waiting for more digits.

[drgeoff]

The call history shows no digits dialed.
SP1 inbound ringing
line1 outbound connected

A PSTN line never receives caller ID it only sends it and caller ID is sent as FSK not DTMF.
When the line receives a "call for service" ( a loop ) it attaches a DTMF receiver to the line. The receiver is not capable of reacting to FSK even if it was sent down the line. However the DTMF receiver is capable of receiving legacy dial pulses. My guess is that there is pulse on the line ( loop, break, loop ) and the receiver interprets as a digit 1 ( if it was a single pulse ). A single digit other than 0 does not match any dial plan and it times out waiting for more digits.

When you set InboundCallRoute to 'ph', the OBi sends the Caller's number, if available, to the phone.  That can be sent to the phone as DTMF tones if that method is selected on the OBi under Physical Interfaces, Phone Port, Port settings, CallerIDMethod.  Yes, DTMF is unlikely to be selected in N.America but your use of the term 'PSTN' rather than 'POTS' suggested (but admittedly does not prove) that you might be elsewhere in the world.  I wondered if, when InboundCallRoute is set to 'Li' it would also attempt to send the Caller's number.

That the Call History does not show any dialled digits is not surprising.

Possibly the explanation is that the OBi does not implement the 'dial through' functionality in the same manner that you are familiar with on the SPA3000.  Most people use the OBi's AA to bridge incoming calls to PSTN dial tone.

I would expect that setting InboundCallRoute to, say, 'Li(1234567890)' would cause 1234567890 to be dialled on the PSTN line but I have not tried to verify that.

[rjdyck]

Don't confuse the functions of the phone port versus the line port. I agree the phone port can send caller ID.
The phone port receives digits from the phone ( DTMF ) ( outgoing call ) and it sends ring ( AC voltage ) followed by callerid ( FSK ) ( incoming call ) to the phone.
The line port sends digits that it received from the phone or VOIP or AA to the telephone line ( outgoing call ) and it receives ring ( AC voltage ) and callerid ( FSK ) from the phone line ( incoming ).

I want to receive the actual dialtone from the PSTN and then call the number of my choice using the keypad of my softphone. In other words the Obi should simply bridge the SP and LI. The DTMF from the softphone simply passes through the Obi to the receiver in the telco central office. The Obi is indeed bridging the SP and the LI because I hear the the telco's recording. The telco has received some unknown digit and is waiting for more to match some dial plan.

[azrobert]

An inbound call on a SP provides 2 pieces of information, callerid and userid. When you bridge a call the OBi will treat the userid as an outbound number. You need to remove the userid before routing the call to Line.

If your userid is rjdyck change the inbound route to: {>(<rjdyck:>):li}
If you want to check for a specific callerid: {8005551212>(<rjdyck:>):li}

Sometimes the provider will send something other than the userid. You can determine what is being sent by temporarily changing the inbound route to "spx($2@192.168.1.99)". spx must be defined as SIP and the IP address is any random number. Place a call to the SP, wait a few seconds and hang up. The OBi will attempt to bridge the call to the IP address. The call history will show what is being sent.

You can try {>(<@.:>):li} but I'm not sure it will work.

[rjdyck]

You can try {>(<@.:>):li} but I'm not sure it will work.

Your transform worked. I had tried something similar (<@.:>:li). I wanted to ensure that any number received by the Obi was not passed on. It was about this point where I started looking a the call history. There were no outbound digits shown yet it still was not working. After your suggestion which was successful, I started looking away back in my call history. ( i'm an Obi newbie, wasn't aware of history ). It turns out that when I had simply LI as the inbound route for the SP the SP's user identity showed up in the outbound history. I don't what purpose is served by this "feature".

To summarize, my transform didn't work but it masked the problem because it didn't show up in the call history. Without the transform the SP user ID shows in the outbound history..

If the syntax you used is in the manual I missed it.

Edit: Sorry I forgot. Thank you so much for taking the time on this.

[azrobert]

I don't what purpose is served by this "feature".

You can call the OBi with a softphone and bridge the call to a provider passing an outbound number. You point the softphone proxy to the IP address of the OBi and the port number of the SP. These calls are also processed by the inbound call route. The syntax for this type of call is:
{userid>outbound_number:dest}

The same syntax is used with the OBion softphone or OBi to OBi calls.

[rjdyck]

You point the softphone proxy to the IP address of the OBi and the port number of the SP.

From the softphone point of view the Obi is the service provider? I assume the softphone should not attempt to register.

[azrobert]

From the softphone point of view the Obi is the service provider?

Correct

I assume the softphone should not attempt to register.

Yes and no. You can't register to an OBi1xx, so the softphone must be able to call without registration.

You can register to an OBi2xx. The parm is "X_Proxy" and it must be enabled (checked). It's found in the SPx Service parms.