OBi110 on local network

[bndwdthseekr]

I'm trying to set up my OBi110 to bridge software phones such as MicroSIP or Phoner to a POTS line. I have it working, but there are a few little things I would like to get ironed out.

I'm currently using Phoner on my PC, and I would like to be able to 'pick up' on an active call on the POTS phone line, but I can't do that at the moment. If its possible without a pbx server I would like to use multiple PCs as handsets on the same line too, so I'm hoping there's a setting to enable this.

Now for a phone connected to the phone port, I would like to bridge that to the line port as soon as the phone is 'picked up' or off the hook so I don't have the delay when dialing numbers. I don't need to call any of the PCs, they should all act as handsets for the POTS line.


here's the obi110 Configuration I'm currently using:
ITSP Profile B > SIP > ProxyServer > 127.0.0.1
ITSP Profile B > SIP > X_SpoofCallerID > Checked
SP2 Service > X_ServProvProfile > B
SP2 Service > X_InboundCallRoute > {line1:li},{ph}
SP2 Service > X_RegisterEnable > UnChecked
SP2 Service > AuthUserName > line1
SP2 Service > AuthPassword > (blank)
LINE Port > InboundCallRoute > ph,sp2(0@192.168.2.108:5060)

[drgeoff]

1.  Concerning your first question, VoIP does not work like that.  You cannot emulate the way that analogue multiple analogue phones can be paralleled across a POTS line.

The OBi110 can transfer a call to another SIP terminal or bring another SIP terminal in to make a 3 way conference call.  Those have to be done manually using hook-flash on the 110's phone.  (Consult the Admin Guide in the documents section of obitalk.com.)  Perhaps use Call Waiting to notify the person who has answered the POTS call that one of the softphones wants to take the call.

2.  Dialling a single # connects the 110's phone to the POTS line.  If that is too much trouble and you never want to dial VoIP calls from that phone you could set up a hotline to automatically dial that immediately when the phone is taken off hook.  Or you can configure a warm line with you choice of delay before that automatic #.  If you start manually dialling before the delay times out, the # does not happen and the manually dialled digits are treated in the normal way.  Check the Admin Guide.   

[azrobert]

There is a feature that allows you to "Barge In" to an active line call, but this only works from the Phone Port.

You dial "*96" (Code29 in the Star Code Profile) then dial "#". Star codes are only recognized from the Phone Port, so this won't work from an SP.

I wanted to do this before the BargeIn feature was available. Someone on the forum suggested changing the voltage or current on the Line Port until the OBi110 didn't recognize the Line port was in use. I don't remember the details. I was able to BargeIn, but the Line port became flaky and unusable. Maybe you can find the post. If you can get this to work, you would be able to BargeIn from an SP.

[bndwdthseekr]

Is there a way to signal the SIP phone(s) that there is an active call either on the phone port, or active on a parallel phone on the pots line and have the option to join the call, or is there even a protocol to do something like this?
If dialing # from the SIP would allow me to do this I guess it would be okay, but having the active call appear on hold after an incoming call is picked up or something like that, it would be awesome.


Can the 110 get the number dialed from the POTS phone parallel to the line port? If so can the 110 notify the sip phone what number the call is going out to and notify the sip phone that its on hold, or ring it at the very least?


I'm open to any suggestions, because I haven't really found any information on setting something like this up in this way.

[bndwdthseekr]

I guess I can't edit my posts :/

Right now I'm getting '487: Cancelled' when the parallel phones pick up a call, but the OBi110 isn't detecting whether or not the line is in use after this. I can see that the voltage is ~7-9v when its in use but the current always shows 0ma. It should detect this condition correct?

[drgeoff]

Is there a way to signal the SIP phone(s) that there is an active call either on the phone port, or active on a parallel phone on the pots line and have the option to join the call, or is there even a protocol to do something like this?

No.

[drgeoff]

Can the 110 get the number dialed from the POTS phone parallel to the line port?

No.

[drgeoff]

I guess I can't edit my posts :/

Right now I'm getting '487: Cancelled' when the parallel phones pick up a call, but the OBi110 isn't detecting whether or not the line is in use after this. I can see that the voltage is ~7-9v when its in use but the current always shows 0ma. It should detect this condition correct?

You can edit your post by clicking the "Modify" button.

The OBi can only detect current that is passing through its LINE socket.  That only happens when the OBi itself has gone "off-hook".

[bndwdthseekr]

Okay, so I was able to join an active call on the line port by dialing # with my PC after setting the following settings lower than the lowest I observed in 'phone and line status' with an active call.
'Physical Interfaces > LINE Port > LineInUseVoltageThreshold'
'LineInUseCurrentThreshold' set to 0

It's been interesting setting this up, and is going to make live MUCH easier. I would love to be able to use VOIP and ditch the old school tech, but for me it's not feasible at the moment due to the bandwidth requirements that go along with working with a team over the phone while sharing large files. I'm probably in a very small minority that still has a pots line huh?

Props to @azrobert  for mentioning this! Maybe in a few weeks Ill make the time to get a Raspbx set up for a less hacked together approach, but even with it set up this way I'm getting superb audio quality, so I can't complain. Any suggestions on pbx settings would be much appreciated too, because I'm a noob to this kind of stuff and I'm still learning, but have a much better idea of how a pbx will integrate into this now.

[azrobert]

I noticed a slight problem with your config. You should remove the "#" before routing to Line. The default config does this in the Phone outbound route.

SP2 Service > X_InboundCallRoute > {line1>(<#:>|@.):li},{ph}

[bndwdthseekr]

Sorry, you've completely lost me there... for some reason I'm having a really hard time visualizing how this works. I didn't see that anywhere in my config except in the digit map right before 911.

Can you please explain how I set up a hotline to dial # when the phone (sip or phone port) is picked up? I've spent some time looking around on the forum, but haven't found anything.

Don't know why I'm so slow picking this up 

[bndwdthseekr]

Oh, I forgot to add (and edit button just has a green 'loading' banner at the top and does nothing) It would be great if I could switch from normal phone to sip/obi phone without having the inband dtmf beep when I pick up the call, if that's possible.

[azrobert]

Hotline – Add following to the beginning of the Phone DigitMap:
<S0:#>|
To wait 2 seconds before # is sent:
<S2:#>|

The format of the inbound route rule is:
{userid>dialed number:li}

If you don't specify the ">dialed number", the rule will match any number dialed.

Your current rule will send anything you dial to Line, including #
If you try to pick up a call by dialing #, you might hear a dtmf tone.
If you try to get a dial tone by dialing #, you will get something like "Invalid Call".
You can correct this by removing the # and therefore sending nulls to line.

I tried to do this with one rule. Maybe you'll understand this better:

{line1>(<#:>):li},{line1:li},{ph}

[drgeoff]

Can you please explain how I set up a hotline to dial # when the phone (sip or phone port) is picked up? I've spent some time looking around on the forum, but haven't found anything.

Read the section of the Admin Guide on Digit Maps and Call Routeing. Hot line and wsrm line are explicitly explained.

[bndwdthseekr]

Okay, I'm starting to understand now. I reset the obi to make sure there wasn't a setting messing me up, then input all the info again. At that point it wasn't working anymore and I didn't figure out why until I noticed the phone port was ringing instead of the line port. I'm not sure how it worked before with the sp2 inbound route going to 'ph' instead of 'li'.


Now I'm using the following and it works... is this correct for what I'm doing?
SP2 Service > X_InboundCallRoute > {line1>(<#:>):li},{li}


I'm not using the phone port at all at the moment, but just to clarify, from what I understand there's no way to automatically create a conference bridge, or any kind of workaround I can do on the obi itself to automatically conference line, phone, and sip client when either the phone or sip phone is picked up (or sip dials in) on an active call. Is this correct? Are there any solutions to reach this outcome using only the OBi?

I'm cool with learning how to set up asterisk if that's what it takes I guess...

[azrobert]

You can route an inbound Line call to a conference room:
Line Inbound Route: sp2(bndwdth@conference.sip2sip.info)

You can route the Hotline and the SIP device to the same conference room, but how do you tie everything together. I think you can ring the other devices when a Line call is sent to a conference room, but you can't have an automatic connection.