SPA112 and AUTO ANSWER

[gpowerz_bureau]

I am testing the SPA112 to be used with our SIP application.

the SPA112 is working perfectly but we need to do an auto answer because we will put headset directly instead of a phone (the user dont need too answer/hang up the call because everything is done by the application on the PC).

Somebody find the way to set the auto answer mode on this box ?

Thanks

Gilles

[Dan Lukes]

The phone device connected to SPA112 can autoanswer incomming call, if it decide to do it.

But SPA112 can't "answer" the client device in any way as far as I know.  POTS line interface is so dumb for such task.

Avoid conversion from SIP to (so dumb) POTS line and use native SIP phone with headset connector.

[gpowerz_bureau]

Yes thanks for your response but If I put the SPA 112 + one phone it will me to very usefull because I can buy an IP phone for the same price.

I am very disapointed than the SPA 112 is not able to auto answer because all soft phone have this functionnality !

By the way they are doing auto answer for the audio streaming why not for regular calls ????

[Dan Lukes]

It seems my english is very bad.

I will try it once more. No SIP-to-plain-dumb-POTS-line gateway, including SPA112, can force your headset do answer the call unless the headset decide to answer the call by self. You told nothing about your's headset features to us, so it's hard to help. What signal your headset expect from line to do auto-answer (most common signal, e.g. "ring" seems not to be sufficient for your device)? What signal your headset expect from line to do auto hung-up (CPC is most common as mentioned by Daniel)? Read your's headset documentation and answer those two questions. Then we can tell you if the SPA112 can be configured to transmit signals recognized by your particular device.

 disapointed than the SPA 112 is not able to auto answer because all soft phone have this functionnality

SPA112 is not IP phone. It's SIP-to-POTS gateway (dumb headset, in your particular case). E.g. device that convert feature-rich SIP protocol to very dumb (and ancient) analog protocol. If you need IP phone features, then you should byu the IP phone, then connect headset to them. But you buyed different kind of device. It's not bug of hammer it's not suitable for screewing.

[gpowerz_bureau]

Your english is perfect but you don't understand exacly what I need to do with it.

I am working in a company who is providing turnkey solution for Call center (hardware and software).

Those solutions are based on numerical/ analog board (Dialogic boards) or on full IP system (Dialogic HMP)

You have to understand than everything is manage on the PC screen (answering call, hold, hook, ..) and the agent are directy connect to the system througt an Headset for analog system and through IP phone and Headset for IP System (headset a mainly from Plantronic).

Everything is working fine but we are not using at all the IP phone (keyboard, LCD screen, )  because we have everything on the PC screen.

We use sometime also softphone and a headset connected to the PC but the quality is not so good as with hard phone.

My goal today, is to find a "black box" only for SIP to analog purpose to plug the headset. I try also to reduce the cost per seat.

I have try with the SPA112 with a regular analog phone and it's working fine but if I buy 1 SPA112 + 1 analog phone it will more complicated that 1 IP phone and the price will almost the same.

I am trying also to reduce the space and the cable for each agent because it could also cost time to support.

Gilles

[Dan Lukes]

My goal today, is to find a "black box" only for SIP to analog purpose to plug the headset. I try also to reduce the cost per seat.
I have try with the SPA112 with a regular analog phone and it's working fine but if I buy 1 SPA112 + 1 analog phone it will more complicated that 1 IP phone and the price will almost the same.
I am trying also to reduce the space and the cable for each agent because it could also cost time to support.

Unfortunatelly, it seems I understood your problem from the start. Well, let's allow me to explain how POTS line work (simplified description).

There are three states of line. Idle state (earphone on hook), ringing state (earphone on hook, incomming call, phone is ringing), call state (earphone off hook, local user dialing or call in progress, parties speaking).

Onhook/offhook state of local phone is signalled to PBX by device (phone) impedance. (Note the SPA112 is the "PBX" for the purpose of this text). High impedance mean "on hook", low impedance mean "off hook". E.g. transition from high impedance to low mean "user picked the earphone", transition from low to high mean "hangup". The impledance is the only local-phone-to-PBX signal avaiable. There is no way to change the impedance of local phone device from PBX side.

Ring is signal from PBX to local phone. It's AC waweform sent from PBX (or SPA112 here) to local phone. Note that "ringing signal" can be sent in off-hook (e.g. high impedance) state only. In advance, it's not possible to place other call to line that's still busy (e.g. off-hook, low impedance state).

As Daniel mentioned bellow, if your local device is smart enougth OR is powered from line, then another PBX->phone signal may work. Short removal of DC power from line mean "call ended, please hang up". But it's still responsibility of local phone to hang up.

Conclusion ? Device connected to phoen line (local phone) must be able to enter "high impedance" state which mean "I'm ready for next call". It must be able to enter "low impedance state" which mean mean "I'm accepting incomming call" in the case the line is ringing or "I'm going to place outgoing call" otherwise. No way to connect local-device that's in low-impedance state permanently. PBX (=SPA112 here) can't place new call to such busy line.

If you wish to have auto-answer/auto-hangup device, then the device you have

1. auto answer:

device should recognize ringing signal (causing high to low impedance transition which mean incomming call accepted)

2. auto-hangup

device should recognize CPC signal (causing low to high impedance transition which mean release of line)

All at all, your gateway and line device must be compatible As SPA112 is POTS gateway, but your line device (headset) seems not to follow POTS signalisation protocol. It's not surprising it doesn't work well. It's possible to make auto-answer/auto-hangup working, but you need line device that follow line protocol. You should either replace your headset with a POTS-compatible device, or you can replace SPA112 gateway with gateway implementing communication protocol used by your headset.

Hope I explained it clearly. Ask in doubt.

[gpowerz_bureau]

Dan Lukes

Your explanation are very clear and I know exacly how POTS line works (I have done electronic devices)  but you didnt't understand than I don't need to answer or hook the line because I need to be "on line" every time.

The only thing I ask is if  there is a way in the SIP protocol, like  some devices to auto reply to the "INVITE"  (I think it's "200 OK") then the server can send the telephony flow to the SPA 112 device.

It's not a hardware way but a software way. It seems than the ATA device is already doing this thing for audio streaming device but I had try and it's waiting for a streaming server.

So may be the will be one day an upgrade on the SPA112 and they will open this kind of feature.

Gilles

[Dan Lukes]

It's not possible to place new call to line from PBX side when local line device is signalling "line is in use" (low-impedance state) in the POTS world.

SPA112 can't send such audio to analog line, as such line is busy. You are asking for very special feature "ignore busy signal on analog line, place new call despite of it". Once more - it's not issue between server and SPA112 as both use strandard SIP signaling. It is issue between SPA112 and your headset, as SPA112 use POTS line signaling, which is not understood by your headset. You are asing if SPA112 can be reconfigure to use other line protocol than POTS. It doesn't play that protocol implemented in your headset is very dumb, the device hold line on low-impledance state permanently. It is line protocol. Both devices must speak the same protocol which is not true now.

I agree that firmware can implement SIP<->dumb_headset interface. I know no SIP<->POTS device implementing such dumb second protocol, either made by Cisco or it's competitors. I'm almost sure it's not possible to reconfigure current SPA112 to replace POTS line protocol with dumb protocol you are speaking about

And, finally, I suspect that such line protocol will not be implemented in near future, as there are other solutions for problem you described (with apropriate equipment). But I can't speak for Cisco, of course.

So sorry, I understand you wished for other answer ...

[Dan Miley]

There is a feature to allow the remote end hang up the call, it's called

CPC, Calling party controll.  Described in the admin guide.

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide/SPA100_AG_OL-25117.pdf

pg 39

but the original answer is correct for an ANALOG phone, the phone needs to have an answer/autoanswer feature.  You cannot just hang a dumb headset to with a 2 wire connection and have call control.  at least Cisco only supports an analog phoneset attached to the ATA.

you might be able to purchase an interface for your headset.  Some will even allow touch tone and pulse dialing, hangup, hookflash, etc.

What type of headset is it?

Dan