Call Waiting SPA 3102

[ess]

I still do not have any luck sending a flash signal to the pstn. I have tried both the flash button and the hookflash.

This feature is not a make-or-break feature, i would just like to get it working .

I am wondering if perhaps the pstn doesn't recognize the tone generated by my phone company for call waiting. Caller ID does not work either. I am wondering if perhaps they use different protocols than the ATT standard here.

Thanks for the help SY

-Tim

 

[Anonymous]

For this to work you have to get the specific settings from you provider and/or country. This thing can be tempramental.

These are the settings you need:

Dial Tone: Country/provider specific

Busy Tone: Country/provider specific

Reorder Tone: Country/provider specific

Ring Back Tone: Country/provider specific

Ring 1 cadence : Country/provider specific
Ring 3 Cadence: Country/provider specific
(With default ring set to 3. Tweak the "1.5/3.4" to suit your CPE if CND doesn't work.) If you don't care too much about CND, just use "60(.4/.2,.4/2)" as Ring 1.

CWT:

Hook Flash Timer Min: .07 Up or down this one, that is specific per PHONE

Hook Flash Timer Max: .13

PSTN Hook Flash Len: .1

Disconnect Tone: Country/provider specific.

On-Hook Speed:

=================================

Re caller ID, that is working now 3CX got a fix for that. You have to allow for the ATA to receive the DTMF for the CLID first, so if 3cx picks up the phone to quick to route it to your deskphones it is not able to collect the CLID in the first instance (but most likely will have it with the retry).

Your SPA settings are:


In PSTN Line
PSTN Answer delay: Minimum of 3
PSTN ring Thru delay: minimum of 1.5 (should always be half of PSTN Answer delay)

 

[ess]

I am not having much luck getting through to someone with that technical information. They are going to call me back. It is a smaller (than ATT) company called Birch Telecom, and I have noticed their tones do sound slightly different.

Just a general question about the feature, the PBX software will detect this call waiting call coming in? Or do the phones find the call waiting? I've noticed there are settings in the hardphone's configuration.

Anyways, thanks very much for helping me out guys, I really appreciate it. This setup is so cool and I'm so close to having it perfect thanks to the support from these forums.

-Tim

 

[Anonymous]

Just to clarify, you have a phone and you want to use hookflash with that phone...

This phone is connected to the 3cx IP-PBX? You might want to check if there



For your CLID

Set in the PSTN the following options:

FXO Timer:

PSTN Answer Delay: Minimum 3
PSTN Ring Thru Delay: Minimum 1.5
All other leave as default setting.

That seems to be working for most 3102 users.

 

[ess]

I have 2 hard sip phones (spa-921) and an analogue phone. I want to be able to use hookflash on the analog phone. The spa-921's dont have a "flash button", but I think the first issue is to make the 3cx PBX recognize that there is another call. To do this, perhaps I need the gateway to recognize the tones.

I was unable to get any tone code lists from my telephone provider, they don't really have too many technical people.

Now, I think this is not a 3cx issue anymore. It sounds like I need my gateway to recognize the call waiting tone.

If I take 3cx out of the equation, and just use the gateway, the call waiting should work correct?

Anyways, if you know of any resources about finding these codes or somehow measuring them myself, let me know.

-Tim

 

[SY]

Configure your FXS port to send DTMF signal (AVT) for HookFlash. It will send DTMF to FXO using RTP and Call Waiting will work as you expect. Just try it.

 

[ess]

SY, I have tried this as you described in an earlier post without any luck. Even if I change the DTMF to AVT, pressing flash brings me to another dial tone on 3cx, not the call waiting. I hope I am not frustrating you guys, I have been actively trying to remedy this.

In my testing I have found that sometimes, just to tease me, Call Waiting caller ID (the caller id of the person calling on call waiting) will be passed through to the FXS port of the gateway after 2 call waiting tones. However, 3cx does not recognize the additional incoming call.

Thanks for your patience all.

-Tim

 

[Anonymous]

Ahh,

There might be an issue that you use to many channels eg you have a conflict on the ports. Not sure how 3CX does do these things actually.

I need some info from you:

1. The CLID works? When a caller calls you can see the CLID displayed on the phone and in 3cx?
2. What country are you in, the call progress tones should be specific for countries.


If need be, dump the screens of your SPA and email them to me.

No promises but when i get time I will have a look at it.

 

[ess]

The caller ID is wierd. It works only when a call waiting call is coming through on the pstn line connected to that gateways fxo. However, when pressing flash, it goes to another dial tone from the 3cx server.

I am in the USA, and I know the call waiting tone should be the same throughout the country, but it is not. On the line the pstn is connected to, there is a low beep then a high beep, and they loop every 10 seconds or so. I can get a recording. The line is through a small provider, but they use the same central office equipment as AT&T here.

On other lines I have used in the US, the call waiting tone is just one low beep looped. So what is happening, is the gateway doesn't recognize this other tone as a call waiting tone, and when I try to press flash on the analogue phone, it simply gives me another dial tone from the 3cx server to setup a conference.

I wish I had a way to copy the call waiting tone I hear (other than putting a microphone up to the headset.) The second beep is so high that it doesn't sound like standard dtmf.

Anyways, this thread has become more like a personal journal for me, and I apologize

-Tim

 

[Anonymous]

Lets focus on one thing first .

You mentioned that the CLID only shows on the PSTN call, does this mean you have an VSP line aswell?

Does the CLID show on that one? (if it is supported that is).

What is it actually what you try to do (lets see if we can find a way arround)? (Sorry for asking that again).

 

[ess]

I'm only using a PSTN line here at the office. We are in a remote location and we have satellite broadband. The satellite broadband isp does not give us a public ip, so we are behind a NAT sharing an external IP with other subscribers. I can't forward any ports or anything like that. So, the only VOIP that goes on stays local here at the office. We are using the system more for voicemail extensions and the digital receptionist more than anything.

Let me try and explain the scenario when call waiting works, because it is rare. If I receive an incoming call on the FXS port's analogue phone, the Caller ID does not work. However if while I am in the middle of a call made from or received by the analogue phone, and a call waiting call comes in, Caller ID for the incoming call waiting call is passed straight through from the gateway's fxo port to the fxs port without being recognized by 3cx at all.

So the gateway recognizes that incoming call waiting call and gets its caller id, but when it comes time to flash over to the call, I can't get to the other call.

Does that make any sense? To me it does, but that means nothing lol.

So again I don't use a VSP at all, only PSTN.

-Tim

 

[Anonymous]

Yep,

Clear as mud.

I will look at it for you when I am back in the office.

If the ATA receives the CLID but you do not get the clid on the phone you might have to change some settings on 3cx for that ATA to get the CLID forwareded too you.

 

[ess]

the ata does not receive the caller id (it doesn't see it in the "info" status page.)

 

[Anonymous]

Are you sure that your provider provides you with the CLID?

If they do not provide it it does not show in the status window.

also calling party might have disabled caller ID.

 

[ess]

I do subscribe to CLID and CWT as well as CWT CLID services through my provider.

I still do not have these functions working correctly with the 3cx system.

To help me troubleshoot I have a few questions, and this will probably be more of a personal journal again.

My provider can't tell me what tones they use, except that they use a "US POTS call progress tones." Is there a way to get a capture of my call waiting tone in a linksys/sipura format/syntax?

For my linksys SIP Phones, are they themselves supposed to recognize the call waiting tone? Or is there a sip message from either the gateway or the pbx to trigger this.

Caller ID is not shown on the gateway status. Is there a way I can somehow monitor the line and see what format the CLID is signaled in?

Thanks for the help guys. Again, I am using Birch Telecom in the USA in Missouri.

-Tim

 

[SY]

I do subscribe to CLID and CWT as well as CWT CLID services through my provider.

I still do not have these functions working correctly with the 3cx system.

To help me troubleshoot I have a few questions, and this will probably be more of a personal journal again.

My provider can't tell me what tones they use, except that they use a "US POTS call progress tones." Is there a way to get a capture of my call waiting tone in a linksys/sipura format/syntax?

For my linksys SIP Phones, are they themselves supposed to recognize the call waiting tone? Or is there a sip message from either the gateway or the pbx to trigger this.

Caller ID is not shown on the gateway status. Is there a way I can somehow monitor the line and see what format the CLID is signaled in?

Thanks for the help guys. Again, I am using Birch Telecom in the USA in Missouri.

-Tim

Tim,

Call Waiting on analog line is not a SIP call until device (FXO) will not convert it. Switching between calls on analog line is controlled by DTMF tones. If your phone (analog on FXS or SIP phone) will understand these signals and will control it then it will work because PBX doesn't influence this process. PBX handles and routes ONLY SIP calls.

I did post this information above:
If your FXS port configured to work with "SIP" Call Waiting, it will be able to receive two SIP calls and will do switching between those calls when you press "flash", if you configure it to handle "analog" Call Waiting it will send DTMF to device(FXO) to switch calls on analog line.

See my previous post about Grandstream HT-488 configuration.

Stepan