Call Waiting SPA 3102

[ess]

I am aware that there are still some issues regarding the 3102 sending caller id info to 3cx. However, have people been able to get call waiting to work?

Right now, I have two 3102's on two separate pstn lines. I also have 2 spa 921 sip phones. I also have an analogue phone as an extension on one of the gateways.

In a call-waiting scenario on one of the pstn lines, what steps should I take to answer the call?

Flash on the analogue phone does not switch over to the other caller, it simply gives me a dial tone. Pressing flash again does switch me back to the original call.

On the SIP phones, I do not have a flash button. Hitting the hook simply hangs up the call (this works on old analogue phones).

It seems as if they are in a world of their own. Similarly, * codes such as *67, etc., won't send, as well as # codes. They do send as DTMF codes during a call. However, while dialing, those are simply a "submit" key on my sip phone.

I guess I am just generally confused about how these SIP phones are actually communicating with the PSTN itself through the server then the gateway. I never really got the dial plans working as they should. I know some people have it setup where #9 will give them a second dial tone. I simply rely on the outbound rules in 3cx, which work for basic calling.

The software does work very well for my setup. I just need some enlightening I guess.

p.s. sorry about the confusion rant :lol:

Thank you very much!

 

[Anonymous]

Are they both pstn calls? From memory I do not think that the spa will allow you to pickup a call when you are on the VoIp line.

I have to look into that actually, its been a while .

 

[ess]

Yes, call waiting is a service trough the PSTN. If another call comes in while you are on the line, there are beeps. If you press flash, you can switch to the other incoming call.

 

[SY]

Change busy detection for extension from "use PBX status" to "Use phone status".

Regards,
Stepan

 

[Anonymous]

Does this work now or do you still have troubles?

 

[ess]

The call waiting still doesn't work. Pressing flash on an analogue phone simply brings me to another dial tone. Is there some way I could tell that the PSTN gateway is in fact recognizing that there is a second call?

I have also tried the gateway with two different telcos that have different call waiting tones. The second one I tried is the standard tone listed as the US tone on the World PSTN Tone Database.

I've also noticed that caller ID only works on the second telco.

I have tried changing things like the hook flash timings on the fxo, etc.

Thanks for all of your help guys,

-Tim

 

[SY]

The call waiting still doesn't work. Pressing flash on an analogue phone simply brings me to another dial tone. Is there some way I could tell that the PSTN gateway is in fact recognizing that there is a second call?

I have also tried the gateway with two different telcos that have different call waiting tones. The second one I tried is the standard tone listed as the US tone on the World PSTN Tone Database.

I've also noticed that caller ID only works on the second telco.

I have tried changing things like the hook flash timings on the fxo, etc.

Thanks for all of your help guys,

-Tim

Did you change settings of "busy detection"?

 

[ess]

yes I have. I changed the extension that is on the fxs port of the gateway and tried it on both PBX detection and phone detection with no luck (phone detection is default).

-Tim

 

[archie]

Are you trying to put a call on hold with analogue phone connected to fxs??

 

[SY]

yes I have. I changed the extension that is on the fxs port of the gateway and tried it on both PBX detection and phone detection with no luck (phone detection is default).

-Tim

ess,

I'm sure that it (Call Waiting) works correct.

Open
"Destination Unreachable / Forwarding >>>" on extension configuration page

on "No Answer" tab setup "No Answer After (seconds)" to longer value to force PBX to don't apply No answer rule too fast (default is about 15 secs)
or disable it by choosing "Continue Ringing".

on "Phone Busy" tab
set "Busy Detection" to "use phone status".
Press "Ok"

Configure your FXS port to support Call Waiting.

That's all.

Regards,
Stepan

 

[ess]

archie: No, not by pressing a hold button on the phone. When I hear the call waiting tone, I try a hook flash or press flash. This brings me to a second dial tone rather than bringing me to the other call on the pstn line. I believe this dialtone is from 3cx. Is there a way I could test by changing the dial tone?

SY: I feel like I am doing something wrong here, but I truly do have it set up as you have described. Flashing simply brings me to another dial tone instead of to the other caller. I have two mobile phones that I have tested call waiting with the following 2 options. (For the sake of simplicity, we'll call mobile #1 "M1", mobile #2 "M2", and the analogue phone connected to the fxs port "FXS".)

I'll call M2 from FXS and answer, then call the FXS from M1.
I'll call FXS from M2 and answer, then call the FXS from M1.

Both result in hearing the call waiting tone, and not being able to send a flash signal or hookflash to the pstn.

Call waiting DOES work through the PBX. If I dial M1 from the FXS and answer, then call from a 3cx softphone to the FXS extension, I can press flash and switch between the M1->FXS(through pstn) and the Softphone->FXS(through)extension call.

It seems that when making or receiving an external call like this, the flash/hookflash are not sent to the pstn. They remain on the 3cx pbx.

This is confirmed with I try using the PSTN's Call Conference feature. When I try and press flash to dial another number to conference with, I see this in the 3cx status:

09:52:42.024 StratLink:nHangUp [CM104001] Call(6): Ln:10000@Home Line hung up call; cause: BYE; from IP:192.168.1.107
09:52:39.913 CallConf:nIncoming [CM003002] Call(7): Destination not available for call from sip:[email protected] to sip:#MOBILENUMBER2#@192.168.1.100
09:52:39.911 CallConf::Rejected [CM103005] Call(7) is rejected: Destination does not exist, or is not registered
09:52:39.824 CT_Lines::selectNextTarget [CM105003] Line Ln:10000@Home Line has more active calls than allowed by Administrator. Skipped.
09:52:39.781 CallConf:nIncoming [CM103002] Call(7): Incoming call from 106 (Ext.106) to sip:#MOBILENUMBER2#@192.168.1.100
09:52:07.592 CallLegImpl:nConnected [CM103001] Call(6): Created audio channel for Ln:10000@Home Line (192.168.1.107:16470) with third party (192.168.1.107:16468)
09:52:07.582 StratInOut:nConnected [CM104005] Call(6): Setup completed for call from Ext.106 to Ln:10000@Home Line
09:52:07.582 CallLegImpl:nConnected [CM103001] Call(6): Created audio channel for Ext.106 (192.168.1.107:16468) with third party (192.168.1.107:16470)
09:52:07.558 CallConf:nProvisional [CM103003] Call(6): Ln:10000@Home Line is ringing
09:52:07.266 CallConf:nIncoming [CM103002] Call(6): Incoming call from 106 (Ext.106) to sip:#MOBILENUMBER#@192.168.1.100

I increased the default (1) number of channels to 2, which resulted in returning a 503 error from the PSTN when trying to conference. I expected this though as it is not truly pressing flash to get to another line.

I *DO* in fact have call waiting enabled in the gateway config and I am not sure what I am doing wrong. Thanks so much for your efforts SY.

-Tim

 

[SY]

Tim,

Could you please describe which kind of equipment is using to receive call and make call from/to mobile phones (M1 and M2) in described scenario?

Thanks,
Stepan

 

[ess]

It is a standard analogue telephone on the FXS port of the SPA3102.

 

[SY]

It is a standard analogue telephone on the FXS port of the SPA3102.

FXS port can't receive call from mobile, there is some PSTN gateway or VoIP provider, isn't it?

 

[ess]

Yes I am sorry I misunderstood the question, the SPA3102 also has an FXO port that is connected to the PSTN. It is the PSTN Gateway. I am not using any voip provider.

 

[SY]

Tim, could you please take a look into PBX log, and check, does new call arrive on PBX when you try to make second call from mobile?

 

[ess]

There are no changes in the log indicating that the PSTN gateway is telling the PBX that there is a call waiting tone.

I have call waiting enabled. Will there be a change in the INFO/STATUS page of the SPA3102 indicating that a call waiting tone is being recognized.

 

[SY]

It is up to your device (FXS port) to send DTMF to alternate call on analog line, or to alternate SIP lines (put on hold current SIP call or switch between two SIP calls). PBX delivers all available information to device.

There is "Hook Flash Tx Method" option on Line configuration page. Try to change it to "AVT".

In this case you will be able to receive just one SIP call, but will be able to alternate calls on analog line by "flash".

I did check it with Grandstream HT-488, but seems SPA should also have the same behavior.

Regards,
Stepan

 

[ess]

For my SIP hardphones (SPA-921), is it possible to have a hook flash sent to the pstn rather than to the pbx?

 

[SY]

By choosing this option you change behavior of FXS port. It will send DTMF flash signal to PSTN instead of sending sequence of SIP packets to SIP PBX to switch active SIP call.

Just try it...

Thanks,
Stepan