[OzDan]

I know this has been discussed somewhere but I cannot seem to find a categoric answer..

Is it possible to Flash between 2 PSTN calls using the telco's Call Waiting Service through the SPA3000?

If so, how?

Thanks in advance all.

[Handsomejacek]

Here's one thread on the topic which might help

http://voxilla.com/index.php?name=PN...r=asc&start=15

I have not been succesful getting it to work though.

[lmatter]

I am also struggling to get call waiting to work. I am working on this from two angles, either one is probably acceptable to me.

Option 1) Configure the sipura so that the pstn line and line 1 are completely separate. Have everything go through asterisk and let asterisk handle call waiting.

Option 2) Let the sipura pass PSTN calls to Line 1, handle all the call waiting stuff.

I can't get either to work.

To test option 1), I disabled "PSTN Ring Thru Line 1", set "Line 1 Signal Hook Flash To PSTN" to Disabled (assuming asterisk was going to take care of it). In the Line 1 tab I set"Hook Flash Tx Method" to AVT.

I also removed the *70 code from the regional settings as that conflicted with the asterisk code for enabling call waiting. After that, dialing *70 would get an asterisk acknowlegement message. BTW, my dial plan is simply (*xx|xx.). Not sure if that's a good one, but for me it is working.

My asterisk system is set to route all incoming calls to the Line 1 extension. I call the voip line from my cell phone and answer with the analog phone on line 1. Then I call Line 1 from a softphone, and I get the CW tone. I hit the flash button *once* and, voila, I am connected to the second extension. However, the first call from my cell gets disconnected. I may be able to fix this with some asterisk tweeking (perhaps in extensions.ael).

When testing option 2)
I enable PSTN Ring Thru Line 1, set "Line 1 Signal Hook Flash To PSTN" to double, call *56 to make sure sipura cw is active, and call the PSTN line twice. Most of the time clicking the flash just gives me a dialtone. Clicking it twice doesn't help, although perhaps adjusting some of the flash times might make this better. One time I was successful at connecting to the second caller, but I couldn't switch back. And having to hit the flash twice really isn't an option for me.

If I make more progress I'll report back.

[mcbeav1]

CWT will work if your on a pstn call, and recive another pstn call. you should be able to flashook between the 2 calls. My 3000 has 2.0.13. I think it will also work with the newest software on the sipura web site. Some of the other software cwt will not work. hopefully you dont have a 3102, because i just spent a MONTH going back and forth with linksys about the same issue, and they told me "Call waiting does not work with:
1. fxo already in a call and get a another call from fxo side. This is not possible as the signal is out of band and we don't support this " . Doesn't make much sense that an old sipura 3000 will work but yet a new 3102 will not.

[mberlant]

Since PABXes by nature provide services like Call Waiting and 3-Way Calling, they are not designed to expect these features on a trunk line. As such, there is no facility in VoIP to pass a hookswitch flash through VoIP, and Asterisk doesn't know how to do it.

Since many SPA3000 owners (mostly here, I would suppose) are using the SPA-3000 in a non-commercial environment, attempting to integrate the SPA with a feature-filled residential phone line, Sipura added the ability to pass a flash to the PSTN Line from the phone attached to Line 1 in Version 3 firmware. When enabled, flashing Line 1 twice in succession will send a single flash to the PSTN Line.

There are two independent timers involved in this process -- the timer that listens on Line 1 for the flash and the timer on PSTN Line that determines the duration of the flash that goes to the phone company. The one that listens for your Line 1 flash should be set for a duration between 350ms and 650ms and the one the SPA sends to the PSTN Line should be set to 450ms for optimal success in most countries.

[lmatter]

It appears there is a facility to pass the flash through RTP (http://rfc.net/rfc2833.html, DTMF event 16). However, it does not look like asterisk fully supports it. I found this bug which seems to apply to my situation, though I am not using a zap channel.

I'm also not sure if the FXO port on the sipura would respond to this event, though I would hope it would.

[mberlant]

The preamble to the RFC makes a specific note that "flash" is not required for compliance with the RFC. This, coupled with my observation above that the function is not required in a conventional PABX, correlates with my never having seen a SIP server or client that supports transporting the flash signal over VoIP and delivering a hook flash to an FXO port.

You will notice that the "bug" you came across was closed in Asterisk without action. This is because in a PABX environment it is not reasonable to disable a PABX function in favor of moving that function out of the PABX and into a (one of many) serving trunk.

The reasoning behind this is that it is unreasonable to expect that a second call (Call Waiting) will be intended for the same extension on a PABX as the extension currently engaged on that trunk circuit. It is also unreasonable in a PABX environment to expect a new caller to be alternated with an old caller in speaking to his destination party within the PABX. For these reasons, trunks attached to PABXes have no such facilities.

As you can see by reading the "bug" text, the poster had to disable recognition of the flash by his PAP2 in order to get the PAP2 to pass the event to Asterisk. In doing so, he effectively disabled his ability to answer Call Waiting, perform a transfer or set up a 3-Way call involving any of his VoIP or additional PSTN lines. Perhaps this fellow has an Asterisk with only one extension, no VoIP services and one PSTN line, in which case what he did was reasonable, but I don't know anyone with such an environment.

[yozh]

Originally Posted by mberlant

Since PABXes by nature provide services like Call Waiting and 3-Way Calling, they are not designed to expect these features on a trunk line. As such, there is no facility in VoIP to pass a hookswitch flash through VoIP, and Asterisk doesn't know how to do it.

Since many SPA3000 owners (mostly here, I would suppose) are using the SPA-3000 in a non-commercial environment, attempting to integrate the SPA with a feature-filled residential phone line, Sipura added the ability to pass a flash to the PSTN Line from the phone attached to Line 1 in Version 3 firmware. When enabled, flashing Line 1 twice in succession will send a single flash to the PSTN Line.

There are two independent timers involved in this process -- the timer that listens on Line 1 for the flash and the timer on PSTN Line that determines the duration of the flash that goes to the phone company. The one that listens for your Line 1 flash should be set for a duration between 350ms and 650ms and the one the SPA sends to the PSTN Line should be set to 450ms for optimal success in most countries.

Hello Since I couldnt get spa-1000 working with call waiting, after reading few posts here I went and got SPA3000, which supposed to allow you to do call waiting with PSTN line, so I went and setup SPA3000 with the wizard here and enabled the double hook flash. Also changed the PSTN Hook Flash Len: to .45 and when I do double hook flash all I get is dial tone and asterisk puts the other phone call on hold. Please help, its very frastrating and I already spend so much money on this. Thank you