Sipura 3000 as PSTN gateway

[zyx]

Hi,

I'm fairly new to asterisk so please bear with me.
I'm using my sipura 3000 as an voip->pstn and pstn->voip gateway.
Everything is running smooth, inbound and outbound calls work perfectly.

My question is how to change the default outbound pstn behavior of my sipura 3000. When i call outbound through the sipura it seems to asterisk the sipura answers the phone before the other pstn receiver answers the phone.

For incoming psnt calls you have the option:
Off Hook While Calling VoIP: no

I want my outbound calls to be handled the same way if possible. Can the sipura direct when a outbound pstn call gets accepted by the other end?

Regards,
Peter

[atheling]

What you want is not possible as the PSTN protocol for a local loop has no way of informing you of the answer at the far end.

At least that is true in the US for residential service. Your "answer signal" is the cessation of the periodic ring back tone and that, by its nature, is really hard to detect by a non-human.

In other parts of the world or for old fashioned payphone or business PBX service in the US the phone company can use metering tones, polarity reversals, etc. to indicate answer. Typically that type of local loop service was used to tell the payphone to swallow your money. Given the ubiquity of T1 for business and the death of the payphone I am not sure if those services are available anymore. If they are available they will definitely put you into business class service with the higher cost that implies.

[zyx]

Would isdn solve my problem because it's digital?

[atheling]

Originally Posted by zyx

Would isdn solve my problem because it's digital?

It has been 10 years since I was working in telephony equipment design, so I a bit "rusty" on a lot of things. But the answer is yes and no.

Yes, the ISDN protocol does have an answer message that will tell your end that the far end has answered.

But no, once you turn that into an analog loop for your SPA-3000 FXO to plug into you have lost the signal.

Alternatively you could look for ISDN to VoIP gateway equipment. However that is an area that I have never researched so I have no clue as to what is out there or what it costs.

[termy_pe]

This has been in the forum before the spa3000 or 3102 does not do answer supervision.

But years ago, someone did the reverse, with the incoming. By calling trought the pstn and then forward it to the line1, who gets into asterisk. The line does not pick up, I heard it works.

for the reverse, someone gave me a clue. But I dont know how to do call fowardinbg with the spa.

For the outgoing, someone told me, call to line 1, forward it to gw0, and then gw0 will call out trough the pstn. That should do it.,

But theres no help on how to setp gw0.

As for sip answer supervision. Theres command on asterisk, for background detect or voice detect. All we need is when a call is made, to park the call dont answer it, make the call. And If theres a voice that comes, then you bridge them together.

I dont know why, but all the application that are working, are patent and are own by cisco or other companys.