Should a SIP->PSTN gateway return "answered" im

I have an SPA-3102 configured as a PSTN gateway for my * server. It works OK, (despite the appaling documentation).
However, one small thing has me stumped...

When I do e.g. Dial(SIP/spa/<number>) the SPA responds Immediately with:

SIP/spa-XXXXXX is ringing
SIP/spa-XXXXXX answered SIP/XXX-XXXXXX

But it hasn't even finished dialling yet. If the call to the PSTN does not complete (e.g. no answer) how can I handle
this in my * dial plan? As far as * can tell, the call was successful even when it wasn't

Is my understanding of how SIP-PSTN gateways are expected to work wrong or is the SPA weird. Anyone know?

Hi

This is how most if not all gateways work when connecting to Analog lines, there is no answer supervision. ISDN gateways can differ.
Basicly you have to think of a alog gateway as a store and forward device.

To handle call progress you have to rely on tone detection.

Ian

ianplain :

Hi This is how most if not all gateways work when connecting to Analog lines, there is no answer supervision. Ian

Ah, thanks Ian, fair enough then.

I guess that's one advantage of internal analogue hardware over external.
Shame, due to my platform not having support for any internal kit I'll just have to get used to it.

[FX. sound of a carefully crafted dial plan being torn up]

Hi

Quote:

I guess that's one advantage of internal analogue hardware over external.

Uhh nope, you see the same, as soon as the call dials its answered...

If you you want answer supervision then user ISDN or SIP/IAX trunks

Ian

Thanks Ian, I understand now.