Re: Asterisk Q.... re answering incoming PSTN line

[Kevin Hawkins <lists@xxxxxxxxxxxxxxxxx>]

Thanks there -and that's helped a couple of things for me... (disabled
PSTN to VOIP for example) - I've never before seen a web interface
(Sipura) with so many configuration options on a tiny device ;-). Still
some bits you suggested to try...

Yes, my answer was being caused by the Sipura but if I extended the
Sipura answer time the group in Asterisk wasn't ever ringing with the
call (my group contains  a couple of SIP Softphones) . They did ring
when Asterisk 'answered' I have found that PSTN Ring Thru Line1= 'Yes'
causes the FXS interface to directly ring the attached phones when the
FX0 (PSTN) interface receives a call ( a type of passthrough)  but the
Asterisk group does not ring as well . - which actually again isn't what
I need. I want to potentially take a call at my desk/PC SoftPhone or SIP
device before dropping it later (if unanswered) onto the wired phones in
the house, hopefully with a distinctive ring based on CID. . However I
didn't want the PSTN caller to actually be answered by Asterisk unless
someone picks up the call somehwhere.  I'm sure it's all possible...  so
many options - and much to read ...  I'm going to play a bit before
asking any more Q's -

Kevin

UKHA wrote:
>> An incoming call is answered by Asterisk and fed to the
appropriate
>> extensions while simulating ringing back to the caller. What I
wanted
>> (expected) to happen was that the call was not answered on the
PSTN
>> incoming line unless an extension picked it up - does/can it not
work
>> this way ??  I intended that voicemail would answer eventually but
not
>> for say 30 seconds.
>>
>
> Here you go:
>
>  set the SPA parameters as follows:
> 1. Line 1 and PSTN line each have distinct and separate User IDs and
> passwords, and there are entries for each in Asterisk's sip.conf, with
> type=friend, and DIFFERENT contexts for each. Make sure that they both
> appear as registered (Registration State on the Info tab).
> 2. PSTN Ring Thru Line 1 is "yes". PSTN-To-VoIP Gateway
Enable is "no"
> (if that is yes, then if Asterisk doesn't answer, the SPA eventually
will, and
> I don't want that, but you might).
> 3. In the User 1 tab, I have Cfwd Sel1 Caller set to the userid of the
PSTN
> line, and Cfwd Sel1 Dest set to an extension in the context for the
PSTN
> line.
> 4. In extensions.conf, in the context for the PSTN line, before doing
> anything else, I do a SetCallerID so that _if_ the call were to ring
or be
> transferred back to Line1, it would not be picked up by the Selective
Call
> Forwarding setting.
> 5. On the PSTN User tab, I have Default Ring set to "Follow Line
1".
>
> A
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> Yahoo! Groups Links
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