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


  • Subject: RE: Asterisk Q.... re answering incoming PSTN line
  • From: "Paul Gale" <groups@xxxxxxxxxxxxxxxx>
  • Date: Thu, 2 Feb 2006 08:52:43 -0000

Kevin,

Are you going to use Patrick's Asterisk xAP connector?

I couldn't get it to work for me :(

I plan on using Asterisk as a glorified answer phone but until I can get
the status out of waiting calls etc to somewhere other than a PC (LED
display etc), it won't really work for me.

Paul.


> -----Original Message-----
> From: ukha_d@xxxxxxx [mailto:ukha_d@xxxxxxx] On Behalf Of
> Kevin Hawkins
> Sent: 01 February 2006 23:17
> To: ukha_d@xxxxxxx
> Subject: Re: [ukha_d] Asterisk Q.... re answering incoming PSTN line
>
> 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
> >
> >
> >
> >
> >
> > Yahoo! Groups Links
> >
> >
> >
> >
> >
> >
> >
> >
> >
>
>
>
>
>
> Yahoo! Groups Links
>
>
>
>
>
>




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