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

[Kevin Hawkins <lists@xxxxxxxxxxxxxxxxx>]

I'm hoping to, but I don't know of anyone who can compile it yet , and
I
have no Linux experience at all so not looking good  , I think P is
looking at it ....    Really I was looking at Asterisk just to
understand what all the noise is about - and I have to say so far I am
very impressed - and I think I've now solved the problem below...


Kevin

Paul Gale wrote:
> 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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> Yahoo! Groups Links
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