[asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?

[Larry Alkoff]

Doug Crompton wrote:

Doug, please forgive me but I'm still having trouble understanding two 
points from your last response.

Can you please post your extension 405 (analog extension on spa3k) in 
sip.conf
and your [sipurafxs1] ?

I finally understand that INRINGSDEV is meant to specify which analog 
and SIP phones to ring at extension INRINGSEXT = 405 and would like to 
see just how you do it.


Larry


> On Wed, 15 Nov 2006, Larry Alkoff wrote:
> 
>> Thank you very much Doug for your detailed response to my question.
>> I'm working on a new sip.conf and extensions.conf using your code as a
>> guide.
>>
>>
>> Questions:
>> In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
>> The comment says "ring analog phones on spa3k fxs but grandstream406
>> seems to refer a Grandstream sip phone, not an analog one.
>>
>> Does INRINGSDEV mean ring a specific sip phone and the analog ones?
> 
> INRINGSDEV is a list of the devices you want to ring when you use this
> variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
> and I have one grandstream 200, at extension 406, named grandstream406.
> The analog extension, fxs on the spa3k, is 405.
> 
>> How would I ring all the _sip_ phones when a pstn call comes in?
>> My macro 'ring-all' ?
>>
> 
> You just add them all together in the ring statement with the & as in my
> INRINGSDEV variable. Actually the use of the variable was taken from
> sample code given to me when I started out. It is probably a good idea
> though. you could just put them all in the dial statement but if you use
> it in more than one place it is handy to just change it in one place and
> use the variable.
> 
> SIP/sipurafxs1&SIP/grandstream406&third&fourth&.....
> 
> 
>> Notes:
>> Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
>> My extension to ring incoming calls is 120 vs your 405.  All ok on these
>> two.
>>
>> I'm nearly there thanks to you.
>>
> 
> OK glad it helped. If you have any other questions let me know. The spa3k
> has a million settings.
> 
>> Larry
>>
>>
>>
>> Doug Crompton wrote:
>>> Below is my config for spa3k fxo. I do not show the settings in the spa3k
>>> which must reflect settings here, port, username, secret, etc.  I have
>>> DTMF set to inband here and in spa3k to fix a problem with DTMF not
>>> working for menus from PSTN. This was discussed earlier and is a problem
>>> in asterisk that may (or may not) be solved in 1.4. I am using earlier
>>> version. Inband must also be specifed in spa3k pstn.
>>>
>>> [sipurafxo1]
>>> type=peer
>>> username=sipurafxo1
>>> secret=xxxxxxxxx
>>> canreinvite=no
>>> context=from-pstn
>>> host=dynamic
>>> nat=no
>>> port=5061
>>> disallow=all
>>> allow=alaw
>>> allow=ulaw
>>> allow=gsm
>>> allow=g723.1
>>> dtmfmode=inband
>>>
>>>
>>> In extensions.conf. This is a little fancy but the bottom line is that it
>>> ends up in either a day or night mode. Only day shown. The spa3k fxo in
>>> sip calls the from-pstn but the pstn-day-time (below) could be relabeled
>>> from-pstn to always go to phones. The night mode basically goes to VM.
>>>
>>> INRINGSEXT and INRINGSDEV are just variables defined to -
>>>
>>> INRINGSDEV=SIP/sipurafxs1&SIP/grandstream406 ; ring analog phones on spa3k
>>> fxs
>>>
>>> INRINGSEXT=405 ; the extension to ring for incomming calls
>>>
>>> The stdexten macro is just the standard one in sample extension file.
>>>
>>>
>>> [from-pstn]
>>> exten => s,1,GotoIf($[ ${day-night} = 0 ]?2:10
>>> exten => s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
>>> exten => s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
>>>
>>> exten => s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
>>> exten => s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
>>>
>>>
>>> [pstn-day-time]
>>> exten => s,1,SetGlobalVar(RingTimeout=35)
>>> exten => s,2,NoOp("${CALLERID}")
>>> exten => s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},"")
>>>
>>>
>>> On Tue, 14 Nov 2006, Larry Alkoff wrote:
>>>
>>>> My SIP phones can dial out through Sipura SPA3k to POTS for local and
>>>> 911 calls _but_ incoming POTS calls are being swallowup somehow.
>>>>
>>>> Am I on the right track with the code snippit below?
>>>>
>>>> sip.conf:
>>>> ---------
>>>> In sip.conf the following code is _supposed_ to ring the SIP phones when
>>>> a POTS line call comes in through Sipuara to Asterisk.
>>>>
>>>> [spa3k-pstn-in]         ; Pots-line-in from Sipura
>>>> ; If you're using Asterisk, this goes into the Incoming settings
>>>> ; For your Trunk
>>>> host=dynamic
>>>>
>>>> type=friend		; should be peer if incoming only ??
>>>>
>>>> context=[macro-ringall]	;ring all the sip phones
>>>>
>>>> secret=xxxxx
>>>> dtmfmode=rfc2833
>>>> disallow=all
>>>> allow=ulaw
>>>> insecure=very
>>>>
>>>>
>>>> extensions.conf
>>>> ----------------
>>>> context to ring all SIP phones when a POTS call comes into SPA3k:
>>>>
>>>> [macro-ringall]         ; ring all SIP phones
>>>> exten => s,1,Dial(SIP/120&SIP/121&SIP/122&SIP/124&SIP/125&SIP/126&SIP/127)
>>>> exten => s,2,hangup
>>>>
>>>> --
>>>> Larry Alkoff N2LA - Austin TX


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux