Hopefully a quick question, using an SPA-3000 is it possible to do the
following? If not is there a similar unit (as in small and not that
expensive) which can, or do I have to go to a PC with trixbox?
Call comes in from PSTN
If call is from my mobile then:
Provide a VOIP dial tone
Else:
Divert call to a VOIP number with no (or near to no) delay
Robert Gauld explained :
> Hopefully a quick question, using an SPA-3000 is it possible to do the
> following? If not is there a similar unit (as in small and not that
> expensive) which can, or do I have to go to a PC with trixbox?
>
> Call comes in from PSTN
> If call is from my mobile then:
> Provide a VOIP dial tone
> Else:
> Divert call to a VOIP number with no (or near to no) delay
It's certainly possible to do both, however, I'm not sure it can do
both at once.....will the divert/forward stop it giving you dial tone
when you call in on your mobile?
I suppose you would make use of the "PSTN Caller ID Pattern:" section
for your mobile on the PSTN Line tab. On the User 1 tab, set up a
forward on no answer, with a shorter delay for other calls.
Will you be diverting to a VoIP number from the same provider as you
have configured on your device?
Jono wrote:
> Robert Gauld explained :
>> Hopefully a quick question, using an SPA-3000 is it possible to do the
>> following? If not is there a similar unit (as in small and not that
>> expensive) which can, or do I have to go to a PC with trixbox?
>>
>> Call comes in from PSTN
>> If call is from my mobile then:
>> Provide a VOIP dial tone
>> Else:
>> Divert call to a VOIP number with no (or near to no) delay
>
> It's certainly possible to do both, however, I'm not sure it can do both
> at once.....
which is exactly why I'm asking.
will the divert/forward stop it giving you dial tone when
> you call in on your mobile?
>
> I suppose you would make use of the "PSTN Caller ID Pattern:" section
> for your mobile on the PSTN Line tab. On the User 1 tab, set up a
> forward on no answer, with a shorter delay for other calls.
>
> Will you be diverting to a VoIP number from the same provider as you
> have configured on your device?
>
>
Sorry should probably have been more precise on that. It will either
forward to a number from the same provider (one of those 7 number
jobbies) or to a static IP.
"Robert Gauld" <zenadsl6355@zen.co.uk> wrote in message
news:44d58900$0$15041$db0fefd9@news.zen.co.uk...
> Hopefully a quick question, using an SPA-3000 is it possible to do the
> following? If not is there a similar unit (as in small and not that
> expensive) which can, or do I have to go to a PC with trixbox?
>
> Call comes in from PSTN
> If call is from my mobile then:
> Provide a VOIP dial tone
> Else:
> Divert call to a VOIP number with no (or near to no) delay
If you do use Asterisk you can do what I do
Call comes in (DDI used specifically for this purpose). Call is not
answered.
Asterisk waits 5 seconds then returns the call to the CLI it just
registered.
You answer the call and Asterisk asks you for a password then gives you
dialtone.
The beauty of this is no charge whatsoever is made to the initiating phone
so for example, I can use my Company mobile to make personal international
calls and the kids can make calls from their PAYG mobiles with minimal
credit
(Asterisk logs everything of course)
Robert Gauld expressed precisely :
> Jono wrote:
>> Robert Gauld explained :
>>> Hopefully a quick question, using an SPA-3000 is it possible to do the
>>> following? If not is there a similar unit (as in small and not that
>>> expensive) which can, or do I have to go to a PC with trixbox?
>>>
>>> Call comes in from PSTN
>>> If call is from my mobile then:
>>> Provide a VOIP dial tone
>>> Else:
>>> Divert call to a VOIP number with no (or near to no) delay
>>
>> It's certainly possible to do both, however, I'm not sure it can do both
>> at once.....
>
> which is exactly why I'm asking.
>
> will the divert/forward stop it giving you dial tone when
>> you call in on your mobile?
>>
>> I suppose you would make use of the "PSTN Caller ID Pattern:" section
>> for your mobile on the PSTN Line tab. On the User 1 tab, set up a
>> forward on no answer, with a shorter delay for other calls.
>>
>> Will you be diverting to a VoIP number from the same provider as you
>> have configured on your device?
>>
>>
> Sorry should probably have been more precise on that. It will either
> forward to a number from the same provider (one of those 7 number
> jobbies) or to a static IP.
Jono wrote:
> Robert Gauld expressed precisely :
>> Jono wrote:
>>> Robert Gauld explained :
>>>> Hopefully a quick question, using an SPA-3000 is it possible to do the
>>>> following? If not is there a similar unit (as in small and not that
>>>> expensive) which can, or do I have to go to a PC with trixbox?
>>>>
>>>> Call comes in from PSTN
>>>> If call is from my mobile then:
>>>> Provide a VOIP dial tone
>>>> Else:
>>>> Divert call to a VOIP number with no (or near to no) delay
>>>
>>> It's certainly possible to do both, however, I'm not sure it can do both
>>> at once.....
>>
>> which is exactly why I'm asking.
>>
>> will the divert/forward stop it giving you dial tone when
>>> you call in on your mobile?
>>>
>>> I suppose you would make use of the "PSTN Caller ID Pattern:" section
>>> for your mobile on the PSTN Line tab. On the User 1 tab, set up a
>>> forward on no answer, with a shorter delay for other calls.
>>>
>>> Will you be diverting to a VoIP number from the same provider as you
>>> have configured on your device?
>>>
>>>
>> Sorry should probably have been more precise on that. It will either
>> forward to a number from the same provider (one of those 7 number
>> jobbies) or to a static IP.
>
> Do you not have one yet?
>
>
No - I'm asking if it can before I buy one - hence the "If not is there
a similar unit (as in small and not that expensive) which can" part of
my original question. I wanted to find out if it would / might work or
if it's already been tried with/without success before parting with my
money.
>>
>> Do you not have one yet?
>>
>>
> No - I'm asking if it can before I buy one - hence the "If not is there
> a similar unit (as in small and not that expensive) which can" part of
> my original question. I wanted to find out if it would / might work or
> if it's already been tried with/without success before parting with my
> money.
Fair enough. There'll be someone along here, at some point, I would
imagine' that'd be prepared to try it on their device.
Have you looked in the Linksys/Sipura user forum at http://voxilla.com
?
In article <44d5ebdf$1_2@x-privat.org>, Graham. <me@privacy.com> writes
>
>"Robert Gauld" <zenadsl6355@zen.co.uk> wrote in message
>news:44d58900$0$15041$db0fefd9@news.zen.co.uk.. .
>> Hopefully a quick question, using an SPA-3000 is it possible to do the
>> following? If not is there a similar unit (as in small and not that
>> expensive) which can, or do I have to go to a PC with trixbox?
>>
>> Call comes in from PSTN
>> If call is from my mobile then:
>> Provide a VOIP dial tone
>> Else:
>> Divert call to a VOIP number with no (or near to no) delay
>
>If you do use Asterisk you can do what I do
>
>Call comes in (DDI used specifically for this purpose). Call is not
>answered.
>Asterisk waits 5 seconds then returns the call to the CLI it just
>registered.
>You answer the call and Asterisk asks you for a password then gives you
>dialtone.
>
>The beauty of this is no charge whatsoever is made to the initiating phone
>so for example, I can use my Company mobile to make personal international
>calls and the kids can make calls from their PAYG mobiles with minimal
>credit
>(Asterisk logs everything of course)
>
>
I would appreciate an example of how you set this up on Asterisk, it's
something I would like to do, but, a working example might just get me
started !
I have an spa3000, had a couple of stabs at getting it configured as
pstn gateway for *, its now back in the box for a while.....
> I would appreciate an example of how you set this up on Asterisk, it's
> something I would like to do, but, a working example might just get me
> started !
>
> I have an spa3000, had a couple of stabs at getting it configured as
> pstn gateway for *, its now back in the box for a while.....
>
This is the Nerd Vittles article that whetted my apatite.
I ended up doing things a little differently, the dialtone gives me access
to the 'from internal' context so I can do anything a local extension can
do.
I use a Sipgate trunk to receive the initial trigger call,
and VoIPCHEAP to make the call-back and the onward call (only one account
needed though).
If I have time I will post my config file later. I will need to censor it
first.
OK here is what I did.
Paste something like this at the bottom of extensions_custom.conf
where: 2462468 is my sipgate number (and incoming user context)
and: 1234 is the DISA (or should that be DOSA:-) password
Point the incomming route for 2462468 to custom app:
custom-ringy-in,6608571,1
OK here is what I did.
Paste something like this at the bottom of extensions_custom.conf
where: 2462468 is my sipgate number (and incoming user context)
and: 1234 is the DISA (or should that be DOSA:-) password
Point the incomming route for 2462468 to custom app:
custom-ringy-in,2462468,1
Jono wrote:
> Robert Gauld formulated on Sunday :
>
>>>
>>> Do you not have one yet?
>>>
>>>
>> No - I'm asking if it can before I buy one - hence the "If not is there
>> a similar unit (as in small and not that expensive) which can" part of
>> my original question. I wanted to find out if it would / might work or
>> if it's already been tried with/without success before parting with my
>> money.
>
> Fair enough. There'll be someone along here, at some point, I would
> imagine' that'd be prepared to try it on their device.
>
> Have you looked in the Linksys/Sipura user forum at http://voxilla.com ?
>
>
Thanks for the link. It appears the options I'm after are mutually
exclusive so can't both be done at the same time.
In article <44d6411d_1@x-privat.org>, Graham. <me@privacy.com> writes
>OK here is what I did.
>Paste something like this at the bottom of extensions_custom.conf
>where: 2462468 is my sipgate number (and incoming user context)
>and: 1234 is the DISA (or should that be DOSA:-) password
>Point the incomming route for 2462468 to custom app:
>custom-ringy-in,2462468,1
>
>
>
>
>
>
>
>;graham one ringy dingy
>[custom-ringy-in]
>exten => 2462468,1,NoOp
>exten => 2462468,2,Congestion
>exten => 2462468,3,Hangup
>
>
>exten => h,1,SetCIDNum(${CALLERIDNUM:1})
>;the ':1' above strips the first digit (0)
>exten => h,2,System(echo channel: SIP/VOIPCHEAP/0044${CALLERIDNUM} >
>/tmp/${CALLERIDNUM})
>;the '0044' above adds prefix
>exten => h,3,System(echo context: custom-callout >>
/tmp/${CALLERIDNUM})
>exten => h,4,System(echo extension: ${CALLERIDNUM} >>
/tmp/${CALLERIDNUM})
>exten => h,5,System(echo priority: 1 >> /tmp/${CALLERIDNUM})
>exten => h,6,System(echo callerid: >> /tmp/${CALLERIDNUM}) ; Your
CallerID
>for your TelaSIP account goes here (just use this line as-is. Graham)
>exten => h,7,System(echo sleep 10 > /tmp/${CALLERIDNUM}.2)
>exten => h,8,System(echo cp /tmp/${CALLERIDNUM}
/var/spool/asterisk/outgoing
> /tmp/${CALLERIDNUM}.2)
>exten => h,9,System(chmod 775 /tmp/${CALLERIDNUM}.2)
>exten => h,10,System(/tmp/${CALLERIDNUM}.2)
>exten => h,11,Hangup()
>
>[custom-callout]
>exten => s,1,Background(silence/1)
>;exten => s,3,Background(asterisk-friend) ; announcment too anoying,
>commented out.
>exten => s,2,Authenticate(1234)
>exten => s,3,DISA(no-password|from-internal)
Thanks Graham,
I had a look a while ago at the nerdvittles article, & had been circling
round it & other similar how-to's, just need to get the grey matter in
gear & some uninterrupted time.
In article <44d6411d_1@x-privat.org>, Graham. <me@privacy.com> writes
>OK here is what I did.
>Paste something like this at the bottom of extensions_custom.conf
>where: 2462468 is my sipgate number (and incoming user context)
>and: 1234 is the DISA (or should that be DOSA:-) password
>Point the incomming route for 2462468 to custom app:
>custom-ringy-in,2462468,1
>
>
>
>
>
>
>
>;graham one ringy dingy
>[custom-ringy-in]
>exten => 2462468,1,NoOp
>exten => 2462468,2,Congestion
>exten => 2462468,3,Hangup
>
>
>exten => h,1,SetCIDNum(${CALLERIDNUM:1})
>;the ':1' above strips the first digit (0)
>exten => h,2,System(echo channel: SIP/VOIPCHEAP/0044${CALLERIDNUM} >
>/tmp/${CALLERIDNUM})
>;the '0044' above adds prefix
>exten => h,3,System(echo context: custom-callout >> /tmp/${CALLERIDNUM})
>exten => h,4,System(echo extension: ${CALLERIDNUM} >> /tmp/${CALLERIDNUM})
>exten => h,5,System(echo priority: 1 >> /tmp/${CALLERIDNUM})
>exten => h,6,System(echo callerid: >> /tmp/${CALLERIDNUM}) ; Your CallerID
>for your TelaSIP account goes here (just use this line as-is. Graham)
>exten => h,7,System(echo sleep 10 > /tmp/${CALLERIDNUM}.2)
>exten => h,8,System(echo cp /tmp/${CALLERIDNUM} /var/spool/asterisk/outgoing
> >> /tmp/${CALLERIDNUM}.2)
>exten => h,9,System(chmod 775 /tmp/${CALLERIDNUM}.2)
>exten => h,10,System(/tmp/${CALLERIDNUM}.2)
>exten => h,11,Hangup()
>
>[custom-callout]
>exten => s,1,Background(silence/1)
>;exten => s,3,Background(asterisk-friend) ; announcment too anoying,
>commented out.
>exten => s,2,Authenticate(1234)
>exten => s,3,DISA(no-password|from-internal)
>
Just working this up using voip.co.uk, they do not require international
format. & to make things interesting, I would like this to include
access from a Spanish 0034 number.
I think I have it cracked, I will test using the UK setup to check for
typo's & then rework a little.
In article <8kcfpVA9Vk2EFAPU@starfire.demon.co.uk>, news1001@starfire.demon.co.uk writes
>In article <44d6411d_1@x-privat.org>, Graham. <me@privacy.com> writes
>>OK here is what I did.
>>Paste something like this at the bottom of extensions_custom.conf
>>where: 2462468 is my sipgate number (and incoming user context)
>>and: 1234 is the DISA (or should that be DOSA:-) password
>>Point the incomming route for 2462468 to custom app:
>>custom-ringy-in,2462468,1
>>
>>
>>
>>
>>
>>
>>
>>;graham one ringy dingy
>>[custom-ringy-in]
>>exten => 2462468,1,NoOp
>>exten => 2462468,2,Congestion
>>exten => 2462468,3,Hangup
>>
>>
>>exten => h,1,SetCIDNum(${CALLERIDNUM:1})
>>;the ':1' above strips the first digit (0)
>>exten => h,2,System(echo channel: SIP/VOIPCHEAP/0044${CALLERIDNUM} >
>>/tmp/${CALLERIDNUM})
>>;the '0044' above adds prefix
>>exten => h,3,System(echo context: custom-callout >> /tmp/${CALLERIDNUM})
>>exten => h,4,System(echo extension: ${CALLERIDNUM} >> /tmp/${CALLERIDNUM})
>>exten => h,5,System(echo priority: 1 >> /tmp/${CALLERIDNUM})
>>exten => h,6,System(echo callerid: >> /tmp/${CALLERIDNUM}) ; Your CallerID
>>for your TelaSIP account goes here (just use this line as-is. Graham)
>>exten => h,7,System(echo sleep 10 > /tmp/${CALLERIDNUM}.2)
>>exten => h,8,System(echo cp /tmp/${CALLERIDNUM} /var/spool/asterisk/outgoing
>> >> /tmp/${CALLERIDNUM}.2)
>>exten => h,9,System(chmod 775 /tmp/${CALLERIDNUM}.2)
>>exten => h,10,System(/tmp/${CALLERIDNUM}.2)
>>exten => h,11,Hangup()
>>
>>[custom-callout]
>>exten => s,1,Background(silence/1)
>>;exten => s,3,Background(asterisk-friend) ; announcment too anoying,
>>commented out.
>>exten => s,2,Authenticate(1234)
>>exten => s,3,DISA(no-password|from-internal)
>>
>
>Just working this up using voip.co.uk, they do not require international
>format. & to make things interesting, I would like this to include
>access from a Spanish 0034 number.
>I think I have it cracked, I will test using the UK setup to check for
>typo's & then rework a little.
>
>Any comments ?
>
>
Got the Uk part right just cannot get past dtmf problems.
what is your dtmfmode for sipgate ?
might have to leave this for a couple of weeks !
--
news1001
>>
>>
> Got the Uk part right just cannot get past dtmf problems.
> what is your dtmfmode for sipgate ?
>
> might have to leave this for a couple of weeks !
I only used used Sipgate to trigger the callback.
I used two VoipCheap trunks for the actual calls
(they need international format)
I found it only worked with inband signalling and
a codec that can carry it ie ULaw or ALaw
Oh yes, one problem I have yet to solve, if both parties are PSTN then the
call doesn't clear-down!
If one or both parties are GSM or VoIP then no problem.
SPA-3000 inbound call from PSTN 
Linear Mode
