|
Home > Archive > Voice Over IP in UK > August 2006 > SPA-3000 inbound call from PSTN
You are viewing an archived Text-only version of the thread.
To view this thread in it's original format and/or if you want to reply to
this thread please [click here]
| Author |
SPA-3000 inbound call from PSTN
|
|
| Robert Gauld 2006-08-06, 7:11 am |
| 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?
| |
| Robert Gauld 2006-08-06, 7:11 am |
| Jono wrote:
> Robert Gauld explained :
>
> 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.
| |
| Graham. 2006-08-06, 1:11 pm |
|
"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)
--
Graham.
%Profound_observation%
| |
|
| Robert Gauld expressed precisely :
> Jono wrote:
>
> which is exactly why I'm asking.
>
> will the divert/forward stop it giving you dial tone when
> 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?
| |
| Robert Gauld 2006-08-06, 1:11 pm |
| Jono wrote:
> Robert Gauld expressed precisely :
>
> 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.
| |
|
| Robert Gauld formulated on Sunday :
> 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...
>
>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.....
--
Mike
| |
| Graham. 2006-08-06, 1:11 pm |
| > 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.
http://nerdvittles.com/index.php?p=73
Look at the heading "One ringy-dingy"
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.
--
Graham.
%Profound_observation%
| |
| Graham. 2006-08-06, 7:11 pm |
| 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
;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[vbcol=seagreen]
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)
| |
| Graham. 2006-08-06, 7:11 pm |
| 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[vbcol=seagreen]
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)
| |
| Robert Gauld 2006-08-06, 7:11 pm |
| Jono wrote:
> Robert Gauld formulated on Sunday :
>
>
> 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/${CALLERIDNUM}.2)
>exten => h,11,Hangup()
>
>[custom-callout]
>exten => s,1,Background(silence/1)
>;exten => s,3,Background(asterisk-friend) ; announcment too anoying,
>commented out.
>exten => s,2,Authenticate(1234)
>exten => s,3,DISA(no-password|from-internal)
Thanks Graham,
I had a look a while ago at the nerdvittles article, & had been circling
round it & other similar how-to's, just need to get the grey matter in
gear & some uninterrupted time.
--
Mike
| |
|
| In article <44d6411d_1@x-privat.org>, Graham. <me@privacy.com> writes
>OK here is what I did.
>Paste something like this at the bottom of extensions_custom.conf
>where: 2462468 is my sipgate number (and incoming user context)
>and: 1234 is the DISA (or should that be DOSA:-) password
>Point the incomming route for 2462468 to custom app:
>custom-ringy-in,2462468,1
>
>
>
>
>
>
>
>;graham one ringy dingy
>[custom-ringy-in]
>exten => 2462468,1,NoOp
>exten => 2462468,2,Congestion
>exten => 2462468,3,Hangup
>
>
>exten => h,1,SetCIDNum(${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
>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 ?
--
Mike
| |
| news1001 2006-08-09, 7:11 pm |
| 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
>
>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
| |
| Graham 2006-08-09, 7:11 pm |
|
> 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.
Good luck,
--
Graham.
%Profound_observation%
| |
| Ivor Jones 2006-08-09, 7:11 pm |
| "news1001" <mike@starfire.demon.co.uk> wrote in message
news:fkgXpiAN7k2EFAKv@starfire.demon.co.uk
[snip]
> Got the Uk part right just cannot get past dtmf problems.
> what is your dtmfmode for sipgate ?
DTMF mode for Sipgate should be "INFO"
Works on my SPA-2000 anyway.
Ivor
|
|
|
|
|