Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work? </edg>
I don't think it can be done, I tried and never succeeded, so did other (search the list), havn't heard of anyone that succeeded. On Tue, 29 Mar 2005 16:41:01 -0800, Ed Greenberg <edg@greenberg.org> wrote:> Anybody using a Sipura 3000 for FXO with Asterisk? > > Mine is working except for one small nit... > > When a call comes in from the PSTN, the Sipura answers it and then passes > it on to Asterisk, which plays extension ring tone. > > I'd prefer for the POTS line to stay on-hook while the extension rings, and > to only be answered by the Sipura when the extension answers. > > Has anybody made this work? > > </edg> > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Ed Greenberg wrote:> I'd prefer for the POTS line to stay on-hook while the extension > rings, and to only be answered by the Sipura when the extension answers.You and everyone else... don't think anyone has found a *good* work around, but if you search there is a solution involving a caller id hack.
Yes I have mine working exactly like this. The following is from the
Voxilla forums:
http://voxilla.com/forum-viewtopic-t-1335-postdays-0-postorder-asc-start-0.h
tml
The text is (in case you don't have web access). There's more posts on
it
but this is the nuts & guts of it. BTW, I used _ instead of A as my prefix -
works great!
--------
I've got a way to get the SPA-3000 to use the FXO port to take inbound from
PSTN (grabs and passes telco caller-ID name/num as well) and pass to
Asterisk for add'l handling.
Sure, the SPA-3000 does a great job of 'front-ending' inbound PSTN
calls,
and can even pass-through to the built-in FXS port, or external VoIP
service, but I needed Asterisk to get the call BEFORE it was
"answered" and
handled/routed by the SPA-3000.
Would seem to be a simple mode of operation, yet everywhere I looked it
didn't seem possible to do just that.
I wanted to use it as a 'simple' FXO <-> SIP gateway to Asterisk
AND also
use the FXS port as an Asterisk extension.
Here's how:
(I'm only detailing the tricky part .. the rest is really basic Asterisk
and/or SPA-3000 setup)
1. Setup Asterisk and SPA-3000 so both the PSTN line (FXO) and Line1 (FXS)
are registered with Asterisk as different extensions (i.e. FXO user ID=10
and FXS user ID=2000) on different ports (5060/5061).
In this example I'll use Asterisk extension "99" as the place I
want to send
the inbound PSTN call to.
2. PSTN Line tab:
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: NO
PSTN Ring Thru Line 1: YES
PSTN CID For VoIP CID: YES
(here's one of the tricks to make it work)
PSTN CID Number Prefix: A
(I used 'A' but I suppose you could pick any ALPHA character that
WOULDN'T
be expected as a valid caller-ID NUMBER)
FXO Timer Values (sec)
PSTN Ring Thru Delay: 3
3. User 1 tab:
Selective Call Forward Settings
Cfwd Sel1 Caller: A*
Cfwd Sel1 Dest: 99
4. In Asterisk (in the context that you've defined exten 99):
exten => 99,1,SETCIDNUM(${CALLERIDNUM:1})
exten => 99,2,Dial(SIP/${exten})
(for example)
Here's what happens:
Call rings FXO port.
Wait three seconds so that caller-ID gets sent (you might need to increase
this, but 3 secs seems to work fine for me) to the SPA-3000.
PREFIX the caller-ID NUMBER with a LETTER before passing it to LINE 1
(so if original caller-ID was 5559991212, it's now A5559991212, not a
'valid' caller ID number, but SPA-3000 and Asterisk don't seem to
care,
thankfully).
SELECTIVELY forward ONLY calls with caller-ID NUMBER that begin with A
(actually this should be EVERY inbound PSTN call) to Asterisk extension 99
As soon as Asterisk gets the call, STRIP the 'invalid' A off and
we're left
with a good, original callerID number. Send the call out to a device (can be
the SPA-3000 FXS (exten 2000) or port if you want!)
The call is still UNANSWERED at this point.
FXS port starts to ring, and original PSTN-provided caller-ID is sent as
usual.
Answer extension 99 (or send it voicemail) and FXO finally goes off-hook.
You can make calls to extension 2000 and not worry about them being bounced
back to extension 99 since no "normal" caller-ID NUMBER should ever
(??)
start with "A"
Above all, I think this could be made a whole lot more intuitive and
fool-proof if Sipura just added a feature into the firmware.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Ed Greenberg
> Sent: Wednesday, 30 March 2005 10:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sipura 3000 FXO with Asterisk
>
> Anybody using a Sipura 3000 for FXO with Asterisk?
>
> Mine is working except for one small nit...
>
> When a call comes in from the PSTN, the Sipura answers it and
> then passes it on to Asterisk, which plays extension ring tone.
>
> I'd prefer for the POTS line to stay on-hook while the
> extension rings, and to only be answered by the Sipura when
> the extension answers.
>
> Has anybody made this work?
>
> </edg>
>
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On Tue, 2005-03-29 at 16:41 -0800, Ed Greenberg wrote:> Anybody using a Sipura 3000 for FXO with Asterisk? > > Mine is working except for one small nit... > > When a call comes in from the PSTN, the Sipura answers it and then passes > it on to Asterisk, which plays extension ring tone. > > I'd prefer for the POTS line to stay on-hook while the extension rings, and > to only be answered by the Sipura when the extension answers. > > Has anybody made this work?What are you trying to accomplish by this? When SPA-3000 answer the phone and pass it over to Asterisk the party that is calling doesn't know about it. So when * answers the call adjust ringing time to whatever you want (5min?). The calling party will hear ring to tone and will never know that the phone was picked up by asterisk. -- #Joseph
> Anybody using a Sipura 3000 for FXO with Asterisk? > > Mine is working except for one small nit... > > When a call comes in from the PSTN, the Sipura answers it and then > passes it on to Asterisk, which plays extension ring tone. > > I'd prefer for the POTS line to stay on-hook while the extension > rings, and to only be answered by the Sipura when the extension answers. > > Has anybody made this work? >The comments about it being an ugly "hack" arent really correct. The Sipura is really built for standalone useage wiht a sip provider however it does work well with asterisk. Follow this thread http://voxilla.com/forum-viewtopic-t-1335.html it works and it works **VERY** well :-) Pete
Ed Greenberg wrote:> Anybody using a Sipura 3000 for FXO with Asterisk? > > Mine is working except for one small nit... > > When a call comes in from the PSTN, the Sipura answers it and then > passes it on to Asterisk, which plays extension ring tone. > > I'd prefer for the POTS line to stay on-hook while the extension > rings, and to only be answered by the Sipura when the extension answers. > > Has anybody made this work?There's something about this on the wiki. Dig it.
Gregory Wiktor - ADCom Corp.
2005-Apr-16 11:28 UTC
[Asterisk-Users] Sipura 3000 FXO with Asterisk
Lucky you, my spa-3000 likes to dial 911. So far the local cops have
been nice about it though. (my mobile number ends in 9110)
I have been having trouble getting quality and tone transmission right.
Seems to be a delay, but there is less than 5ms ping time. G729 is 'ok'
and ulaw seems unbearable...
Oddly, this is only if I dial direct. If I have the spa dial the call,
things work. So lets say I dial 9303 to pick up the fxo, that's where I
experience oddities.
Greg
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Razza
Sent: Saturday, April 16, 2005 6:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
All,
Further to my note below, I now have incoming working - yipee! (and seem
to have identified a problem with the G711A codec in the latest sipura
firmware - although need to do some checking). This box sounds great
compared to the echo ridden FXO and gives me an FXS for very little more
cash.
I now have a really strange issue for outgoing calls, everything seems
ok including the SIP messages (i.e. <dialled number>@<sipura ext>)
but I
am always getting through to a wrong number (fortunately I'm doing this
on a Sunday and it's a business number so I'm just getting their answer
machine).
I have included excepts from my test extensions.conf and sip.conf files,
could someone please confirm these are ok (for my own sanity)? The other
strange thing is the sipura info tab tells me 'Last Called PSTN Number'
is correct.
I assume I have got something very wrong with the sipura config,
although have not changed anything - so assistance on the sipura would
be greatly appreciated.
-------------------------
*** extensions.conf ****
-------------------------
[general]
static=yes
writeprotect=no
[globals]
CC=UK
CONSOLE=Console/dsp
[sip_home]
exten => 100,1,SETCIDNUM(${CALLERIDNUM:1}) ; strips leading character
added to CLI by the SPA3K to frig no answer issue
exten => 100,2,Dial(SIP/budget1,25,tr)
exten => _0X.,1,Dial(SIP/${EXTEN:0}@101,60,r)
exten => 105,1,Dial(SIP/budget1,20tr)
-------------------------
******* sip.conf *******
-------------------------
[general]
%< ------ SNIP ------- >%
[101]
;PSTN
type=friend
regexten=101
username=983
secret=razza
context=sip_home
port=5080
host=dynamic
nat=no
canreinvite=no
disallow=all
;allow=alaw
allow=ulaw
[budget1]
type=friend
regexten=105
username=budget1
secret=razza
context=sip_home
callerid="Kitchen" <105>
host=dynamic
nat=no
;canreinvite=no
disallow=all
;allow=alaw
allow=ulaw
Regards,
Ray
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Razza
Sent: 16 April 2005 00:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Pete wrote: > The comments about it being an ugly "hack" arent really correct.
The
Sipura is really built > for standalone useage wiht a sip provider
however it does work well with asterisk.>
> Follow this thread
>
>http://voxilla.com/forum-viewtopic-t-1335.html
>
>it works and it works **VERY** well :-)
>Pete
Help!!!
I have spent the whole day trying to get this to work and simply cant,
I'm aware the instructions are very simple but there is no sip traffic
generated to the * server from the SPA-3000 when I call my PSTN number
(outgoing from sip phone to spa-3000 through * is fine) - are there
other settings I am missing?
As I am in the UK I have also changed the line impedences according to
http://www.sinet.bt.com/351v4p2.pdf and have changed the 'Caller ID
Method' (in regional tab) to 'ETSI FSK WithPR (UK)' but still
nothing.
Anyone able to send me screen dumps of their config or advise?
Ray.
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