[Asterisk-Users] Multiple Sipura 3000

[Tim P]

I saw this in the list awhile back, it helped me setup my sipura 3000s
to act as trunks
Setup the PSTN side of the Sipura 3000 as a trunk within Asterisk
In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA,
ensure that port is 5060 and context is from-internal. It should be
named SatelliteOut. Disable voicemail and directory.  Add a second
extension (e.g. 201) for PSTN Line on SPA, ensure that port is 5061
and set context to from-pstn. It should be named SatelliteIn.  Disable
voicemail and directory.

In Trunks add a Sip trunk and copy the Outgoing block as follows (just
leave Incoming as it is - do not delete the any defaults, but you do
not need to change them either).:

Trunk name SatelliteOut

context=from-pstn 
fromuser=201 (or whatever extension you used) 
host=IP address of you SPA (needs to be fixed IP) 
port=5061 
secret=your password 
type=peer 
username=201 (or whatever extension you used) 

Inbound User context SatelliteIn 

Leave defaults in Inbound box and leave Register String blank. 

In DID Routes, add DID with a unique string (I used S followed by the
PSTN number that the SPA is attached to - e.g. S8887776666

Set an outbound route using the new SatelliteOut trunk. 

On the SPA 3000: 
Do the following configuration in admin login, advanced mode: 
In Line 1, make sure SIP port is 5060, & proxy points to your * Box,
NO outbound proxy. Fill out subscriber info with settings above e.g.
User ID = 200
Password =1234
Display Name = SatelliteIn

In PSTN Line, ensure SIP Port = 5061 & proxy = Asterisk Box IP, NO
outbound proxy. Fill out subscriber info with
Display Name = SatelliteOut
User ID = 201
Password =1234
It is vital that you Set Dial Plan 8 to (S0<:S8887776666>) (for the
string you used for the DID route in Asterisk).

Ensure that both VoIP-To-PSTN Gateway Enable and PSTN-To-VoIP Gateway
Enable are set to yes.
Set PSTN Caller Default DP to 8. 
If you want incoming calls to all be sent to * then set PSTN Ring Thru
Line 1 to no.
Set PSTN Answer Delay to the number of seconds that you want the phone
to ring for before sending it to your * box. Set it to 1.

Leave other settings on the SPA at factory defaults until you really
know what you're doing and want to fine-tune things.

Lastly, make sure you plug into the line jack into the SPA and not the
jack marked phone! I know this seems obvious, but I've missed this
simple step before!

The only kink with inbound using the settings posted is that you can't
have it ring to a phone plugged into the Sipura's phone port. You can
still call out, and the system will still pick up the call if you have
auto attendant recieve the calls. But, if you set the inbound calls to
ring extension 200, your calls will just go directly to voicemail.

That aside, you can have any other phone on the system ring for
inbound calls directly, or set a ring group.


On 6/17/05, Rich Adamson <radamson at routers.com> wrote:
> > If I have multiple Sipura 3000 device how can I dial out properly? I
> > can receive call without any problem and that's working really well.
> > Caller ID is shown and when someone call he get's the welcome message
> > the same way I have it configure with the X100P card. I don't seem to
> > have any echo problem with the Sipura 3000 (but I do with X100P cards)
> >
> > My main concern is for outgoing call. Can I create a group like I did
> > in zaptel for Sipura 3000 device? Like if the FXO port of the first
> > Sipura 3000 is busy it will switch to the second and if second is
> > also busy then to the third one, and all the way until all the Sipura
> > 3000 are in used before saying that there's no line left?
> >
> > The only configs I saw on the wiki were with 1 Sipura 3000 but I
> > couldn't find anything on how to setup multiple Sipura 3000 devices
> > in asterisk for outgoing calls.
> 
> If I understood what you're trying to accomplish, try something like
> this.
> 
> In sip.conf, define each spa3k something like this:
> [3021]                  ; PSTN side of SPA3000
> type=friend
> host=dynamic
> username=3021
> secret=myspa1
> context=from-sip
> canreinvite=no
> group=17
> pickupgroup=2
> deny=0.0.0.0/0.0.0.0
> permit=216.21.194.0/255.255.255.0
> 
> and be sure to include "group=17" in each spa3k definition.
> 
> Then in extensions.conf, use a dial statement like this:
> 
> exten => _9.,1,Dial(SIP/g17/${EXTEN:1}
> 
> Pick whatever group number you want instead of =17 in the above.
> If I recall correctly, you can have up to 32 groups (or something
> like that).
> 
> When the spa3k first hit the market, someone recommended using port
> 5060 and 5061 in the spa definitions. I have never had to do that
> with any spa3k. Rather, I leave both the fxs and fxo definitions
> in the spa3k default to 5060 and use different userid & secrets
> for the fxs and fxo definitions. The above definition for x3021
> is the actual one in use right now, which functions correctly.
> I've added the "group=17" in the above as an example; I don't
> actually use that right now (for different reasons).
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>