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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

AAH Sipura 3000



 
 
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  #1  
Old November 21st 05, 09:26 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default AAH Sipura 3000

Please can someone give some help with getting my Sipura 3000 working
with my AAH box?

I've tried the tools and voxilla and the Nerd Vittles tutorial, but I
still can't make or recieve calls over the pstn line.

Can someone give some advice?

Many thanks


Matthew

  #2  
Old November 21st 05, 09:34 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default AAH Sipura 3000



Matt wrote:
|| Please can someone give some help with getting my Sipura 3000 working
|| with my AAH box?
||
|| I've tried the tools and voxilla and the Nerd Vittles tutorial, but I
|| still can't make or recieve calls over the pstn line.
||
|| Can someone give some advice?
||
|| Many thanks
||
||
|| Matthew

Which version of [email protected]?

Which firmware on the spa3k? I'm using v3.1.5(GWb) on both mine.

are you trying to make outbound calls with a phone connected directly to the
SPA, or via [email protected] and from there to the SPA?

Post your [email protected] trunk settings for your SPA-PSTN.

Have you got anywhere you can upload the html files for your SPA? (minus
passwords, of course)?


  #3  
Old November 21st 05, 10:25 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default AAH Sipura 3000

Hi Jono,

Its AAH1.5

Firmware is 2.0.13(GWg)

Trying to make calls via the AAH box out through the SPA and the PSTN -
replacing an xp100 clone (very echoey)

Will upload the html files in few minutes

AAH Extension

[199]
username=199
type=friend
secret=xxxxxx
record_out=Never
record_in=Never
qualify=no
port=5061
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="PSTN" 199

[99]
username=99
type=friend
secret=xxxxxx
record_out=Never
record_in=Never
qualify=no
port=5062
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="PSTN Incoming" 99

[pstn]
username=asterisk
type=peer
secret=xxxxxx
port=5061
insecure=very
host=192.168.0.121
fromuser=asterisk
dtmfmode=rfc2833
context=from-internal
auth=md5


Many thanks for you help.

Do you know if the SPA 3000 will detect distinctive rings - like BT
Callsign?

Cheers,


Matthew

  #4  
Old November 21st 05, 10:41 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default AAH Sipura 3000

Files are at:

http://www.emsolutions.co.uk/voip/

Thanks

Matthew

  #5  
Old November 21st 05, 10:50 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default AAH Sipura 3000



Matt wrote:
|| Hi Jono,
||
|| Its AAH1.5
||
|| Firmware is 2.0.13(GWg)
||
|| Trying to make calls via the AAH box out through the SPA and the PSTN -
|| replacing an xp100 clone (very echoey)
||
|| Will upload the html files in few minutes
||
|| AAH Extension
||
|| [199]
|| username=199
|| type=friend
|| secret=xxxxxx
|| record_out=Never
|| record_in=Never
|| qualify=no
|| port=5061
|| nat=never
|| host=dynamic
|| dtmfmode=rfc2833
|| context=from-internal
|| canreinvite=no
|| callerid="PSTN" 199
||
|| [99]
|| username=99
|| type=friend
|| secret=xxxxxx
|| record_out=Never
|| record_in=Never
|| qualify=no
|| port=5062
|| nat=never
|| host=dynamic
|| dtmfmode=rfc2833
|| context=from-internal
|| canreinvite=no
|| callerid="PSTN Incoming" 99
||
|| [pstn]
|| username=asterisk
|| type=peer
|| secret=xxxxxx
|| port=5061
|| insecure=very
|| host=192.168.0.121
|| fromuser=asterisk
|| dtmfmode=rfc2833
|| context=from-internal
|| auth=md5
||
||
|| Many thanks for you help.
||
|| Do you know if the SPA 3000 will detect distinctive rings - like BT
|| Callsign?
||
|| Cheers,
||
||
|| Matthew

For a start, it is much simpler to have the same asterisk extension number
for both the Line 1 & PSTN Lines in the SPA3K. You seem to have two
extension numbers. I'm sure there's a valid reason for it, however, I like
simplicity.

Rather than posting your sip_additional.conf, could you cut & paste the
Inbound, Outbound & Dial Rules directly from the trunk in AMP?

The spa3k cannot make use of the distinctive ring from your phone company
(AFAIK).

My sip_additional looks like this:

[204]
username=204
type=friend
secret=xxxxxxx
record_out=Never
record_in=Never
qualify=no
port=5060
nat=never
[email protected]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callgroup=1
callerid="SPA-3K" 204

[from-internal]
type=friend
secret=xxxxxxx
insecure=very
host=dynamic
dtmfmode=inband
disallow=all
context=from-internal
allow=ulaw

However, when looking at the trunk in AMP, this is what i have:

Dial Rules:
1886601274+NXXXXX
18866+01XXXXXXXXX
18866+02XXXXXXXXX
18185+07XXXXXXXXX
18866+00.
0800.
0500.
0808.
0845.
188660+1XXXXXXXXX
188660+2XXXXXXXXX
181850+7XXXXXXXXX
0+8X.
999
00000000+9.
1280|X.

Outgoing Settings:

Trunk Name pstn-spa3k

auth=md5
context=from-internal
dtmfmode=inband
fromuser=asterisk
host=192.168.1.6 ; this is the IP of the spa3k
insecure=very
port=5061
secret=xxxx
type=peer
username=asterisk

Incoming Settings:

User context from-internal

allow=ulaw
context=from-internal
disallow=all
dtmfmode=inband
host=dynamic
insecure=very
secret=xxxxxxx
type=friend


  #6  
Old November 21st 05, 10:52 PM posted to uk.telecom.voip
Sparks
external usenet poster
 
Posts: 156
Default AAH Sipura 3000

"Matt" wrote in message
oups.com...
Files are at:

http://www.emsolutions.co.uk/voip/

Thanks

Matthew

I have it all working here (with help form Jono and a few web sites!)

I can't however, seem to get the calls from the BT line route via a DID rule
(they only route via the default "Incoming Calls" rule).

I can PDF the relevant bits for you if you like!

Sparks...


  #7  
Old November 21st 05, 10:59 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default AAH Sipura 3000



Sparks wrote:
|| "Matt" wrote in message
|| oups.com...
||| Files are at:
|||
||| http://www.emsolutions.co.uk/voip/
|||
||| Thanks
|||
||| Matthew
|||
|| I have it all working here (with help form Jono and a few web sites!)
||
|| I can't however, seem to get the calls from the BT line route via a
|| DID rule (they only route via the default "Incoming Calls" rule).
||
|| I can PDF the relevant bits for you if you like!
||
|| Sparks...

Hi sparks,

You can simulate the DID by pointing the default dial plan on the PSTN tab
to a ring group. Mine points to SO:1


  #8  
Old November 21st 05, 11:01 PM posted to uk.telecom.voip
Andrew Gabriel
external usenet poster
 
Posts: 227
Default AAH Sipura 3000

In article .com,
"Matt" writes:

Do you know if the SPA 3000 will detect distinctive rings - like BT
Callsign?


I don't believe so.
It can generate distinctive ring-thru's based on callerid.

--
Andrew Gabriel
  #9  
Old November 21st 05, 11:09 PM posted to uk.telecom.voip
Sparks
external usenet poster
 
Posts: 156
Default AAH Sipura 3000


Hi sparks,

You can simulate the DID by pointing the default dial plan on the PSTN tab
to a ring group. Mine points to SO:1


Ah, thanks for that, what do I then need in the DID route?

Sparks...


  #10  
Old November 21st 05, 11:24 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default AAH Sipura 3000



Matt wrote:
|| Files are at:
||
|| http://www.emsolutions.co.uk/voip/
||
|| Thanks
||
|| Matthew

Differences noted so far:

Line 1 Tab

My dial plan looks like this:
(192:08000192190|118xxx:08000192190|[2-79]11:@gw0|xx.|*xx.|**xx.|#,:xx.:@gw0|#,:*xx :@gw0)

PSTN Tab:
Under Proxy & Registration, you have Register set to Yes (I have No)
You have Use Outbound Proxy, Use OB Proxy In Dialog, Make Call Without Reg,
Ans Call Without Reg all set to No (I have YES)

Under Subscriber Information, yours is completely blank. This is where you
put your PSTN extension number, so in mine I have user ID 204, password
XXXXXXX. If you also put a username, this will over-ride the caller display
on any x-lite soft phones.

Under Dial Plans you have set SO:99 as the dial plan (I have SO:1 which
points to Ring Group 1, and therefore rings all the phones)

Under VoIP-To-PSTN Gateway Setup, you have Line 1 Voip Caller DP as 1 (I
have none)

Under VoIP Users and Passwords (HTTP Authentication) , you have nothing to
authenticate calls from your server set up. (I have user Asterisk, Password
xxxx, Auth method HTTP Digest, DP 1)

Under FXO Timer Values (sec) you have set the delay to 16 seconds. Mine's
set to 1 second, as I want * to handle all the calls.

If you upgrade the firmware, you can select to stop the spa answering the
PSTN line before the VOIP side answers. In other words, all your *
extensions can ring without the SPA taking the PSTN line off-hook first.


 




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