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Home > Archive > Voice Over IP in UK > November 2005 > AAH Sipura 3000
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| 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
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|
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 a@h?
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 a@h and from there to the SPA?
Post your a@h trunk settings for your SPA-PSTN.
Have you got anywhere you can upload the html files for your SPA? (minus
passwords, of course)?
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| 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
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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
mailbox=204@default
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
| |
| Sparks 2005-11-21, 5:45 pm |
| "Matt" <matthew.humphreys@gmail.com> wrote in message
news:1132612910.564707.194950@g43g2000cwa.googlegroups.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...
| |
|
|
Sparks wrote:
|| "Matt" <matthew.humphreys@gmail.com> wrote in message
|| news:1132612910.564707.194950@g43g2000cwa.googlegroups.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>
| |
| Andrew Gabriel 2005-11-21, 5:45 pm |
| In article <1132611925.794372.138090@g43g2000cwa.googlegroups.com>,
"Matt" <matthew.humphreys@gmail.com> 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
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| Sparks 2005-11-21, 5:45 pm |
|
> 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...
| |
|
|
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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|
Sparks wrote:
||| 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...
Nothing.
Create a ring group that includes (just) the SPA extension number & point
the ring group to whatever you wanted to point the DID to. Then you don't
actually need the DID.
| |
|
| Cheers Juno,
The separate extensions were picked form the Nerd Vittles tutorial -
the reasoning did not make a lot of sense to me either!
>From AMP
Trunk SIP/pstn
Dial Rules
Blank (I tend to do them all in the outbound routing)
Trunk name: pstn
Outgoiing:
Peer Details:
auth=md5
context=from-internal
dtmfmode=rfc2833
fromuser=asterisk
host=192.168.0.121
insecure=very
port=5061
secret=xxxxxx
type=peer
username=asterisk (Could not work out where this is matched in the
sipura)
No incoming or registry settings
Many thanks
Matthew
| |
|
| Cheers for that, I'll do the firmware upgrade and try again tomorrow.
Many thanks for all your help.
I wondered about putting a ring group in the dial plan, but thought I'd
stick to the rules - I'll go for a ring group tomorrow.
Ah - now I see where the asterisk user comes from!
What does the FXO Timer value do? I want asterisk to answer all calls,
so I'll copy your settings.
Cheers again,
Matthew
| |
|
|
Jono wrote:
|| Sparks wrote:
||||| 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...
||
|| Nothing.
||
|| Create a ring group that includes (just) the SPA extension number &
|| point the ring group to whatever you wanted to point the DID to. Then
|| you don't actually need the DID.
Or................set up an unused extension number put this extension
number in the ring group, then set your voip dial plan (on the PSTN tab) to
<SO:456> where 456 is the unused extension number.
Then, as this phone (456) will always be offline, any calls coming in on
your SPA will observe the rules of the ring group where you tell it what to
do if the call goes unanswered.
| |
|
|
Jono wrote:
|| 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.
Also, on your Line 1 tab, under Subscriber Information, your user
credentials (205) bear no relation to the extension numbers you appear to
have set up in asterisk (99 & 199)
| |
|
|
Matt wrote:
|| Cheers Juno,
||
|| The separate extensions were picked form the Nerd Vittles tutorial -
|| the reasoning did not make a lot of sense to me either!
||
||| From AMP
||
|| Trunk SIP/pstn
||
|| Dial Rules
||
|| Blank (I tend to do them all in the outbound routing)
||
|| Trunk name: pstn
||
|| Outgoiing:
||
|| Peer Details:
||
|| auth=md5
|| context=from-internal
|| dtmfmode=rfc2833
|| fromuser=asterisk
|| host=192.168.0.121
|| insecure=very
|| port=5061
|| secret=xxxxxx
|| type=peer
|| username=asterisk (Could not work out where this is matched in the
|| sipura)
||
|| No incoming or registry settings
||
||
|| Many thanks
||
||
||
|| Matthew
You're only half way there!
If you review all my posts, you should have enough to get it all working.
How did you save the sipura html pages in the way that you did? I could
upload mine somewhere (tomorrow)
You need inbound settings to handle your inbound calls. The asterisk user
name & password needs to go in the sipura to authenticate outbound calls.
| |
|
|
Matt wrote:
|| Cheers for that, I'll do the firmware upgrade and try again tomorrow.
||
|| Many thanks for all your help.
||
|| I wondered about putting a ring group in the dial plan, but thought I'd
|| stick to the rules - I'll go for a ring group tomorrow.
||
|| Ah - now I see where the asterisk user comes from!
||
|| What does the FXO Timer value do? I want asterisk to answer all calls,
|| so I'll copy your settings.
||
|| Cheers again,
||
||
|| Matthew
The timer means that (in your case) the spa won't answer the line, nor pass
the call to asterisk until the 16 seconds have elapsed. There was a long
winded fudge to pass the call to * before the line was answered, which
involved adding letters to the CLID, then passing the call to * which
stripped the letter & rang the phones.
With the new firmware, it's a simple yes/no to "Off Hook While Calling VoIP"
This means that I can process inbound pstn calls with *, the CLID is
captured by * and the phones ring, all without the caller knowing.
Previously, the spa would answer the line, the callers would then hear my
internal ring tone (that was generated by *) and found it off-putting.
To get your head around the spa3k, you have to bear in mind that there is no
direct path from the pstn line to the line 1 Voip (which your handset may be
plugged into)
For a PSTN call to be answered, a VOIP call is placed within the unit
between the two lines (AFAIK)
| |
|
|
Matt wrote:
|| Cheers for that, I'll do the firmware upgrade and try again tomorrow.
||
|| Many thanks for all your help.
||
|| I wondered about putting a ring group in the dial plan, but thought I'd
|| stick to the rules - I'll go for a ring group tomorrow.
||
|| Ah - now I see where the asterisk user comes from!
||
|| What does the FXO Timer value do? I want asterisk to answer all calls,
|| so I'll copy your settings.
||
|| Cheers again,
||
||
|| Matthew
The timer means that (in your case) the spa won't answer the line, nor pass
the call to asterisk until the 16 seconds have elapsed. There was a long
winded fudge to pass the call to * before the line was answered, which
involved adding letters to the CLID, then passing the call to * which
stripped the letter & rang the phones.
With the new firmware, it's a simple yes/no to "Off Hook While Calling VoIP"
This means that I can process inbound pstn calls with *, the CLID is
captured by * and the phones ring, all without the caller knowing.
Previously, the spa would answer the line, the callers would then hear my
internal ring tone (that was generated by *) and found it off-putting.
To get your head around the spa3k, you have to bear in mind that there is no
direct path from the pstn line to the line 1 Voip (which your handset may be
plugged into)
For a PSTN call to be answered, a VOIP call is placed within the unit
between the two lines (AFAIK)
| |
|
| Many thanks guys - all working as expected.
Just need to find a way around callsign..
thanks again
Matthew
| |
|
| Just found 1 issue...
If a call out over the pstn, I get a ringing tone before the unit has
dialed.
So if I call an engaged number, I get the ringing tone for 1 cycle and
then the engaged tone.
Very odd.
Any thoughts?
Thanks
Matt
| |
|
| Another one.... sorry!
I assumed that the Cfwd Sel1 Caller:00* prefixed the cli passed to
asterisk with a 00.
This it does, but if I change it, to 0* or Z* it still passes 00 - am
I looking in the wrong place?
I'd really not like any prefix, as I use the ring groups in AAH to add
a prefix, depending on which ring group is called.
Thanks
Matthew
| |
|
|
"Matt" <matthew.humphreys@gmail.com> wrote in message
news:1132659434.647788.269900@o13g2000cwo.googlegroups.com...
> Just found 1 issue...
>
> If a call out over the pstn, I get a ringing tone before the unit has
> dialed.
>
> So if I call an engaged number, I get the ringing tone for 1 cycle and
> then the engaged tone.
>
> Very odd.
>
> Any thoughts?
>
>
> Thanks
>
> Matt
>
Yep, that will happen.................can't think of anything else to say!
| |
|
|
"Matt" <matthew.humphreys@gmail.com> wrote in message
news:1132664423.589182.173080@g49g2000cwa.googlegroups.com...
> Another one.... sorry!
>
> I assumed that the Cfwd Sel1 Caller:00* prefixed the cli passed to
> asterisk with a 00.
>
> This it does, but if I change it, to 0* or Z* it still passes 00 - am
> I looking in the wrong place?
>
> I'd really not like any prefix, as I use the ring groups in AAH to add
> a prefix, depending on which ring group is called.
>
>
> Thanks
>
> Matthew
>
Are you saying the cli isn't being displayed properly?
My SPA presents the CLI exactly as received by the trunk......what's yours
doing?
| |
|
|
Jono wrote:
> "Matt" <matthew.humphreys@gmail.com> wrote in message
> news:1132664423.589182.173080@g49g2000cwa.googlegroups.com...
>
> Are you saying the cli isn't being displayed properly?
>
> My SPA presents the CLI exactly as received by the trunk......what's yours
> doing?
A reboot seems to have fixed it - thanks.
| |
| Andrew Gabriel 2005-11-22, 8:45 pm |
| In article <SvGgf.13466$Lw5.5230@text.news.blueyonder.co.uk>,
"Jono" <no@no.co.uk> writes:
>
> "Matt" <matthew.humphreys@gmail.com> wrote in message
> news:1132659434.647788.269900@o13g2000cwo.googlegroups.com...
>
> Yep, that will happen.................can't think of anything else to say!
Well it doesn't for me, dialling through the spa-3000
(I don't have AAH).
--
Andrew Gabriel
| |
| Andrew Gabriel 2005-11-22, 8:45 pm |
| In article <Uctgf.13265$Lw5.7309@text.news.blueyonder.co.uk>,
"Jono" <no@nospamblueyonder.co.uk> writes:
>
> To get your head around the spa3k, you have to bear in mind that there is no
> direct path from the pstn line to the line 1 Voip (which your handset may be
> plugged into)
>
> For a PSTN call to be answered, a VOIP call is placed within the unit
> between the two lines (AFAIK)
Correct. It affectively has 2 VoIP units in it, one on the FXO
and one on the FXS. To call through the unit between the two
ports, the call gets converted to VoIP and then back again.
--
Andrew Gabriel
| |
|
| I guess some settings are different between our 2 units? Any chance of
some html files of your sipura settings?
Many thanks
Matthew
| |
|
| 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...
*****************
yes please - I think you have my email address already!
Matt
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