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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.

Configuring Sipura SPA3102



 
 
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  #1  
Old February 23rd 08, 12:24 PM posted to uk.telecom.voip
Chris Davies
external usenet poster
 
Posts: 444
Default Configuring Sipura SPA3102

So I've taken the plunge and, despite warnings about its dire
documentation, purchased a (Linksys) Sipura SPA-3102.

I've got Sipgate working on Line 1, but I can't seem to get my ideal
configuration. I was rather hoping someone here could help...

* Inbound PSTN from BT to ring the phone using Ring Cadence #1
(BT Ring)

* Inbound VoIP from Sipgate to ring phone using Ring Cadence
#2 (Call sign)

* Outbound phone to use VoIP via Voipcheap

* Outbound phone to use PSTN if prefixed with, say, 121

* CLID to be presented to phone from PSTN and VoIP

I don't want any of this VoiP-PSTN gateway stuff where you can enter
a PIN to get gatewayed from one to the other. Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone myself.

Can anyone help? I'll happily offer a beer or two

Thanks,
Chris
  #2  
Old February 23rd 08, 02:07 PM posted to uk.telecom.voip
Brian A
external usenet poster
 
Posts: 1,037
Default Configuring Sipura SPA3102

On Sat, 23 Feb 2008 12:24:33 +0000, Chris Davies
wrote:

So I've taken the plunge and, despite warnings about its dire
documentation, purchased a (Linksys) Sipura SPA-3102.

I've got Sipgate working on Line 1, but I can't seem to get my ideal
configuration. I was rather hoping someone here could help...

* Inbound PSTN from BT to ring the phone using Ring Cadence #1
(BT Ring)

* Inbound VoIP from Sipgate to ring phone using Ring Cadence
#2 (Call sign)

* Outbound phone to use VoIP via Voipcheap

* Outbound phone to use PSTN if prefixed with, say, 121

* CLID to be presented to phone from PSTN and VoIP

I don't want any of this VoiP-PSTN gateway stuff where you can enter
a PIN to get gatewayed from one to the other. Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone myself.

Can anyone help? I'll happily offer a beer or two

Thanks,
Chris

I don't know how much you know already so I will assume not so much.
I will just provide you here with the basic tools to solve some of the
problems you might encounter.
It is useful to know that, in addition to the PDF on the SPA-3102,
there is a general PDF covering such things as extra in info. on dial
plans. It was on the Sipura.com site so I assume that it is still
there.

Incoming calls can only be received from PSTN and the single, default,
in/out provider.

You have 4 gateways, in addition to the in/out bound provider.
To route calls to any provider you can choose, in your dial plan:-
1. No stated gateway - calls will be routed via your default
in/outbound provider.
2. gw0 - calls routed via PSTN
3. gw1,gw2,gw3,gw4 - Calls routed via the approriate gateway.
Include this in any rule as appropriate:-
:@gwx where 'x' is the gateway number.

You can route outgoing calls by number type.
For example:
:00441274[2-9]xxxxx:@gw1
This will route local (Bradford 01274) calls via gw1
In th eabove local calls are defined as starting with any number from
2 to 9 inclusive followed by five other numbers. 00441274 is
automatically inserted. If the number of digits following [2-9] is
unknown, or flexible, just write 'x.' (without the quotes) to replace
xxxxx
Further dial plan info. on my crappy looking web site:-
www.leafcom.co.uk
Access only from your home computer - I am sharing space with another
site that has nothing to do with me.

I don't use a PSTN so I'll leave that bit to someone else, but I hope
that the above will help you get started.






---
Remove 'no_spam_' from email address.
---
  #3  
Old February 23rd 08, 03:36 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default Configuring Sipura SPA3102

on 23/02/2008, Chris Davies supposed :
So I've taken the plunge and, despite warnings about its dire
documentation, purchased a (Linksys) Sipura SPA-3102.

I've got Sipgate working on Line 1, but I can't seem to get my ideal
configuration. I was rather hoping someone here could help...

* Inbound PSTN from BT to ring the phone using Ring Cadence #1
(BT Ring)


Make sure you've UK-ised your SPA.

whilst logged in as admin/advanced, go to the Voice section & choose PSTN user page. at the bottom, there should be an option to set the default ring for the PSTN - under "PSTN Ring Thru Line 1 Ring Settings"

* Inbound VoIP from Sipgate to ring phone using Ring Cadence
#2 (Call sign)


As above, but look on the User 1 page.

At the bottom, there should be the Default Ring setting.

* Outbound phone to use VoIP via Voipcheap

* Outbound phone to use PSTN if prefixed with, say, 121


Both of the above are dealt with using the Dial Plan on the Line 1 page. What dial Plan have you set?


* CLID to be presented to phone from PSTN and VoIP


What have you got set as the CLI type on the Regional page? (try ETSI FSK with PR(UK))


I don't want any of this VoiP-PSTN gateway stuff where you can enter
a PIN to get gatewayed from one to the other.


OK.

Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone myself.


On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While Calling VoIP no.



Can anyone help? I'll happily offer a beer or two


Below, I've pasted {{{{Welcome}}}}'s regular post on regional settings for the UK

Admin Login
Advanced View

Under Regional Tab.

Dial Tone: [email protected],[email protected];60(*/0/1+2)
Second Dial Tone: [email protected],[email protected];60(*/0/1+2)
Outside Dial Tone: [email protected];60(*/0/1)
Prompt Tone: [email protected],[email protected];60(*/0/1+2)
Busy Tone: /1)
Reorder Tone: [email protected];20(*/0/1)
Off Hook Warning Tone: /1+2)
Ring Back Tone: /2/1+2)
Confirm Tone: /1)
SIT1 Tone: /0/3,0/1/0)
SIT2 Tone: /0/3,0/4/0)
SIT3 Tone: /0/3,0/4/0)
SIT4 Tone: /0/3,0/4/0)
MWI Dial Tone: /1+2)
Cfwd Dial Tone: /1+2);10(*/0/1+2)
DND Dial Tone: /2);10(*/0/1+2)
Holding Tone: /1)
Conference Tone: /1)
Secure Call Indication Tone: /2)
Feature Invocation Tone: /1)

Ring1 Cadence: 60(.4/.2,.4/2)
Ring2 Cadence: 60(1/2)
Ring3 Cadence: 60(.25/.25,.25/.25,.25/1.75)

Ring1 Cadence = The UK Ringing pattern.
....

CWT1 Cadence: 30(.2/.2,.2/4.4)

Ring Waveform: Sinusoid (though Trapezoid may help problematic phones to ring).
Ring Frequency: 25
Ring Voltage: 80 (You can use 70, if problems try 75, 80, 85, 90)
CWT Frequency: [email protected]
Synchronized Ring: Yes

Hook Flash Timer Min: .06
Hook Flash Timer Max: .2
Callee On Hook Delay: 0
Reorder Delay: 5
Call Back Expires: 1800
Call Back Retry Intvl: 30
Call Back Delay: .5
VMWI Refresh Intvl: 0
Interdigit Long Timer: 10
Interdigit Short Timer: 3
CPC Delay: .5
CPC Duration: .1

....

Time Zone: GMT
FXS Port Impedance: 370+620||310nF
Daylight Saving Time Rule: start=3/-1/7/2:0:0;end=10/-1/7/2:0:0;save=1:0:0
DTMF Playback Level: -16
DTMF Playback Length: .25
Caller ID Method: ETSI FSK With PR(UK)
FXS Port Power Limit: 3
Caller ID FSK Standard: v.23
Feature Invocation Method: Default
More Echo Suppression: yes

In Line 1 (or Line 2)

Line Enable: yes
NAT Mapping Enable: yes
NAT Keep Alive Enable: yes
Network Jitter Level: low (Depends on how good your boadband is / route to VoIP server)
Jitter Buffer Adjustment: Up and Down


Proxy and Registration
Proxy: Your Voip Proxy address (for example sip1.sipdiscount.com or sip.voipcheap.com or sipgate.co.uk etc)

Subscriber Information - (where you enter your VoIP Account details).

Supplementary Service Subscription - This is where you enable services such as Call Waiting, 3-Way Calling, block anonymous caller etc)

Audio Configuration - This is where you set-up which codecs you want to use or only use.
If you have limited broadband speed (mostly if upload is poor) then you probably only use one or more from G729a, G723 or G726)
If you want the best quality line, then you'd choose G711a (which is the standard used on a standard line)

(Myself, I use G711a only, have that set to my preferred codec and use that one only, but you may find other codecs are fine).

DTMF Tx Method: (This will depend on the VoIP service you use, trial and error and depends on codec) Soon know if it don't work when
trying telephone banking for example and it doesn't recoginse what numbers you are pressing).

Hook Flash Tx Method: None


  #4  
Old February 27th 08, 07:02 PM posted to uk.telecom.voip
Chris
external usenet poster
 
Posts: 9
Default Configuring Sipura SPA3102

Jono wrote:

Chris Davies wrote


Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone myself.


On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While Calling VoIP no.


Can an SPA-3000 also be configured not to answer PSTN until the VOIP
call is picked up? That would be really handy. But there seems to be
no "Off Hook While Calling VOIP" option in the SPA-3000's menu.

Chris
  #5  
Old February 27th 08, 09:00 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default Configuring Sipura SPA3102

Chris wrote on 27/02/2008 :
Jono wrote:

Chris Davies wrote


Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone myself.


On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook While
Calling VoIP no.


Can an SPA-3000 also be configured not to answer PSTN until the VOIP
call is picked up? That would be really handy. But there seems to be
no "Off Hook While Calling VOIP" option in the SPA-3000's menu.


It appears in (presumably) a later firmware revision than yours.

Just upgrade your firmware to the latest & greatest!

the latest on the sipura site is
http://www.sipura.com/Documents/spa3k-3.1.10d.zip (version 3.1.10d)

I'm using the Linksys branded firmware version 3.1.20(GW) which can be
downloaded from the linksys.com support pages.

This
http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayo ut&cid=1169083367861&pagename=Linksys%2FCommon%2FV isitorWrapper&lid=6786143006B03
seems to be almost the closest I can get to a link to the firmware. (or
http://tinyurl.com/3ynf6n if the long one wraps)

Even though v 3.1.20(GW) is apparently for a "Linksys" SPA3000, it
works perfectly well on my "Sipura" SPA3000.

HTH.


  #6  
Old February 29th 08, 12:34 PM posted to uk.telecom.voip
Sinna
external usenet poster
 
Posts: 42
Default Configuring Sipura SPA3102

Jono wrote:
Chris wrote on 27/02/2008 :
Jono wrote:

Chris Davies wrote


Also, I don't want the
SPA3102 to answer an incoming call unless I've picked up the phone
myself.

On the PSTN Line page, set PSTN Ring Thru Line 1 to yes and Off Hook
While Calling VoIP no.


Can an SPA-3000 also be configured not to answer PSTN until the VOIP
call is picked up? That would be really handy. But there seems to be
no "Off Hook While Calling VOIP" option in the SPA-3000's menu.


It appears in (presumably) a later firmware revision than yours.

Just upgrade your firmware to the latest & greatest!

the latest on the sipura site is
http://www.sipura.com/Documents/spa3k-3.1.10d.zip (version 3.1.10d)

I'm using the Linksys branded firmware version 3.1.20(GW) which can be
downloaded from the linksys.com support pages.

This
http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayo ut&cid=1169083367861&pagename=Linksys%2FCommon%2FV isitorWrapper&lid=6786143006B03
seems to be almost the closest I can get to a link to the firmware. (or
http://tinyurl.com/3ynf6n if the long one wraps)

Even though v 3.1.20(GW) is apparently for a "Linksys" SPA3000, it works
perfectly well on my "Sipura" SPA3000.

HTH.


Sipura was acquired by Linksys, acquired by Cisco.

Sinna
  #7  
Old February 29th 08, 12:39 PM posted to uk.telecom.voip
Chris Davies
external usenet poster
 
Posts: 444
Default Configuring Sipura SPA3102



Brian A wrote:
You have 4 gateways, in addition to the in/out bound provider.
To route calls to any provider you can choose, in your dial plan:-
1. No stated gateway - calls will be routed via your default
in/outbound provider.
2. gw0 - calls routed via PSTN


These now work well; thank you to you and Jono (and {{{Welcome}})


3. gw1,gw2,gw3,gw4 - Calls routed via the approriate gateway.


I've hit all sorts of problems with these additional gateways. Reading
around it seems that what Linksys/Sipura don't tell you is that the
SPA3102 is broken with respect to these gateway definitions.


:00441274[2-9]xxxxx:@gw1
This will route local (Bradford 01274) calls via gw1


Theoretically, yes. In my case I have Sipgate as my primary provider
and I'm trying to use Voipcheap for outbound dialing via gw1, but it
doesn't work. (Read on for details.)

Here's some specifics; perhaps if one of you see that I've dropped a
clanger they can help me out:

1. Default provider

Proxy: sipgate.co.uk
Outbound proxy: sipgate.co.uk
Use outbound proxy: Yes
Use OB Proxy in dialog: Yes
Register: Yes
Make call without reg: No
Answer call without reg: No

Display name: Chris Davies
User ID: 12345678 (my Sipgate ID)
Auth ID: 12345678 (my Sipgate ID)
Use Auth ID: Yes


2. Gateway accounts

Gateway 1: :5060 (previously verified)
GW1 NAT mapping enabled: Yes/No (doesn't seem to matter)
GW Auth ID:
:5060 (my Voipcheap ID)

Gateway 2:
:5060 (previously verified)
GW2 NAT mapping enabled: Yes/No (doesn't seem to matter)
GW Auth ID:
:5060 (my Voipcheap ID)


3. Dial plan

( **1 : xx. :@gw1 | **2 : xx. :@gw2 | **3 : xx.
:5060;usr=ABCDEF;pwd =****** | xx. )

I've also found references to "usrid" instead of "usr", but that seems
to make no difference.


What seems to happen is that regardless of the authentication details I
provide - whether in the GW1 and GW2 configuration or in the dialplan
itself - the authentication details for the primary provider are used.

Looking at the wireshark network trace, I can see attempts to authenticate
to voipcheap using my sipgate userid. Obviously voipcheap doesn't want
to play ball as it doesn't have a clue who this is.

This stuff's hard enough to get one's head round without buggy software,
too.

Chris
  #9  
Old February 29th 08, 05:24 PM posted to uk.telecom.voip
Chris Davies
external usenet poster
 
Posts: 444
Default Configuring Sipura SPA3102

Nick wrote:
Gateway 1: :5060 (no idea if + or 00 matters,
but I use +))


About 1/2 hour ago I found this, which makes it all happiness and light:
http://www.aoakley.com/articles/2008-01-08.php,

Thanks for your help.
Chris
  #10  
Old February 29th 08, 05:30 PM posted to uk.telecom.voip
Chris Davies
external usenet poster
 
Posts: 444
Default Configuring Sipura SPA3102 [thanks!]

Chris Davies wrote:
Can anyone help? I'll happily offer a beer or two


I don't have email addresses for all of you, but if you (Brian A, Nick,
and Jono) would like to email me direct, I'm sure we can work out how
to get beer (money) to you. Please make sure there's no "-usenet" in
the email address.

Many thanks indeed,
Chris
 




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