Configuring Sipura SPA3102
Chris Davies
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
Brian A
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.
---
Jono
> 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: 350@-19,440@-22;60(*/0/1+2)
Second Dial Tone: 420@-19,520@-19;60(*/0/1+2)
Outside Dial Tone: 420@-16;60(*/0/1)
Prompt Tone: 520@-19,620@-19;60(*/0/1+2)
Busy Tone: 400@-10;30(.375/.375/1)
Reorder Tone: 400@-20;20(*/0/1)
Off Hook Warning Tone: 480@-10,620@0;90(.125/.125/1+2)
Ring Back Tone: 400@-20,450@-20;*(.4/.2/1+2,.4/2/1+2)
Confirm Tone: 600@-16;1(.25/.25/1)
SIT1 Tone: 950@-16,1400@-16,1800@-16;20(.330/0/1,.330/0/2,.330/0/3,0/1/0)
SIT2 Tone: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
SIT3 Tone: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
SIT4 Tone: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
MWI Dial Tone: 350@-19,440@-22;10(.75/.75/1+2)
Cfwd Dial Tone: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)
DND Dial Tone: 350@-19,440@-19;2(.2/.2/2);10(*/0/1+2)
Holding Tone: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)
Conference Tone: 350@-19;20(.1/.1/1,.1/9.7/1)
Secure Call Indication Tone: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
Feature Invocation Tone: 350@-16;*(.1/.1/1)
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: 425@-20
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
4fzn@yahoo.co.uk Chris
>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
Jono
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%2FLayout&cid=1169083367861&pagename=Linksys%2FCommon%2FVisitorWrapper&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.
Sinna
Sinna
Chris Davies
Brian A <no_spam...@hotmail.com> 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: 0044142...@voipcheap.com:5060 (previously verified)
GW1 NAT mapping enabled: Yes/No (doesn't seem to matter)
GW Auth ID: ABC...@voipcheap.com:5060 (my Voipcheap ID)
Gateway 2: 0044777...@voipcheap.com:5060 (previously verified)
GW2 NAT mapping enabled: Yes/No (doesn't seem to matter)
GW Auth ID: ABC...@voipcheap.com:5060 (my Voipcheap ID)
3. Dial plan
( <**1 :> xx. <:@gw1> | <**2 :> xx. <:@gw2> | <**3 :> xx.
<:0044142...@voipcheap.com: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
Nick
>
> 2. Gateway accounts
>
> Gateway 1: 0044142...@voipcheap.com:5060 (previously verified)
> GW1 NAT mapping enabled: Yes/No (doesn't seem to matter)
> GW Auth ID: ABC...@voipcheap.com:5060 (my Voipcheap ID)
>
> Gateway 2: 0044777...@voipcheap.com:5060 (previously verified)
> GW2 NAT mapping enabled: Yes/No (doesn't seem to matter)
> GW Auth ID: ABC...@voipcheap.com:5060 (my Voipcheap ID)
>
I would use
Gateway 1: +441423...@voipcheap.com:5060 (no idea if + or 00 matters,
but I use +))
NAT enabled = yes (I have no idea if it matters)
GW Auth ID: ABCDEF
Chris Davies
> 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
Chris Davies
> Gateway 1: +441423...@voipcheap.com: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
Brian A
I can really only speak for myself, but I think others will be the
same, just glad to have been of help and don't expect anything in
return. I hope that you are sucesssful in getting your SPA to work
just how you want it to. It is always good to know if the information
supplied has been useful.
You will always find help in this group providing you make an effort
to find readily available information first. Though sometimes, when
you are starting out, it isn't always obvious how to search for what
you want.
Jono
> Sipura was acquired by Linksys, acquired by Cisco.
>
Erm, OK, thanks.....
Jono
No probs. Glad you've got there.
Not a drinker myself, so no need for any beer tokens. Thanks for the
offer anyway!
Chris Davies
> I can really only speak for myself, but I think others will be the
> same, just glad to have been of help and don't expect anything in
> return.
No worries; that tends to be my approach, too. (A usenet poster since,
erm, the mid-90s, I think.)
> I hope that you are sucesssful in getting your SPA to work
> just how you want it to.
Yes, I think it's pretty much there, now, thanks.
Chris
Nick
Someday, and that day may never come, I'll call upon you to do a service
for me. But, until that day, accept my help as a gift.
4fzn@yahoo.co.uk Chris
>Chris wrote:
>> 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!
Thanks for your suggestion to upgrade the firmware. I did just that,
and lo and behold the option "Off Hook While Dialling appeared".
Interestly, now when the SPA3000's PSTN port detects ringing, it rings
my IP phone at home without going off-hook as I wanted, but it doesn't
even go off-hook when I answer it! If I leave it to go off-hook whilst
calling (as it used to), then the call is fine. It just means my PABX
won't hunt to the next PABX extension if I don't answer the IP phone
at home. Ho Hum!
I've also set up IP back-to-back dialling from one to the other, which
seems to be more reliable than going through a SIP server.
Thanks for your help,
Chris
Jono
> Interestly, now when the SPA3000's PSTN port detects ringing, it rings
> my IP phone at home without going off-hook as I wanted, but it doesn't
> even go off-hook when I answer it! If I leave it to go off-hook whilst
> calling (as it used to), then the call is fine. It just means my PABX
> won't hunt to the next PABX extension if I don't answer the IP phone
> at home. Ho Hum!
What's your complete configuration? (pbx, phones, etc)
Chris
At the office:
The PABX is a GDK16.
The SPA3000 is hanging off an analogue extension of the GDK16.
Preferred Codec G711a
Use Pref Codec only YES
Dial Plan 8: (xx.)(S0<:IPaddress:5060>)
PSTN To VoIP Enable YES
PSTN Caller Auth Mehtod NONE
PSTN Ring Thru Line 1 YES
PSTN Caller Default DP: 8
Off Hook While Calling VoIP: YES or NO!
Line 1 Signal Hook Flash to PSTN: DISABLED
PSTN Answer delay: 0
PSTN Ring Thru Delay: 1
PSTN Ring THru CWT Delay: 2
PSTN Dialling Delay: 0
PSTN Hook Flash Len: .25
Detect CPC YES
Detect Polarity Reversal YES
Detect PSTN Long Silence NO
Detect VoIP Long Silence YES
Mim CPC Duration 0.09
Detect Disconnect Tone YES
Disconnect Tone 400@-30,400@-30;2(*/0/1+2)
FXO Impedance 600 (seems to work best for in-band DTMF tones)
Ring Validation Time 256mS
On an incoming PSTN call, firstly Ext 102 (my office) rings for 10
seconds, then if no answer Ext 106 rings (the SPA), then if no answer
finally hunts to Ext 104 (the shop itself). Thats why I had hoped to
leave the SPA on-hook unless or until I picked up my IP Phone at home.
The SPA3000 connects into a Draytek router with all conceivable ports
routed through to it.
At home:
Simply a Grandstream BT-101. Again all ports routed to it ok.
Jono, unless its obvious to you from the above, thanks but please
don't worry - I am able to remotely enable the PSTN to VOIP gateway of
the SPA from home when I'm working at home, and turn it off again
remotely when I've had enough for the day! I suspect an SPA3000 at
each end would be far more compatible and the best solution. And I
don't wanna take up too much bandwidth of this group.
Cheers, Chris