[rcilink]

I have a sipura SPA3000 setup with an asterisk pbx (1.0.9 release). Until last week, I had the PSTN line go directly into a $6-7 cheapo modem card. It works, but does have some echo problems, cant detect disconnect, etc.

New config:

PSTN <----> FXO spa3000 <---> asterisk <---> ATA (kitchen phone)

So, now that I have spent several days working on the configuration details on the spa3000 and on asterisk sip.conf files, I have a big question.

Has anyone been able to get the spa3000 to behave correctly with the CallerID, when it is marked P (private) or O (out of area)? I get normal callerID just fine, but do not get the "p" or "o" or anything telling me it was blocked..

Instead, the spa3000 inserts it's own callerid number (the 'username' from the FXO sip registration).

This is not acceptable. I need to know that an 'out of area' call just came in, as I have scripts written specifically to handle these calls. The incoming 'private' calls do not go the same route.

Because the spa changes the CallerID, it breaks the Asterisk "privacy manager" option, and the "Zapateller(nocallerid)" options.

Has anyone been able to get an incoming call (either blocked, or 'out of area') to pass through to Asterisk with the CallerID remaining intact (not replaced by SPA3000 FXO SIP username)... ?


------- Below is screen-copys (text) of the Sipura 3000 and the sip.conf parts-
* Note: both FXS and FXO of Sipura 3000 register with asterisk. I can call out on FXO (through asterisk) and can receive calls from FXO through asterisk.

------

(((FROM SIPURA 3000)))
PSTN Line/SIP Settings

SIP Port: 5061
SIP 100REL Enable: no
EXT SIP Port:
Auth Resync-Reboot: yes
SIP Proxy-Require:
SIP Remote-Party-ID: no
SIP Debug Option: none
RTP Log Intvl: 0
Restrict Source IP: no
Referor Bye Delay: 4
Refer Target Bye Delay: 0
Referee Bye Delay: 0
Refer-To Target Contact: yes
Sticky 183: no


PSTN Line/Proxy and Registration

Proxy: 192.168.1.76
Use Outbound Proxy: no
Outbound Proxy:
Use OB Proxy In Dialog: yes
Register: yes
Make Call Without Reg: no
Register Expires: 3600
Ans Call Without Reg: yes
Use DNS SRV: no
DNS SRV Auto Prefix: no
Proxy Fallback Intvl: 3600
Proxy Redundancy Method: Normal

PSTN Line/Subscriber Information

Display Name:
User ID: spa-pstn-in
Password: abcdefg
Use Auth ID: no
Auth ID:


PSTN Line/Dial Plan

Dial Plan 2: (xx.)
Dial Plan 3: (S0<:s@192.168.1.76>)


PSTN Line/VoIP-To-PSTN Gateway Setup

VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: none
VoIP PIN Max Retry: 3
One Stage Dialing: yes
Line 1 VoIP Caller DP: 2
VoIP Caller Default DP: 2
Line 1 Fallback DP: none
VoIP Caller ID Pattern:
VoIP Access List:


PSTN Line/VoIP Users and Passwords (HTTP Authentication)

VoIP User 1 Auth ID: spa-pstn-out
VoIP User 1 DP: 2
VoIP User 1 Password: abcdefg


PSTN Line/PSTN-To-VoIP Gateway Setup

PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN PIN Max Retry: 3
PSTN CID For VoIP CID: yes
PSTN CID Number Prefix:
PSTN Caller Default DP: 3
Off Hook While Calling VoIP: no
Line 1 Signal Hook Flash To PSTN: Disabled
PSTN CID Name Prefix:
PSTN Caller ID Pattern:
PSTN Access List:


Regional/Miscellaneous

Caller ID Method: Bellcore(N.Amer,China)
FXS Port Power Limit: 3
Caller ID FSK Standard: bell 202


-- Asterisk Configs (important parts)--------------------------
-- taken from SIP.CONF

[spa-pstn]
; outgoing to POTS on Sipura 3000 FXO
;
type=peer
insecure=very
fromuser=spa-pstn-out
username=spa-pstn-out
secret=abcdefg
host=dynamic
canreinvite=no
port=5061
dtmfmode=rfc2833
nat=no
context=intern

[spa-pstn-in]
; incoming from POTS on Sipura 3000
;
type=peer
insecure=very
fromuser=spa-pstn-in
username=spa-pstn-in
secret=abcdefg
host=dynamic
canreinvite=no
port=5061
dtmfmode=rfc2833
nat=no
context=incoming

[gurudc]

Here in Australia we do not get name with CLID - apart from "MOBILE" for mobile calls. The way I way I would solve your problem is to give your SPA a unique ID and set your Privacy scripts to run when they see this ID. Not the "proper" solution I know - but the outcome is the same.

[dualarrow]

rcilink,
I have recently done a similar thing to what you have done so I can divert blocked CID numbers to a specific extension.

Yes, it can be done. There are some hints in other posts on this site with regard to CID which gave me the hints I needed.

In the SPA PSTN tab under "PSTN-To-VoIP Gateway Setup" there is an option "PSTN CID Number Prefix:". I have set mine to "00" (thats 2 zeros) although you could set it to an alpha as another post had. This will prefix all incoming CID numbers with "00", so if you received a call from "12345678" it would be passed on as "0012345678". If a blocked call were to come in, you would get the sipura's internal details as you stated. I doubt the details would start with "00".

All you have to do now is test for the 1st 2 digits being "00". If they are, then you have a valid CID following them. If not, you have a blocked CID.

Here is a section from extensions.conf that I used (I hope I have supplied enough)

exten => 111,1,NoOp(CID ========================= ${CALLERID})
exten => 111,2,SetCIDName(${CALLERIDNAME})
exten => 111,3,GotoIf($["${CALLERIDNUM:0:2}" != "00"]?111,5)
exten => 111,4,SetCIDNum(${CALLERIDNUM:2})
exten => 111,5,LookupCIDName
exten => 111,6,NoOp(NEW CID VALUE SET TO ============== ${CALLERID} =========)
exten => 111,7,GotoIf($["${CALLERIDNUM}" != "110"]?111,11)
exten => 111,8,SetCIDName(Unknown Name)
exten => 111,9,NoOp(======= Unknown Caller =========)
;Goto(NoCIDContext|s|1)
exten => 111,10,GoTo(111,21)

exten => 111,11,NoOp(======= CID CHECK ====${CALLERIDNUM}====${CALLERIDNAME}======)


111,3 checks if the 1st 2 digits are "00". If they are not, it jumps to 111,5
111,4 strips the 2 fake "00"'s from the CID leaving the original CID as the new CID
111,5 looks up the number to add a CID name if it's known (details at http://www.voip-info.org/wiki/index....+LookupCIDName )

111,7 checks if it's the same as my SPA's extension (ie it had CID blocked) and jumps to 111,11 if it wasnt blocked
111,8 replaces a blocked CID with one of my choosing

The rest you should be able to figure out. The NoOp's are for debugging.

hth,
Andrew