Need help to configure SPA 3102 on PBXes in order to use PSTN line as a Trunk

Hi evereone.

I'm pretty new on PBXes and I need help to setup the sys so a user can call make a local call (I'm Brazil) using my PSTN line.

Certain calls are toll free depending on the prefix of the number called (ie: 0800).

I'm trying to set a trunk that enables PBXES user (or extension) to access my SPA 3102 and then placing a PSTN call. So if I'm in Japan I can make a local call in Rio de Janeiro using a softphone.

Here are the scenarios needed to work:

01) A call from an outside user rings to the local PSTN number connected to SPA-3102 is automatically transferred to a PBXes' extension >> Extension rings on a Softphone configured for this extension.

02) A PBXes User wants to make a toll free call, emergency call or another type of local call not supported by VSP. So he may dial some extension or code on its SIP extension and the call is sent to the SPA-3102 under a static Ip (ie: dyndns). Caller hears a tone sound to make a local call using the PSTN line connected to SPA 3102. He places the call like he was at home even being overseas.

03) A PBXes user wants to make any other call not listed above. So this call is handle by PBXes.

The three ways I already have acheived once but certainly I did something wrong. It worked good only once and then, never again.


Is it possible some how?

My configs are:

My router: SAGEMCOM f@ST 1704 v1.0
NAT SETUP:
Opened ports: 5060 to 5061 - protocol TCP/UDP - Wan Interface ppp0
DMZ: 192.162.1.2 (SPA3102 IP)

ATA: SPA3102 Linksys

VSP: callcentric

====================
* SPA-3102 Configs *
====================

==========
Line 1 TAB
==========

Line Enable: YES

SAS Enable: NO
SAS DLG Refresh Intevl: 30

NAT Mapping Enable: YES
NAT Keep Alive Enable: NO

SIP Transport: UDP
SIP Port: 5061
Auth Resync-Remote: YES
SIP Remote-Party-ID: NO

Proxy: pbxes.org
Outbound Proxy: pbxes.org
Use Outbound Proxy: YES
Use OB Proxy In Dialog: NO
Make Call w/o Reg; YES
Ans Call w/o Reg; YES
Register: YES
Register Expires: 600
Proxy Redundancy Method: Normal

User ID: ethos-101
Pasww: ***
Auth ID: ethos-101

Dial Plan: "blank"

=========
PSTN Line
=========

Line Enabel: YES

NAT Mapping Enable: NO
NAT Keep Alive Enable: NO

SIP Transport: UDP
SIP Port: 5060
Auth Resync-Remote: YES
SIP Remote-Party-ID: NO

Proxy: pbxes.org
Outbound Proxy: pbxes.org
Use Outbound Proxy: YES
Use OB Proxy In Dialog: NO
Make Call w/o Reg; YES
Ans Call w/o Reg; YES
Register: NO
Register Expires: 900
Proxy Redundancy Method: Normal

User ID: ethos-102
Pasww: ***
Auth ID: ethos-102

Dial Plan 1: (xx.)
Dial Plan 2: (S0<:101)

VOIP-To-PSTN Gateway Enable: YES
VoIP Caller Auth Method: NONE
One Stage Dialing: NO
Line 1 Voip Caller DP: 2
VoIP Caller Default DP: 1

PSTN-To-VoIP Gateway Enable: YES
PSTN Ring Thru Line 1: NO
PSTN CID For VOIP CID: YES
PSTN Caller Auth Method: NONE
PSTN Caller Default DP: 2

=================
* PBXES Configs *
=================

SIP Extension 101 (normal settings)
SIP Extension 102 (normal settings)

===============
Trunk SIP/SPA3102
===============
dtmfmode: rtf2833
SIP server: something.dyndns.org:5060 (Static IP through dyndns)
Domain: "blank"
Register: NO
Options: "all blank"
Dial rules>> Prefix: 9

===============
Trunk Callcentric
===============
dtmfmode: auto
SIP server: callcentric
User & passwd>> fine
Domain: "blank"
Register: YES
Options & Dial rules: "all blank"

Outbound Route 0: CallSPA
Trunk Sequence: SIP/SPA3102 (only)
Custom Dial Patterns: 9|0800XXXXXXX

Outbound Route 1: Callcentric
Trunk Sequence: SIP/Callcentric (only)
Valid for all numbers

Inbound Rules are ok, transferring calls to extension 101.

Well, making calls through callcentric, just fine.

Calls received on my SPA local number are normally transferred to extension 101.

Somehow it worked. From my Softphone I called 908007048383 and it transfer the call to my SPA, giving me a dial tone.

I believe it is simple to configure it but I've try every way according to my limitations.

Can anyone help me on it or lead me to a tutorial or any other source to get it done correctly?

Thanks in advance.

My best regards.
Marcello Patelli

This post has been edited 2 time(s), it was last edited by eth on 05.05.2012 at 16:51.

RE: Correct SPA 3102 config to use PSTN line as a Trunk

My solution:


pbxes to do:

1,creat sub-pbxes (USER ID: ethos-102/Password: whatyoulike) for SPA3102 pstn line.
2,creat outboud routing: 9(Numbers starting by,select Separate prefix),select trunk sip/ehos-102

spa3102 to do:

1, pstn line use data of sub-pbxes (USER ID: ethos-102/Password: whatyoulike)
2, pstn line/VoIP-To-PSTN Gateway Setup
VoIP Caller Auth Method: none
One Stage Dialing:yes
VoIP Caller Default DP:none



Please note, began to set, delete your original pbxes corresponding set.

This method has been used for a long time, very perfect.

pbxes great.

12.May.2012


SIP RUI:11012017705@ideasip.com

This post has been edited 4 time(s), it was last edited by ok1 on 15.05.2012 at 19:45.

RE: Correct SPA 3102 config to use PSTN line as a Trunk

Can this also be made with the newer Cisco SPA122?

I would like to make our analogue line at the company PBX available at home during weekends threw a sip and pbxes...