[SOLVED] PSTN - Linksys-3102 Configuration problem.

[pw44]

Hia,

trying to configure the linksys spa-3102 for pstn line use only, i did follow the http://wiki.linuxmce.org/index.php/Sipura/Linksys_spa3000_pstn_interface page.
But something must be changed since than.
I successfully created the trunk, inbound route and outbound route.
The problem is after using the "temporary hack" described below, the inbound route for broadvoice is not created.
Trunk contains ZAP/G0, SIP/House Line and SIP/broadvoice
Outbound Routes contains 0 House Line and 1 broadvoice
Inbound Routes contains 2122498618 / any CID , but no broadvoice entry.

Code:

Next, go to Wizard->Devices->Phone Lines (on the left pane in the LinuxMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!) Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode! (Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).

How could it be solved? I do not have voip service (yet), only pstn service.

TIA

Paulo

[pw44]

Hia
One more question: to make Linksys SPA-3102 work with LMCE, which network jack shall be used? Internet (WAN) or Ethernet (LAN)?
TIA
Paulo

[pw44]

Anyone?

[stallione]

Paulo,
What are you trying to do? You mention that you don't have any voip service and yet you are trying to configure broadvoice?

I have broadvoice service and I use LMCE asterisk/freepbx to access the SIP trunks. on the Inside network (or eth1 network) of the Core, I have a linksys ATA connected with two lines that have two extenstions, but in order to set that up, you only need very minimal configuration on the webadmin and dont' really have to mess with freepbx.
Could you please provide more details on what you are trying to accomplish, and I use the WAN port on the Linksys ATA to connect to the LMCE internal network.

Stallione

[pw44]

Hi Stallione,
thx for answering. i am trying to setup the configuration described at: http://wiki.linuxmce.org/index.php/Sipura/Linksys_spa3000_pstn_interface, but i'm getting problem in having the broadvoice hack working.
The wiki does not describe which interface is to be used, lan or wan, and by the time it was written, i don't know if something has changed and the hack does not work anymore.
My planned configuration is initially for PSTN only.  If all works well, i will use voicecheap for calls to Europe, having pstn and voip.
Paulo

[pw44]

Hia,
partially solved the problem.
1) the spa-3102 firmare was outdated - updated it to 5.1.7 (5.1.10, the latest, have bugs, according linksys community list)
2) i was connecting the spa-3102 to the network via LAN port - does not work (but the wikis does not point this detail out - so i updated one of them). Connection via WAN port worked.
Partially solved because there is still a strange problem:
- calling from outside to the house line and picking up the call - when hangup, line free - that's the right behavior.
- calling from house line to outside and call picked up - when hangup, line free - that's the right behavior.
- calling from outside to house line and NOT picking up (call dropped) - line hangs - that's the WRONG behavior.
Did try to change the disconnect tone pattern, to accomplish brazilian standards, but did not work.
It seams that asterisk makes the voice prompt, and do not hang the line after called number disconnect.
Does anyone faced this problem before and may help?
Unfortunatelly, i'm too new to asterisk (started just one week ago to handle with it).
TIA
Paulo

[pw44]

- calling from outside to house line and NOT picking up (call dropped) - line hangs - that's the WRONG behavior.
solved by changing PSTN Line -> Disconnect Tone = 425@-19,425@-19;5(.25/.25/1+2)
425@-19,425@-19;5(.25/.25/1+2) will work for brazilian pstn lines.