[zzpaf]

Hi there,

Approaching desperation...

I am trying to setup the 3102 so that from my IP PBX (FrontRange IPCC) I can dial out to the PSTN when a user dials a "0" before the number and when a call is received on the PSTN, it will automatically ring on ext. 501, which is the Contact Center IVR.

I managed to route all calls from the VoIP to the PSTN, but I have been struggling for 3 days with the incoming calls.

I have read
in the "Sipura SPA user guide" that "Only 2-stage dialing is possible with PSTN-To-VoIP gateway calls" Meaning that the incoming call would need to go through Line 1, but I see that asterisk users have the setting:
PSTN Ring Thru Line 1: No

Actually, how can I make it so that all the calls received on my PSTN get routed to ext 501 on my PBX? No authentication needed.

Right now my SIP PBX is on 10.0.0.20, my Sipura is 10.0.0.100
And my registration is:

Suscriber Information
=================
- User ID: 0
- Password: password

The user registers properly on my box and I get a log saying:
"Registered 0 at 10.0.0.100:5060(local: 127.0.0.1 rport:None) for 3600.0 seconds"
And the VoIP calls that start with 0 get dialed through PSTN properly.

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

Dial Plan 1: (S0<:501>)


Thanks for the help!

[bcalder01]

> Actually, how can I make it so that all the calls received on my PSTN get routed to ext 501 on my PBX? No authentication needed.

This is exactly what I'm trying to achieve as well with my SPA 3000.

[patwa]

Hi. I might be completely on the wrong track here but, try setting the dial plan to use an explicit gateway. By default, it will try to ring 501 through amd using the credentials set up on the Line 1 page.

I am not sure how your SPA connects to the extension. Do you have more phones connected to the SPA that you want to ring? Do you have an Asterisk box with phones connected to that, or do you want to use a VoIP provider to call 501?

In any case, you would use:
(S0<#9:501><:@gw0>) for dialling out the PSTN, and
(S0<#9:501><:@gwx>) for dialling out using other credentials, where x is a gateway between 1 and 4.

Hope this helps.

Hussein.

[zzpaf]

Thanks for the note.

Actually, I have a system similar to Asterisk which has many extensions:

The setup would be:

1. Ext's -> SIP Proxy (Similar to Asterisk) -> SPA -> PSTN (This happens when from the extension they dial a 0 before the number). This already works by using:

PSTN Line
============
Dial Plan 8: (<0:>xx.)
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: none
VoIP Caller Default DP: 8
(This works perfect)

2. PSTN -> SPA -> 501@10.0.0.20 (IVR)
This is not working now

Do I need to use Line 1?
Can I use the routing from the PSTN Line directly?
I have tried about everything but can't get Ext 501 to ring.

Thanks again for the help!

[hwittenb]

Have you tried the 3102 configuration that comes up using the Voxilla Device Configuration Tool for the 3102 under Asterisk? You can find it at Voxilla - Home. I believe you do what you want using the default dial plan for the PSTN-to-Voip Gateway.

[zzpaf]

No luck, the tool is not helping either, again, I get the VoIP -> PSTN OK, but PSTN -> VoIP does not work...

Any more ideas?

Thanks for the help!

[hunok99]

Hi!
I think I have a simular problem. I'm also trying to forward all incoming calls through FXO port of SPA3102 to my PAP2 with the default dialplan (S0<:PAP2 FWD number>) under PSTN tab but it's not working.
The only way I was able to forward the calls is through User1 tab - Selective Call Forward Settings. Here I put my PSTN line FWD number to my PAP2 FWD num.
This is only wotking if PSTN Ring thru Line1 is yes under PSTN tab.
Its very strange, but at at least is somehow working. Of course this way I had to give up my CID, because now all my incoming calls from FXO port are showing only the PSTN line FWD number, not the real callers name and number.

[zzpaf]

hunok99, can you post your config?

This would work for me for the time being as long as I can get the routing to the SIP extension, even if it is through line1, although the optimum would be directly on the PSTN tab, but until we figure it out...

thx,
-pablo

[hunok99]

I hope this helps. Just make sure Line 1 and PSTN are separate accounst. Then forward your PSTN account to your destination with selective call forward inder USER 1 tab. Just remember now your CID will show always the PSTN account info.
Good luck!