[sunshine757]

We have two Panasonic KX-TD1232 PBXs, 2km apart, with 2 Sipura 3000s to configure to allow phone extensions from either PBX to VoIP over a private LAN to extensions on the other PBX.
ie Extn dials a code, gets dial tone from far end, dials far end extensions.
This is tricky. I would very much appreciate help here.

Call types 3(paths 7>6>5)& then call type 4(paths 5>6>7).
Start by getting it working in one direction:
Xtn 225 on PBX "A" is wired to a Sipura FXO port, callers dial 225 & should then hotline via the local LAN to the distant Sipura and, via its FXO port, to a CO line off that PBX ("B"). This will allow this incoming call to dial extensions on that PBX.

When an A xtn dials 225 the A Sipura answers with reorder tone. I would expect dial tone.
Dialling any digit breaks tone momentarily, then the tone continues.
Your help in getting this to work is definitely appreciated.

(Both the 3000s give reorder tone not dial tone to a phone in the FXS port. Is this normal? However the IVR does work.)

Info page for A says :
Last PSTN Disconnect Reason: VOIP Call Failed
VOIP State : Invalid
Last Called VOIP Number: (always empty)
VOIP Peer # : 9901@192.168.0.99:5060

Settings are:
A 192.168.0.96
Gateway 192.168.0.99
UserID: LINE1 9601, PSTN 9602
PSTN Line - Dial Plan 1: S0 <:9902@192.168.0.99:5060>

B 192.168.0.99
Gateway 192.168.0.96
UserID: LINE1 9901, PSTN 9902
PSTN Line - Dial Plan 1: S0 <:9602@192.168.0.96:5060>

Both have:
SIP port: PSTN 5060, LINE1 5061
PSTN Caller Default DP: 1
VoIP Caller Default DP: 2
PSTN Line - Register No, Make call without Reg Yes, Ans Call without Reg Yes, PSTN Ans Delay 0.

Idle Line Voltage is 29v so LineInUse Voltage set to 22v.

The full config .htm file for A is attached.
Thank you !

[mberlant]

Your problems seem to be in your Dial Plans. First of all, all Dial Plans require exactly one set of parentheses around the entire Dial Plan. Second, the "dial immediately" code is S0 (big-ess-zero), not SO.

Try cleaning these things up and see if your success improves.

[sunshine757]

mberlant, Thank you.

PSTN Line Dial plan is now
(S0 <:9902@192.168.0.99:5060>)
and at far end
(S0 <:9602@192.168.0.96:5060>).
However caller into FXO port still gets Reorder tone, with Info - PSTN Line at A end saying :
Last PSTN Disconnect Reason: VOIP Call Failed
VOIP State : Invalid
Last Called VOIP Number: (always empty)
VOIP Peer # : 9902@192.168.0.99:5060
and no indication on Info page at far end of it being seized.

I'm now suspicious of the Sipura - PBX conection at the far end. Will recheck as soon as I can get out there, may be a while, and update post.

Is it normal for an FXS port phone, newly off-hook, to get reorder tone rather than dial tone? Both the Sip3ks give this.
Chris