Cisco SPA8800 DialPlan

3cxcisco08 · ‎12-01-2009

Hello Guys,

we have a Cisco SPA 8800 4 Port Analog PSTN gateway here and i am getting a bit mad with it.

May you can save me.

First, is it possible to switch of the DHCP Server on the AUX Port?

This is just a bit annoying but not my really problem...

Secondly i have a issue connecting the gateway to my PBX and make 4 sim. outbound calls.

I am not using asterisk (3CX Phone System). Here i can only define per gateway once the SIP Port to communicate to.

But this brings me always to line 1 of the device. then this one is busy my outbound calls fail to work.

Can i set somewhere in the device or in the dial plan to use the next line when line 1 is full.

if line 2 is full then line 3 and so on?

I have seen Fallback options and tried them to select in Line 1 Fallback to 4, for 2 into 6, for 6 into 8 and for 8 none.

But this does not work either.

I am thankful for any suggestion.

3cxcisco08 · ‎12-02-2009

Heavily reading this document right now,

but i even fail to trigger a new line to dial when i write for "Line 1" DialPlan 1: (xx.|<#2,:>)

I would understand now what ever i dial i use line 2 of the gateway, or is here already i thought issue?

http://www.cisco.com/en/US/products/ps10033/products_qanda_item09186a0080a35a44.shtml

Patrick Born · ‎12-02-2009

Hi 3cxcisco08,

Why do you want to switch off the DHCP server on the AUX port? The AUX port is designed to *only* be used for administration tasks so it should not affect you in any way. Connect the SPA8800 to the network using the ETHERNET port. Do not connect the AUX port.

Here's the SPA8800 admin guide, perhaps it will be helpful:
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf

Linked from: https://www.myciscocommunity.com/docs/DOC-2149

Perhaps this Asterisk-specific document will help you: https://www.myciscocommunity.com/docs/DOC-7654 It deals with outbound dialing.

You say:

I am not using asterisk (3CX Phone System). Here i can only define per gateway once the SIP Port to communicate to.

Each port of the SPA8800 is individually registered, 5061, 5161, 5261, and 5361 for ports 1-4 respectively.

Let me know if I misunderstood your questions and I'll try again.

Regards,

Patrick

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3cxcisco08 · ‎12-03-2009

Hello Patrick,

thx for your post.

The both links i have read before and was not able to use them to solve my issue.

Basically 3CX creates one Gateway with then 4 ports.

The gateway is defined once, so i can enter once the IP and the SIP port for the 8800.

Then i have 4 time the sip registration.

If i proceed as normal i can make 4 inbound calls and route them to the right destination per port.

So far so good.

But the Cisco SPA8800 only dials on port one out (where the sip port of the gateway-3cx and the Line is the same)

Line 1 5061, Line 2 5161...

But i have no possibility to trigger 5161 from the 3cx due to i can simply can not configure it there...

All the other gateways either work then on a global dial plan or use the sip invite with the sip registration to route the calls.

Basically i need to make a way to send the sip invite to 5061 and from there the box should handle all outbound calls and free lines.

So if the matching line 5061 is busy go to Line 2, busy go to Line 3, busy go to Line 4, if line 4 is as well busy send the "403 service unavailable".

This i tried to work out with the fallback DP, didn't worked, also i find no introduction what it is for in the manual, and i tried to use the dial plan to make it work in each Line, still was not able to make it work. I guess the solution is in the DP and VoIP-To-PSTN Gateway Setup but i cant work it out.

Do you have a solution for me?

That would be perfect.

Thx already

Stefan

Alberto Montilla · ‎12-07-2009

Dear Stefan;

No solution. SIP trunking is not yet supported on SPA8800 so you have to address the 4 independent SIP accounts/ports (5060, 61 and beyond). The alternative (I have not checked this on 3CX but in Asterisk and it is documented on the SPA8800 - Asterisk config guide) is that you configure the 4 accounts independently and then create a hunt group for routing the calls.

SIP Trunking is a roadmap feature for SPA8800, thought we dont have yet a date for delivery.

Regards
Alberto

3cxcisco08 · ‎12-15-2009

Dear Alberto,

thx for your answer, at least i am not more thinking i am going carzy and cant set up an dial plan ;-)

So i will wait for an FW update for that BOX and will provide for 3CX clients the "how to" with 4 gateways as an solution.

A bit work for them... but working

THX!!!!