[solarflare]

I've now three different ITSPs set up in my SPA9000, thru line 1 to line 4, and my SPA-1001 second line is registered as an extension 101.

Registration	Station	User ID	IP Address	Reg Expires(s)	User-Agent	
								 101		 192.168.1.1	243				 Sipura/SPA1001-3.1.8(SEc)

Auto attendant is on and its dial plan is (x:10x|xxx.) which should be just about right. The purpose is that the SPA-1001 would ring when the incoming caller presses one.

The AA script is the default:

<aa><form id="dir" type="menu"><audio src="prompt1" bargein="T"/><noinput timeout="10" repeat="T"/><nomatch repeat="F"><audio src="prompt3" bargein="T"/></nomatch><dialplan src="dp1"/><match><default><audio src="prompt2"/><xfer name="ext" target="$input"/></default></match></form></aa>.

The menu plays correctly and the call is forwarded, but then it just ends. Curious. What I'm doing wrong in here? I am trying to forward it to my SPA-1001, aren't I?

Two other questions as well.

(1) Can I disable or set the auto attendant somehow to pass thru either line 2 or a predefined caller id?

(2) FXS1 and FXS2 setup screens are a tad confusing. There are just the User ID and Display Name fields. No ITSP settings for proxy registrations at all? What if I want my FWD account in the FXS1 port?

(3) Do I really have to spend a whole line for associating my SPA-1001 with my SPA9000? I thought one could set up a manual static extension in the SPA-9000 PBX side, so that it wouldn't neccessarily waste a whole ITSP registration appearence.

[humba3]

Can I disable or set the auto attendant somehow to pass thru either line 2 or a predefined caller id?

I've never tried the callerid thing but I have all my lines equipped with a Contact list that directly routes calls to all the phones I want to ring.. just replace the 'aa' element in the contact list with the numbers you want.. e.g. my line1 contact list looks like this: 2?, 3?, 40. Meaning all phones with user ID 2x, 4x and 40 ring when a call comes to line 1.

What if I want my FWD account in the FXS1 port?

You can steer that via Group Lines. Create a group in which you put your FWD line and the userid of your fxs port. It's no different than using groups for IP phones.

Do I really have to spend a whole line for associating my SPA-1001 with my SPA9000? I thought one could set up a manual static extension in the SPA-9000 PBX side, so that it wouldn't neccessarily waste a whole ITSP registration appearence.

Wouldn't a hotline cutover DP on the SPA1k do the trick? Refer to the SPA9000 sticky for SPA9k and SPA3k integration.. it outlines the procedure.

[solarflare]

Ok, tnx for the answers. Sounds promising, though I seem to have serious trouble on understanding the Group User IDs and Group Lines thing. Since the Linksys quick setup application is for Windows only, it's unusable in my computer so this really has to be done the hard way in case it was trivial with that piece of software.

I'm still trying to get the aa working properly. My FXS1 User ID is foo. My imagination has met its limits with the AA Dial Plan variations.

(<800:foo@FXS1>S0)
(<800:foo>S0)
(<800:foo@>S0)
(<800:foo@127.0.0.1>S0)
(<800:foo@127.0.0.1:5080>S0)

I wonder why none of these doesn't work AA Dial Plan as intended. So that extension 800 would ring the FXS1 port, which it doesn't right now. (<800:>foo) on the other hand says that the extension itself isn't valid.

The phone accepts the extension saying "Call forwarded" but the FXS1 doesn't ring.

This has to be very simple, right? Anyone?

[humba3]

I configured my contact lists manually and it's really simple when you read through all the documentation (which in itself is a tremendeous undertaking).
Each group consitst of 1 elements, the Lines, and the members. Here's my group 1:
Group 1 Line: 1,2 Group 1 User ID:2?,3?,40,80

That means the following: Extensions 20 - 29, 30 - 39, 40 and 80 can call outside using the accounts configured in the Line1 and Line2 tab. In the Group X Line, you put a comma separated list of lines on your SPA9k.. so it can contain up to 4 elements and each element is either 1,2,3 or 4. The order indicates in which order a line is being used. Say your Line1 can make 2 outgoing calls simultaneously (that's set by your IPTSP, and if you want to be more restrictive than your IPTSP you can go to your Line X tab and configure the Call Capacity: to set the number of calls you want to allow on that line), then if 20 makes a call, 21 makes a call, those two calls will go out via Line 1. If while these calls are in progress, 22 makes a call as well, that call will go through the Line 2 account.. and so forth.

Keep in mind this only concerns outgoing calls.. incoming calls is a whole different ballgame and is configured via the Contact List field in Line 1 - Line 4.

I have to ask, why on earth do you use alphanumeric user ids? This is a telephone system after all.. telephones user numbers and except for Alcatel system phones, phones don't have a useable keyboard that makes typing a feasible option. Why not just use 800 as the user ID? Imagine.. you have an IP hardphone (e.g. the SPA942) which wants to call the phone on the phone on the FXS1 port. Now instead of simply typing 800 and be done with it, it has to resort to IP dialling and type: 333666 pause 666. Of course there's an internal phonebook and shortdials but the bottom line for me is to keep non digit characters out of telephony. After all, you can still control a caller will see on his screen by appropriately configuring the Display Name: in the FXS1/2 tab.

[solarflare]

Originally Posted by humba3

I have to ask, why on earth do you use alphanumeric user ids? This is a telephone system after all..

Some VoIP operators seem to like the alphanumeric user ids, so I just copied them. Now there is only numerical ones.

The aa menu asking for the extension works almost correctly. It rings the right phone now, but the voice still seems to be one way only. I'd expect to figure this one myself in the near future though.

I seem also to receive the incoming calls now in the FXS1 port, using the contact list "aa," and the extension number from the FXS1 User ID field. I've copied my old SPA-1001 dial plan L:20,S:7,( x | [x*][x*]. <:@sipbroker.com> | <#:>[x*][x*]. ) as both the FXS1 and Line1 dial plan, and under the SIP PBX Phone Parameters' Phone Dial Plan as well. But the problem is that I get thru with the 613 echo test number only when dialing out. It doesn't call out properly with a six digit FWD number.

Do I actually need all these different dial plans? Basically I'd be very happy for a default of calling out with FWD (at Line1) and otherwise the sipbroker ENUM gateway for now.