On the 3102 you have to arrange for it to call a given 'user' via the sipx 
proxy.
   - Under PSTN Line, PSTN Caller Default DP ==> 2
   - Under PSTN Line dial plan 2 ==> S0<:201> , which roughly means 'wait 0 
seconds for dialing to end, then substitute nothing (the empty string before 
the :, which is what you get from the PSTN) with '201' the SIP username
   - Set up the sip proxy as your sipx installation

On the sipx side:
   - Create a user '201' (if you're manually setting up your telephones click 
on advanced settings and get the password for that user)
   - On the telephone you want to ring on inbound calls set it up to register 
with sipxecs as user 201
   - Alternately you can set up a hunt group (from Features / Hunt Groups) with 
an extension number and program the 3102 to call into that number, then you can 
redirect inbound calls to other extensions using the hunt group rules (a bit 
more flexible)

I know I sound like a broken record, but the 3102 is just dead broken and may 
not work right with some phones or sipxecs in some scenarios. What you're 
describing though isn't one of them, I think all you need to do is change 
S0<:0> to S0<:201> and register a phone as user 201. You can use tcpdump or 
wireshark to look at the INVITE message from the 3102 and see what its really 
putting out, there should be a 'To:' header with [email protected] in it.

-Eric

On May 5, 2010, at 8:39 AM, Dave C. Lake wrote:

> Hi Eric,
>  
> I’ve been trying what you said but don’t seem to understand. I think if you 
> could do the step by step guide it would be apprieciated. I get the calls 
> coming to AA but not to any other extension.
>  
> Regards,
>  
> Dave
>  
>  
> <image001.jpg>
>  
> Innovative Business Solutions
> P.O. Box GR5102
> Unit 11, Bois D' Orange Commercial  Complex,
> Gros Islet, St. Lucia
> Phone:   (758) 4504427
> Fax:       (758) 4582759
> Mobile: (758) 7248839
> Email:    [email protected]
> Web:      www.ibsstlucia.com
>  
> From: Eric Varsanyi [mailto:[email protected]] 
> Sent: Tuesday, May 04, 2010 7:32 PM
> To: Dave C. Lake
> Subject: Re: SPA-3000 as PSTN Gateway
>  
> base href="x-msg://78/">
> You don't have to put the extension into the dialing plan. You create a user 
> with a given extension number (the username == the extension number), then 
> program the appearance buttons on your phones (or the 'account' setup on your 
> softphones, etc)  to register as that user (using the generated SIP password 
> you get when you create the user). Any devices that are registered for a 
> given user should ring when the 3102 tries to INVITE via the main sipx proxy.
>  
> You can make the FXS side of the 3102 register as some user, so if you want 
> the FXS on some other 3102 (or even the same one) to ring you have to program 
> it to register as the user (ext) that you created in sipxecs.
>  
> The dialing plan is needed to make outbound calls through the FXO side of the 
> 3102 (ie: if someone dials '9' route the call to the 3102's unamanged 
> gateway).
>  
> If this doesn't make sense I can try to make a step-by-step example to 
> illustrate the concepts.
>  
> -Eric
>  
> On May 4, 2010, at 5:45 PM, Dave C. Lake wrote:
> 
> 
> Thanks,
>  
> I’ve tried putting another extension in the dialing plan but it doesn’t seem 
> to work. It just keeps ringing. Have you ever experienced this?
>  
> Dave
>  
>  
> <image001.jpg>
>  
> Innovative Business Solutions
> P.O. Box GR5102
> Unit 11, Bois D' Orange Commercial  Complex,
> Gros Islet, St. Lucia
> Phone:   (758) 4504427
> Fax:       (758) 4582759
> Mobile: (758) 7248839
> Email:    [email protected]
> Web:      www.ibsstlucia.com
>  
> From: Eric Varsanyi [mailto:[email protected]] 
> Sent: Tuesday, May 04, 2010 6:35 PM
> To: Dave C. Lake
> Subject: Re: SPA-3000 as PSTN Gateway
>  
> base href="x-msg://57/">
> In dial plan entry 2 is a '0' -- that's what the incoming calls go to, so if 
> you want it to go elsewhere use another extension there.
>  
> To add a gateway, go to devices / gateways and in the pulldown add an 
> unmanaged gateway. If you need to use other than port 5060 (I don't recall if 
> the 3102 uses 5061 on the FXO side) click on advanced settings and the port 
> number is there. Here's what one of mine looks like for example (this is to a 
> patton, I returned my 3102's after I discovered how broken they were):
>  
> <image002.png>
>  
> On May 4, 2010, at 5:26 PM, Dave C. Lake wrote:
> 
> 
> 
> Hi Eric,
>  
> Thanks for your help. I have been playing around with your config and have 
> gotten incoming calls to the auto-attendant working but have a few questions. 
> Is there a way to get the calls terminating somewhere other than the AA? How 
> exactly do I set up the unmanaged gateway? Is just the IP address and the sip 
> port sufficient?
>  
> Regards
>  
>  
> <image001.jpg>
>  
> Innovative Business Solutions
> P.O. Box GR5102
> Unit 11, Bois D' Orange Commercial  Complex,
> Gros Islet, St. Lucia
> Phone:   (758) 4504427
> Fax:       (758) 4582759
> Mobile: (758) 7248839
> Email:    [email protected]
> Web:      www.ibsstlucia.com
>  
> From: Eric Varsanyi [mailto:[email protected]] 
> Sent: Tuesday, May 04, 2010 12:19 PM
> To: Dave C. Lake
> Cc: [email protected] users
> Subject: Re: SPA-3000 as PSTN Gateway
>  
> For the 4 corners:
>    FXO, inbound calls (telco->sip): set up the routing in the 3102 to dial 0 
> to go to the AA (if you let north american style callerid (with spaces) 
> through sipx will drop your calls due to the bad headers from the 3102)
>    FXO, outbound calls (sip->telco): created an unmanaged gateway pointing at 
> the 3102  and modified the existing dialplan rules for local and 911 calls to 
> use it
>    FXS, inbound calls (sip->phone): I created a user ("205") and programmed 
> in the resulting sip password in the configuration so the 3102 would register 
> for inbound calls
>    FXS, outbound calls (phone->sip): insure that user 205 has permission to 
> make LD calls outbound.
>  
> -Eric
>  
> On May 4, 2010, at 11:09 AM, Dave C. Lake wrote:
>  
> > Hi Eric,
> > 
> > Thanks for your quick response. Just a few questions though. Is there 
> > any configuration to be done on the SipXecs side of things. I notice 
> > you have the PSTN line as 0 where is this setup. How does the dialplan work.
> > 
> > Thanks
> > 
> > Dave
> > 
> > 
> > 
> > 
> > Innovative Business Solutions
> > P.O. Box GR5102
> > Unit 11, Bois D' Orange Commercial  Complex, Gros Islet, St. Lucia
> > Phone:   (758) ( 4504427
> > Fax:       (758) ( 4582759
> > Mobile: (758) ( 7248839
> > Email:    [email protected]
> > Web:      www.ibsstlucia.com
> > 
> > -----Original Message-----
> > From: Eric Varsanyi [mailto:[email protected]]
> > Sent: Tuesday, May 04, 2010 11:41 AM
> > To: [email protected]
> > Subject: Re: SPA-3000 as PSTN Gateway
> > 
> > Dump of a "working" config below.
> > 
> > The Spa3102 has many problems that no config will fix, it doesn't 
> > implement the SIP standards properly and Cisco has abandoned it 
> > support wise (the newest firmware is still broken and is quite old).
> > 
> > The FXS side works more or less, though DTMF may not work depending on 
> > your ITSP. The FXO side kind of works as long as you don't allow it to 
> > import caller ID from calls (it corrupts the From: header, though I 
> > wrote a patch to sipxecs to work around the corruption), there was 
> > also a problem with REFER that did something bad to the AA.
> > 
> > -Eric
> > 
> > 
> > On May 4, 2010, at 10:32 AM, [email protected] wrote:
> > 
> >> Hi Eric,
> >> 
> >> I saw your post in the forum where you had setup an SPA -3102 as a 
> >> PSTN
> > gateway. would you happen to have the configuration for this. I am 
> > attempting to do something similar.
> >> 
> >> Thanks
> >> 
> >> Dave Lake
> > 
> > 
> > 
>  
>  

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