[mberlant]

I would not be shy about playing with the various voltage settings. These are settings that affect only the voltages at which the various sensors engage and disengage, not any true input sensitivity. The one I would adjust (downward, probably) is the Tip/Ring voltage adjust.

I will do some measurements today and follow up with some specific recommendations.

Michael

p.s. It's encouraging (sort of) that you got the same result attached to the PBX.

[mberlant]

I have measured Tip/Ring Voltage on NTT as nominally -48.6vdc and Line-in-Use as -9.5vdc. I would recommend setting Tip/Ring Voltage to 45v and Line-in-Use to 12v and see if that works.

I hope these numbers help you toward success.

Michael

[bernart]

PROBLEM: Does not work when trying to make call from * via FXO where the * and the SPA are not on the same lan segments and using outbound proxy to get around blocked 5060 ports

Here is comfiguration.

1. Configured SPA to communicate via FXO in both directions
2. Port 5060 is blocked by ISP
3. SPA is behind NAT
4. * is not behind NAT
5. Using Outbound proxy (Fwdnat.pulver.com:5082) to compensate for
blocked ports

Result

1. line 1 works well rcv and make calls
2. PSTN works in only 1 direction, receives calls from PSTN and onpasses it to * for processing.
3. PROBLEM: Does not work when trying to make call out from * via FXO.

basically :
exten => _1212.,1,Dial(SIP/${EXTEN}@outbound proxyort)

nor

exten => _1212.,1,Dial(SIP/${EXTEN}@spa3k addressort)

does not work.

Couple of questions

1. When using an outbound proxy what is the correct address that belongs in the dial string? Is it the outbound proxy address, the address for the device, or something else.

2. If the typical 5060 and 5062 are blocked then if you use the device's address, then when it registeres via the outbound proxy with the proxy's address, then how does the SPA find the * when it was not contacted via its outbound proxy name.

[mikey2006]

I'm using the configuration that phoneboy described above, and can make out going calls via Asterisk to the PSTN line on the SPA3000 without a problem.

However when I try to get an incoming call from the SPA3000, to the Asterisk I get the following error in asterisk:
May 16 15:59:23 NOTICE[18588]: chan_sip.c:8578 handle_request_invite: Failed to authenticate user My Incoming CID <sip:2222222222@MYVOIPSERVER>;tag=b542fda9909e61ea o1

If I disable PSTN CID for Voip CID, then the calls will go through from PSTN to the Asterisk extention they are suppose to, any idea what I'm missing here

[PhoneBoy]

Sounds like you need to add an insecure=very to the appropriate place in sip.conf on the Asterisk server.

[arielb27]

Argh I have been trying to get my spa3000 working with asterisk@home and there Amp interface. I am able without any problems in getting line 1 which is just like any other setup for asterisk and spa 2000 type. But no matter what I do for following all the steps here I can't get the pstn side to be used at all. I am using the latest firmware 3.1.3(GWa)

I even tried the setup in the GeekGazzett. Which looks like they spend more time getting it working but still no go.

[masonc]

I have several SPA3k installed and working with Asterisk servers, I used the wizard and it works very well. However, for one installation where I have added a line that was previously just a seperate apartment with an analog phone, I would like to set up the dial plan so that a call placed from that extension, 1005, calls out to the pstn line without going through Asterisk, if the number is 911|XXXXXXX, but other numbers go to Asterisk. I don't see why I want to run local calls through the pbx just to have Asterisk send them back to the PSTN line.
In other words, for some dial strings, I want to bypass Asterisk. How would I do that?

[mberlant]

You need to configure this functionality into your SPA3000's Line 1 Dial Plan. Look at the appropriate threads here to see how to use the @gw0 suffix to send what you want out the PSTN Line port.

[masonc]

([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx< :@gw0>)
The dialplan constructed by the wizard seems to send everything to gw0. However, everything from line1 goes to Asterisk. I'm confused.

[mberlant]

This Dial Plan sends only x11 calls and calls prefixed with # to gw0. Everything else goes to Asterisk. In order to send local calls out via PSTN you will need to remove some ambiguities. I must assume that all Asterisk extensions are in the ranges 10XX and 11XX, so that they do not encroach on the North American Numbering Plan. I must also assume that all 7 digit calls are truly local, or that the tenant is smart enough to dial toll calls within the same Area Code as an 11 digit call. Based upon these assumptions I recommend this:

I also don't happen to believe that Skype has superior function to SIP. I value the seamless interoperability with the PSTN and the way a telephone connected to an ATA behaves just like it would if it were connected to the PSTN. And, apparently, there are many people (especially in Texas) who are so at ease with the telephone-connected-to-ATA solution that they have coerced the FCC into mandating E911 support for users in the US (even those of us who neither want the feature nor want to pay for it). That's pretty strong evidence in my book.

I also don't happen to believe that Skype has superior function to SIP. I value the seamless interoperability with the PSTN and the way a telephone connected to an ATA behaves just like it would if it were connected to the PSTN. And, apparently, there are many people (especially in Texas) who are so at ease with the telephone-connected-to-ATA solution that they have coerced the FCC into mandating E911 support for users in the US (even those of us who neither want the feature nor want to pay for it). That's pretty strong evidence in my book.

([2-79]11S0<:@gw0>|1800[2-9]xxxxxxS0<:@gw0>|1888[2-9]xxxxxxS0<:@gw0>| 1877[2-9]xxxxxxS0<:@gw0>|1866[2-9]xxxxxxS0<:@gw0>|[2-9]xxxxxxS0<:@gw0>|*xx.|**xx.| 1[01]xxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]xxxxxx.|<#,:>xx.<:@gw0>|<#,:>*xx<:@gw0>)

With this Dial Plan x11 calls, toll free calls and 7 digit local calls go via gw0. Asterisk extensions, * and ** codes, 11 digit North American calls and international calls go via Asterisk. Beyond this, the # prefix forces a call to go via gw0.

Good luck.