Newbie and home/home office setup

[itgeekbyday]

I hesitated posting here because I am sure I am going to get "this information is all here if you look" and I have no doubt that it is..however I have been reviewing for several days but there are mixed messages as well as part of what I want to do in one post and part of what I want to do in another post, etc. but combining all of them into what I want to do really confuses me due to lack of experiance and knowledge to read between the lines...so having said all that I am asking that you be gentle with me <ducking head>.

Firstly what I am running:

PBX in a flash purple (fresh install last night 12-17-10) with all the latest google fixes, etc. Followed this:
http://nerdvittles.com/?p=712

DID's etc:
One single google voice account
One single PSTN (AT&T) basic phone service (No LD service)

Hardware:
One Aastra 6739i
One Linksys SPA-3102
Analog phones elsewhere in the house (connected to PSTN)

Understanding and use so far:

I have been using PBX in a flash purple and google voice since the release of purple (I have been through all the working and not working situations of the google voice issues) and have implemented all the fixes along the way. I did a clean install last night just so there is no confusion on my end. I have been using it in this capacity for the last month or so as my main buisness line for incoming and outgoing calls. In other words everything coming from and going to the 6739i via google voice (no analog line or other voice line). I have also integrated the SPA-3102 as a analog interface in order to use a polycom soundstation as a conference phone.

Description of the solution I am trying to achieve:

I want to continue to use a seperate phone line (my main google V number) for my personal buisness and route this to the 6739i on line 1. I also want to integrate the AT&T number (basic PSTN analog phone service) and route this to ring all the analog phones in the house (via the SPA-3102) but also ring Line 2 on the 6739i. That is the incoming side.

For outgoing I want to route local calls (within my area code and 800 service) out the local PSTN and ALL long distance for all phones out google V. I want to make sure NO long distance ever goes out local PSTN due to charges incurred. I also obviously want 911 to route out the PSTN with my main AT&T number as the caller ID. 911 should route this way and show this number no matter the phone used.

whew! sorry long post. ok so I would say I have the experiance on the SPA3102 to make the basics work as well as some fundamental understanding of freePBX, etc but when I try to get my head around all my configuration needed I get overwhelmed. I also start implementing the different aspects of what is needed and since it is not a complete installation (pieces missing) it is hard for me to understand if I am moving forward or backward. What i would love is for someone to have this type of installation done and documentated so that I could follow that procedure but I find that unlikely, so what I would be happy with is someone pointing out the different aspects of the config's needed or pointing me in the right direction on each of the different pieces.

Thanks in advance!
M

 

[itgeekbyday]

Forgot to mention

I also would like different IVR's for each line..so when someone calls on my google line..they get a business specific message and when they call on the PSTN (home number) they get a different IVR or home message

 

[Linetux]

First, if I'm reading your request correctly, you're going to want 3 extentions in PIAF. We'll call them 101, 102 and 103 for simplicity.

I want to continue to use a seperate phone line (my main google V number) for my personal buisness and route this to the 6739i on line 1. I also want to integrate the AT&T number (basic PSTN analog phone service) and route this to ring all the analog phones in the house (via the SPA-3102) but also ring Line 2 on the 6739i. That is the incoming side.

So 101 and 102 will go onto your 6739i. Line 1 will register as 101, line 2 will register as 102. Your SPA-3102 will be 103.

You'll use a ring group for your incoming AT&T number, which will then route the call to 102 and 103.

For outgoing I want to route local calls (within my area code and 800 service) out the local PSTN and ALL long distance for all phones out google V. I want to make sure NO long distance ever goes out local PSTN due to charges incurred. I also obviously want 911 to route out the PSTN with my main AT&T number as the caller ID. 911 should route this way and show this number no matter the phone used.

For this, it's a simple route. You'd just set the dial rules to only allow L/D calls to route out of the GV.

911 will route out your PSTN, as well as any local calls (if you want that to happen).

If you're unsure of what I've just said, ask for clarity and you'll get it, I'm sure

 

[MisterQ]

You may also want to consider an adaptation for the Long Distance call barring. If your Voice Provider (Google) is having problems, you might want the ability to fail back to the PSTN for that. Perhaps with a voiceover to advise in the process.

 

[itgeekbyday]

First, if I'm reading your request correctly, you're going to want 3 extentions in PIAF. We'll call them 101, 102 and 103 for simplicity.

So 101 and 102 will go onto your 6739i. Line 1 will register as 101, line 2 will register as 102. Your SPA-3102 will be 103.

Ok that is good info and I understand how to create the diff extensions however I am now not sure how to route the second line to the 6739i

For this, it's a simple route. You'd just set the dial rules to only allow L/D calls to route out of the GV.

What do those dial rules look like and where do they go exactly?

911 will route out your PSTN, as well as any local calls (if you want that to happen).

That is exactly what I want but again not sure how to do that exactly

If you're unsure of what I've just said, ask for clarity and you'll get it, I'm sure

I did mention I was a noob with respect to PBX's so I will have lots of questions I am sure before this is working for me>

 

[itgeekbyday]

You may also want to consider an adaptation for the Long Distance call barring. If your Voice Provider (Google) is having problems, you might want the ability to fail back to the PSTN for that. Perhaps with a voiceover to advise in the process.

Good idea. However for the home side if google was down we just would not have LD service and we have cell phones for backup. for the business side I also have a cell phone as well as other avenues for communication. So I am willing to work around being down, etc if needed.

 

[kh40s]

I also obviously want 911 to route out the PSTN with my main AT&T number as the caller ID. 911 should route this way and show this number no matter the phone used.

All you have to remember is that outbound routes are processed in top down order. So you would start with the most specific routes first, leaving the more general, wildcarded routes at the bottom. For example, you could safely put the 911 route first. The dial pattern would be a simple "911". So your first outbound route might look something like the following, assuming your area code is 202 (the last pattern will get local numbers dialed without an area code)

911
1800XXXXXXX
1202XXXXXXX
NXXXXXX

while your more general (and long distance) routes would be the last to be defined and would be wildcarded

011.

caveat: I'm not in the US, so someone else may have better and more specific US dialing patterns they can share with you.

 

[Linetux]

Ok that is good info and I understand how to create the diff extensions however I am now not sure how to route the second line to the 6739i

How were you planning on putting "Line 1" on the 6739? I'm not sure how much you know about it, but basically it's either editing the config file (for automatic configurations) or using the built-in web server in the phone to adjust the settings for "Line 2".

 

[itgeekbyday]

No PSTN incoming calls

I am stuck..I think the problem is either in my SPA3102 config or my inbound rules in PIAF? I can place calls from an analog phone connected to PIAF via the SPA3102 and those go out the GV interface however when I call the analog number (not the GV number) it rings and rings. Not sure which are to look?

 

[itgeekbyday]

How were you planning on putting "Line 1" on the 6739? I'm not sure how much you know about it, but basically it's either editing the config file (for automatic configurations) or using the built-in web server in the phone to adjust the settings for "Line 2".

Line 1 is already working and tied to GV. I used the XML files from aastra and followed the document on how to implement in PIAF to get the phone working so it is all automatically configed via t*f*t*p. I will look at the config files.

 

[vespaman]

where are you sending inbound calls on line 2 ? Set up your IVR and send them to that first. When thats working, in the IVR have it time out to your aastra phone

 

[itgeekbyday]

Maybe start simple?

Ok I am overwhelmed at this point. too many areas to look at and not sure which area is the problem. So..what I would like to do is have some help with respect to setting up a very simple PSTN call in to PIAF (single analog phone) and back out.

So in essence call order: (in) PSTN > SPA3102 > PIAF > Analog phone....(out) Analog phone > PIAF > SPA3102 > PSTN

In order to do this I would like some help in the following

1. A way to temporarily disable GV (already working) so it is not involved in the calls (hard to troubleshoot analog when this is working)
2. A copy of someone's SPA3102 config that is already doing this? Or at the minimum some help in the config between PIAF and SPA3102
3. Specifically the areas in PIAF that are involved. In other words: trunk help, extension help (specific SPA3102), dialing rules in/out, etc.

** I really appreciate all the help and comments so far and some of them have led me to the point I am so far with respect to a partial working system however like I said I get confused as to where to look or what to change **

 

[Astrosmurfer]

1. Go to the Google Voice trunk configuration screen under Trunks in FreePBX. You will find a disable trunk check box.

2. Below is a copy of the config I posted on the trixbox forum.

This example sets up the analog phone, plugged into the FXS port on the SPA-3102, as Asterisk extension 2000. When the power fails, that analog phone/extension 200 changes to the POTS line. The SPA-3201 bridges the FXO and FXS ports on power fail. This is an excellent feature of the 3102!

Asterisk SIP extension 2000 Should look like this:

2000
secret: mypassword
dtmfmode: rfc2833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: no
port: 5060
qualify: yes
dial: SIP/2000

The Line1 configuration on the SPA3102 should look something like this:

Line Enable: yes
SIP Port: 5060
Proxy: trixboxIP
Register: yes
Make Call without Registry: no
Ans Call Without Registry: no
Display Name: 2000
User ID: 2000
Password: mypassword
User Auth ID: no
Preferred Codec: g711u
Use Pref Codec Only: no
DTMF Tx Method: Auto
Dialplan: (2xxx|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx<:@gw0>)
Enable IP Dialing: Yes

The PSTN Line configuration of the SPA3102 should look like this:

Line Enable: yes
NAT Mapping Enable: no
SIP Port: 5061
Proxy: trixboxIP
Use Proxy: yes
User ID: 5555
Password: mypassword55  ;Must match trixbox trunk config.
Register: no
Make Calls without Reg: yes
Preferred Codec: G711u
Use Pref Codec Only: no
dial plans 1 through 7: xx.
dial plan 8: (<S0:5555>)
PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: yes
PSTN CID For VoIP CID: yes
VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: Yes

VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID: 5555
VoIP User 1 Password: mypassword55

Trunk SPA3102 Asterisk configuration should look like this:

Trunk Name: SPA3102
type=peer
auth=md5
host=SPA3102IP
port=5061
secret=mypassword55
username=5555
dtmfmode=rfc2833
insecure=very

User Context: 5555
type=user
context=from-trunk-sip-SPA3102
nat=no
username=5555
secret=mypassword55
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

 

[itgeekbyday]

1. Go to the Google Voice trunk configuration screen under Trunks in FreePBX. You will find a disable trunk check box.

2. Below is a copy of the config I posted on the trixbox forum.

This example sets up the analog phone, plugged into the FXS port on the SPA-3102, as Asterisk extension 2000. When the power fails, that analog phone/extension 200 changes to the POTS line. The SPA-3201 bridges the FXO and FXS ports on power fail. This is an excellent feature of the 3102!

Asterisk SIP extension 2000 Should look like this:

2000
secret: mypassword
dtmfmode: rfc2833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: no
port: 5060
qualify: yes
dial: SIP/2000

The Line1 configuration on the SPA3102 should look something like this:

Line Enable: yes
SIP Port: 5060
Proxy: trixboxIP
Register: yes
Make Call without Registry: no
Ans Call Without Registry: no
Display Name: 2000
User ID: 2000
Password: mypassword
User Auth ID: no
Preferred Codec: g711u
Use Pref Codec Only: no
DTMF Tx Method: Auto
Dialplan: (2xxx|xx.|*xx.|**xx.|<#,:>xx.<:@gw0>|<#,:>*xx<:@gw0>)
Enable IP Dialing: Yes

The PSTN Line configuration of the SPA3102 should look like this:

Line Enable: yes
NAT Mapping Enable: no
SIP Port: 5061
Proxy: trixboxIP
Use Proxy: yes
User ID: 5555
Password: mypassword55  ;Must match trixbox trunk config.
Register: no
Make Calls without Reg: yes
Preferred Codec: G711u
Use Pref Codec Only: no
dial plans 1 through 7: xx.
dial plan 8: (<S0:5555>)
PSTN-To-VoIP Gateway Enable: yes
PSTN Ring Thru Line 1: yes
PSTN CID For VoIP CID: yes

Trunk SPA3102 Asterisk configuration should look like this:

Trunk Name: SPA3102
type=peer
auth=md5
host=SPA3102IP
port=5061
secret=mypassword55
username=5555
dtmfmode=rfc2833
insecure=very
 
User Context: 5555
type=user
context=from-trunk-sip-SPA3102
nat=no
username=5555
secret=mypassword55
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

You Rock! A couple questions:

"context: from-internal"
"context=from-trunk-sip-SPA3102"

Are these custom contexts that need to be created in asterisk in the "custom contexts" menu?

If so any other commands to screens that would need to also reference these custom contexts?

 

[Astrosmurfer]

There is NO need to create any contexts.

The from-internal context already exists and is the default container of all internal extensions.

The from-trunk-sip-SPA3102 is created by FreePBX when you create and name the trunk. The context name is based on the trunk name. So, if you named your trunk cheese, rather than SPA3102 FreePBX would create a context called from-trunk-sip-cheese and you would use that as your context in the trunk's User section.

Note: Others may well present different trunk configs. I know that there are many other ways to do the trunk config, including without a User section at all. But, this is an established working config. As requested.

 

[itgeekbyday]

There is NO need to create any contexts.

The from-internal context already exists and is the default container of all internal extensions.

The from-trunk-sip-SPA3102 is created by FreePBX when you create and name the trunk. The context name is based on the trunk name. So, if you named your trunk cheese, rather than SPA3102 FreePBX would create a context called from-trunk-sip-cheese and you would use that as your context in the trunk's User section.

Note: Others may well present different trunk configs. I know that there are many other ways to do the trunk config, including without a User section at all. But, this is an established working config. As requested.

Ok..thank you for that explanation.

Currently what I have is all trunks are disabled except the new SPA3102 trunk that I created. If I dial out I get "all circuits are busy" and if I dial in I get a ring and then a dial tone. I can then dial a number and it says "thank you for calling..please hold while we find someoone to take your call"

 

[Astrosmurfer]

Did you create an outbound route that goes to the SPA3102 trunk?

The dial in problem says that your 3102 is not properly configured. Recheck.

 

[itgeekbyday]

Did you create an outbound route that goes to the SPA3102 trunk?

The dial in problem says that your 3102 is not properly configured. Recheck.

ok..set the SPA to factory defaults and setup again per your instructions...now calls come in fine!! I did forget to create an outbound route. I have now created that and assigned emergeny and local seven digits..One other issue..if I dial out I get a busy signal. I dont want anything going out this route but local calls within my area code and 911.

 

[itgeekbyday]

For 911 define your first outbound route with CID rule of 911 and point to your analog trunk only.

Create a second outbound route for local calls. You need this tool which will allow you to set rules to catch all of your local numbers. You can even structure the rules to allow 7 digit dialing. Set the order of your trunks as analog first then voip (or just analog if you wish, but you will only be able to make one local call at a time)

Create additional outbound routes for long distance, etc

Lorne

I dont think it is a "route" issue..unless I am wrong, and I prob am! But my thinking is if it was a route issue that I would be able to manually do the route? In other words if a local number needed to prepend a 1 or an area code, I should be able to dial 1+area+number and the call would go through instead of trying to dial a local 7 digit? Currently 7 digit dial gets a busy signal and either 1 + area + number or aree + number gets "all circuits busy"

 

[Astrosmurfer]

I'd recommend routes as lgaetz describes but, let's get it working for now. Once you have proven your trunk and device configuration, then you can diddle with call routing and trunk selection.

The busy signal could mean one of three things.

1. No matching route.
Create a top level outbound route with 9 as a selector digit and a catch all dial pattern as below. Set it with only the SPA3102 trunk.

9|.

This dial pattern matches any dialed number starting with a 9 and it strips the leading 9 from the dial digits that it sends on to the trunk. The '.' matches any and all numbers. Make sure that you dial 9, to select the route, and then your local number. You will replace this route with something more specific after you have confirmed that dialing in and out work as expected.

2. No Asterisk to SPA3102 connection.
Check the Asterisk logs for clues about this. Rejected connections are likely due to a user/password mismatch. Triple check the config of the trunk and SPA3102.

3. Improper dialed digits.
The above route should eliminate this as a possibility. But again, the Asterisk log will tell you what digits are being sent.