Possible to use OBi110 to as an FXO port on an Asterisk server?

[azrobert]

I've never setup a non-FreePBX Asterisk, but I was able to help someone with a plain Asterisk. I configure Asterisk a little differently than the above example and the person was able to convert the code to vanilla Asterisk.

Setup an Asterisk trunk without registration:
Trunk Name: OBi110
PEER Details:
type=peer
username=twinclouds
host=192.168.1.110
port=5060
canreinvite=no
insecure=invite,port
qualify=yes

Host = IP address of the OBi110
Port = port number of SP1

OBi110
Setup a dummy SIP definition on SP1 like this:
Service Providers -> ITSP Profile A -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile A -> SIP -> X_SpoofCallerid: Checked
Voice Services -> SP1 Service -> AuthUserName: anything
Voice Services -> SP1 Service -> X_RegisterEnable: unchecked
Voice Services -> SP1 Service -> X_ServProvProfile: A
For testing leave X_InboundCallRoute at default: ph

Now send a call to the OBi110 trunk. If the call reaches the OBi110, the phone port will ring. You don't need a phone attached to the phone port, the phone port LED will flash.
If this test is successful, change the X_InboundCallRoute to:
{twinclouds>(xx.):li}
Or just:
li

I route inbound PSTN calls to Asterisk with a Line InboundCallRoute:
sp1(1000@xx.xx.xx.xx:5060)

xx.xx.xx.xx:5060 is the IP address and port of Asterisk.

In Asterisk setup an inbound route with DID = 1000

[twinclouds]

@azrobert:
thank you very much for your quick response.  Really appreciate for your help.   I will give it a try and let you if it works.  I may bug you more if I have encountered any issues.
update:
Looks like everything works.  I made some changes from your suggestions.  Since everything was working for freepbx, I only changed the ip address from the box of freepbx to that of asterisk.  In asterisk, I added the section of trunk taking from the freepbx to the sip.conf.  It is essentially the same as yours except the username and password was their.  I have to use "host=dynamic" rather than to specify the obi110's ip address.
In the extensions.conf.  I added the dial-out statement for the obi trunk.  I have also added a dial in section as:

[obi-in]
exten => 8585551212,1,Dial(SIP/101&SIP/102&SIP/103&SIP/104,60,tr) ; phone must be registered
exten => 8585551212,2,Hangup

For the obi, the InboundCallRoute: should be sp2(8585551212), the same as for freepbx.

Thank you for your help.  Once done, it is actually pretty simple, but I will need to test more.

[MrMoxy]

I followed the guidelines here to use my OBi110 as an FXO port to connect my Comcast POTS line to FreePBX. The OBi110 is connected to Asterisk via SIP on SP2. (SP1 is not being used.) I can send and receive calls with no problem, EXCEPT that the OBi appears to be changing the caller ID. I have caller ID on my Comcast line, but the caller ID appears as the trunk name = AuthUserID after it goes through the OBi.

Service Providers --> ITSP Profile B SIP
 ProxyServerPort: 5060
 RegistrarProxyPort: 5060
 OutboundProxyPort: 5060
 X_SpoofCallerID:  Checked

Voice Services  --> SP2 Service
 X_InboundCallRoute: LI
 AuthUserName: ObiTrunk1
 AuthPassword: 123456

Physical Interfaces --> Line
 InboundCallRoute:  SP2(13035551234)
   (Note: I tried putting in just SP2 or SP2(13035551234;ui=$1) here but then the call does not route)
   RingDelay: 2000
        (3500 ms is too long for my system and the extension does not ring)

In my trunk I have
Trunk name: ObiTrunk1
CID options: Allow any CID
PEER details:
username=ObiTrunk1
secret=123456
host=(my OBi's IP)
type=friend
context=from-trunk
qualify=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw&g729

User context: 13035551234
User details:
secret=123456
type=user
context=from-trunk


When I dial into my Comcast line and it rings on the FreePBX extension, the caller ID shows up as ObiTrunk1 (the trunk name). Does anyone have any suggestions on how to get the callerID to come through?

[CALYTA]

Michigan,

Yes, actually I used your guides a  Tutuapp 9Apps ShowBox s a starting point .  However, I didn't like your method for several reasons.

1.  Your guide suggests using the phone number as the trunk name and SIP Username.  I found that doing so causes problems if you call in on another trunk and your CID is the trunk #.  You end up getting a disconnect recording.  Not sure why.  I found that if you change that from the phone # to OBITRUNK1, everything works as it should.

2.  Your guide includes parameters that don't seem to be essential making the thing work.  The user parameters, for example, are unnecessary.

3.  Your guides are awfully long.  I'm trying to condense the steps down into a very short guide.

4.  I didn't see that your guides allowed easily routing calls to and from both Google Voice and the line port.  As I recall, it was an either/or situation.  I'm working on coming up with something that allows you to do both, i.e. to route calls to the line port or Google Voice, based upon what number you dial.

As you'll see, what I've come up with fulfills that function.  You can dial 81 to get the line port, 85 to get Google Voice, or dial regularly and route according to some default rules.

9.   If you need more PSTN lines, you can setup more OBi devices like this; just assign a different AuthUserName (userid) to each SP1 interface, such as 1000, 1001, 1002, 1003, etc.

[CALYTA]

Michigan,

Yes, actually I used your guides as a starting point.  However, I didn't like your method for several reasons.

1.  Your guide suggests using the phone number as the trunk name and SIP Username.  I found that doing so causes problems if you call in on another trunk and your CID is the trunk #.  You end up getting a disconnect recording.  Not sure why.  I found that if you change that from the phone # to OBITRUNK1, everything works as it should.

2.  Your guide includes parameters that don't seem to be essential making the thing work.  The user parameters, for example, are unnecessary.

3.  Your guides are awfully long.  I'm trying to condense the steps down into a very short guide.

4.  I didn't see that your guides allowed easily routing calls to and from both Google Voice and the line port.  As I recall, it was an either/or situation.  I'm working on coming up with something that allows you to do both, i.e. to route calls to the line port or Google Voice, based upon what number you dial.

As you'll see, what I've come up with fulfills that function.  You can dial 81 to get the line port, 85 to get Google Voice, or dial regularly and route according to some default rules.

Then work on the call route to send calls in/out of SP1 directly to the FXO line (Li)

[sherylb]

Thank you so much. I really appreciate for your help.

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[Taoman]

delibelly is a SPAMMER.  Lots of invisible URLs in the quoted text.

As is sherylb.

I don't get it. If it's hidden so you can't see it and there's no hot link so you can't click on anything what's the point of doing this in the first place? What am I missing?

[SteveInWA]

delibelly is a SPAMMER.  Lots of invisible URLs in the quoted text.

As is sherylb.

I don't get it. If it's hidden so you can't see it and there's no hot link so you can't click on anything what's the point of doing this in the first place? What am I missing?

It does seem like a particularly stoopid way to spam.  You can hover your mouse cursor over the words in this sentence (in the original post) and see the links:  "Yes, actually I used your guides a To know more Smarttip Click Heres a starting point . "

[Taoman]

You can hover your mouse cursor over the words in this sentence (in the original post) and see the links:  "Yes, actually I used your guides a To know more Smarttip Click Heres a starting point . "

Hmmm. Didn't notice that.

I see sherylb came back and edited her/his/its post. If you move your cursor down in the post you will see horizontal lines that are now hot links. But it's invisible unless you move your cursor to that point. How many people are going to do that and then click on the link?

As you said..........stoooooopid!

[technologywell]

I've never setup a non-FreePBX Asterisk, but I was able to help someone with a plain Asterisk. I configure Asterisk a little differently than the above example and the person was able to convert the code to vanilla Asterisk.

Setup an Asterisk trunk without registration:
Trunk Name: OBi110
PEER Details:
type=peer
username=twinclouds
host=192.168.1.110
port=5060
canreinvite=no
insecure=invite,port
qualify=yes

Host = IP address of the OBi110
Port = port number of SP1

OBi110
Setup a dummy SIP definition on SP1 like this:
Service Providers -> ITSP Profile A -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile A -> SIP -> X_SpoofCallerid: Checked
Voice Services -> SP1 Service -> AuthUserName: anything
Voice Services -> SP1 Service -> X_RegisterEnable: unchecked
Voice Services -> SP1 Service -> X_ServProvProfile: A
For testing leave X_InboundCallRoute at default: ph

Now send a call to the OBi110 trunk. If the call reaches the OBi110, the phone port will ring. You don't need a phone attached to the phone port, the phone port LED will flash.
If this test is successful, change the X_InboundCallRoute to:
{twinclouds>(xx.):li}
Or just:
li

I route inbound PSTN calls to Asterisk with a Line InboundCallRoute:
sp1(1000@xx.xx.xx.xx:5060)

xx.xx.xx.xx:5060 is the IP address and port of Asterisk.

In Asterisk setup an inbound route with DID = 1000

ah thats great information. keep sharing such information!