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Author Topic: Possible to use OBi110 to as an FXO port on an Asterisk server?  (Read 908313 times)
azrobert
Hero Member & Beta Tester
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Posts: 3203


« Reply #20 on: June 09, 2015, 09:07:31 pm »

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
« Last Edit: June 09, 2015, 10:16:32 pm by azrobert » Logged
twinclouds
Jr. Member
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Posts: 28


« Reply #21 on: June 09, 2015, 09:51:26 pm »

@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.

« Last Edit: June 09, 2015, 11:28:24 pm by twinclouds » Logged
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