NOOB help setting up PSTN/POTS

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NOOB help setting up PSTN/POTS

Postby dbs » Tue Sep 23, 2014 3:16 am

Hi Guys

I have installed Asterisknow 11.10.2 on a dedicated machine

I have got extensions working and can make calls.

But I can seem to work out how to setup the connection of the PSTN/POTS lines. I have Patton SN4114 boxs. Where or how do I get the asterisk box to see the Patton box's. It has me stumped as I cant see in any if the menu's how to do this. The answer is most likely obvious so I must be missing something.

I would preffer to do it via the GUI, but can CLI if needed


Am I running the correct version that has the DAHDI modules ?

The end system will be running 8 PSTN lines in line hunt that point to a ring group pilot with about 15 extensions.

This is more a learning system for me to get up to speed with Voip systems so its not for commercial use


I do have a 20yr background in Telephony and CCT switched PABX systems, so the Asterisk box is new and different to me

Any help would be much appreciated
dbs
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Re: NOOB help setting up PSTN/POTS

Postby dbs » Tue Sep 23, 2014 4:27 am

Ok...have done a bit more research and I think I need to set it up as a SIP trunk to the PAtton box

Would I be correct in assuming this ?
dbs
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Re: NOOB help setting up PSTN/POTS

Postby Lexus45 » Tue Sep 23, 2014 10:16 am

UPDATE: only now noticed that the topic is about Asterisk Now.
You'd better consult those who know AsteriskNow. I configured it a couple of times but don't have expert knowledge. As I know, there are additional configs, which could be edited by hand (not via GUI), their names end with _custom (extensions_custom.conf). Original configs should not be edited manually, the changes to them are applied through GUI. Anyway, consult AsteriskNow experts.


Hello.
I searched about Patton SN4114. So, it's a pstn/voip gateway with 4 FXO ports and 1 Ethernet.
In this case you have to configure a SIP trunk between Asterisk and Patton (though both support h323, I think you'd better use SIP).

Add to your sip.conf on the Asterisk system something like:

Code: Select all
[patton]
type=peer                               ; read about types in default sip.conf
directmedia=no                           ; make RTP go through Asterisk
host=<patton box ip address>
context=from-patton                   ; inbound calls go to this context
qualify=yes                          ; send OPTIONS packets to see peer's status


Then add the context 'from-patton' to your extensions.conf. Let's imagine that you have an internal device 'voipphone':
Code: Select all
[from-patton]
exten => _X.,1,NoOp( ### This is a call from Patton ### )
    same => n,Dial(SIP/voipphone)


And your 'voipphone' will ring if any extension was dialed from Patton gateway.


Assuming that 'voipphone' is in context 'internal', add to it (or create one) also in extensions.conf:

Code: Select all
[internal]
exten => 123456,1,NoOp(### imagine that it's a landline number in your city ###)
    same => n,Dial(SIP/patton/123456)
Lexus45
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