Can SIP phone calls be terminated through Digium T1 PRI card


#1

The setup I have in mind is T1 coming in from phone company which will be connected to *. Asterisk will be connected to a hub/switch and SIP phones will connect to the hub/switch. The question I have is: Can I make calls out through the Zap channels that the T1’s are connected to through the sip phones?

Any help would be appreciated,
knight


#2

that is exactly how we are set up. works great.


#3

Will the extension for the sip phones go in the sip.conf or zapata.conf because it’s a sip phone but uses zap channels to dial out.


#4

SIP phones are configured in sip.conf. Zap lines are configured in zapata.conf. Extensions and dialplan rules in extensions.conf. This is exactly what Asterisk is designed for. :smile:


#5

connect the sip phones through a ‘hub/switch’

don’t even think about using a hub, make sure you use a switch.


#6

[quote=“p_lindheimer”]connect the sip phones through a ‘hub/switch’

don’t even think about using a hub, make sure you use a switch.[/quote]

Very true !
100 MBits, full duplex !

ANd yes, same setup here:

40-ish sip phones, a few faxes on ATAs and softmodem (IAX modem on hylafax), TE110 interface jumpered to E1 (Europe).

A charm… /drool

You might want to read EXSESSIVLY :smiling_imp: all documentation, wikis and forums.

Asterisk is a really GREAT pbx with endless possibilitys.
I sucked up every bit of infos for almost 3 month and still learn every single day about asterisk and some tricks !

Sip.conf:
The file holding your sip confguration (nomen est omen)

zapata.conf:
The config for your zap interface, your T1 card

Extensions:
The brain of asterisk: What to do how and wheren and where and why
The dialplan, the heart of all callprocessing.

You will curse, you will cry, you wish you were a fish only worrying about food…and never choosed to setup an asterisk box, you will miss sleep, you will get divorced: But at the end you will have one of the best and functionrich PBX out there… :wink: