Can SIP phone calls be terminated through Digium T1 PRI card

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,

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

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.

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:

connect the sip phones through a ‘hub/switch’

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

[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 !

The file holding your sip confguration (nomen est omen)

The config for your zap interface, your T1 card

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: