Displaying 5 results from an estimated 5 matches for "steiltech".
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sailtech
2006 Feb 17
1
Outbound ZAP Dialing
I have server with a total of 6 Analog ports...using TDM04B and TDM02B.
I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have
worked through getting the DIDs to work and route to the
extensions...now what I need to do is when Extension 1111 picks up the
phone to dial, I would like them to use their DID analog line first,
unless someone has called in on it and they are trying to
2006 Feb 02
0
POTS lines vs. using T1 to connectphoneservices?? HELP
...honeservices?? HELP
A fractional T1 is what I would suggest and it is easy to setup and
configure. You should only need to plug in the T1 line directly into
the T1 Card on the server. The provider will supply the equipment to
terminate the line on your premises.
On 2/2/06, Kevin Steil <kevin@steiltech.com> wrote:
> Need help...I need to install a card to terminate 7 lines...I
> have not order the phone lines yet...I can either do analog lines 1FBs
> or order a fractional T1...any suggestions on what hardware would be
> easier to install and configure...also if I went with...
2006 Mar 03
9
Program Buttons on Cisco 79xx Phones
Does anyone have a good resource to learn how to program the soft and
hard buttons on a Cisco 7940 or 7960 phone? Using SIP
Firmware...thanks.
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2006 Feb 01
1
RE: Asterisk-Users Digest, Vol 19, Issue 10
Need help...I need to install a card to terminate 7 lines...I have not
order the phone lines yet...I can either do analog lines 1FBs or order a
fractional T1...any suggestions on what hardware would be easier to
install and configure...also if I went with a T1...do I need an external
CSU/DSU or anything or does it just plug into the T1 card...thanks..
-----Original Message-----
From:
2006 Jan 12
0
How to register a SIP phone on Asterisk behind NAT
I currently do this for about 30 different cisco 79xx's connecting to
some hosted Asterisk servers.
Asterisk listens by default for any SIP connection on UDP port 5060.
And will use RTP UDP port 10000 to 20000
The phones use UDP Port 5061 for incoming connections (from Asterisks or
other SIP Devices) and use for RTP, UDP port 10000 to 20000.
Now, if you are going to have the two remote