I would like to tie outbound calls from specific extensions to specific
zap channels...I have multiple clients in an executive suite and would
like to be able to tie lets say extension 1234 to Zap Channels 1 and 2
and extension 5678 to channels 3 and 4 and so on...
This so that their caller ID
YES YES YES
-Original Message-
From: Steve Murphy [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 19, 2006 2:32 AM
To: Asterisk User List
Subject: [Asterisk-Users] An FXO version of IAXy?
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like
Sounds great..thanks...
-Original Message-
From: Greg Oliver [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 04, 2006 8:25 PM
To: [EMAIL PROTECTED]; Asterisk User List
Subject: Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted
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 Firmwarethanks.
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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
Firmwarethanks.
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Anyone have a great reference for configuring the PAP2-NA with Asterisk?
-Original Message-
From: Ed Greenberg [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 19, 2006 11:57 PM
To: Michael J. Liberatore
Cc: Asterisk User List
Subject: Re: [Asterisk-Users] Fwd: Which ATA device do you
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 picks up the
phone to dial, I would like
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
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 [EMAIL PROTECTED] 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 a T1...do I need an external
CSU/DSU or
You can do a SIP debug and see what they are advertising as available
codecs...and that should prove that they are not requesting the correct
codec.
-Original Message-
From: Brent Torrenga [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 31, 2006 4:20 PM
To: Asterisk User List
Subject:
I use VMWare, but will start testing XEN...I use VMWare to slice up some
nice big servers to provide dedicated hosted PBXes. We also use the VMs
for easy deployment and is a vital part of our DR Plan...
Now, we are full VoIP...not T1 or PRI cards...
-Original Message-
From: Paul
Does anyone know how to change the defaults for Call Waiting in the
Asterisk database...? I am currently using AAH and I am running cron
job that updates this...but I would like to have Callwaiting on by
default...its set in the config for the cisco phones, but that does not
seem to request the
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 1 to 2
The phones use UDP Port 5061 for incoming connections (from Asterisks or
other SIP
Depends if you have reinvite on or off. On yes off no. At least that
is what I have read...you can verify with a network sniff on the
Asterisk server...use
tcpdump -ln host ip.off.asterisk.server and not tcp port ssh (telnet or
whatever protocol you are connecting to the astersisk server
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