Displaying 20 results from an estimated 35 matches for "clwade".
2004 Jul 16
6
Asterisk + NEC Electra Elite IPK Integration
Hi,
I'm am currently in the process of trying to integrate an * box with an
NEC Electra Elite IPK.
Currently, we have 7 POTS lines coming into our building. These lines
are plugged into our NEC using the appropriate analog line interface
card from NEC. The NEC effectively has NO configuration done to it,
other than to make all the internal phones ring when a call comes in.
We also
2004 Aug 31
3
Cisco 79XX SIP Ring Tones
Hi all,
Has anyone gotten custom ring tones to work using ALERT_INFO with the
Cisco 7940 SIP phone? I've read the wiki, but just can't get this to
work. I'm currently using the 7.2 SIP image.
Thanks,
Chris
2004 Oct 07
1
'set debug' problems
...ump absolutely everything to one file, this
also affects the verbose output to that file as well.
Any help would be appreciated.
Thanks,
Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Dec 01
1
[OT] [slightly] app lever vs driver level implementation...
...rgive me if i've been posting incomplete thoughts lately, my various
project deadlines have been getting to me, i'm mentally exhausted.]
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Dec 06
5
two questions
Hi people,
question one
i see that asterisk is now in 1.x release. having tried it in the past
i want to know if i can use a voice modem as an outgoing line.
i know in the past that was not possible/supported so im just asking
in case the option is now available.
question two
im planing to use asterisk as a pure voip solution with sip phones and
h323 phones no need for digium/dialogic hardware
2004 Nov 11
6
cisco poe
...r from the 99.999% I just stated, but
it also seems to be the only low-end solution for poe. Am I right, or
just plain blind?
Thanks,
Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Aug 23
2
Cisco 7940 Question
Hi all,
I know this is a stupid question, but it is one I've been trying answer
for quite some time. Exactly how many simultaneous calls can the Cisco
7940 have, considering you can be talking to one, and have XXX others on
hold? Using SIP, is XXX only 1? I've found documents in various places
indicating different values in regard to the max number of calls the
phone can handle.
2004 Nov 22
0
new application swait...
...njoy,
Chris
PS. my C is very rusty, almost gone even, so I give no warranty that
this will not crash your entire system. Use at your own risk.
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Nov 30
0
park app vs. extension 700
...deas. (writing this just before going home for day, so it'll be
tomorrow before I can respond with further details if needed)
Thanks,
Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Dec 07
1
Ringing multiline phone
Is there a way to ring selective line on multi-line phone.
For example if I'm on the phone talking internally on line 1 and the
calls comes-in the line 2 will automatically ring.
The phone P104 allow extension to be assign each line.
Is there a way to call certain line (example line 3) on multi-line phone
instead of line 1 when the phone is not busy?
For example the Sip phone P104 has
2004 Dec 07
1
Inoming caller id withheld, move to new context, possible?
Hi,
now I've got caller id working on my BT line in the UK, I'd like to
play a different
message to those pesky sort who with hold their outgoing number.
How can I do this in my extensions.conf for my
[incoming-analog]
context?
I realise some people may call who are unable to change the way that
their system
withholds the outbound number, so I'll give them chance to leave a voice
2004 Dec 13
2
Incoming Toll-Free
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Anyone know of a service provider offering this?
-Mark
707-735-1038
2005 Jan 14
1
gotoiftime - different hours
If I have different opening hours on different days, can I accomodate
that in a single gotoiftime, or will I need to filter them out one by one ?
For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00
can this be done something like
GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or
something like that, or do I have to do:
2005 Feb 16
0
How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
> -----Original Message-----
> From: Chris Wade [mailto:clwade@sparco.com]
> Brian Roy wrote:
> > I think that my PBX does this too. Is there any way I can get the
> > Zaptel drivers to disconnect on that tone too? I would love
> to replace
> > my existing voicemail with * but I can't get my PBX to signal a
> > disconnect pr...
2004 Sep 23
3
app_valetparking / parking in general
Does anyone have Music-On-Hold and valet parking, or regular parking
working together? No matter how I configure it, I cannot get moh to
continue to play after I park a call using either valet parking or
regular parking. The only thing I can think of is that I might need to
use # transfer instead of sip native transfer?
Shouldn't this just work? If needed I can post the config for one
2004 Oct 05
1
Phantom calls on FXO
I'm getting these "calls" at 16 and 46 minutes after every hour. The
SIP phone rings, and if we pick up, we get a dial tone. If we don't
pick up, we get the dial tone in a voicemail message. An analog phone
connected to the incoming POTS line doesn't ring (whether or not *
remains connected to the line). It's like the horror movie where the
babysitter is getting
2004 Oct 07
2
recent 's' and 'n' priorities and lables
...they might be needed/wanted (namely, all the different goto's).
If this already exists, please hit me with the clue stick ;)
Thanks,
Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administrator dba Sparco.com
Email: clwade@sparco.com 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400 Millington, TN 38053
Fax: (901) 872 8482 USA
2004 Nov 29
2
Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
I've been banging my head against a brick wall for the last hour and I'm
sure this is one of those easy to solve things - just that I can't see the
wood for the trees.
I'm trying to do:
-----------
[some-context]
Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass)
[macro-dodial]
Exten => s,1,SetCallerID(${ARG2})
Exten => s,2,SetMusicOnHold(${ARG3})
Exten
2004 Sep 20
6
SER + Asterisk
Hi there,
I've seen people using SER with Asterisk. I took a look at SER
website, and I didn't see the point in using it, since Asterisk
already handles SIP very well (apparently, at least).
But, as I'm starting, and some of you (more experienced) use it, I
know that there's something there... So I would like to know why to
use SER. Is it because of scalability, performance,
2005 Aug 15
7
Switch between FXS ports
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3 ..so that I can keep channel 4 open for incoming call.
Thanks,