and 5.
Does that make sense? Anyone have the solution?
Jimmy Ezell
From: asterisk-users-boun
mailbox=100
[incomming3]
type=friend
context=internal
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
mailbox=100
===
Jimmy Ezell
Assistant IT Manager
(408) 487-2200
http://www.hmhca.com/
image002.jpg___
-- Bandwidth
Sorry I mean to say cisco 7960 phone.
From: Jimmy Ezell
Sent: Tuesday, August 11, 2009 9:15 AM
To: 'asterisk-users@lists.digium.com'
Subject: Cisco 1760 Multiline phone
I have a cisco 1760 phone
Of Jimmy
Ezell
Sent: Tuesday, August 11, 2009 12:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 1760 Multiline phone
Sorry I mean to say cisco 7960 phone
on really busy days
will calls make it up to lines 4 and 5.
Does that make sense? Anyone have the solution?
Jimmy Ezell
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Thursday, June 11, 2009 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
Still no luck getting this to work. I have been looking
Still no luck getting this to work. I have been looking at the
CallManager Logs but so far that is worse then useless. Anyone out
there have any luck connecting Asterisk 1.4 and Cisco CallManager
3.3(5)?
Jimmy Ezell
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
As you can see below I am striping off the 8 before it ever goes to CCM
in the extensions.conf file.
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
I have the H323 gateway in CCM configured to use the same Calling Search
Space as my phone extensions.
Jimmy Ezell
dtmfmode=rfc2833
nat=no
canreinvite=yes
qualify=yes
extensions.conf file
==
[globals]
CISCOTRUNK=H323/callman02
[cisco]
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
exten = _8XXX,n,Congestion()
exten = _8XXX,n,Hangup()
Jimmy Ezell
Converting CCM to Asterisk
http
company looking for a long term, low maintance,
basic phone system (Calls, Hold, Transfer, Park, Conference), what is
the best stable release of Asterisk to use?
Even worse question to ask, what is the best Linux ditro to run Asterisk
on?
Jimmy Ezell
In article
e77ab304d41c084090d498682873252fc1b...@nts-10.ca.hmhengineers.com,
Jimmy Ezell jez...@hmhca.com wrote:
Sounds like the workaround for 4.7 is to add this symlink
that you mention. What directory
does the symlink need to be in? What should it look like?
Where should it point
On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
multi-processor machine ( I had to remember to specify smp
for the kernel)
I repeat: why bother with such an old system? Really?
Recall the comment from the book. That book had nothing really specific
to Centos 4. Why do you shoot
and MeetMe Help
On reading your mail more carefully, it looks like you do have
ztdummy, do depmod -a and then try to modprobe it.
Same thing happens
[r...@localhost ~]# depmod -a
[r...@localhost ~]# modprobe ztdummy
FATAL: Module ztdummy not found.
Jimmy
on Monday 05/18/2009 Jimmy Ezell(jez
and MeetMe Help
On Monday 18 May 2009 19:26:21 Jimmy Ezell wrote:
make[1]: Leaving directory
`/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686'
[ `id -u` = 0 ] /sbin/depmod -a 2.6.9-78.ELsmp || :
Here is your problem. Your kernel name is 2.6.9-78.ELsmp
and the kernel
name you just compiled
and MeetMe Help
On Mon, May 18, 2009 at 05:26:21PM -0700, Jimmy Ezell wrote:
make[2]: Entering directory
`/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686'
You seem to be using RHEL4.7 or Centos 4.7 .
Is this a new installation?
--
Tzafrir Cohen
Centos 4.7 and yes I installed
, May 19, 2009 at 11:11:40AM -0700, Jimmy Ezell wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Tzafrir
Cohen
Sent: Tuesday, May 19, 2009 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial
On Tue, May 19, 2009 at 01:12:02PM -0700, Jimmy Ezell wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Tzafrir
Cohen
Sent: Tuesday, May 19, 2009 12:21 PM
To: asterisk-users@lists.digium.com
Subject: Re
Time to spotlight my ignorance here.
I am up to post 9 on my Cisco Gateway to Asterisk Step by Step posts.
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
The next post is about MeetMe but I am running into trouble. I am pretty sure
the ztdummy thing is the problem but I
I would modprobe ztdummy and see what happens and look at the log and
lsmod after you do that.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
cov...@ccs.covici.com
John,
modprobe comes back right away with ztdummy
: FATAL: Module zaptel not found.
May 18 14:46:59 localhost zaptel: Loading zaptel framework: failed
On boot I also see Waiting for zap to come online...Error: missing /dev/zap!
on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote
I would modprobe ztdummy and see what happens
Yes (I assume you have it configured for SIP)
Press the Setting button (square with check)
Scroll down to the Unlock Config
Enter the password of cisco
Press the accept softkey
Now you can scroll up to Network Configuration / SIP Configuration and manually
change settings.
Jimmy M. Ezell
Jonathan And Dan,
Thank you both for the responses. But the problem turned out to be that - I'm
an idiot. I was placing both calls to our company number for the voicemail
system which I thought was part of a hunt group. As it turns out, only one
call can come in at a time on that number, and
Dan thank you, yes that seems to help. It looks like the bridging is happening
now and I see the light come on in the second FXO port, but then I get a busy
signal after that and the call still does not complete. If I set the second
line as priority 1 it completes the first call on that line
I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my
searches have failed me, I hope someone can help. (By the way, I added an
asterisk index for quick navigation on the blog
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.)
Here is the snag and I
asterisk-users@lists.digium.com
Message-ID:
3de056a30904171820x5f67217fp2b469bd312193...@mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1
On Fri, Apr 17, 2009 at 2:47 PM, Jimmy Ezell jez...@hmhca.com wrote:
Our small company is replacing Cisco CallManager with Asterisk (because we
Our small company is replacing Cisco CallManager with Asterisk (because we are
tired of sending them money) and I am documenting the process as I go on my
blog. I am trying to make the notes as easy as possible in hopes that I can
ease someone else's pain. Here is the link:
vi. I use it because that is what I know (at least enough to
get by). I did add the nano suggestion onto the blog posting.
Thanks,
Jimmy
-Original Message-
From: Doug [mailto:d...@natel.net]
Sent: Friday, April 17, 2009 12:36 PM
To: asterisk-users@lists.digium.com
Cc: Jimmy Ezell
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