Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then
splitting it up so that 6 channels went to a video system (h.320) and 17
channels to our PBX. You can also convert the signaling or send out on
different type of connections like v.35. Pretty cool device and rock solid.
lled on the phones. Looks amazing though- any
> > idea on pricing?.
> >
> >
> > On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote:
> > > Anyone use the LDAP feature yet on the polycom phones? If
> > so how well
> > > does it
Anyone use the LDAP feature yet on the polycom phones? If so how well
does it work? How are you using it in your environment?
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati
ons/corporate_directory_access.html
Roy Anciso
Director of Technology
Manistee Intermediate Scho
I understand the maximizing pricing and branding aspect of phones but
when you look at feature set it just doesn't make sense. And as far as
purchasing the phone you can get it without a contract at the same
price.
When I starting thinking about it, can anyone else see a time when desk
phones ar
I'm trying to understand something that just doesn't seem to compute.
How can companies like Cisco justify selling their hard phones for as
much as they do? I know there is a matter of recouping R&D costs but
when you look at the iPhone with all its amazing features for less than
$500.00 it just do
Is there a limit on how many phones you can use? I couldn't find
anything on the website about this.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Wilson
Sent: Wednesday, March 12, 2008 1:43 PM
To: asterisk-users@lists.digium.com
Subject
Hello List,
After seeing a few positive responses for the Linksys SPA-942 phones I
was hoping to get some answers on the following questions:
* How do the phones handling system wide paging? Is it similar to
the Polycom phones?
* Can a corporate directory be configured with the phones u
Hello List,
Not sure if this will be helpful but I made changes to the original
Cisco directory.php.txt script and applied them for use on the Polycom
phones. This will create an extension directory and alphabetize it
based on the sip registrations you have setup in sip.conf. Note that
this only
Heres what I do for this:
exten => *85,1,VoicemailMain(${CALLERID(NUM)})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of arkda
Sent: Tuesday, January 22, 2008 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [aster
launch the
followme app.
On Jan 22, 2008 10:25 AM, Anciso, Roy <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I've been reading up on followme app for asterisk 1.4 and I have it
working
> but I was wondering if the following was possible:
>
> Based on followme.conf pre
I've been reading up on followme app for asterisk 1.4 and I have it
working but I was wondering if the following was possible:
Based on followme.conf present the caller with the option to locate the
person:
Call comes in (external or internal) and rings extension with followme
configured. Before
Behalf Of Christian
Pinedo Zamalloa
Sent: Friday, January 18, 2008 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk
On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote:
> Now that you have your 7911g phone up r
Now that you have your 7911g phone up running, would you mind checking
the audio quality when leaving a voicemail for on another local asterisk
user from this phone? I have a 7911g and I hear loud audio taps from the
phone. The 7961g phone doesn't have this issue. I'm just trying to
rule out the
Just wondering if this is possible:
Make a call from a registered sip extension (Doesn't matter if it's
internal or external) during the call press a key sequence let say *90
to start recording call. When the call ends the recording automagically
goes to their voicemail.
Thanks
Roy Anciso
Although it's not LDAP I used a script that I found on the voip wiki and
changed it so it looked at only sip configuration files. It also
alphabetizes the output so it can be displayed that way on the phone.
Below are my notes on the subject. If someone is willing to post this
to the wiki and send
erisk and CTPSEP
odyssee
That is not the name the phone requests
When uping my 7960, the empty file did the trick
I so far am unable to go beyond 7.1 however, as Asterisk rejects
anything I dial with 7.3
Anyone have SIMPLE sample config files?
John Novack
Anciso, Roy wrote:
> Try naming t
6.tlv from
> 10.10.10.10
> NOT 09:29:02.687805 TFTP: [25]:Finished --> rcvd 1 bytes
> NOT 09:29:02.691794 SECD: ctlRequestFile: tftp Status 0 rcv'd
> ERR 09:29:02.693428 SECD: ctlVerifyFile: CTL file too small:
> /usr/tmp/CTLFile.tlv
> NOT 09:29:02.695315 SECD: updateCTL: fin
29:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL
processing
FAILED** ctl-err 12 (file is too small)
NOT 09:29:03.227508 DHCP: Restart - delay = 1
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Anciso, Roy
> Sent: Frida
I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without
any problems.
As far as the CTLSEP File (Straight from Cisco):
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i
pp7960/addprot/mgcp/frmwrup.htm#wp1047292
The CTLSEP MAC file is a certificate trust lis
I believe you can create a blank file to keep the phone from
complaining.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Friday, December 21, 2007 10:16 AM
To: Asterisk -Users
Subject: [asterisk-users] 7970 CTLFile.tlv?
Chad,
You might want to upgrade to the latest firmware. I have 7961g on
8-3-3SR2S and works very well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad
Osmond
Sent: Thursday, December 20, 2007 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [as
For those using Cisco 7911g phones, I am running into an issue with one
the Cisco demo phones we have. The 7961 works great with asterisk no
problems However, the 7911g gets audio clipping when recording
voicemails or the unavailable message. Also when a call is transferred
using the 7911g the mu
Is there a way to tell asterisk to beep every few seconds rather than
play MOH.
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
__
Hello List,
We purchased a TE120P card from Digium and it works great. The only
problem is that we are still experiencing echo on some calls. I've tried
various echo cancellers (right now we are using OSLEC) and still no
luck.
My question has anyone gone from the TE120P to a Sangoma A101D-X
Asterisk version 1.4.13
Also when I listened in on a transfer it sounds like the moh is trying
to start but then immediately stop and tries to start again.
Below is my musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/moh
random=no
-Original Message-
From: [EMAIL PROTE
I'm having this problem. Here is my output with verbosity on 10:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/2524-099012b0", "SIP/2523|15")
in new stack
-- Called 2523
-- SIP/2523-09905220 is ringing
-- SIP/2523-09905220 answered SIP/2524-099012b0
-- Packet2Packet bridging SIP/2
Hello List,
For those of you with Cisco phones and XML directories and large user
bases, how do you handle the 32 directory object limit? I know you can
create multiple xml files with 32 objects in each but this just seems
really sloppy. I would like to have one large directory.
Thanks
Roy
Hello List,
I'm looking at the page command. I was wondering if there was a way to
set a wild card to dial all registered sip devices. For example page all
1XXX extensions.
Thanks in advance
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Man
: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML
Files
2007/11/15, Greg Oliver <[EMAIL PROTECTED]>:
On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote:
>
>
> 2007/11/14, Greg Oliver <[EMAIL PROTECTED]>:
> On Tue, 2007-11-13 at 08:44 -
Sorry forgot the images:
http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anciso,
Roy
Sent: Thursday, November 15, 2007 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial
ey XML
Files
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> Hello List,
>
> Does anyone have access to the soft key configuration files for the
> Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
> didn't find much up there.
>
> Thanks
>
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML
Files
Anciso, Roy wrote:
> Hello List,
>
> Does anyone have access to the soft key configuration files for the
> Cisco 7911/7941/7970/7971 phones? Checked up on t
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn't find much up there.
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49
The version I have is 2.1.2.0. It makes for a really nice software sip
phone:) The other thing I should note is that you only need the
SEPXXX.cnf.xml file and dialplan.xml file in your tftp directory.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uw
I'm not sure if anyone has done this before or not but, I was able get
the Cisco IP Communicator soft phone to work with Asterisk using SIP.
Thought I would share my experiences. The key is on the installation. To
have the software use the SIP protocol type the following command:
"msiexec /i CiscoI
] Selecting OSLEC for zaptel-1.4.6
Dave Fullerton wrote:
> Anciso, Roy wrote:
>> Hello list,
>>
>> Can someone outline the steps for selecting OSLEC canceller in 1.4.6?
I
>> know there was a bug fix for this but I can't figure out how to
select
>> it.
>
Hello list,
Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I
know there was a bug fix for this but I can't figure out how to select
it.
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-72
I do this to tie extensions to a particular number:
exten => _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD<2317231516>)
exten => _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology<2317234264>)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Turbo
Fredriksson
Sen
That is correct. Our Cisco rep is sending us a 7911G and 7941G so we can
test with asterisk. We plan on converting them over to SIP for testing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, October 25, 20
For those of you running Cisco phones, did you start out with a Cisco
CallManager and move to Asterisk? And if you did switch do you find that
you or your users are missing features they once had? How have you
handle the issue?
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate
Hello List,
For those of you using Cisco phones, did you have to purchase a 'SIP
license' for each phone?
Thanks
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
___
t: Re: [asterisk-users] What web GUI are people happy with?
Anciso, Roy wrote:
> Just wondering what web GUI people like for asterisk. I installed
> asterisk from source and I was looking at possibly installing web GUI
> for system management. So far freepbx.org looks promising anybody
el
Just wondering what web GUI people like for asterisk. I installed
asterisk from source and I was looking at possibly installing web GUI
for system management. So far freepbx.org looks promising anybody else
have any suggestions.
Thanks
Roy Anciso
Director of Technology
Manistee Intermed
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