We're mostly Cisco CallManager with some SIP and Asterisk.
I want someone at one of our locations to be able to dial and number
and have Asterisk simultaneously dial several Call-Manager extensions
which are set to auto-answer and talk into the phone creating a sort
of paging system.
We have
We have occasional problems with failed transfers. The PSTN call comes
into Cisco Call Manager, then to asterisk over a SIP trunk and then to
an asterisk controlled SIP phone. The SIP phone transfers back to a
CallManager controlled SCCP phone which sometimes fails.
Is there a wait to let
In a CallManager environment (currently 4.0, moving to 6.1 in the next
few months), can Asterisk completely replace Unity
as a voicemail system?
What works and what doesn't? MWI? Call Handlers? Does everything
work via a SIP trunk? Who has done this
and is willing to contact me?
Thanks.
)? Can you call the phone from
CallManager?
Peter Pauly
http://www.usbtests.com
On 3/11/08, Aaron Fransen [EMAIL PROTECTED] wrote:
Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1,
Asterisk is running strictly VoIP over the network and using CCM as the
trunk.
Calls
I don't see any evidence that queue metrics can push data to the
phone. I'm really looking for a home-grown solution that pushes
XML/HTML to a phone during a call, like the 7960's.
On 12/13/07, Dovid B [EMAIL PROTECTED] wrote:
Queue Metrics
- Original Message -
From: Peter Pauly
I have been unable to get callerid name passed from Cisco Callmanager
over a SIP trunk to Asterisk. Only the number is displayed. Has anyone
been successful getting callerid name?
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Are there any phones whose display can show queue statistics, ie:
calls waiting, etc, on the phone itself without too much trouble with
Asterisk? Especially while the phone is in use (on a call)?
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Does anyone know how to customize the order of the soft keys on a 7960
running SIP? All the documentation I could find is CallManager
related. Specifically, I want to move the transfer function to the
first set of buttons during a call.
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Try this:
queue-thankyou = /dev/null
On Nov 30, 2007 10:02 AM, [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Short of replacing a sound file with a sound file containing only a
short period of silence, is there any way to suppress certain sounds
from playing during queue
I currently have the following setup:
exten = 2000,1,Playback(/var/lib/asterisk/sounds/Greeting)
exten = 2000,2,Queue(Qabcdef|t)
exten = 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy)
exten = 2000,4,Hangup
exten = 2000,103,Hangup
What happens is, that the greeting in step one is played
I'm monitoring my tftp servers' logs and my Cisco 7960 test phone
won't download dialplan.xml to the phone. I know this from the logs
and from the behavior of the phone. I see it downloading other files
like the ring tone file, etc.
Is there something that needs to be set in the cnf files to
The screen on the 7960 is a rather low resolution one, and therefore
does not display much data. Pressing the directory button (and selecting
Resolutions and color depth on the phones are as follows:
7905/7912 192x53 Grayscale, Depth=1
7920 128x59 Grayscale, Depth=1
Are there any cheap SIP phones (like the Grandstream
for example) that support power over ethernet?
What is necessary to support SIP phones in a
Cisco Call Manager environment?
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[EMAIL PROTECTED]
I assume there are several people on this list that
have Cisco Call Manager implementations under their
belt
We are beginning a call manager implementation and
the first question I asked Cisco was, should we use
SIP or Skinny. Cisco is pushing me towards Skinny,
saying that I will lose some
Does anyone know what would be involved in making
Asterisk work as a voicemail system in a Centrex
environment? We have a Centrigram voicemail system
that belongs in the Smithsonian. There are analog
lines coming into the box and a 56KB data feed from
the phone company's switch.
Peter
The Indianapolis Marion County Public Library has put out a
request for proposals on its website for a local dialtone/voice
system. I know there are several people on this list that
run consulting companies that specialize in implementing
Asterisk systems.
A PDF file describing the RFP
Are there any companies/consultants in the Indy area that
are Asterisk experts? Please contact me via email. THanks.
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
The PDF on the website says that this thing
supports a downloadable ring-tone. This
makes me somewhat suspicious - does
this thing generate ringing voltages
and actually ring the attached analog
phone?
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[EMAIL PROTECTED]
I'm currently running firmware version 3.2 on my
Cisco 7960. I've seen on the list that several
people are running the 5.x latest versions.
I've avoided going to higher firmware versions
because I'm worried about potential problems
or issues with the encryption mechanism used
in the later
I thought you guys would be interested to know:
eWeek has a short article about Cisco bringing out
a new IP phone: 7970G. It has a high resolution
color touch-screen display with support for XML and
can act as a mini-browser to allow the development
of vertical applications.
But get this: the
On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote:
Come on people! Fork out $50 for a discman and another few bucks for some
royalty free library music and have that on hold instead.. You're in
control, you know what your callers are listening to, and you're also legal
Why go to all
I don't know if this has been mentioned yet:
Voicepulse is now offering wholesale pricing and
IAX2 connectivity for Asterisk users. No fees, pay
as you go. They also
offer incoming calls for $7.99 per month. See
wholesale.voicepulse.com.
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On Thu, Sep 18, 2003 at 07:02:42AM -0700, TC wrote:
well, i have same problem...
it sounds like nufone is not allowing calling of #800.
anyone from nufone care to comment?
I have seen nufone die, if the callerid is not
a cid from us 48 try setting your sic to
I added SetCallerID and
Is anyone else having trouble dialing 800 numbers
through Nufone? I'm getting the SIT tones no matter
what number I dial. Normal long distance works fine.
I don't think it's my dial plan (it was working previously).
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I'm looking for a source for 50-pin amphenol
cables, the ones used to connect Adtran's to
punch down blocks. Preferably, one that's
mail order and takes orders over the internet.
Thanks.
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[EMAIL PROTECTED]
On Thu, Sep 11, 2003 at 09:30:35PM -0700, John Todd wrote:
Before running any application that has sound playback (Playback,
Background, VoiceMailMain2, etc.) it would be wise to execute an
Answer first, then a Wait(2) to allow for VoIP channels to fully
establish and settle.
Adding
On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote:
nope
when I click on something on the left I get a FQDN not just the pne you had
Hmmm.
Further info: it works with Microsoft Internet Explorer. It
does not work with Mozilla 1.4 under Linux. It also does
work with
I'm using a Cisco 7960 with asterisk and any recording
on the machine, be it voicemail prompts, time of day,
echo test message, etc, is cut off for the first 1/4 to
1/2 second. I've tried setting the phone to gsm but
it still happens.
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On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote:
hmm works for me... its the exact same code that is installed on the sample server
listed below and I dont get the problem there. lemme know more info and ill look
into it
Dave
Well, there is no such domain as phpconfig.
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the
On Tue, Sep 09, 2003 at 11:41:17AM +0100, Skuse, Phil wrote:
Yes. They are on the same subnet.
I solved my sound problem with X-lite by using the latest
CVS version and compiling that. I had been using the
stable and unstable versions out of Debian.
On Tue, Sep 09, 2003 at 11:04:34AM +, WipeOut . wrote:
Where did you get access to X-Ten.com's CVS server?
I didn't know they had the source code for x-lite available..
Sorry, I should have been more clear - I used the latest
version of Asterisk via CVS.
Can I configure an Adtran channel bank with a mixture
of FXS and FXO cards and have them come into a single
T100P T1 card? It seems like this would be a cheaper
solution than trying to load a bunch of PCI cards into
a PC.
Also, when shopping for an Adtran (on ebay) - what do I need
to watch out
For the benefit of others having this problem - I
installed the latested CVS build and the problem
went away - I can hear audio now from X-lite.
I was using the debian unstable package.
Here's what I did:
cd /usr/src
mkdir asterisk
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs
Does the Digium FXS card support modems (and Tivo devices)?
If so, to what speed have they been tested?
Also, on a somewhat unrelated question:
How does the FXS card generate ringing voltages
if the PC only supplies 12 volts?
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Asterisk-Users mailing
On Thu, Sep 04, 2003 at 06:26:03PM -0500, James Sharp wrote:
On Thu, 2003-09-04 at 17:22, Peter Pauly wrote:
Does the Digium FXS card support modems (and Tivo devices)?
If so, to what speed have they been tested?
Assuming that you can do native zaptel bridging (Going from an FXS port
adding nat=yes to the sip definition made no difference.
Does Asterisk use the DSP in your sound card to do the
audio processing?
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I started to suspect the X-lite client in my problem
(I was getting no audio when calling into asterisk)
because after I would make test calls to asterisk,
setting X-lite back to my FWD account - I would get
no audio with FWD either, even though the sound card
was working and I got dial-tone,
I have been using X-Lite on FWD without any troubles
and recently became interested in trying asterisk.
I am able to register from X-Lite and dial a number -
I've tried dialing some of the sample numbers in the sample
extentions.conf file, like 500 and 1234, they appear to dial
correctly from
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
correctly from X-lite but nothing else happens - no audio is
heard. My understanding is that I should hear some sort of
I am using x-lite with the asterisk demo no problem. All I modified was
sip.conf
Is the asterisk
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