Displaying 20 results from an estimated 10000 matches similar to: "Asterisk to Cisco Unity"
2004 Dec 04
2
XML to monitor queues on Cisco display ?
Jean-Louis curty to Asterisk
More options 4:38pm (7 minutes ago)
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
when it gets too busy :-)
I was wondering if it was poss to display this info on a display of a
2005 Jan 22
4
chan_skinny and firmware upgrade
Hello all,
I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that
chan skinny is possibly capable of doing that and would like to make
sure.
I have P00405000600 firmware which I have put in version in
skinny.conf. the phone basiclaly stops at verifying load. tcpdump
shows nothing happening apart from small amount of traffic to port
2000 (skinny).
Does anyone
2005 Jun 09
1
Asterisk to Cisco Voip System Unity
Hi all, first post. My company's office in the UK is soon going to get a
Cisco VoIP solution system. What I am interested in, and couldn't find
googling, is if it is possible to connect an Asterisk solution to the
Cisco system and have all the nice advantages of it (mainly calling the
extensions and directly reach the other office).
Thanks, have a nice day
Simone
2004 Jul 14
5
ACD Issues
Alright, folks. I just deployed * into full production at my office.
We have around 50 7905's, 5 7940's, and a handful of soft clients. We
run a call center with around 15 agents. I also have a queue set up for
the receptionists so that they don't get bombarded with calls.
Everything seems to be working with a very few minor glitches.
I firmly believe that the few problems we are
2004 Dec 13
2
Echo on one E1 line, but not the other
We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.
We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.
During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is
2008 Feb 28
2
Asterisk and Cisco Unity?
Has anyone here any experience in getting an Asterisk box to talk to
a Cisco Unity system? I have a potential customer who would like to
add a conference bridge to their existing Cisco Unity setup.
The digging I have done so far suggests that it should be possible to
talk SIP between them, but I'd be interested in any stories of success
or failure.
Cheers
Tony
--
Tony Mountifield
Work: tony
2006 Jan 23
1
Asterisk SIP phones to Cisco Unity via CCM 4.0 SIP Trunk
Hi,
I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and
about 45 SCCP phones on the ccm, and 200 users on unity. we want to
migrate all users to IP Phones to ditch our ancient phone system. I would
love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet
and run sip to an asterisk server, but have their voicemail on Unity.
these phones are $150 each,
2004 Jul 09
4
Dell 6450 / TE405p
I'm having some trouble here - need some help!
I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell
6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses
and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two
64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)"
I cannot get the card working in any of the slots.
2004 Jun 18
5
Problems with faxing via TE405P/Asterisk
Skipped content of type multipart/alternative
2006 Feb 01
1
Asterisk SIP phones to Cisco Unity via CCM4.0SIPTrunk
> thanks, using your example, and this url:
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00800dea82.shtml
<http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note
09186a00800dea82.shtml>
> I got it to work... then I realized that there's no way the SIP
phone > on asterisk is going to get the MWI ( message waiting
2005 Mar 27
3
missing ring-tone
Hi there
I've got a rather irritating problem with my Asterisk server.
Whenever someone tries to call me, the don't get the usual "ring-tone" when
they wait for me to pickup the phone.
I don't know if I've disabled this feature somewhere in my configuration
files.
Since I'm in Denmark, I've got an entry in the indications.conf file
pointing to
2006 Jan 30
1
Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk
It can be done.
1. Setup a new Vm profile on CCM with a mask of XXXX
2. Setup a CTI route point:
a. Set the directory number to a pattern. I use *27XX
but any pattern that you can send from * is good, ie. 88XXX
b. Set the VM profile to the newly created profile
c. Set the line to forward all calls to VM
3. Change the dialplan in * to append the extension called to
the
2004 Dec 28
6
Music instead of Tunes
Hello,
more and more operators in Europe offer music instead of ring tunes.
E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo,
or Mozart.... Currently I will have to answer the line to do that. Is
there a way to do this with asterisk?
Regards,
Marc
--
CTO Marc Storck
MS Networks SA mstorck@luxadmin.org
Internet Service
2005 Aug 31
4
/etc/init.d/asterisk barfing
Ok, starting to get cheesed off and feeling rather silly.
cvs head as of 5 minutes ago.
#root asterisk -vvvvvvvc
works, no problem.
#root safe_asterisk
works no problem
#root service asterisk start
Starting asterisk: [ OK ]
#root asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
/var/run/asterisk.ctl and
2004 Jul 01
2
Providing Telewest in the UK with per extens ion outbound callerID
Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd
accept just our DDI if that was all I could get.
Steve
-----Original Message-----
From: Storer, Darren [mailto:starusers@comgate.tv]
Sent: 01 July 2004 09:35
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per
extension outbound callerID
Hi Steve,
2004 Sep 03
2
mpg123 - multiple instances, taxing CPU
Is there any reason why there should ever be more than 1 instance of mpg123
running on a * server?
I just did an 'uptime' and noticed all 3 of my loads where over 3.00.
'top' showed 8 mpg123 processes all processing the same 3 songs (our
background music).
I tried to kill one of them but another one spawned in its place.
Any ideas?
Thanks,
Matthew
2006 Mar 21
6
FAX over PRI
We are doing this with the latest spandsp, iaxmodem and hylafax.
Seems to work very well for us so far.
-Jonathan
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Michael Gaudette
> Sent: Tuesday, March 21, 2006 3:34 PM
> To: 'Asterisk Users Mailing List - Non-Commercial
2007 Jun 07
2
Bridged PRI calls - processor involvement?
On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does the processor have?
We're now seeing chunks of missing audio and I can't tell whether this is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade.
I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any other ideas on how I could go about tracking this down?
I'm
2008 Oct 23
1
recursibve listing of file owner, possible?
Hi,
I'm writing a utility that needs to smbmount various shares from servers in numerous domains (no problem, all working) and then list the contents of the directories (no problem again) and obtain the windows file owner in a textual form.....
Any ideas how I can achieve the last part efficiently?
I see that smbcacls can do it 1 file at a time, I really need a way of doing it
2004 Jun 15
5
PRI problems (telewest -> * -> LG GDK 186)
Hi,
?
I'm trying to figure out what the issue is splicing Asterisk between our
Telewest PRI and a GDK-186 with a PRI card.
?
We're using the Digium TE405P
?
Our telco provider is Telewest, and Telco directly into switch is fine.
?
When I splice Asterisk in, I can make and receive calls from Asterisk
extensions, I can make outbound calls from the GDK, but inbound calls do not
seem to pass