similar to: Asterisk and Cisco Unity?

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,

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

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

> 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

On Wed, 23 Jul 2003, Yifang Dai wrote: > I wish! My company just spend a lot $$ on the shinny CCM phone system, so I > don't think I can change that easily... But if I can get asterisk to > talk to CCM via h323, and prove it's usefulness, I might have a chance > to use * in the branches... Well, good luck, then! > By the way, do you know if we can get *'s VM to

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

> Hrm, I made a new extension, made sure that I could dial it from a ccm phone -> > asterisk. then I set that as an alternate MWI extension ( it always put X in as an > additional entry ). the AGI I have behind that extension never runs ( in fact, > unity never calls asterisk ). > any other suggestions? You mentioned that a CCM phone can dial the Asterisk extension. Do

Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the

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

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

OpenBSD 5.6 nut-2.7.2 (bin package) Liebert IntelliSlot Unity Card IS-UNITY-DP ups.conf: ...... ? mibs = ietf ? snmp_version = v1 ...... The problem is that the card responds to other snmp (v1,2c,3) requests normally but not to the 1.3.6.1.2.1.33.1.1.1.0 (UPS-MIB.upsIdentManufacturer). # snmp-ups -a laes2ups -DDDDDDD Network UPS Tools - Generic SNMP UPS driver 0.72 (2.7.2) ?? 0.000000???? debug

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for

Hello, I noticed that in the last period, many gaming companies started to build free games for the online platforms. I think that in the next period of time, they will appear allot of 3d games on social sites. My example is N.O.V.A. from facebook (developed by Gameloft). From my pov it's a very good start for online 3d games. So, what's my problem: I want to play nova on linux, using

Installed unity perfectly, no error whatsoever, although when i run it, whatever path i put in it to create a new project in, it says "Specified path is not valid(should be a name of non existing or empty directory). I also tested using Unity Loader, and it shows the splashscreen until it crashes Ubuntu. I'm using Ubuntu 10.10, Maverick Meerkat. I can run it fine on Windows 7, without

re, all. Here's a bug filed against Google Chrome for the same problem: http://code.google.com/p/chromium/issues/detail?id=80856 and here's some forum users complaining about the same problem, and regrettably their solution is to stop using Ubuntu's Unity mode: http://ubuntuforums.org/showthread.php?t=1742354. I work for VMware and for our Unity mode (which has the unfortunate name

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated