similar to: PSTN -> CCM3.2 -> Asterisk CLID

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

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

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma support suggested changing the RXGAIN in zapata.conf and it worked. Now using Trixbox 2.0 and have upgraged to latest software Asterisk 1.2.16 Zaptel 1.2.15

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

> 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

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,

Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement ("number is not in service" etc.) when I'm calling a disconnected number through

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

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

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

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

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

Hello all, I've been using the following qplot command: qplot(pixX,pixY, data=som, geom="tile", fill=rgb) + scale_fill_identity() + opts(aspect.ratio = .75) + facet_grid(unitX ~ unitY) Now I would like to convert it into the explicit ggplot grammar, so I can remove the extras: axes, labels, background, borders, facet labels, and extra white-space around the plot. (If anyone has

I am attempting to run a Poisson EWMA model using Patrick Brandt's source code. I get the following error when I run the code: Error in dimnames(x) <- dn : length of 'dimnames' [1] not equal to array extent Dimnames(x) looks like this: [[1]] NULL [[2]] [1] "mip" "div" "nom" "unity" "mood"

A couple of years ago I was building an online multiplayer game doing the frontend in Flash MX 2004 and using Colin Moock''s Unity server (http://moock.org/unity/) for the backend. This was to be along the lines of JungleWar and the Hidden and Dangerous mini games for those whom remember these: http://moock.org/unity/showcase/screenshots/hiddenanddangerous.gif

i ve downloaded asterisk-oh323-0.6.6.tar.gz I am getting this and anybody know howto fix this? #tar zxvf asterisk-oh323-0.6.6.tar.gz oh323]# cd asterisk-oh323-0.6.6 asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpm TESTS BUGS COPYING README rules.mak wrapper asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Joao, Thanks for sending the Installation tips as pasted below. It works. Seshu ---------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz Get asterisk-oh323 from