similar to: Cisco to use * as a gateway?

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

Please I need help, I don't know why,almost every time I dial on my dect phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm using two dlink voip gateways, MGCP: DL102s. Any ideas? thanks in advance michelle matis ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis!

I've read through the archives and have picked up that * does not need a gatekeeper to talk directly with an H323 handset to send and receive calls. I'm trying to go PSTN----*-----H323 and all the examples that I can find use a gatekeeper. Are there any examples or hints for doing it without the gatekeeper? many thanks in advance Brian

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

I've been trying to figure this one out for a while, but to no avail. I have my cisco ubr924 setup for MGCP with Asterisk as the call-agent. I have manually registered the endpoint in mgcp.conf. When I pick up the phone, I get no dialtone and debug shows errors. IOS on the ubr924 is 12.2. Any help is appreciated. from mgcp.conf: [ubr924] host=65.37.86.203 context = from-sip (just as a

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

asterisk*CLI> load cdr_unixodbc.so Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend) == Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing '/etc/asterisk/cdr_unixodbc.conf': Found -- cdr_unixodbc: dsn is MySQL-asterisk -- cdr_unixodbc: username is root -- cdr_unixodbc: password is [secret] -- Connected to MySQL-asterisk it

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

I'm not sure what gave me this idea, but I decided to comment out the swap partition in /etc/fstab and reboot my laptop. I did not run swapoff directly at any time. I'm running more things now than I would ever dream of to hammer my 500MB of memory, but I still notice NO slowdown. This is the best desktop experience I've ever had for any OS on any hardware! Here's the first 12

Hi All, I am in the process of writing an R extension in c++ and am using several STL containers (e.g., vector<double>, map<int, double>, multimap<int, double>). I make sure to clear all these containers at the end of the .Call. Everything compiles and runs just fine, but I'm a bit worried since I haven't found any other packages that use STL. So, my question: is it

Hello, I've noticed that with g++ 4.3 I get the following "warning: deprecated conversion from string constant to ?char*?" when I call e.g., Rprintf("Hello world"); or when I try to define a const char e.g., char lower = 'L'; I am wrapping the c++ code in extern "C" {}. This seems to be something new with g++ 4.3, because the same code compiled

Greetings all, I'm trying to create a CentOS 4.4 kickstart CD (not a network install), duplicating what I've done for Fedora Core 3. I am having a cirular dependency for initscripts, which causes initscripts not to be installed (no /etc/inittab when the boot gets to INIT) The cascade is: initscripts-7.39.25.EL-1.centos4 requires /sbin/nash /sbin/nash is in mkinitrd-4.2.1.8-1

I have data from a design in which items are completely nested within subjects. Subject is the only second-level factor, but I have multiple first-level factors (IVs). Say there are 2 such independent variables that I am interested in. What is the proper syntax to fit a mixed-effects model with a by-subject random intercept, and by-subject random slopes for both the 2 IVs? I can

My App is hosted currently on Heroku, and I attempted to use elevation/event_calendar <https://github.com/elevation/event_calendar> to display events because it supported multiday events. I had it working locally and looking great, but when I tried to push it to the server, it caused error during the assest precompile. This in turn brought the whole site down. I had to roll back a few

Hello, I feel like I understand how the dial plan works pretty well with one exception. It seems like queues are using the stdexen macro to ring the agents/extensions. Is this normal? Is there anyway to configure this differently? I realize this is a newbie question, but I have searched google/archives and haven't been able to find the answer. Thanks, Elliot --------------