similar to: Remote MWI for Central Voicemail?

Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363&highlight= Regards, Scott -----Original Message----- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate

I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to

Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Hi All, First time caller - tried WINE today for the very first time, on OpenSolaris no less. Good news - was shocked and stunned how easy it was to get an old copy of Unreal up and running - everything (almost...) "just worked" with practically no configuration, got lovely smooth rendering using my crappy built-in Intel X3100 adaptor. However, I cannot get any audio output

Hi all, Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal: 1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk application associated with it. 2. Receiving a fax at asterisk works fine, at least appears to be working

Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

dear rsyncists, rsync is superb! but did i actually find a bug? i did: $ rsync -vRHSPa rsync://rsync.mirror.ac.uk/updates.redhat.com/7.3/en/os/SRPMS . into an empty local directory the result was 140 bogus hardlinks of the same file: -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24 Canna-3.5b2-62.7.3.src.rpm -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console

Hi Forum, I am new and learning this research tool. was trying to run some tests on a censored dataset using the NADA library. Everything is hunky dory except 'censummary' throws an error and I am trying to figure out a reason for this. Please help! below you will see the error statement... thnx --nk > library(NADA) Loading required package: survival Loading required package: splines

Hola Dale, Bumping it up to 3 bits makes everything work hunky dory on the PC side of my project. Would you like me to commit that? Also, after speaking with some VStudio folks, it looks like that issue will likely remain for VS2k8 as well. Another trick that was suggested to me was to try something along these lines: Enum E : unsigned int { A, B, C, D }; ... Class ... { E e : 2; Unsigned

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI

Hello, I am new to the list and this is my first posting to the list. I have an ADS running on Win2k3 in Native Mode. I have a user created in ADS. While creating the user I have specified the "Win2K name of user" as testbug and "Pre-Win2k Name of user" is bugtest. Essentially they are not same and do differ. I have a samba 3.0.25d running on Fedora Core 7 which has joined

I have a D-Link DNS-323 NAS Drive which I'm accessing from a desktop PC running Ubuntu 8.10. When I first mount the DNS-323 share, everything is hunky-dory. I can access folders and files and save them. However, when my desktop PC has returned from standby I have problems saving files, which manifests itself in a number of ways: (1) If I "drag and drop" copy a file from another

I've got a small network at home, currently consisting of 2 '98 machines and a 2k-pro box. All was hunky-dory under Network Neighborhood, until I got a DSL line installed two days ago, and I thought it best to disable file sharing on all the boxes until I had a firewall inplace, which I now do. At any rate, I'd had need to access some files stored on one 98 box with the other two. I