similar to: customize the pager email

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss > Sent: Monday, October 10, 2005 5:58 AM > To: Asterisk Users

Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to

Hi all! I've got a machine setup to be an "RSYNC Server", i.e. running rsync in daemon mode waiting for connections from various other machines on my network. This machine is running Debian Sarge and rsync 2.6.3. For the past several days, I've been getting notices like this in my backup logs: ============================================================================ =====

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

My apologies if this is considered somewhat off-topic but I presume that this information may be may also be of use to others using these filters. Is there someone with a better understanding of COM than I have who can tell me where I'm going wrong with the following C# interface definition? I can set and recover a quality setting using this interface but not the isUsingQualityMode flag. Not

Hi; Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Best

Question for all: Right now i have a Table in a mySQL DB that has a row called Ingredients. When the data is entered into the DB its enter like so from a text area: 1 1/2 lbs. beef top sirloin, thinly sliced 1/3 cup white sugar 1/3 cup rice wine vinegar 2 tablespoons frozen OJ concentrate 1 teaspoon salt 1 tablespoon soy sauce 1 cup long grain rice 2 cups water 1/4 cup cornstarch 2 teaspoons

Hi all, I am new to R. I downloaded and installed R 2.15.1 I tried typing R.exe --help at the DOS Command line C:\", but I keep receiving: [quote] R.exe is not recognized as an internal or external command [/quote] I tried many variations of R.exe --help, but roughly the same response Any ideas? thx w -- View this message in context:

I have an application written in c# which uses two instances of the theora encoder in a single filtergraph. Each one is followed by an ogg multiplexer for output to separate files. When I run the filtergraph the smaller video's buffer is intermittently written to the start of the larger video's buffer. I am using the latest filters from opencodecs_0.84.17338.exe and the smaller video is

Since the list community has done so much for me in my humble asterisk beginnings I have put together a simple little script written in php that serves as a paging reminder script. If anyone is interested in a copy of it contact me off list and I'll forward you a copy. The basics of the script are as follows: It queries an asterisk inbox of your choosing for the existence of a file that

Ahoy, I made this pager, " def list @item_pages = Paginator.new self, Item.count, 10, @params[''page''] @items = Item.find :all, :conditions => "category_id = #{params[:condition]}", :limit => @item_pages.items_per_page, :offset => @item_pages.current.offset @categories = Category.find_all