similar to: Emailed voicemail

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

>how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH.

Guys. Anybody has developed and asterisk brochure for commercial purposes (consultant, etc) that I might be able to take a look at?

Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have let's say a reception that is comprised of 2 zap extensions and a mobile phone to dial using ISDN through Capi. I want to have a delay before starting dialing the mobile phone so that it rings only when the call has been unanswered for say 25 seconds. I tried to use Capi/2106994444:ww6935555555 but without any success. There is any way to do it or the code has to be modified ? Thanks

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

For anyone playing around with IAXy(S100i) devices, I am making the following available: Windows IAXy Provision v1.00 This is a from-the-ground-up development of a means of provisioning IAXy devices using a Windows environment. For some users, being bound to Linux for IAXy provisioning is not viable or convenient in some cases. This application provides a GUI data entry for the various IAXy

Richard, I feel a little stupid now. Our spam filter (GWAVA) was blocking the emails because I had WAV files in the block list. One of those things that doesn't occur to you until you've had a little bit of sleep. Thanx for the help! A *-------------------------------- Date: Fri, 25 Mar 2005 06:03:42 -0800 From: "Richard J. Sears" <rsears@americanIS.net> Subject: Re:

Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several

[cross-posted from www.tek-tips.com] Dear samba forum, I've now wasted most of a weekend on this issue, so now it's time to swallow my pride and ask for help! Background: My main home computer died. I can't afford to replace it right now, so I'm attempting to retread an old AMD PC into a linux print/file server. Clients are a mixed bag of Windows XP home/professional PCs. The

Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? Thanks

Hey Everyone - I was going to create a visio diagram outlining how my extensions will flow out. I was just wondering if anyone on the list may have an example they have already done up so I can get some ideas. Thanks ****************************************** Richard J. Sears Vice President American Internet Services

Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a "sip reload" then a "sip show peers" I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get

I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from

I'm running Wine 1.0.1, trying to get it to run various old programs: fractint, civ II, and most important: Pagemaker 6.5 . Even when I specify a graphics screen for the fractint, all I get on it is a text cursor which won't curse. Civ II installs nicely in the Wine "C" drive, but won't play --------------------------------------------------------- and the Pagemaker

Hi all, I''m using ruby 1.8.2-15 and a recent rails 1.0 download. I''m trying to implement the Classic/Custom pagination in a list action. Here''s my code: def list getSearchTerms # NOTE: @params[:search_results] is an Array of product_data objects # returned by a search action. @results = @params[:search_results] if @results.nil? then # The default