similar to: Best cheap card to use for home Asterisk system???

I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called "business phone service". They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are "cutting out" at times. What

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. The new versions of 'register' and 'asthostid' can be

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hi All, I am trying to upgrade an old Asterisk installation to 1.4.2 (it's currently running CVS-08/02/04-15:15:26) but have hit a couple of problems. The first was easily fixed. I got "storage size of sin isn't known" errors whilst compiling streamplayer.c, but after seeing http://bugs.digium.com/view.php?id=4908#32012 I manually added "#include <netinet/in.h>"

Hi, I have to install an Asterisk PBX for a customer and he wants something like logic supply's fanless computers. Can anybody advise about how good will they work, are they compatible with the Asterisk system? I'll also be installing a sangoma 4 port FXO card in it. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

We currently have most of our voicemail forwarded to user's email addresses, but the message is coming in at a way low volume. It sounds great when you listen on the phone, but it's very hard to hear when you listen on the computer. Does anyone know of a way to increase the gain on the file before sending it off? Aaron

Hi everyone, I am trying to get iaxmodem up and running. This is a very basic setup, which at this moment should only answer incoming faxes. What I did: zapata.conf (rest of it should be fine): faxdetect=incoming group = 1 channel => 1-2 context=from-pstn iax.conf: [200] username=200 type=friend callerid="Fax" <200> secret=dooo host=dynamic notransfer=yes allow=all

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way

Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a "set it and forget it" type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). After

Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing & Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED & CONFIDENTIAL CLIENT COMMUNICATION    *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may

Hi, I'm relatively new to Asterisk, and I'm looking to build a tiny system for home use. For context, at home I've got a line from Vonage (last I heard, they won't give out SIP credentials and let you use random hardware/software), which comes out as an analog line with a dialtone, and I'll be treating that as if it were a regular PSTN connection. I also work from home,

Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. Peder

I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended