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I'm trying to install Therm 5.2 from http://windows.lbl.gov/software/therm/therm.html but I see: Code: $ wine THERM52Setup.exe fixme:actctx:parse_depend_manifests Could not find dependent assembly L"Microsoft.Windows.Common-Controls" xcb_io.c:452: _XRead: Assertion `dpy->xcb->reply_data != 0' failed. wine: Assertion failed at address 0xffffe410 (thread 0009), starting

...some testing before switching completely to an Asterisk based system), is there something I can try to improve the percentage of calls that are disconnected? I would use silence detection in voicemail.conf but wanted to know if there was some other method. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC 877.901.3113

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:

...d RSS (podcast/Vodcast) can be created per 'playlist' 4) We would like to install the server on a Mac OS X server Can Icast do all this, maybe by adding some 3rd part apps or something? Tor Berg The Danish School of Journalism Olof Palmes Alle 11 DK-8200 Aarhus N IT-Developer e-mail: tb@djh.dk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/icecast/attachments/20061013/9f959620/attachment.htm

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

...to setup aliases for SIP? For example, direct sales@mysipdomain.com to 55544@mysipdomain.com If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

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

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:

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

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

-----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

...'ll need to check two more items: Upgrade firewall plugins: upgrade-arno-firewall check upgrade-arno-firewall upgrade Install sound files: **NOTE sound files are not installed by default starting with 0.7.1** upgrade-asterisk-sounds upgrade core en ulaw Enjoy! Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

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, 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

Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how

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

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 Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like "Password" or "Call from 205-456-2222". Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run "top" and there is no heavy load on CPU or

Hi all - I'm building an Asterisk system (Trix2.2) for the house- I'd like to do the following things: I have a single phone line (happens to be Charter Communications VOIP, but I have their ATA and they've connected to red/green pair in the house wiring) What I'd like to do is this: Get some low-end but reliable card/external adapter which would connect to