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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:

Hi, Is there any specific made embedded hardware designed VoIP software or Asterisk? I want to build a router that have VoIP enabled, so that I can use it connect to a VoIP ISP. Thanks Sam

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 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>"

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

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

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

...r 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

-----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

The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page:

...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

The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash "ShellShock" bug. New in 1.2.0: * New Linux Kernel 3.2.x * "igb" ethernet driver for Intel Atom C2000 * Enable AES-NI support * New "sip-user-agent" firewall plugin * New versions of Asterisk 11 and 1.8 * Bash "ShellShock" security fixes A

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

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