search for: volfman

Does anyone have the MySQL add-on as a binary for OS X? Or am I getting it wrong and MySQL is installed by default? Thanks. Michael

Is there a way to send a voicemail to two different email addresses when a caller leaves a message? Thanks a bunch! Randy

...sk.keystreams.com/ . Yes some of the threads are doubled up, but that sort of conversion is not perfect so just use the search feature. It currently has 2002,2003, 2004, and Jan/Feb of 2005. It will *not* be updated in real time (at least not for now) Please direct flames/questions/comments to volfman@keystreams.com. -- Roman Volf Keystreams Internet Solutions volfman@keystreams.com

...;> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > Post your extensions.conf and what's on the CLI (asterisk -r) > > -- > Roman Volf > Keystreams Internet Solutions > volfman@keystreams.com > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only

I'm just curious why Sipura isn't using free IAX protocol with their devices instead of SIP? With IAX NAT traversal would have been easier, so why are they using SIP. Is there any politics in it? -- #Joseph

Hi, I have a little problem running Asterisk. I just got the asterisk, zapttel and libpri sources from cvs. I built and installed it. Next I installed the sample configuration. The problem arise when I try to start Asterisk. Running asterisk -vvvvc I get the following error [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf':

I have done a cvsup of the openssh port. It builds correctly, but refuses to install with the following: ===> Installing for openssh-3.6.1_5 ===> openssh-3.6.1_5 conflicts with installed package(s): ssh2-3.2.9.1_1 They install files into the same place. Please remove them first with pkg_delete(1). *** Error code 1 Stop in /usr/ports/security/openssh. I was unable to

Hey all, I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD -

Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when