similar to: MySQL Support For OS X

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

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is

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

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

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

If user1 creates a file on the share, why with this configuration can user2 delete that file created by user1? Thanks, Michael [global] idmap gid = 16777216-33554431 idmap uid = 16777216-33554431 path = /var/www/ unix password sync = yes workgroup = cmny os level = 20 auto services = advertising editorial null passwords = yes encrypt passwords = yes winbind use default domain = no

For those newbies who seem to not know google exists, I've setup a searchable forum located at http://asterisk.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

We have a basic application that runs a SIP channel to pick up a call and process it. We are using Broadvoice and it's been working great. We recently rebooted the machine and now when a call comes in Asterisk picks up the call but does not process it. Asterisk seems to send the call back to Broadvoice. Nothing at all has been changed in the configuration to warrant this. Below is the

Does anybody know if Avaya has a test SIP firmware available for 4620 and 4640 IP phones? The 46xx SIP image from their website is a combo download with SIP for the 4602, and h323 for the the 4620 and 4640. It looks like they demo'd a SIP image for the 4640 as far back as 2004: http://www.sip.org/von/2004/boston/slides/DSC_0042.php Thanks, Andy -------------- next part -------------- An

Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph

I am trying to configure RealTime Voicemail with MySQL. I downloaded compiled and installed the CVS HEAD for asterisk, and for asterisk-addons. MySQL seems to be loading correctly (the cdr table is recording incoming calls). But the RealTime Voicemail doesn't seem to be checking the database table for the voicemail users. When trying to login to voicemailMain if I use a user in the

I have an IVR application that works fine from multiple DID sources, unless the call to that DID was from a Vonage service user. In this case about half the DTMF tones never get recognized by Asterisk. Has anyone else seen this? Suggestions? I'm running 1.0.9.

Hoping someone can give me a pointer on this: I'm trying to connect a Mac OS X 10.2.6 machine to Samba 2.2.8 hosted on a Linux 2.2.20 machine. I can authenticate, connect to a share, navigate through. I can even delete and move files. But, whenever I try to save a file in most programs, or copy a file in the Finder, I'm told that I don't have the privileges to do it. Strangely

I am having trouble with FC3. After doing a yum update (of 1264 packages) I still cannont compile 1.2.1 from source: make[1]: `libedit.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline' make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: `libdb1.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/db1-ast'

We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing

Hi, What would you say that the best compression format is for voice recordings on Asterisk? The tradeoff being the file's size. I like GSM because of the small files size but the quality isn't great. What are people finding as a good setting for GSM? Thanks, Michael