similar to: Asterisk Won't Process Call

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

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

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

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'

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

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

Is there anyway to NOT allow the incoming caller to leave a voicemail message for a certain mailbox? I would like the caller to hear the message and then have the option to "press 1"(for example) to call the user (make an outgoing call), but not to be able to leave the message. Even if after the unavail message is played the caller gets kicked back to another menu that has the option

Is there any way to have the user confirm the extension they are looking to go to before transfering? i.e. "You pressed 5 4 3 3 2. Is this correct?" 1 - GoTo extensionPressed 2 - Enter extension again Thanks! Michael

Hi, I am a newbie and just discovered AGI (after learning a lot about extensions.conf's language). Before putting in a lot of time on AGI/Perl/PHP I would like to know if its possible to do most of the functionality performed in extensions.conf through AGI. Can AGI be used as a replacement for scripting a dialplan? Thanks, Michael

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.

Hello everyone! It has been a while since the 1.0.7 release, and I have fixed a lot of stuff since then. I think it's about time to make another release. I realize that there are still some outstanding issues, but it's nearly impossible to bring that down to zero. However, I'm open to discussion on anything that someone may feel is a "show-stopper". I am on IRC as

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

How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the

Hey all, I''ve had these functions for some time now, and would like to offer them as two new dollar-sign functions - elements to be extended by Prototype geniuses. :-) makeText(string) as $T() - return text node element Does just what it says... I''m sure someone could extend it nicely when via Prototype. (example) var x = $T(''hello world'');

When I do t.test on two distributions (see example below), it outputs numerous data about the t.test. What I'd like to do is individually capture some of this data and assign it to other variables. However, I am unable to find anything in the help section. In the example below, the t value is -4.0441 and the p-value is 0.006771 How can I assign these values to two variables, let's

I have configured asterisk with the AMP php configuration utility. I am able to make outgoing calls through broadvoice but incoming calls are sent to BV's Voicemail and never actually enter the IVR. When I show sip debug info through the asterisk prompt it actually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified

Hi I wish to install asterisk 1.2 (the latest tar.gz from the site ....not the CVS version) on an HP box with a TE110P (single port E1/T1) My question is which OS would be preferred in this configuration Fedora Core 1 or Fedora Core 3, and are there any install guides out there that are recent enough for asterisk 1.2 I am also open to suggestions for other Operating Systems if any of you feel

---------- Forwarded message ---------- Date: Wed, 07 Jul 2004 00:31:21 -0400 From: Declan McCullagh <declan@well.com> To: politech@politechbot.com Subject: [Politech] House bill exports analog phone regs to VoIP http://www.politechbot.com/docs/boucher.voip.bill.070604.pdf There's a new bill in the House of Representatives to regulate phone calls made over the Internet. It was