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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

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

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.

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

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

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

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

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

In our IVR we have a user enter a 6 digit number and information if returned. Our problem is that no all of the digits that the user presses are being recevied correctly. Its not as if the first digits are being cut off or the last, its just some digits aren't coming through. Our setup is that we have an IAX channel coming in from VoicePulse going to our Asterisk box via the T1 coming in to

I am trying to use swift in PHP/AGI. function swift($text, $escape_digits='', $frequency=8000, $voice=NULL, $fnameIn='') During swift speaking some text I want the caller to be able to press 1, 2 or 3 to do thing 1, thing 2 or thing 3. How are these digit defines and then caught? Thanks, Michael

I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI> sip show registry Host

I have a new share setup that I would like to do some pretty simple things with. I would like to have multiple people connect to it. Everyone can read and write to it but the only creator of a given file can delete that file. Also, when a given file (say Word, Quark or Photoshop docs) is open by user1 it should be considered read-only by the other users. If user1 creates a file user2 (in the

We are developing an IVR application and when I am testing locally on my machine using a softphone (iaxcomm) the digits I press for GET DATA work every time. I am testing with a local extension that goes right into my routine. However when I try to call in to the system using an analog or cell phone GET DATA drops some digits that are pressed. There doesn't seem to be a pattern to which

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

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'