similar to: Confirmation Of Extension Before Transfer?

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

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

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

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

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

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

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 all, I'm creating a site where people can share their benchmarks. If you are interested, the site is at (I just started on it so it has the stock graphics and color scheme still): www.dcsnow.com/mambo I would like some thoughts on what would be a good way to standardize the testing so the results are more comparable. Is Bonnie+ a good program for hard drive speed testing? Is there

Using CentOS 5.8: Currently on my workstations, when I press the power button the computer immediately does a 'shutdown -h now' (per /etc/acpid/events/power.conf). Is there a way to change it so that a confirmation dialog comes up, rather than an immediate shutdown? I assume that I am going to need to change that power.conf file to tell some program that the power button's been

We need incoming calls to simultaneously ring SIP phones, and a cell phone which is called via a SIP or IAX trunk. When the cell phone answers we'd like a brief prompt played (e.g. "press # to accept call") and if # is pressed connect the incoming call to the cell phone. ZAP trunks have some of this functionality with the call confirmation option, but we must use SIP or IAX trunks.

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

I was wondering how the Dell SC420 will perform under normal office to office communications. We would equip each server with a T1 card to make office to office SIP calls. They will integrate into our existing PBX systems. Does anyone on this list use this hardware currently???? Thanks!

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Hi, I've been using Asterisk for several months now with great success. I'm working on a system that tries to deliver a recorded message to a user, as follows: 1. a call file is placed in /var/lib/asterisk/outgoingcalls 2. This triggers a call to be placed 3. When answered, the caller hears the recorded message 4. After the message, the caller must confirm by pressing 1 5. If they

Hello all, This email is intended rather informative than questioning. While developing some script-generated dial plan, we figured out that there are differences between Snom 190's and BudgeTone 101's relating to transfers. It appeared that the 190's will have their own 'Caller ID' set as the 'CALLERID' variable in astersisk when transfering a call, while 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

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