similar to: How to generate core dump?

Anyone using this feature of asterisk's voicemail? I'd never heard of ADSI, and saw it as I was perusing the voicemail source this morning. Is it some kind of visual way of managing voicemail on your phone's display, or does it require a terminal of some kind? Thanks! j

Hello, I have some zaptel cards, and I would like to install them in some user's computers. Is there any way to connect those cards with asterisk server (which is in another computer)? All manuals I have read explain how to connect asterisk and zaptel cards in the same computers, but not on different ones. Thank you very much.

Hi! I see lots of examples using , but core show application displays | So what is the correct character to use to separate parameters for application, functions and macros? thanks klaus

So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in.

Hello, I just bought a TE121 T1/E1 card, and now trying to install it on a 1.4.23.1 asterisk with dahdi 2.1.0.4 Actually first everything went on well and i managed to see my card on dahdi. Here's the output: #asterisk# dahdi_hardware pci:0000:04:08.0 wcte12xp+ d161:8000 Wildcard TE121 and this is the scan: -------------------------- asterisk# dahdi_scan [1] active=yes alarms=RED

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Hey, all. I'm planning on migrating a W2K3 server to a Linux solution. It needs to be AD-aware, support ACLs, etc. This isn't something I'm doing Right Now(tm), so I can wait a little bit. A couple questions: 1) Are there any known issues with BTRFS? 2) Which version of Samba would be most appropriate for this? 3) AD integration: I've never really done it (with success); any

I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks!

Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

[Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones "inside?" [In other words, I obviously need a NIC -- but what would be the

I got dragged away from Asterisk (somebody made me an offer I couldn't refuse for system administration), but I'm thinking about seeing if I can't get it deployed at my new employer. Regardless, there are two things about older voicemail that used to annoy me: - Dial by name. Has anyone made it so it can be first or last? - Jump to voicemail; you used to have to actually dial the

Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use "rescue

Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between "split" and "combined" firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, all. I've got a fax extension in my extensions.conf, but spandsp never sends my faxes there. Both applications -- txfax and rxfax -- are registered by Asterisk, so they compiled and installed correctly. I've got a Sangoma A104 card, and (as some people had suggested) have loaded ztdummy. Everything seems fine, except that it never gets recognized as a fax. I've even turned off

Hi, all. Got my Uniden UIP200 today (ordered from thetwistergroup.com), and was very excited to set it up... until I came to the realization that there were no docs with it whatsoever. There was, however, a sheet of paper with the stock warnings (don't use the phone in the tub, etc.), AND a URL -- for documentation. Score! Well, no. The URL it gave me, bcs.uniden.com, does indeed have