similar to: Not answering/script.

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!

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

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

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

I'm RTFM'ing, but I can't figure out how the dhcpd.conf file specifies the boot server, and how it differentiates between whether it's FTP or TFTP. I've tried option 66/next-server, and option 150, to no avail. And the docs just don't -- leastwise, in the way I'm reading them -- make sense. I've checked the admin guide, I've checked the wiki... no dice.

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

[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

> 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

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

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.

I've a little problem starting winbindd. using It on a redhat 9 linux, compiled from source. I've configured nsswitch.conf with winbind and kerberos. Naturally joined my ADS realm with the following command: net ads join -U administrator successfully. now the problem is that smbd and nmbd work correclty but I can't start winbindd due the following error and I can't manage why,

Hi, all. I'm thinking of provisioning my (non-PRI) T1 to be part data, part voice (currently, it's only voice). With the T100P, how do I get it to "do" data? Just load up the HDLC module, and it gets an IP address? Or...? I tried RTFM'ing, but there just doesn't seem to be much on it (leastwise, that's readily findable). Thanks much, Ken D'Ambrosio

Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0 sources that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the exact same- config files for both; I've tried enabling/disabling ULAW, ALAW and G279 on the

Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hey, all -- I'm looking to do some LDAP/Samba 3.x integration. I've found a bunch of docs, but a lot of them either are outdated, incomplete, or reference materials I haven't been able to find. Is there a one-stop-shopping document I should be looking at? Thanks! Ken D'Ambrosio

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

I bought a couple Polycom Soundpoint 300's, and have them working nicely with SIP... but I'd like to be able to do automatic config via FTP, but it requires some XML config files. The docs discuss them in detail, but I can't seem to d/l them from Polycom. [No, it doesn't appear to be on the CD that comes with the phone.] I've created an account at Polycom's tech support

Hi, all. I'm trying to create a PDF queue... or, rather, I already have a homebrew solution I hacked together with an lprng filter, and it doesn't work too well. The solution at www.linuxgazette.com/issue72/bright.html, which uses the smb.conf and "print command" paramaters, appears to be much more elegant, but has one major shortfall: I don't see any way to get 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