similar to: No audio on PRI.

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

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

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 there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

> 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.

Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only

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.

[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

Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

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. 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

Hello, i'm using a TE410P on some E1/PRI with EuroISDN and experiencing a few audio quality problems with current CVS (both zaptel and asterisk) and the following network ISDN public SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local

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

I have semi-successfully provisioned 2 domain controllers on one domain with (I think) working replication. I followed the Samba wiki and used the sernet-samba-ad repos and packages on CentOS 6.4 x64. Questions: Replication status has times for outbound neighbors as @ NTTIME(0) on both sides. What does this mean? dc1: http://pastebin.com/atxPraCA dc2: http://pastebin.com/V9JkwbUq Both reports end

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 was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is

It's just come to my attention that newer phones from both Snom and Grandstream support OpenVPN. Is this a new trend or something? Since OpenVPN, in one swell foop, deals with both NAT issues and securing communications, I'd be very interested in hearing if other phone vendors were embracing OpenVPN as well. (Other VPN solutions are all well and good, but I really like the flexibility

Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no "dahdi" command -- not from the base, nor as a subset of the "core" commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? Thanks... -Ken