similar to: Headset Recommendation

Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jmann at txhmg.com

Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 "lines" registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com ________________________________ This e-mail, facsimile, or letter and any files or

I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com

I'm running some mysql queries on the standard sql logging of calls, and am interested if anyone has any they'd like to share to get good statistics. I'm interested in # of calls per day, based on DST. Number of Calls per day based on SRC, avg duration of calls, etc.. Thanks. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999

Any advise on troubleshooting this: Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF It happens nightly, and I have to reset asterisk to "clear" it. Zap/Dahdi channels wont' work

Using the default features.conf setup, if I include parkedcalls in my dialplan, and a call gets parked, and times out, where does the call go? Does it go to a timeout extension in parked calls, or does it go to a timeout extension in the original context? (Using an AEL based dialplan similar to below). -- context internal { ... ... t { jump 600 at

Looking for two things: 1. Anyone that has dialplan logic for an executive assistant. My owners want their extensions to ring on her phone, and be very obvious to her which extension is ringing. They also want her to have presense. She's got Polycom IP 650 with sidecar, they have IP 550 phones. Thusfar I've got her registering to 4 extensions. Each extension is labeled with an

Is there a way to override sip peers defined in users.conf with respect to their context and hints? Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored. Thanks for any help. nurscarepbx*CLI> core show version Asterisk 1.4.22

[priv] type=3Dfriend dbsecret=3Ddundi/secret context=3Dlongdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > >

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

RJ11 Plantronics 1 3 2 4 3 1 4 2 RJ11 Pin 1 is on the left when looking at the contact points. Plantronics Pin 1 is on the left when looking at the contacts (through the plastic sheild) My multimeter battery is low, so YMMV, but: Pin's 1,4 are connected with ~160 ohms Pin's 2,3 are connected with ~1400 ohms On my Plantronics head set. Chad -----Original Message----- From:

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). ________________________________ This e-mail, facsimile, or letter and any files or attachments

Tom - here are a few suggestions Plantronics 510SL Bluetooth - they make a bundle that comes with a Bluetooth transceiver and a handset lifter. This is nice if you have users that already have Bluetooth headsets for their mobile phones, as it allows you to sync with their office phone as well. http://www.plantronics.com/north_america/en_US/products/cat29880043/cat2 9880054/prod5460016

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

On 06/05/2015 03:48 PM, John R Pierce wrote: > On 6/5/2015 1:44 PM, g wrote: >> for what it would take to do so, you are better off getting buying >> better quality at a little more cost. > > I've had pretty good luck with the basic models of Plantronics > headsets. They tend to be well made. The Logitech stuff I've > bought has often broken within a

Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a packet out of? I've got multiple NICs in my box, each with it's own public IP. I need the SIP messages to originate from any one of the IPs depending on which number was originally called(and therefore where the packet originally came from). My fear is that it will listen on all IPs fine, but only respond

I've got a couple of extensions in users.conf that have both SIP and IAX access(IAX softphone, SIP hard phone). I'd like to setup my dial string to "check" to see which they are actively registered with, and send the call appropriately. Right now I have: Exten => _4xx,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN}) But not all phones have both techs, so there is a lot of

Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux 2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz Anyone have a workaround or patch? Error below ==================== Building modules, stage 2. MODPOST CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M] /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.ko make[1]:

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: