similar to: Plantronics and Snom RF feedback

I have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset everytime you finish a call. I have even purchased the Aastra 9120 which sais it has on-hook

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

I have a system that is using Intel HDA as its output, and a Plantronics USB headset as the microphone input. I would like to have my wine configuration emulate this setup, I currently have been unable to confirm that it isnt... I just cant get Steam to choose the plantronics headset as the default voice device (this is the goal). Would appreciate any insight!

Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions?

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of J. Oquendo > Sent: Thursday, April 26, 2007 6:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk brute force watcher (was FYI) > > Steve Totaro wrote: > > I suspect that

I am looking for SIP with power over ethernet desktop phones that will work with asterisk and Plantronics HL-10 Handset Lifter for Remote Answering. Any suggestions? I am considering buying about 150 of these desktop phones for a new call center.

Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061122/054eb688/attachment.htm

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

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:

So I have my extensions.conf (http://www.infiltrated.net/exten.stupidity.conf) shortened in case someone wants to look. Has someone encountered the following? I've racked my brain on this for too long... I have two contexts, day and night... Caller (Daytime) --> Dials an extension --> Caller hears extension ring on receiver --> Call goes through Caller (Night) --> Dials an

Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'

So I was bored yesterday and tried solving a few problems with one stone: 1) Notify me of potential brute forcers (multiple attempts to register multiple numbers from one address) 2) Notify me of (l)users who are having password issues So I whipped up a simple script to run in cron and notify me that UserX from X_IP_Space had X amout of password issues. I'm currently running this from cron

Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also,

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one

Curious to know if anyone's created something similar to the following, if so and you'd care to share an AGI or dialplan, much appreciated. I will be eventually write a script to answer the nTH call. (if I can't find it (why reinvent wheels). Looking to do some testing sending anywhere between 50-200 calls to a machine. I'd like a Snom/Polycom/whatever to pick up after the nTh