similar to: Polycom IP500 - problems with multiplesimultaneous calls

Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:

Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When there is an incoming call the phone will display two buttons "answer" and "ignore". If you press "ignore" the call is dropped instead of sent to voice mail. The following is the log: -- Called 111 -- SIP/111-00001c14 is ringing -- Got SIP response 486 "Busy Here" back from

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

hello, i can't figure out how to set position of panels of my stripplot - i`d like the panels of one level of the factor stage (nr. of panels within each stage, A: 12, B: 12, C: 12, D: 4, each panel representing a site) to be in one column, with A to D from left to right and with descending site.nr at each row. like: A1 B1 C1 D1 A2 B2 .. .. A3 .. .. .. how is this achieved? any help

On my 7905s I can configure a "voicemail" number, which in turn activates a "Messages" softkey. When pressed, it goes to that exten, which in turn is configured to go to VoiceMailMain. Great, works like a charm. However, the phones also have a "go to voicemail timeout" after which, the phone diverts the call to voicemail via a "temporarily moved". This

Hi All, I have a data frame with a group of variables named b1, b2, b3, ..., b18. These variables take the value 1, 2 or NA. For each observation, I want to do some computation by looping over the values for the group of variables: b1 to b18. In STATA I would do: forval i=1/18 { --- use b`i' for computation ---- } How can this be done in R? Deepankar

Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

Hi, I just got and setup a new ip500 yesterday and it worked for about 15 minutes. Then it froze during a reboot.Now, when power cycled, the logo comes on for 3 seconds and then the screen is blank and nothing further happens. 468* factory reset doesn't work. I am about to send the phone back, but wonderd if anyone had a suggestion first.

I've had an IP300 for a while now and it's been working fine. I just got an IP500 and when it connects to the FTP server it downloads the new bootrom and says error loading. The bootrom is fine and works on the 300... In addition, I downloaded a new copy to be sure and it still doesn't work. Can anyone give me some advice?

I've been looking at Asterisk for a while now, and want to set up a small installation at my house. I did originally want to get a Cisco IP phone, but due to price and it not being easy to obtain the SIP firmware I have decided again them. I am very interested in the Polycom IP500. I can find many places in the USA that sell it but nowhere in the UK. Also can anyone recommend some good

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

Greetings, I'm wondering if it's possible to display line breaks with caller ID display. I have the Polycom ip500 phone, and what I am trying to accomplish is instead of the phone saying 'Incoming call from: name/number' i want it to appear on the phone like this Incoming Call from: Menu Context last in Name Number I tried using \n and \\n between the variables (${VAR} \\n

Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

tuning for ulaw g.711 - Polycom IP500 I've got QoS ironed out at this point (dedicated section of bandwidth with VoIP having priority), yet ulaw is still unusable due to small 'holes' in the audio. Running iperf tests shows packet loss at very low percentages (< 0.02 %), and when we do drop, it's only 1 packet at a time. I thought ulaw was supposed to handle this OK.