similar to: Newbie Polycom: IP601 console with expansion module

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header. > -----Original Message----- > From: Warren (mailing lists) [mailto:warren-lists@icruise.com] > Sent: Friday, October 27, 2006 5:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Taking a

Hello people, I am having difficulties figuring out the POE power draw in watts from a Polycom IP601. I want to know how many IP601's can be powered from the Cisco Catalyst 3560G-48PS. The IP601 wallwart has: Input 120VAC 60hz 19W, Output 24VDC 500mA. I assume the output is appropriate value to figure out how many phones can be powered. The Cisco 3560 datasheet says "the 48-port PoE

Hello, Can anyone provide me with an example of how they have set up their dialplan and handset for a receptionist desk that handles a large volume of concurrent calls? I'm having a problem with transferring calls while several calls are either answered or coming into a receptionist's telephone at the same time. Thanks, Colin -------------- next part -------------- An HTML attachment

Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

Hi, I have a problem with my printing setup of a windows XP client with a samba server. The windows driver seems to use different ways of smb/printer communication for printing in normal/duplex mode and for printing brochures. The latter failes silently. normal/duplex printing uses: SMB Open Print File Request brochure printing starts with: SPOOLSS OpenPrinterEx request I recorded the network

Hey all, Here''s my situation: I have a pair of controllers with associated models (called Services and Testimonials) that are quite similar. Because their CRUD behavior is executed via AJAX, the "templates" for the actions are all short .rjs files. Now, because of the similarity of the models, most of the templates are exactly the same, with only the object names changed. That

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

I'm having a problem with polycom ip601. If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom rings, and stops ringing as soon as I hang up. But if the phone is called via a queue, the polycom continues to ring long after I've hung up. Other phones in the queue (grandstream, cisco) don't have this problem, and stop ringing properly when I hang up. polycom bug

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED & CONFIDENTIAL CLIENT COMMUNICATION    *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may

Hi, Quick question. With an old phone system a receptionist receiving a call has 1 button to push to transfer calls to a specific extension, with Asterisk, a receptionist would actually put the caller on hold, pick up another line, call the extension, ask if the person is available, hang up pick up the caller again and transfer. To me it's seems a long way to simply do a receptionist

Hi! I've got a number of extensions (about 50) on a working Asterisk setup. For each user, I have two extensions configured (for example 11021 for a Cisco 79XX phone and 11022 for X-Lite), and a ring group that ties the two extensions together (for example, 1102). Reason being that if the user is away from his/her desk or working offsite, they can answer the soft phone on the PC. From

Hey all... Scenario (INTERNAL) 1 Call comes in to receptionist and gets transferred to someone 2 No one picks up that transfer 3 Call goes back to receptionist Now when the call goes back to the receptionist, how can I change either the ringer, the callerID or both? * TIA