similar to: Transfer via CTI

I found this solution from the web and figured I'd share it because it affects all phones getting input from IIS. Map .gif, .jpg, .css etc (in my case I used .xhtml for the Polycom 601) in IIS under your sites: Properties -> Virtual directory tab-> Configuration -> Application configuration -> Mappings tab. Make ASP DLL [..\inetsrv\asp.dll] to handle these files. This

Hello all: Can anyone help me with finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/6b26b235/attachment.htm

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

I'd like to setup an autodial that will do the following: 1. First try call my multiple SIP phones at my house - if no answer within n seconds, try my mobile phone. Alternatively call both SIP phones at home + mobile at the same time. 2. Once someone answer in step 1 (on any phone) initiate calling to a specific phone number/extension Currently I have it working except that in step 1 I can

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi When using a call file to place a call I can't seem to figure out how to get the variable alert_info passed to the actual channel (in my case a SIP phone) that an agent is logged in at. Please can someone give me a pointer in the right direction ;) Thanx! Probably best illustrated in an example: Below works great and tells SIP/123 to pick up the call from asterisk then it dials the

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave

Hi, Is there a reliable way to auto-dial SIP phones (specifically Polycom) with some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the user's phone ring, and when picked up dials the desired number, but I (and more to the point, many of my customers) find this annoying. I'd like the phone to autodial on speakerphone (or headset if there is one), without any human

On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield <tony at softins.clara.co.uk > wrote: > > If I do this from an NEC digital extension I get 141496920000, but if I > do > > it from an NEC POTS extension I get 1942124000 > > That looks like when you pick up the analogue phone and dial 9, it > immediately opens the outgoing line and sends the 141 acces code, but >

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Hello list, I try to configure auto dial from asterisk (called server B) to another asterisk (server A) using SIP but I have a strange problem ! (Softphone connected to server B calling clients of server A works) On server B, I have : sip.conf : [to_serverA] type=peer username=from_serverB fromdomain=domainB fromuser=from_serverB host=server_A_IP secret=xxxx insecure=very nat=no

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple