similar to: running asterisk as non-root

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

Dear All, Just hope your guys out there can help me through..since I've been playing for serval hours....and still not able to resolve it... The workflow as I've created an .call file for Asterisk to call out and it's working fine with outdial, passing variable to asterisk..But the problem is when the calls reached Context and execute AGI script, the script didn't get any

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

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

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

I have an application that requires I be able to dial into an asterisk box, then from there dial out to another user through a PSTN. I'd like to be able to both 1) record this call and 2) let another user dial in using something like ChanSpy to listen to the conversation. I can get this working by executing an auto-dial script to connect one end of a call to an outside Asterisk box which

Hello, I am using asterisk to autodial a phone number once an hour to verify an answering IVR script is working by listening for a beep played by the IVR, and that is working well. The one thing I have not been able to find is a way to trap if the call didn't connect at all, like if the entire IVR is down or the phone line is dead, since it doesnt enter the menu until the call is

Hello, I'm writing a small PHP application that generates calls automatically and tries to store call details on a Mysql Db, using manager API . When making an autodial call, I noticed that I couldn't read $DIALSTATUS values; since I can't evaluate dial status (BUSY, CONGESTION, NOANSWER), I can't understand when a receiver was busy or not. Nobody seems to have solved this

http://bugzilla.mindrot.org/show_bug.cgi?id=109 ------- Additional Comments From andrewb at cs.mcgill.ca 2002-02-12 08:16 ------- Silly me I forgot to mentiont the version of Solaris that we are running. SunOS 5.7 Generic_106541-14 sun4u sparc SUNW,Ultra-25 Sorry about that. Andrew. ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching

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 used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk "outgoing" directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York

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 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

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 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