similar to: Internal Extensions Busy

I have been struggling with my Asterisk setup for 3 days now and I think I have done well...apart from the small detail that I cannot dial out on my phone (PSTN) line. My setup is: Suse Linux 9.0 1 fxo card connected to a BT(UK) line 1 Cisco ATA186 sip v3.0 with two analogue phones attached to it Asterix CVS-HEAD-05/30/04-06:56:31 with the UK Userid patch applied. Asterisk loads without any

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

Hi all, the lists seems to be littered with disconnect problems using various equipment (TDM 400,Linksys etc etc.) My question is very simple and could make for good solution to Asterisk users. Since * can detect various tones according to different country standards would it be possible to disconnect on the 'off-hook' warning tone? This tone is: 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz, at

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

hi list, I'm running a newly installed asterisk@home an i registered two soft phone. both soft phone are registered 8901/8901 x.x.x.x D 255.255.255.255 50710 Unmonitored 8900/8900 y.y.y.y D 255.255.255.255 6281 Unmonitored but when I call one another, they are always busy and directed to its voicemail Sorry, if this was posted before TIA

IPManager version 1.6 has just been released. Below is a list of some of the features you will get on your Asterisk server using IPManager to generate your configuration files. Download: http://ipsoftware.thorben.dk <http://ipsoftware.thorben.dk/> PBX Features The following features will be available to users of the PBX if you are using IPManager to configure your PBX. *

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Has anyone seen the following problem? Until recently, I couldn't understand why some extensions on my * system would have a "congestion tone" as soon as I picked up the handset. A little sleuthing through the logs and the source code led me to understand that * thought it had seen the extension go off-hook, send some DTMF tones, and then wait. * treated this situation as a

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

---------- In?cio da mensagem original ----------- De: asterisk-users-admin@lists.digium.com Para: asterisk-users@lists.digium.com Cc: Data: Mon, 12 Jul 2004 11:48:05 -0500 Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide

I have recently built my first asterisk system and am very impressed with its capabilities. However, I have run into one problem that hopefully someone can help me with. I am trying to use the DIALSTATUS function to route incoming calls to the appropriate Voice Mail (busy or unavailable) or to an Unavailable Number recording if the number is not assigned. However, I find that DIALSTATUS

Hi We run a hosted VoIP service for multiple customers off the same server and I'm having an odd issue with just one customer in particular. We're using realtime in a MySQL DB and this is their dialplan *************************** 1. row *************************** context: pcsu-Identifier exten: s priority: 1 app: Answer appdata: *************************** 2. row

I'm sorry if this has been answered before. I seem to remember reading a similar thread a while back, but for the life of me I can't find it (after 30 minutes of intensive googling). I have a voicetronix openline 4 card in an * server running CVS HEAD as of 1st March. Everything is working pretty much as as expected (with a little more echo than zaptel?). The question I have is

We have an Asterisk server acting as a hosted PBX system for many clients, and we're going through an upgrade to Asterisk 1.6 by moving our most important (and complicated) clients one at a time. But we're having a problem with one customer that I really can't explain. I can place calls directly to one phone at the customer's location (they also have an IVR that asks for an

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.