similar to: Call Progress Analysis

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the <transfer> tag and setting bridge="true" (i.e <transfer name="transfer1" bridge="true" connecttimeout="10s"> ) but as soon as

Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe000000 (32-bit, non-prefetchable)

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

Greetings listers, I've got 1.4.21.2 using Polycom 501 phones and Zap lines. Most of my calls come in and go out fine with the exception of Mechanized answering devices. When I call my 401K plan (1-800-777-401K) the call will last exactly one minute. The call never bridges, so even though the connection is made, Asterisk hangs up at the end of the Dial command.

Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its

How can I stop the following warning from occuring? invalid factor level, NAs generated in: "[<-.factor"(`*tmp*`, iseq, value = structure(1, .Label = "12", class = "factor")) The Label messages are for "5", "8", "12" and "46". I want the NAs to be generated as needed. Is this causing R to slow down by generating the warning

dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] exten =>

NENA i2 The NENA i2 architecture was designed to support the interconnection of Voice over Internet Protocol (VoIP) domains with the existing Emergency Services Network infrastructure. This overview will describe the functional elements and call flow of a VoIP 9-1-1 call over the i2 architecture. The NENA i2 architecture was also designed to utilize existing 9-1-1 voice and data links to all

Hi I'm having a problem with ipfw, I think. For some reason it denies packets randomly for example: PING 196.14.239.2 (196.14.239.2): 56 data bytes ping: sendto: Permission denied ping: sendto: Permission denied 64 bytes from 196.14.239.2: icmp_seq=2 ttl=63 time=0.258 ms 64 bytes from 196.14.239.2: icmp_seq=3 ttl=63 time=0.233 ms 64 bytes from 196.14.239.2: icmp_seq=4 ttl=63

Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and