similar to: Make a transfer for external line.

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891

Sorry for the very newbie-like question. I have the FXS part straight. The part I don't understand is the FX0 part. Will I need the FX0 card if I am connecting to a service like FWD? My goal is to get rid of my phone line all together. I am under the impression I will only need an FX0 if I'm connecting to the central office side of the phone connection or to an existing PBX. Bottom

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi all, I am using c++ functions in R by Rcpp and inline packages. The code is quite simple, but the R session always automatically crash after some running time. Does anyone here familiar with Rcpp and inline? What¡¯s the problem in the following code? I have checked the input values, no NA and other strange value exists. Thank you for your attention! > mkc <- cxxfunction(

Hi all, I am using c++ functions in R by Rcpp and inline packages. The code is quite simple, but the R session always automatically crash after some running time. Does anyone here familiar with Rcpp and inline? What¡¯s the problem in the following code? I have checked the input values, no NA and other strange value exists. Thank you for your attention! > mkc <- cxxfunction(

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

Dear all, I have an FX0 card installed in * and connected to a PBX. Calling works ok ( both in bound /out bound) but after the call, I have to press the '#' key to terminate the call, otherwise the line stays busy. Anybody has a fix for that? Thank you. Anthony

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

We obtained some disturbing results from convolve() (inaccuracies and negative probabilities). We'll try to make the context clear in as few lines as possible... Our function panjer() (code below) basically computes recursively the probability mass function of a compound Poisson distribution. When the Poisson parameter lambda is very large, the starting value of the recursive scheme ---

Hi: I have a machine with four interfaces connecting four different networks. I am learning to use nmap and trying to force the nmap working only one interface. As nmap man page states, I use -e option and it would not work: nmap -e fx0 -v -sP 192.168.128.0/23 Starting Nmap 3.95 ( http://www.insecure.org/nmap/ ) at 2006-02-04 14:04 CST getinterfaces: Failed to open ethernet interface (el0)

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

Hi I have installed 2 fx0 cards into a machine 1 for pstn (out) and the other for pstn (in) However I can get the first card (pstnout to work fine) I have swapped the cards and both work I am also unable to make outgoign on the second card ztcfg -v shows r2d2:/etc/asterisk# ztcfg -v Zaptel Configuration ====================== 2 channels configured. My /etc/zapdata.conf file # fxsks=1,2

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

Hi, I am trying to do the world's most simple install. I have a Wildcard TDM400P with 3 ports: 1 FXS on port 1 and 2 FXOs on ports 3 and 4. (i'm not using port 3 for now, put want it for expansion purposes) I simply (to start with) am looking to have the FXS phone ring when an FX0 port is dialed. I would also like to be able to place outgoing calls on the FXS through the FXO. Right

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer