similar to: Relaying calls to another SIP extension

I have a repeated error in the asterisk console. Incoming call: Got SIP response 400 "Bad Request" back from 10.0.2.136 It is repeated every few seconds and never stops until I restart Asterisk. It starts after I make a call to a certain extension, 202 which transfers the call to another extension all phones then hang up. The SIP trace looks like:

Firstly can I ask when the documentation site will be online again? I'm struggling here without it. Further to my recent post I have tried to simplify things a little. I have used a VoiceXML app to simple call an asterisk extension. EG: <form id="transfer"> <block> <call name="xfer" dest="sip:101 at 10.0.4.147:5060"/>

I have a SIP voice server which I want to place an Asterisk server in front of to handle call routing. At the moment I can call the apps on the server fine, but it cannot transfer to another extension via asterisk. Even attempting a call from the voice server to an asterisk extension results in a "407 Proxy Authorisation Required" error on Asterisk and a bridge_reject

Hi, I am building 2.1.0 (re-release version, if that matters) on a Windows XP machine. Following the instructions 3.1 "Building from source" in R-admin.html, I managed to get up to 3.1.6. But when I try to build the recommended packages, I get C:\hiro\codes\proj\R-2.1.0\src\gnuwin32>make recommended --- Unpacking recommended packages ---- VR make[1]: *** No rule to make target

hi, can anyone confirm the following problem? when i do dos> rcmd rd2dvi --pdf my-package-name i get dos> Can't open perl script "c:\PROGRA~1\r\rw1091/bin/rd2dvi": No such file or directory might the problem be in (double back slashes rather than forward slashes) R-1.9.1\src\gnuwin32\front-ends\rcmdfn.c(251): strcat(cmd, RHome); strcat(cmd, "/bin/Rd2dvi.sh");

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

in ?arima (R-1.9.1), the return value component 'convergence' should be 'code'? (it's a pity there is no reliable way to check return value documentation consistency with the code, or is there?) h. ---------------------------------- Hiroyuki Kawakatsu School of Management and Economics 25 University Square Queen's University, Belfast Belfast BT7 1NN Northern Ireland

in ?arima (R-1.9.1), the return value component 'convergence' should be 'code'? (it's a pity there is no reliable way to check return value documentation consistency with the code, or is there?) h. ---------------------------------- Hiroyuki Kawakatsu School of Management and Economics 25 University Square Queen's University, Belfast Belfast BT7 1NN Northern Ireland

hi, in 1.9.1, the return value from ARMAacf(pacf=TRUE) is not named by lags, contrary to ?ARMAacf. the simple fix is to move names(Acf) <- down after if(pacf), with an appropriate starting lag as pacf=TRUE appears to start at lag 1 (whereas pacf=FALSE starts at lag 0). for consistency, one could argue to append 1 for lag 0 for pacf=TRUE (or start pacf=F at lag 1). however, given the

hi, i noticed that se.fit from predict.lm is the same whether interval="conf" or interval="pred". it is not clear to me from ?predict.lm whether this is intended or not. i suggest that se.fit should match the type of interval requested, if interval is specified. suggested change in lm.R line 700 if(se.fit || interval != "none") se <- sqrt(ip) to if(se.fit

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

Hi: Let me see if I understand what you're doing. You're using winamp (or whatever) to stream to a shoutcast server which pushes a relay on to live365 (or live365 pulls from shoutcast, same deal really). Now you've got a second winamp with samcast trying to pull shoutcast or live365 and pass it to icecast. Is that right? OK. Firstly, there is no source connecting to icecast. The

I think my sendmail is not correctly set. It says the default is to disallow relaying. I am running centos 4.8 and I think sendmail is relaying for anything. Based on this: http://www.sendmail.org/~ca/email/chk-89f.html#RELAYING I added|: FEATURE(relay_hosts_only) to my sendmail.mc file and restarted sendmail. I am still seeing connects that get relayed and are not in my /etc/mail/access

At 10:06 AM 9/4/02 +0200, you wrote: >Hi, > >Can someone enlighten me, how does icecast2 stream relaying work? Is it >same (pull-style) setup like in icecast 1.x? in the config file, I see >elements like: > > <master-server>127.0.0.1</master-server> > <master-server-port>8001</master-server-port> >

Does anyone have a working configuration of relaying using icecast-2.0.0 and ices-2.0-Beta4? Also, anyone using multiple streams from the master server to the slave? I had relaying and multiple streams working great using a CVS download sometime ago and tried setting it up with the latest builds and it is appears to work, but no sound is coming from the slave box. Also, some of the syntax has

Hi, Can someone enlighten me, how does icecast2 stream relaying work? Is it same (pull-style) setup like in icecast 1.x? in the config file, I see elements like: <master-server>127.0.0.1</master-server> <master-server-port>8001</master-server-port> <master-update-interval>120</master-update-interval>

On Tuesday 11 May 2004 11:28, Ian Andrew Bell wrote: > I also tried installing Icecast 1.31 ... the config file claims its > relaying capabilities aren't very well tested and I can verify this. > But now I'm concerned that even 2.0 won't work for me in relaying. > > I'm trying to relay multiple streams from a cluster these hardware > encoding devices: > >

Hi; Has anyone successfully relayed an Icecast stream using Shoutcast? I realize that this is backwards! However, several organizations that want to use our streams use Shoutcast, I really don't want to setup shoutcast and Icecast, but unless I can relay an Icecast stream with Shoutcast, I'm going to have to. Thanks, Fred -------------- next part -------------- An HTML attachment was

Well, I've hit another problem: Icecast crashes if I define more than 10 listen-socket ports. I have 44 streams that I'm trying to relay via Icecast. It works just fine until I try to add the listen-socket port definitions. If I put the 11th one in, it will crash on startup without entering anything in the log files (log level on 4). I've use sysinternal tcpView to verify that the

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling