similar to: IAX2 to SIP

Hi, I'm setting up multiple asterisk servers and trying to do the classic DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15) After googling a bit, I fell on a discussion about putting this in a variable so that adding additionnal servers would be easy. I can't seem to find the link anymore, but it went something like this: extensions.conf:

Hi everyone, I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me. I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating). I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hi, > R.version.string [1] "R version 2.0.1, 2004-11-15" I have found two possible bugs in function validObject. Details below. Cheers, Scott ================================== Scott D. Chasalow Principal Statistician Statistical Genetics and Biomarkers Bristol-Myers Squibb Company Email: scott.chasalow <AT> bms.com ================================== ------ BEGIN R

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

What I am trying to do is the following: A call is sent to the * box via a SIP invite. The * box answers via an IVR menu system with " Enter the extension you want to dial" so I enter in my 5 digit extension and get the below message. Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No channel type registered for 'SIP)' Jan 18 10:10:03 NOTICE[-1380238416]:

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Dear friends we have the following problem. We have Operating system=OpenBsd installed computer and an other one with windows 98 when we change the password via smbpasswd myname ... then we have the following problem. My sambapassword is really changed. Since when I try to communicate samba server through windows 98 installed computer I see there is no problem. (Since we can see the shared

I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found ==

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Dear Rusers, I have some data in .csv file like "2004-8-1" and "2004-10-1", and i need to convert them into days from the origin (January 1, 1960). I have tried the function date.mmddyyyy(), but cannot get it. Anybody can show me how to handle the date data? Thanks very much! My dataset like: time 2004-8-1 2004-10-1 2001-9-1 2002-9-1 -- With Kind Regards,

Dear Ruser, I use abline() function to add the reference line successfully, but i can't display the values corresponding to the reference line on the x/y axis, anybody knows how to display it? *My simulated programs:* y<-rnorm(50) plot(x,y) abline(v=0.5) *#my question is how to display x=0.5 in the x axis?* Thanks in advance. -- With Kind Regards, oooO::::::::: (..):::::::::

Hi ! While browsing Linux manpages (man 3 syslog) I noticed that the manual says that the LOG_AUTH facility is deprecated use LOG_AUTHPRIV instead. Is there a good reason why OpenSSH doesn't have an option to use LOG_AUTHPRIV facility ? (Looks like that tcpd/telnet etc. use the AUTHPRIV facility (in RH6.2)). Shouldn't be too hard to add the AUTH_PRIV facility ? Cheers, -Jarno --

Dear friends, The following is an example to explain my question. I want to get a legend which will show the z-values according to different colors in image() function. x<-sort(runif(10)) #x-coordinates y<-sort(runif(10)) #y-coordinates z<-matrix(runif(100),nrow=10) #attributes values image(x,y,z,col=gray((6:3)/6)) # legend(x,y,legend=z,col=gray((6:3)/6)) #error. the colors should

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

I can not dial out Error 1242952640]: File channel.c, Line 1642 (ast_request): No channel type re gistered for '' NOTICE[1242952640]: File app_dial.c, Line 516 (dial_exec): Unable to create channel I believe this to be the device in phone.conf Am I heading in the right dirrection ??? Regards Mick