similar to: Storing a number to Dial

I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA

Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension => 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension => 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for

Hi all I am a relatively new R user so please excuse this question if it has been covered some where else, just tell me where to find it. I have a simulation model that out puts dates in a standard dd/mm/yy format R reads this as a factor and I cant find anything that will allow me to convert them to a date. In S+ I have used a chron() function that required you to specify the format of the

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hello I'm using your program R in a course I'm taking at the University of Oslo, and thereby I need to download it to my PC. Unfortunately I do have some problems, I do not know whish files to download and how I do it. Can you please send me a list of what to download and a like where to find it? Thanks a lot, best regards Maria Befring Hovda PhD-student Norconserv AS, Seafood

Dear Rdev, in R 2.9.0 the doc of function stars() does not state that it returns invisibly the location of atomic graphs. This is a valuable information as it may help to set a value for the key.loc parameter of this function. My whish is just that the "value" section in stars.Rd should be documented. Best, Pr. Jean R. Lobry BTW, the URL:,

Dear All, I've been?searching for?appropriate codes to compute the rate of change and the curvature?of ?nonparametric regression model whish was denoted by a smooth function?but?unfortunately?don't manage to?do?it. I presume that such characteristics from a smooth curve can be determined by the first and second derivative operators. The following are the example of fitting a

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: ---------------------------------------------------------------------- <--- SIP read from 82.101.62.99:5060 ---> SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

Hello, I'm writing from OpenSC project (OpenSSH used to include OpenSC support for smart cards, it has been removed now and PKCS#11 is used instead, whish is nice), we're planning to have a "Security / hardware crypto keys" themed devroom at FOSDEM next year. Are people on this list interested in participating, and trying to tackle the problem of "Why OpenSSH does not work

Hi list, I try to keep a local file tree synchronized with the remote one. However, using the --delete option does not work as expected or at least as I whish it would. Here is an example of the local tree I'd like to sync: /dirA/subDirA /subDirB /dirB/subDirC On the remote machine I have: /dirA/subDirA /subDirB /dirB/subDirC /dirC Here is the rsync call I use: Executing rsync

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

Dear listers, I have written a function to facilitate the drawing of altitude profiles with x (distance), y (altitude) and a z parameter (altitude magnification). profplot<-function(x,y,z=10,...){ op <- par()$mai par(mai=c(0.95625,0.76875,0.76875,0.95625)) plot(x,y*z, type="l",asp=1,las=1,xlab="",ylab="",yaxt="n",...)

Hi to all, I developed a VoIP application using Speex. Now i want to record the conversation to a file. As i have two indipendent thread transmitting e receiving voice i whish to mix both audio streams in one stream before saving the conversation to a file. I'm using Speex in narrow band mode, 8 bits per sample, mono. i found at this link http://www.vttoth.com/digimix.htm a simple way to

I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [root@tomo ~]# ping sip1.voipbuster.com PING sip1.voipbuster.com

Hi, I was successfully using VoIP Buster via IAX2 for several weeks now. Yesterday/today it spontaneously stopped working. Using the "real" client the connection works well though. Anybody else experiencing this problem? Or asked differently: Is there anybody for whom it is still working? Can anybody tell me what the problem could be from this: -- Executing