similar to: asterisk-users Digest, Vol 82, Issue 27

Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 => 1234,Gama Operator,Operator at gama.com 500 => 1234,Operator,Operator at gama.com 501 => 1234,Employer Name,employer_email at gama.com 502 => 1234,Employer Name,employer_email at gama.com Asterisk version is 1.8 and currently I am getting this

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Appreciate the kindly help and advise. Regards Bilal --------------------- > > Bilal, > > I suggest you turn on logging on your tftp server to see > what files are actually being requested, and if the the tftp > server is dishing them out... Try adding a few v's to your > tftp setup: > > File: /etc/xinetd.d/tftp > Line to change: server_args = -s /tftpboot -v

Dears; To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install? Or I need to compile the dahdi and asterisk also? If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version? Regards Bilal ----------- > bilal ghayyad wrote: > > But I am afraid it is a bug because I

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

Dears; I am looking for the card that does not need an electrical power, which one? Is the PCI express doing this? Regards Bilal -------------------------- > While we're at it, can someone please tell me whether I > should be using > vi or emacs? ;-) > > Many thanks, > > Tom > > PS: Bilal: You have asked a nearly unanswerable question. > Some prefer >

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

+ > p2<-function(r){ + gama=0 + for(i in 1:1000){ + c=caminho[[4]] + for(i in 1:caminho[[3]]+1) { + c=c+caminho[[i+3]]*((r[i])^(i-1)) + d=(abs(c))*exp(-(x^2/2))} + gama=gama + ( d/(h(r[i])) ) } + return(gama)} > e3<-p2(r) OBS: r is a rnorm(1000,0,1) > caminho theta_chapeu f_estrela k a0 a1 a2 a3 1 3.2 1.2 3 2 1 4 5 > question i wanted gama to be

I got one email from eric asked me to Lower the rxgain and txgain on your Zap channels. But actually it is already the voice volume is low and I was looking to increase the gain (currently it is 0.0), so I do not know if eric was mean to reduce it less than 0.0, but I can not do that due to the low volume that is already existed, so any more reduce will make the voice not hearable well, even if

Regards Bilal ------------------------- > El 18/07/11 18:03, bilal ghayyad escribi?: > > Dears; > > > > If I need to login using as agent using the > AddQueueMember(team,....) then what to be the second > paramter? How to be written? > > > > For example, if the agent id is 8000 then it will be: > > > > AddQueueMember(CustomerSupport,Agent/8000)

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Hy, can anyone recognize that pixel noise in the playbackfile recorder file: http://www.megafileupload.com/en/file/135429/FMODTestRecording-wav.html playback file: http://www.megafileupload.com/en/file/135431/FMODTestPlayback-wav.html i have no idea what that is anymore. i try everything i know, from changing the way of copying data to different encode/decode algorithms the recorded file is

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

Hy, I'm totaly out of ideas now. here are links to the code I use. codec.cpp http://barvanjekode.gama.us/temp/1078354945.html codec.h http://barvanjekode.gama.us/temp/135707080.html Variables I use are: int samplerate 32000 uint quality 10 uint complexity = 2 I get that wierd noise after I use speex encoder/decoder. It's like there where empty spaces between each encoded

Dear Doug; But I am afraid it is a bug because I read something this in the below link: https://issues.asterisk.org/view.php?id=17270 But maybe this was for old driver .. again, I am afraid if it is a bug. DAHDI Version: 2.4.1 libpri-1.4.11.5 Any advise if the below message is a bug? [Jun 15 16:14:00] WARNING[2773]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to