similar to: msic while ringing

what does the parameter -vvvvvvvgrc meanand are there any others as well? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/20195567/attachment.htm

Whenever some call comes in i want it to be automatically picked up and then it plays some message "Welcome to xyz, Press 1 for sales and 2 for support" and then it takes it to the particular extension of sales/support. can i achieve this thing using asterisk? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been googling for the RPM kernel-source-2.4.25-040218.i386.rpm or kernel-source-2.4.25-040218.i686.rpm for the last 59 hrs and couldn't succeed. Can someone suggest me some good Redhat Linux 9.0 rpm repositories. And will the Debian deb work with redhat or not? Kindest Muhamnmad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -------------- Feb 17 10:59:14 WARNING[18726]:

i have got a music file with extension mp3 and it is not workign with background() is there any way to convert the mp3 to gsm or any other codec? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/2ba5a4f0/attachment.htm

My asterisk environment is: ... -> [Asterisk PBX1] -> [Asterisk PBX2] -> [SIP Clients] Where the "..." are the normail landlines from where i am getting calls into my PBX1. As soon as i recieve a call into the PBX1 i use: exten=>BLAHBLAH,1,Dial(IAX2/PBX,10,tr) to forward it to the PBX2 but on the PBX2 side where i am supposed to choose between a number of sip clients, i

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

i have several large .tar backup file on the server it's about 2 GB and 4 GB the question is, if i syncronize using rsync to other computer, rsync will re-transfer the whole 1 big file, or only transfer part of the file ? or maybe i should reduce the size, by splitting the file into 650 Mb each. or it's the same as if transfering via FTP ? thanks Luqman.H ---------------------

I encountered same error when use Xen3.0.2(source file) on CentOS4.2, after I downgraded to Xen2.0.7(source file), this error disappeared. now, I am using Xen2.0.7 instead of 3.0.2, it works well. --taylor -----邮件原件----- 发件人: xen-users-bounces@lists.xensource.com [mailto:xen-users-bounces@lists.xensource.com] 代表 xen-users-request@lists.xensource.com 发送时间: 2006年6月15日 5:24 收件人:

Bad scaling will waste a lot of everyone's time. I put the data in a data frame mdat, then library(nlmrt) mdat<-read.csv("muzzamil.csv", header=T) fmn <- nlxb(y~a * (x^b), data=mdat, start=c(a=1,b=1), trace=T) fm <- nls(y~a * (x^b), data=mdat, start=c(a=1,b=1), trace=T) fmn2 <- nlxb(y~a2 * ((x-1979)^b2), data=mdat, start=c(a2=1,b2=1), trace=T) fm2 <- nls(y~a2 *

On Wed, Nov 26, 2014 at 10:15:13AM +0000, Daniel Sanders wrote: > > From: Daniel Sanders > > Sent: 25 November 2014 17:39 > > To: Eric Christopher; Tom Stellard > > Cc: LLVM Developers Mailing List (llvmdev at cs.uiuc.edu) > > Subject: RE: [LLVMdev] Proposed patches for Clang 3.5.1 > > > > > > > > I'd also like to propose the inclusion of

Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root at astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw---- 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw---- 1 root root 196, 253

Hello! Now i'm using Wine since 2 years and i always played GTA San Andreas with Wine, together with other games. Every Wine version is getting better and better and i saw a lot of games working simply by waiting it was implemented... Wine is now IMO becoming good for new games, they are more and more supported, less things to patch... But games are working, but getting slower and slower! A

On Thursday 07 April 2016 16:11:12 Timo Sirainen wrote: > On 06 Apr 2016, at 17:29, Pali Roh?r <pali.rohar at gmail.com> wrote: > > > > On Monday 04 April 2016 19:35:22 Pali Roh?r wrote: > >> On Monday 04 April 2016 19:31:06 aki.tuomi at dovecot.fi wrote: > >>> Can you please send them directly to me, please? > >>> --- > >>> Aki

On Monday 04 April 2016 19:35:22 Pali Roh?r wrote: > On Monday 04 April 2016 19:31:06 aki.tuomi at dovecot.fi wrote: > > Can you please send them directly to me, please? > > --- > > Aki Tuomi > > Sent. Received? Are patches OK now? -- Pali Roh?r pali.rohar at gmail.com -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc

> On April 4, 2016 at 8:18 PM Pali Roh?r <pali.rohar at gmail.com> wrote: > > > On Monday 04 April 2016 19:12:56 Pali Roh?r wrote: > > Hello, > > > > I borrowed dovecot parser for email addresses and going to use it in new > > perl module as replacement for Email::Address. During implementation and > > testing I found bugs in dovecot parser &

Hello, I borrowed dovecot parser for email addresses and going to use it in new perl module as replacement for Email::Address. During implementation and testing I found bugs in dovecot parser & generator. So I'm sending my patches together with tests. Pali Roh?r (7): lib-mail: message_address_write: Fix generating empty group list lib-mail: message_address_write: Fix generating group

After palying around to set this correctly (or so I think, please correct me if I got something wrong) I decided to share it with everyone. For Polycom phones in sip.cfg change the following line from: <SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="" tcpIpApp.sntp.gmtOffset="" tcpIpApp.sntp.daylightSavings.enable="1"

On Tuesday 05 April 2016 21:31:52 Pali Roh?r wrote: > On Tuesday 05 April 2016 16:54:14 Guenter Roeck wrote: > > On Tue, Apr 05, 2016 at 07:10:07AM +0200, Juergen Gross wrote: > > > Use the smp_call_on_cpu() function to call system management > > > mode on cpu 0. > > > Make call secure by adding get_online_cpus() to avoid e.g. suspend > > > resume

On Tuesday 05 April 2016 21:31:52 Pali Roh?r wrote: > On Tuesday 05 April 2016 16:54:14 Guenter Roeck wrote: > > On Tue, Apr 05, 2016 at 07:10:07AM +0200, Juergen Gross wrote: > > > Use the smp_call_on_cpu() function to call system management > > > mode on cpu 0. > > > Make call secure by adding get_online_cpus() to avoid e.g. suspend > > > resume