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 ---------------------

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 *

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

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:

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?

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

describe MyController do integrate_views describe "A common base senario" do it "no longer integrates views" do be_careful end end end integrate_views affects an attribute in the class formed by the describe factory method. The second describe generates its own class, so integrate_views is OFF at that level. I''ve already spent far, far too much

Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much

Hi, I am just a little confused of mian effect in the analysis of variance (ANOVA) when you include or do not include an interaction term. Let's assume a simple case of 2-way ANOVA with 2 factors A and B, each with 2 levels. If it shows that main effect for A is significant when the interaction between A and B is NOT included, and the main effect for A is NOT significant when the interaction

Trying a different approch. Senario Raid 1 setup Bootable raid drive failed Mirror has been working for almost a month and then rebooted Now can't boot mirror drive grub not mirrored from other drive. I Fixed bootable drive. Question? Can I hook up both drives and boot fixed drive then rebuilt mirror from nonbootable drive to bootable drive? Does the raid automatically rebuilt when I

Hi, Not sure exactly how I would word the subject line so appologies in advanced. We are trying to accomplish the following scenario: Location 1: PDC: fs01.loc1.example.com IP: 10.0.0.1 Location 2: SDC: fs01.loc2.example.com IP: 10.0.1.1 Clearly when we join the SDC to the PDC there is a naming conflict. The end result would be to have clients at each site resolve the fs01 name to

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already