similar to: noice sip to sip only???

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 want to setup a senario in which the callers hears to some music file while the phone is ringing and as soon as the line is answered the music is stopped palying. i.e. instead of the rings the caller listens to some music. Is is possible with asterisk? Kindest Muhammad 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

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

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

And someone goes and redesigns the look and operation of the Trac. Quite noice, but a couple of comments: * The main toolbar ("Wiki", "Documentation", "Timeline") ends up under the puppet logo on the left hand side, which is a pest. * I can''t click on any link, text box, or button in the main part of the page. All of the navigation links work fine, and I

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

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next

Hi I use acts_as_ferret on an app that is in Danish and English. In Danish english words like "and" and "under" has meaning. Is it possible to make ferret search for these words? As it is now a seach for "under" returns nothing even-though I know the word is present in the index. Cheers Mattias

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

Hello all! First of all, this is my first post to the list. I've tried to find my answers in the forums and by Googling , but no luck. My apologies if this question has been answered before. I've set up an Asterisk box with four local SIP users. The Asterisk box uses a SIP provider for placing external calls and receiving incoming calls as well. In other words, there's no PSTN

Hello all, I am facing as serious problem when running asterisk in HP server.We are developing application to make the outbound calls in PRI lines .We normally uses IBM machine as our servers ,and it was working fine for all installation.For the cost reduction we this time tried with HP server. Model(HP proliant ml110). When we make the calls the there is a lots of disturbance in the sound even

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Aaaargggg.... The S-1-5-18 AND sid S-1-5-32-544 did not resolve. But sid S-1-5-32-544 first not then later on it works.? Sorry about the noice, but that one i wanted to point out also. I hate it when im almost done with typing and mr Penny comes first. ;-) :-p Greetz, Louis > > > > > -----Oorspronkelijk bericht----- > > Van: samba [mailto:samba-bounces at