similar to: R: music on hold error

I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani

Dear R-help, I have a data set consisting of measurements made on multiple subjects. Measurement sessions are repeated for each subject on multiple dates. Not all subjects have the same number of sessions. To create a factor that represents the session, I do the following: data <- read.csv('test-data.csv') # data appended below data$date <- as.Date(data$date,

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give. I use realtime asterisk. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

Hello, This problem is regarding the configuration directive called 'LoginGraceTime'. Problem Description: Tests were done with OpenSSH -3.6.1p2 and 3.7.1p2 on HP-UX. sshd is started with LoginGraceTime as 1 minute.Three windows were used to initiate the ssh client.After launching two clients wait for a sometime without issuing the password so it exceeds the grace period for login.when

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG

has any one installed this, i just tried this on a test server, i get voice but it's corrupted, i do not get the natural voice any idea why -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/af5a3bb2/attachment.htm

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

On Sat, 24 Jul 2010 13:41:42 +0200 Alexandre Colucci <timac at timac.org> wrote: > Hi, > > I am currently studying the possibility to make the transition from > gcc 4.2 to llvm-gcc 4.2 for the projects I am working on. Since you are switching compilers, why not switch to clang instead of llvm-gcc? Best regards, --Edwin

Good morning to all, I work for a bank in Italy, I want to know if i can install R and relative add on like Rbloomberg for free or my company has to pay some fee. tanks to all. Stefano Colucci ------------------------------------------------------ Scegli infostrada: ADSL gratis per tutta l?estate e telefoni senza canone Telecom http://click.libero.it/infostrada

I opened a bug regarding this problem. It seems that some patches were applied in 20050324 that messed up the mouse registering clicks. For Diablo 2, the mouse isn't rendered to the right/bottom 1/8th of the screen. And in both Diablo 2 and Warcraft 3, it seems that mouse events aren't being processed in those areas. I've opened a bug: http://bugs.winehq.com/show_bug.cgi?id=3067

Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-00000000' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines,

Hi, llvm-gcc 4.2 generates warnings when I compile inline assembly code that contains used labels with -Wunused-label. The generated code seems to work yet. gcc 4.2 doesn't generate those warnings. I haven't found any bugs regarding this issue in the llvm bug database. Does anyone know if this is a known llvm issue? Is it a warning that I can ignore and does not affect the generated