similar to: how to backup asterisk installation for upgrade

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Is there a key sequence to stop a call as its ringing, before the call is answered? The * key stops a call after it is answered, but I'd like a way to cancel the call during the ringing phase. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

My CDR is displaying wildly inaccurate results. When I make a call the CDR records the time between connecting into the server and hanging up, instead of recording the time between dialling from the server to the PSTN destination via VOIP termination. It is alright to log the duration of the connection to the server, but why it does not log calls for termination via voip provider is the main

I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The

I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. 2. what directories and files it must have read and right access to 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Are these some info sources which go into these areas in depth?

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix

I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. 2. what directories and files it must have read and right access to 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Are these some info sources which go into these areas in depth?

I have compiled the OH323 module for my system. When can I find some info on how to properly configure it? I haven't read any info for its configuration, and I need some starting info. Were do I start? Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix

I am interested in how much CPU and RAM asterisk requires for call handling. 1. What is the minimum CPU required for asterisk to manage 60 concurrent calls without transcoding. 2. Handle calls on a 75% no transcoding, 25% transcoding 3. How many calls can it connect per second ie from one VoIP -> VoIP? All the above refer to a VoIP setting. 4. Is there a difference between bridging and

Hi folks, I'd like to make a full backup up of my server, about as near to a Ghost or Acronis type of backup as is possible using programs on the server itself. What type of rysnc script can I use for it? I know Linux has some pseudo directories and files that are not part of a backup, will I have to exclude from the backup, or can rsync automatically handle them? Does rsync have option

Hi I'm working on the XCore target and am having difficulty building libgcc. Background: If I use a libgcc built by llvm3.0-gcc with my current clang-llvm3.3 compiler, exceptions 'seem' to work. Trying to rebuild libgcc however breaks exception handling - they aren't caught! I thus assumed I needed to focus on the unwind code and particularly functions that call

How does TDM work, how do you connect to it? I have the impression it can't be routed like ethernet, but a cable from your switch has to be plugged into the providers equipment. I have seen the Asterisk info about TDMoE - does this mean that the Asterisk card will modulate the signal on the Ethernet cable to allow it plug directly into a proper TDM connection? Will someone please enlighten

How does one check what codec translations are in use in a call? I am connecting to sip system which says 488 "4XX Not Acceptable Here". I don't know what is stopping the call from being accepted and I'd like to know if there are codec issues involved. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet

Hi Rowland, On Sun, 26 May 2019 at 19:54, Rowland penny via samba <samba at lists.samba.org> wrote: > On 26/05/2019 18:28, Viktor Trojanovic via samba wrote: > > Small update: I added "lo" to the interfaces in smb.conf, now smbclient > > works also on localhost. But both the other problems remain as described > > for now. > > > > On Sun, 26 May

Hello all, I have a question. I have an asterisk box setup with an x100p card installed. I have about 3 VoIP phones connected to it. The x100p is connected to my pots line. I also have regular phones connected the the pots line as well like most normal people would. When I get an incomming call all the regular phones ring and my VoIP phones ring. This is good. My question is this. Say

The code has changed a lot over the years. Looks like at some point of time the assumption was broken. calculateCallsInformation() may have eliminated the pseudo set up instructions already. // If call frames are not being included as part of the stack frame, and

Thanks, I'll look into that. Still, the case where the function does not call anything remains---in such a situation there are no ADJCALLSTACK pseudos, so regardless of what that function you pointed at does, there won't be any target-independent information about the SP adjustment by the time the frame index elimination runs. Would it make sense to have ADJCALLSTACK pseudos every

Consider this example: --- ex.ll --- declare void @bar() ; Function Attrs: nounwind optsize define void @main() { entry: %hin = alloca [256 x i32], align 4 %xin = alloca [256 x i32], align 4 call void @bar() ret void } ------------- Freshly built llc: llc -O2 -march=x86 < ex.ll -print-before-all # *** IR Dump Before Prologue/Epilogue Insertion & Frame Finalization ***: #