similar to: Need ISDN call generator

On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the carrier) with the ability to go to pass through should the Asterisk PBX (software or hardware level) fail. I did not see this feature in the Digium, Sangoma, Allo, or

Hi all, We know that 'ItaniumDemangle' has been added into llvm lib to resolve demangling related issues. And if we dump mangled function name '_Z3duti' into 'char *llvm::itaniumDemangle(const char *mangled_name, char *buf, size_t *n, int *status)', we will get 'dut(int)'. So my question is: How can I just get the function name 'dut' without parameters,

Hi Fangqing, llvm::itaniumDemangle() doesn't yet provide this functionality, but I'm working on adding it in now for the LLDB folks (http://lists.llvm.org/pipermail/lldb-dev/2018-January/013186.html). I should have a patch up Very Soon! If you would like I can CC you to it. Thanks, Erik On 2018-03-16 2:15 PM, Fangqing Du via llvm-dev wrote: > Hi all, > > We know that

Hi, I found the source of my problem with set_nt_acl_no_snum: fset_nt_acl returned NT_STATUS_INVALID_OWNER (I hope). The problem is dut to winbind. In my ldap tree, I have uid and gid. Why samba 4 don't use theses informations ? In attached file : output of classic-upgrade (debug level 5) I don't want to use winbind. I just want to use pam_ldap or nslcd for have unix information

Hi Karen, Thanks much for your comment and for sharing of your experience. icFuzz has a core that is "really" random, but does not cover the entire C space. The tool was designed from scratch to be extensible, and comes with a couple of extensions that target some of compiler optimizations optimizing compilers typically do: CSE, loop interchange, vectorization, etc. But even in the case

On 1-May-09, at 8:35 PM, Chris Lattner wrote: > I still don't understand why this is a problem, but I decreased the > default to 2 bits. Please verify that this helps, I think I've figured out what's going on, and why no assertions are caused by this. It doesn't necessarily have anything to do with User* pointer alignment per se (although I believe that could still

On May 1, 2009, at 3:40 PM, Stefanus Du Toit wrote: > Hi Nicolas, > > Looks like Preston and I have found the cause of the problem. The > issue is with PointerLikeTypeTraits<T*>::NumLowBitsAvailable. This > is set to 3, which basically assumes that unless the traits are > specialized for a particular pointer type, objects of that type are > allocated with

hi, I'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I

Hello everyone, I can not get the name of the recoding file of agents calls. I set agents.conf as following: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ; The format to be used to record the calls (wav, gsm, wav49) ; By default its "wav". ;recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By

dear R users, I am using sweave to generate report for my data analysis. I recently updated R ro 2.8.0, and now I have the following results when compile the the tex file generated from R. This is pdfTeXk, Version 3.141592-1.40.3 (Web2C 7.5.6) %&-line parsing enabled. entering extended mode (./Lajos.tex LaTeX2e <2005/12/01> Babel <v3.8h> and hyphenation patterns for english,

Hai, Thats a setup i really dont advice. Should work, but expect problems. You can happy run both in VMs. Can you explain maybe why your setting up like this, so we can maybe help you to get a better setup? Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens Stefan G. > Weichinger via samba > Verzonden: woensdag 14

I want write article by russian language using Sweave. For cyrillic text LaTeX use T2A encoding \usepackage[T2A]{fontenc} But in Sweave.sty we find: \RequirePackage[T1]{fontenc} It is source of critical problem. For example Rnw file $ cat estimation.Rnw \documentclass[A4paper]{article} \usepackage[T2A]{fontenc} \usepackage[utf8]{inputenc} \usepackage[russian,english]{babel}

I did a test shown as following diagram. (The DUT is running Linux with NAPI ethernet driver.) I add 1000 routes into DUT then use the Smartbits port1 to send packets (with destinations to all the 1000 networks) to the DUT then receive them back from the port 2. The packets are sending in a rate that is much lower than half of throughput of the DUT. The DUT will lose some packets if the DUT has

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Hi, I'm trying to get Asterisk working with the AVM B1 card. I've tried every instruction set I can find, but to no avail. I think I'm getting mixed up as to what Zaptel/CAPI configuration to use. If someone is currently using one of these cards, would it be possible to mail me a few bits of the important config files? Sorry if I should have included something else too, but let me

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten M?ller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the