similar to: cant get incoming calls in asterisk

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

Package: xen-hypervisor-4.0-amd64 Version: 4.0.1-5.2 Hi, We're seeing weird behaviour regarding network traffic between a Xen dom0 and domU. I reported this issue yesterday to the openvswitch-discuss mailing list, but it seems this could also be a (regression?) bug in the xen hypervisor or the linux kernel... (we've upgraded xen-hypervisor-4.0-amd64 4.0.1-4 -> 4.0.1-5.2 and

I use rsync with cmd line: "rsync.exe -vrtlcd /cygdrive/c/test/rsyncroot/111111/001/ 127.0.0.1::root/111111/001 --delete " It reports me: rsync: mkdir "111111/001" (in root) failed: No such file or directory (2) Looks like it will not craate the non-exist parent directory . How to resolve the problem? thanks in advance for any information. -- ÀîÖ¾¼á

<div><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <div>Hi,</div><div><br></div><div>I have the following instruction in my IR-</div><div><b>%call2 = call i8* @strcpy(i8* %1, i8* %2)</b> #2</div><div><br></div><div>I intend to change call to

I am running a Samba 3.6.20 for my primary domain controller (+ main file server) and my back up domain controller. Each domain controller has an LDAP backend- the LDAP servers configured for multimaster replication. I have domain trusts established with a Windows 2003 AD domain ("WINDOMAIN") . I have enabled nested groups in smb.conf. Winbind is enabled to support

https://bugs.freedesktop.org/show_bug.cgi?id=107381 Bug ID: 107381 Summary: [NV117] Hang with 'fifo: write fault' and thousands of 'TRAP UNHANDLED 00000020' Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal

hi... where usually disable-ticketing is avaliable in vm xml ???? i use this command virsh edit vmname and i can edit any options and added but when add this option to <graphics> as example : <graphics disable-ticketing='yes'> or <graphics> <disable-ticketing /> </graphics> nothing happened the file return to first status only this option ... if i add

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I sent post few days ago with question: why I can't add win2k3 to samba domain? NO answer ;) But today we were able to add it. This is strange but pure 2k3 throws message "The specified user already exist". This is not the first time I saw this message so thought that maybe this has something to do with machine account. But account in LDAP for this server was the same like

On 2020-04-22 5:29 a.m., Johannes Rohr wrote: > Dear all, > > what are the key strategies for intrusion prevention and detection with > dovecot, apart from installing fail2ban? > It is a pity that the IMAP protocol does not support 2 factor > authentication, which seems to stop 90% of intrusion attempts in their > tracks. Without it, if someone has obtained your password and

Dear r-helpers, Here I would like to have your kind helps on generating Venn diagram. There are some packages within R on this task, like venneuler, VennDiagram, vennerable. But, vennerable can not be installed on my Mac book. It seems VennDiagram can not work on my data. And, venneuler may have generated a wrong Venn diagram to me. Do you have any experience/expertise on those Venn diagram?

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Friday, May 8, 2015 5:12:28 PM > Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Hello, > > I am

Hi all, I have just started using RMark to analyse capture-recapture data. I am trying to analyse a simulated data set using the Robust Design (two primary periods with three secondary sessions in each) to estimate apparent survival. On specifying the time intervals (that tell R about the primary and secondary sampling sessions), this is the error I get: > time.int <- c(0,0,1,0,0) >

Dear all, I have a data frame with 2 columns and 102500 rows. The data looks like below V1 V2 8098 110000....... 9099 001100....... 0023 110011....... 6545 111111....... . . . . . . . . . . I want to split the colums like this ; V1 V2 V3 V4 V5 V6 v7 8098 1 1 0 0 0 0 ...... 9099

I''m seeing seriously degraded performance with round-robin SAS multipathing. I''m hoping you guys can help me achieve full throughput across both paths. My System Config: OpenSolaris snv_134 2 x E5520 2.4 GHz Xeon Quad-Core Processors 48 GB RAM 2 x LSI SAS 9200-8e (eight-port external 6Gb/s SATA and SAS PCIe 2.0 HBA) 1 X Mellanox 40 Gb/s dual port card PCIe 2.0 1 x JBOD:

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222

Does llvm have a pass that minimizes the bitwidth of llvm instructions? For instance:   %8 = and i32 %7, 63 63 is 111111 in binary. So the 'and' instruction only requires 6 bits. We could rewrite the above code as:   %8 = trunc i32 %7 to i6   %9 = and i6 %8, 63 Since we only need the lower 6 bits we could also propagate this change backwards to reduce the bitwidth of prior

Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia

Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==>Zap * ==>FXS * ==>Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure

I am just getting started with Rails, don''t know any Ruby, and don''t quite even get object oriented programming yet. I have tweaked my schema to The Rails Way and have generated a bit of scaffolding. I am trying to create a new Part, which references a PartName and a PartNumber. I can create a PartNumber on the fly, but the PartName has to exist. I have an inelegant solution