similar to: Problems with the voice quality under load

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

B.H. Hello, all. We have noticed many calls on our PBX get "stuck" - the other end sends BYE, and our side sends ACK but the call remains active (no hangup event on AMI, the call is listed in 'core show channels') and it's impossible to hang up until asterisk is restarted. Asterisk's log shows lots of messages like this: chan_sip.c: Autodestruct on dialog .... with

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. Our software is written

B.H. Hello, all :-) We have some OPENVOX D410P PRI cards and we are successfully using them with Asterisk boxes which are based on stock ubuntu 12.04 DAHDI and Asterisk packages. The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it uses wct4xxp driver. Now, i'm trying to run this hardware with DAHDI 2.6.2 package which is available from asterisk.org site and looks

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

B.H. Hello! We have several Asterik boxes that are connected to PSTN using PRI cards and they are interconnected using IAX2 trunks so that incoming calls are delivered from PSTN to the servers they belong to. In past we were using asterisk 1.4 on the server that is receiving IAX connections and everything worked as expected. Recently, we have switched to a newer box with asterisk 1.8.22 and

On Mon, 2 Mar 2015, Mordechay Kaganer wrote: > When a particular server gets about 500 concurrent calls, the sound > quality begins to degrade, the sound plays slowly and with clicks. As > far as i understand, it's because asterisk is unable to send the voice > stream in time i.e. the server is overloaded. > > What i don't understand is, at the time that the server

Hi, Recently i have worked on re-writing beryl's transparent cube, and ported 3d plugin to compiz. I'm attaching a patchset here that includes the transparent cube patches (i'll post the 3d plugin when i fix some problems that didn't happen in beryl). Patching order: btf-ftb.patch, clip-planes.patch, plugin-events.patch, cube-paint-order.patch, transparent-cube.patch. Special

Hello there I''m just testing a module (see merging directories) and it is quite involved, in that it: __generic__ 1) creates a user 2) creates a base directory 3) copies generic files (directory with recursion) 4) generate a configuration file from a template 5) creates a symlink __os specific__ 6) copies os specific files (in my scenario two folders, and an init script) 7)

Hello Ron, Which kernel do you run Asterisk/Freepbx with ? Cheers 2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwheeler at artifact-software.com>: > CentOS 7 works well with Asterisk. > Install latest CentOS7 with updates install asterisk > > I am running FreePBX on CentOS 7. > > Ron > > On 14/12/2017 10:38 AM, Olivier wrote: > > Hello, > > I'm used to

Olivier If you installed asterisk from source, you need to recompile it after kernel version upgrade. This will compile & install asterisk modules with latest installed kernel sources. -- regards, abdul basit On 19 December 2017 at 08:01, Ron Wheeler <rwheeler at artifact-software.com> wrote: > Linux x.y.com 3.10.0-693.5.2.el7.x86_64 #1 SMP Fri Oct 20 20:32:50 UTC > 2017

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it

Hello, I'm used to install Asterisk on Debian stable platforms. A customer is asking how I would proceed on a CentOS platform. After a short research (see [1] as an example), I'm wondering what are general kernel practices on CentOS regarding Asterisk and when targeting stability: - Is it recommended to upgrade kernel version(s) (ie moving from linux 3.10 to 4.3) just after OS

The hvm domain which is being debugged sometimes crashes with the following message: (XEN) Failed vm entry (exit reason 0x80000021) caused by invalid guest state (0). (XEN) ************* VMCS Area ************** (XEN) *** Guest State *** (XEN) CR0: actual=0x000000008005003b, shadow=0x000000008005003b, gh_mask=ffffffffffffffff ...[snip]... (XEN) DebugCtl=0000000000000000

Hi in! I need to know how to tell samba not to cache files. Thanks for your help! [BTF]KaZeR http://kazer.homeip.net

I was wondering if anyone can give me any pointers or insights of whether or not to have an asterisk server behind a firewall. I have always ran Asterisk on a public IP but was wondering if I should move it to a local IP behind a firewall. I am looking to set up a location with 300 SIP phones. Normally, I would put the Asterisk server on one public IP and let the SIP phones get DHCP from a

This was a request from the systemd guys. They need a quick and easy way to get all devices attached to a Btrfs filesystem in order to check if any of the disks are SSD for...something, I didn''t ask :). I''ve tested this with the btrfs-progs patch that accompanies this patch. Thanks, Signed-off-by: Josef Bacik <josef@redhat.com> --- fs/btrfs/ioctl.c | 64

Hi, Currently streaming ogg isn't practical in this situation. That was one of the first things i checked into. WHen i looked i didn't see a streamer that did both ogg and mp3. Thanks. Dave. ----- Original Message ----- From: "Daniel Ballenger" <lpmusix@gmail.com> To: "Dave" <dmehler26@woh.rr.com> Cc: <icecast@xiph.org> Sent: Friday, December