similar to: Skype 4 Linux

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the

I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much

https://bugs.freedesktop.org/show_bug.cgi?id=58806 Priority: medium Bug ID: 58806 Assignee: nouveau at lists.freedesktop.org Summary: failed to create kernel channel, -12 on G4 PPC QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: stefan at

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Is there a way to prevent hwaddr/mac address spoofing between DomU''s? So in a way ''binding'' a mac-address on boot time with a virtual interface? (with something like ebtables/arptables/etc?) Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.7 (GNU/Linux) Comment: Using GnuPG with Mozilla -

Hi, If I want to encode two audio channels that are not related, audio from two different sources not stereo, is it best to create two instances of a single channel encoder or use one encoder with two channels? thank you, Paul.

Hello, i have just migrated machine to xen server where only 2 cpus are available. I have copied config from machine where domU was located before. That machine had 4 CPUS (QUADCORE), current machine has 2 CPUS (1 XEON 2cores). I have forgot to change vcpus from 4 to 2 in config, but .. what really surprised me ... machine started . How it is possible? It was really slow alltought and not

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

I have a new x3655 IBM and whenever the /etc/xen/scripts/network-bridge script starts, the ethernet would no longer work. This post isn''t about the cause, but the fix. I did a lot of tcpdumps and Googling, but I''ll spare you that. All they do is prove that yes, there is an issue. :) I tried Debian 4.0 i386/amd64 and Ubuntu 7.04 Server i386/amd64 and they all exhibit the

Hi all, How can I boot a XEN domU guest into safe mode? One of our servers was forcefully shutdown, and it seems like one of my VM''s could be corrupt, or something. When I boot it up, I get the following on the console: Creating root device. Mounting root filesystem. kjournald starting. Commit interval 5 seconds EXT3-fs: mounted filesystem with ordered data mode. Setting up other

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike.

Hi, I'm new here so an introduction. I'm Stefan de Konink, I have a diverse past, present and future in coding, content creation and a background in speech research. As introduced on #celt I want to do some promotion for the codec. I was positively surprised by the availability of something 'almost done' and totally usable in the form of celtclient.c, attached is a tiny

Hi Jean-Marc, It was discussed before, when multichannel streams are encoded; taken multiple instances of the encoder where the input is not correlated. Is there a way to pack the result of all channels again? Or maybe do something more efficient in the first place on multichannel streams? Stefan

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hello, I'm trying to implement a function available on the PSTN net here, if you dial a number which is busy and you press 5, you will be called back when the busy party hangs up. Figuring out if a SIP user is busy isn't to hard, ${DIALSTATUS} produces a BUSY message, however, how can I implement the call back? IE, I dial to extension 712, but that extension is busy, I dial 5 and

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

Hi people, As I said before: I did the IDCT to run on the FPGA. My friends from university did the Reconstruction routines running on the FPGA. I'm helping with the LoopFilter, and it is almost there. (all VHDL) I did a small profiling of the libTheora running on a Altera Stratix II device: The processor used was the NIOS II with 8Kb of data and instruction cache, branch prediction and