similar to: Poor IO performance in hosts and VMs - XCP 1.0 b42052

Hi Folks: I am having trouble creating a Fedora 14 domU on a XCP 1.0 dom0 hypervisor and am hoping that someone can help me figure out what I am doing wrong. Here is what I did (per the user''s guide and reference manual): ## Step 1. Create the VM: # uuid=$(xe vm-install template=''Other install media'' new-name-label=''test'') ## Note: I also tried

Dear all, I have ended up in a situation where CentOS 5.0 does not work for me - is it feasible to downgrade from 5.0 to 4.6 while the servers are up, or would the most sensible option be to just reinstall from scratch? Thanks in advance + best regards Jan

Hi list, I am still a newbie and struggling with tweaking the dial plan to my requirements. I have tried googling for this specific problem, and apologies if I have overlooked the obvious answer already. If you could please be so kind as to point me in the right direction, that would be most appreciated. What I am trying to do, is get rid of the initial "+" in phone numbers coming in

Hello everybody! I’m trying to define the optimal number of surveys to detect the highest number of species within a monitoring season/session. To do this I want to run all the possible combinations between a set of samples and to calculate the total number of species for each combination of 2, 3, 4 …n samples events, so that at the end I will be able to define which is the lowest number of

Hi, I am wondering if you may be able to help me. I have an application that has been developed by a third party and is being used in some field locations. My challenge is that I have no way of independently testing the product across the WAN at this point. All testing they have proposed required manual intervention and subjective analysis. I am not comfortable deploying the solution in this

The Wine development release 1.1.10 is now available. What's new in this release (see below for details): - Support for virtual memory write watches. - Workarounds for the WINAPI compiler bug on Mac OS. - Several fixes for the 64-bit build. - Some more GdiPlus functions. - Various bug fixes. The source is available from the following locations:

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Hello Everyone, I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model. We are looking to interconnect with the PSTN world, and our supplier has given us a few options. We can either do this over traditional PRIs, A-Links or the SS7IP new. I am really interested in SIGTRAN, and was wondering how some of you have integrated it into your architecture. Can Asterisk handle

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have been using Red-fone foneBridge2 Quad T1/E1 for last few years. As these devices are not available anymore, we are looking for alternatives. Are there any similar devices available ? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads.

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support

All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply