similar to: RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

Dear Mitul, I already told my boss about it, I really want a single box, no virtual, but my boss insist. He said that openvz use less resource then KVM (or other virtual for cloud). I really need a solid analysis to argue with him. On the other hand, dahdi cannot be installed in openvz virtual server. I dont have any experience with openvz at all. Thx, On Tue, Apr 7, 2015 at 8:47 PM, Ikka

Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta,

I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani <mitul at enterux.in>: > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka

Dear all, I have PBX with asterisk 13.x a couple of IPPhone that connect to that asterisk PBX send an alphanumeric dialed phone number. for example, in my CDR table, field DST, it show dialed phone number like - 0C81318304632C (it should be 081318304632) - 08D11157112 (it should be 0811157112). Why it's happening ? and how can I prevent it to happen ? Thanks in advance, Ikka Jakarta

Dear all, is asterisk capable to make a call for 24 hour without break ? My dial string in extension.conf is : Dial(SIP/[ext_no]@[pbx_name]) I dont use any dial parameter. The problemm is, my customer complain that the call was cut after 4 hours. Thanks in advance, Ikka Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Harry, Thx for the explanation. My team manage building's PBX that use Asterisk 13.x. We use Asterisk PBX for this buildings that have apartment and office customer. >From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter (cisco SPA112). Others are using PBX like panasonic analog, audiocodes SBC, etc, and we use ATA Converter to convert from SIP to Analog (CO Line)

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo

Hi, Is there anyone here who has experience connecting Asterisk (ver 13.8) with PBX Panasonic KX-TDA600 ? The architecture more less like this : Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax Thanks in advance, Regards, Ikka - Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Dovid, thx for the input. for timer in sip.conf, I used default setting. This is some of the result for "sip show settings" RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer

Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password.

Hi all! After installing 5.3 I've noticed some change in firewall logging. Prior (on 5.2) rules gave me what I needed: trimed to 3 of the same connection. Every new connection on the same rule gave new log line up to 3. I have in kernel: FIREWALL FIREWALL_VERBOSE FIREWALL_VERBOSE_LIMIT=3 Now, all connections on the same rule are trimed to 3. Is it possib- le on 5.3 to have all

On 04/07/2015 10:48 AM, Johan Wilfer wrote: > Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >>

I am having problems mounting a DFS share, even though it works in smbclient fine. After mounting, the mount point (in ls) has question marks for everything, and when ls'ing I just get permission denied. Turning up debug and verbose don't seem to help. Any ideas ? [tom@charles-compaq@1306 /home/tom/Projects/gbb-core-app ] smbclient //exchsvr/dfs -U tchiverton -W BLUEFINGER Password:

If you want to use dahdi dummy driver inside asterisk for timer then this is possible with openvz based container virtualization. We have tested vicidial in this mode for 5-10 agents and it worked well. Mitul Limbani On Apr 8, 2016 8:52 AM, "Pete Mundy" <pete at fiberphone.co.nz> wrote: > List, > > Might as well throw my hat in the ring! > > I can't say

Hi all, I have a running Samba4 Server. I am able to authenticate Windows and Linux Clients very. (1) I want to use samba4 as SSO. In this regard my next step is to authenticate our web site users from samba4 server. In this web site, at home page our corporate users give their e-mail address username at companydomain.com and password (not e-mail password). (2) Our E-mail server is hosted on

I have a perferctly working shorewall system, with basic configuration (external real IP, one private address internal network with some forwarded services), and log handling with fwlogwatch. My problem is that I can''t find out how to make something like this with shorewall (TCP-connections only): - Allow protocol x connections from IP x.x.x.x without logging - Allow protocol x

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten => s,1,NoOp(**** FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ****) exten => s,n,Set(FAXOPT(ecm)=yes) exten =>

I''ve got two recent examples of SSDs. Their pristine state from the manufacturer shows: Device Model: OCZ-VERTEX3 # hexdump -C /dev/sdd 00000000 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................| * 1bf2976000 Device Model: OCZ VERTEX PLUS (OCZ VERTEX 2E) # hexdump -C /dev/sdd 00000000 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff |................| *

Show him this freaking thread, or else ask him to prove it otherwise. We all here have decades of exp dealing with asterisk. Mitul On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" <ikka.tirta at gmail.com> wrote: > Dear Mitul, > > I already told my boss about it, I really want a single box, no virtual, > but my boss insist. > He said that openvz use less resource then

I was wondering if there was a way to determine the number of samples in a Vorbis packet without having to decode the whole thing? The reason I'm asking this is because I'm thinking about implementing the current Vorbis RTP draft for Helix. The problem is that the Ogg pages only contain a single timestamp. If I want to split up the Vorbis packets on different boundries than the Ogg