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...oss 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 Tirtawidjaja <ikka.tirta at gmail.com> wrote: > 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...

...N-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, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150407/21d44ac2/attachment.html>

...sue. 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 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 KVM (or other virtual for >> cloud)....

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

...we use FXS or ATA Converter, its going to have a lot of that, because this tenant going to use about 60 ext / sip line. Replacing asterisk PBX on my (company) side or replace TDA600 on my customer side is not acceptable. So we need to find a "win-win" solution for this. Thx in advance, Ikka On Wed, Sep 14, 2016 at 12:40 PM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > On 09/13/2016 06:51 AM, Ikka Tirtawidjaja wrote: > > Hi, > > Is there anyone here who has experience connecting Asterisk (ver 13.8) > with PBX Panasonic KX-TDA600 ? >...

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on Asterisk 1.8.4 Dear all. I'm using MixMonitor command in my dialplan, and I used the &quo...

...erisk 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 - Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170330/ca994aea/attachment.html>

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/28388a8f/attachment.html>

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 KVM (or other virtual for > cloud). > I really need a solid a...

...uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 Dear Josua, I need to check my server (my settings) first before i complain about it to my provider. Thx to all, Regards, Ikka Jakarta-Indonesia On Wed, May 11, 2016 at 7:39 PM, Joshua Colp <jcolp at digium.com> wrote: > Ikka Tirtawidjaja wrote: > >> Dear all, >> >> is asterisk capable to make a call for 24 hour without break ? >> >> My dial string in extension.conf is : >&g...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160913/e3fb9778/attachment.html>

...ion. So, we can not move / change those IPPhone to the new PBX. Is there a way for us to retrieve / decrypt those SIP secret ? Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100) Any suggestion are appreciate, because I'm really desperate. Thanks in advance, Regards, Ikka Jakarta - Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160629/888ddd39/attachment.html>

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...

...N-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, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150407/caeab0cd/attachment.html>

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) > As lon...

...0 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Tue, Apr 7, 2015 at 9:57 PM, Jeff LaCoursiere <jeff at jeff.net> wrote: > 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 +/- >>&gt...

You can learn the source code of predictive dialer like vicidial, osdial, etc. You have to discover how they work, then you can implement it on your asterisk. But it's going to take a lot of time. Sorry for my bad English Regards, Ikka Vertika Jakarta - Indonesia Rusty Newton <rnewton at digium.com> wrote: >On Sat, Jun 21, 2014 at 5:57 AM, Inventions <research at businesstz.com> wrote: >> Can anyone tell me how to implement a popup URL native asterisk when >> making outbound call? >> >>...

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

...52-00000059", "FAXOPT(status) : SUCCESS") in new stack -- Executing [h at inboundfax:7] NoOp("SIP/192.168.100.252-00000059", "FAXOPT(statusstr) : OK") in new stack I'd tried searching @ google all day, but cannot find the answer. Thanks in advance... Ikka -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141223/622166b7/attachment.html>

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