similar to: Popup URL for outgoing calls.

What CRM your going to use? With regards N.Prakash From: Rusty Newton Sent: ?28-?06-?2014 01:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Popup URL for outgoing calls. 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

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

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 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

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

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)

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, 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,

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 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

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 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.

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) >>

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

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 long as you don't overload the server it works great. I've used

With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in

Hi I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI hardware is required. According to some recommendations like http://osdial.org/howto/? "Internal timing is very critical with Asterisk when it is under load" and we must use DAHDI hardware or "USB Voice

> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >> I have reviewed and shall

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,

<div>* call-limit on PBX is triggered</div><div>š</div><div>02.07.2015, 15:49, "royj@yandex.ru" <royj@yandex.ru>:</div><blockquote type="cite"><div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute