search for: nitesh

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

...ss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call quality you can use tools like VoIP > Monitor (it is not free). > > Regards > > > On Tue, Sep 27, 2016 at 12:03 PM, Nitesh Bansal > <nitesh.bansal at gmail.com <mailto:nitesh.bansal at gmail.com>> wrote: > > Hello all, > > The question isn't directly related to Asterisk, but I'm looking > for recommendations > for a monitoring tool to monitor the health of Aster...

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digiu...

...monitoring tool to monitor the health of Asterisk instances running in Production. Ideally, the tool should be able to generate monitoring traffic (OPTIONS ping or INVITE), use the response/no response from Asterisk to store the health of an Asterisk instance running somewhere in the DB. Thanks, Nitesh Bansal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160927/8f7db63b/attachment.html>

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

...and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problem is that Request-URI isn't populated correctly. I'm using Asterisk 13. Thanks, Nitesh On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp at digium.com> wrote: > Nitesh Bansal wrote: > >> Hello, >> >> I want to use Asterisk to use Kamailio as an outbound proxy for routing >> calls to remote SIP end points, one option could be to use a default &...

On Tue, Jun 20, 2017 at 01:29:00PM -0400, Rik van Riel wrote: > On Tue, 2017-06-20 at 18:49 +0200, David Hildenbrand wrote: > > On 20.06.2017 18:44, Rik van Riel wrote: > > > > Nitesh Lal (on the CC list) is working on a way > > > to efficiently batch recently freed pages for > > > free page hinting to the hypervisor. > > > > > > If that is done efficiently enough (eg. with > > > MADV_FREE on the hypervisor side for lazy freeing, &gt...

On Tue, Jun 20, 2017 at 01:29:00PM -0400, Rik van Riel wrote: > On Tue, 2017-06-20 at 18:49 +0200, David Hildenbrand wrote: > > On 20.06.2017 18:44, Rik van Riel wrote: > > > > Nitesh Lal (on the CC list) is working on a way > > > to efficiently batch recently freed pages for > > > free page hinting to the hypervisor. > > > > > > If that is done efficiently enough (eg. with > > > MADV_FREE on the hypervisor side for lazy freeing, &gt...

Hi Nitesh, Take a look at this http://www.microappliances.com/site/html/index.php?section=Products&page =clienthowto.php I've never implemented it though so I would appreciate some feedback on if it works. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [ma...

...eps the hypervisor from >> throwing >> away good data? > > That looks like it may be the wrong API, then? > > We already have hooks called arch_free_page and > arch_alloc_page in the VM, which are called when > pages are freed, and allocated, respectively. > > Nitesh Lal (on the CC list) is working on a way > to efficiently batch recently freed pages for > free page hinting to the hypervisor. > > If that is done efficiently enough (eg. with > MADV_FREE on the hypervisor side for lazy freeing, > and lazy later re-use of the pages), do we still...

...eps the hypervisor from >> throwing >> away good data? > > That looks like it may be the wrong API, then? > > We already have hooks called arch_free_page and > arch_alloc_page in the VM, which are called when > pages are freed, and allocated, respectively. > > Nitesh Lal (on the CC list) is working on a way > to efficiently batch recently freed pages for > free page hinting to the hypervisor. > > If that is done efficiently enough (eg. with > MADV_FREE on the hypervisor side for lazy freeing, > and lazy later re-use of the pages), do we still...

...21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

...tting is a bit messed up, It looks as follows: *sip:sip:xyz at xyz.com <sip%3Asip%3Axyz at xyz.com>*, it seems that Asterisk added an extra '*sip:'* in the To-header and it breaks. I'm using Asterisk 13. I'm wondering if this behaviour is intended or a potential bug? Thanks, Nitesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160422/fbf546e4/attachment.html>

...[pbx_config] 9. Playback(digits/1) [pbx_config] [end] 10. Noop(Waiting for input) [pbx_config] Include => 'app-blacklist-add-custom' [pbx_config] hyperion*CLI> Thanks, Nitesh

...u wish to call followed by #". And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Cheers, Nitesh

...low the instructions provided by Grandstream support but it doesn't seems to be working... http://www.grandstream.com/documents/GXW410xwithAsteriskConfiguration.pdf OS: Ubuntu 8 Asterisk 1.4.22.1 Anyone implemented Grandstream GXW410x IP Gateway with Asterisk and can share config? Cheers, Nitesh

...er '9099993456' is now UNREACHABLE!" and "Peer '9099993456' is now REACHABLE!"... I changed my maxexpiry and defaultexpiry to 3600 in sip.conf but still it didn't help. I am using Asterisk 1.2.18 with Real-Time config. Any help will be appreciated... Cheers, Nitesh

...er is calling, now the second part is to set the timezone based on the area code and prompt the users correct time. For example, if 248 (MI) user dials into the system, then time clock has to prompt EST time and if 714 (CA) user dials in then prompt PST time. Any suggestions... Thanks Cheers, Nitesh