search for: rana

I tried the suggested changes, but still haven't been able to compile the branch. Here's the log: http://pastebin.com/HR17USXR Thanks, Satwant Rana On Sun, Mar 30, 2014 at 2:37 PM, Gaurav Arora <gauravarora.daiict at gmail.com>wrote: > Hello Satwant, > > This seems to be problem with doxygen installation in the bootstrap > script. Source seems to be broken and not able to build. > > If you already have doxygen instal...

I was able to successfully make the master branch of xapian, but I can't do the same for svn/clustering branch. The bootstrap fails with this log: http://pastebin.com/D1hbLp7k Can someone who has successfully installed the clustering branch tell me what am I doing wrong here? Thanks Satwant Rana -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xapian.org/pipermail/xapian-devel/attachments/20140330/7804f892/attachment-0002.html>

...has rules written in it. This is because file handling is not possible in Snowball, and so I have written a C++ code that generates the Snowball code (Code-ception :P). Since the algorithm has many steps, my codes might have some mistakes. This is where they are located: https://github.com/satwantrana/codes I will be integrating this in my Xapian fork, and release a patch soon. Meanwhile, if someone finds a bug/mistake in this, please respond. Also, I hope this implementation helps my GSoC application. Thanks, Satwant Rana -------------- next part -------------- An HTML attachment was scrubbe...

...nt" --algo bm -j DROP ?Its something like this Registration from '"30" <sp:30 at my_public_ip:5060> failed for '192.168.xxx.xxx:6373' - Wrong Password? ?and there are approx 10 request per minute of this type. Please suggest some way to stop this.? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

...ppended to file created in above step. Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions please? Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140928/d4e3652e/attachment.html>

...only accept just first digit and terminates even before considering second digit. Error message : WARNING[5726][C-0000000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please suggest what might be wrong. Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140908/e8ea5bbb/attachment.html>

...get the predicted N concentration value for each day. However, I am having trouble finding the right spline equation, since there are many forms on the internets. I know it won't be a simple one, but can some one direct me to the equation that would be best to use for ns? Thanks a lot, Ranae -- View this message in context: http://r.789695.n4.nabble.com/Do-YOU-know-an-equation-for-splines-ns-tp4632440.html Sent from the R help mailing list archive at Nabble.com.

...ying that the lengths of the two vectors don't match. Suppose X and Y are two factors. X has 5 levels and Y has 7 levels. This is what i tried doing >temp<-chisq.test(x,y) but got error "the lengths of the two vectors don't match". any help will be appreciated -- Regards, Rana Shoaaib Mehmood

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

...still got confused so can you share me the link of the standard document or mail me that document so that i can go through it and implement on the given scenario. Or if you can share any example configuration then it will be great. I hope for help from you and all experts. Thank You Regards Uttam Rana On Mon, Feb 13, 2012 at 12:45 AM, <samba-request at lists.samba.org> wrote: > Send samba mailing list submissions to > samba at lists.samba.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.samba.org/mailman/listinfo/samba > or...

...hat extension hangup. for example -> [some-context] /// something here which call AGI script no matter what extension receive call. exten => 111,1,Dial(SIP/111) exten => 112,1,Dial(SIP/112) exten => h,1,AGI(pt.py) ;; executes no matter what extension hang up ?Thanks? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

...erfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can be done to mitigate the problem? Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140713/75717b52/attachment.html>

...I am initiating a call using call files. In 'h' extension I am trying to collect the value of ANSWEREDTIME variable but it is returning null. While It works fine when call is not generated using call files instead is generated from softphone. any idea what might be wrong? thanks Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140917/69749bfe/attachment.html>

...using cached data and ignoring the database, but the registration info for phone B never got into the cache. I need rtcachefriends enabled, because I need MWI to work. I've seen this problem discussed before, but I haven't come across a solution. Anyone have an answer? Thanks in advance. Rana Dutt Softel Solutions, Inc. rdutt@softelinc.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050711/624ff9bc/attachment.htm

I am trying to upload the constants to my shared server but built in FTP in Ubuntu is not working -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe from this group, send email to

...ion retry interval? Or can you make calls through B as soon as you pick up the phone and dial, because the INVITE message through A fails, and the phone re-sends the INVITE through B? Thanks for any help. We've been trying this with Aastra 480i phones and SJ Phone without much luck so far. Rana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050720/3968a454/attachment.htm

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

Hi friends, I haven't done such a simulation before and any help would be greatly appreciated. I need your guidance. I need to simulate end to end data for Reliability/survival analysis of a Pump ,with correlation in place, that is at 'Transactional level' or at the granularity of time-minutes, where each observation is a reading captured via Pump's sensors each minute. Once