search for: deepika

Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype: deepika-nijhawan W: <http://www.oxygen8.com/> www.oxygen8.com This communication contains information which is conf...

...cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on "core show translation recalc 10". Can anybody please tell if missing some step in this. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100806/eacc651e/attachment.htm

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110228/e1d37c6d/attachment.htm>

Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype: deepika-nijhawan W: <http://www.oxygen8.com/> www.oxygen8.com This communication contains information which is conf...

Hi, I have installed R on my computer with windows 7 . I also installed reshape software, but I am not being able to work with melt cast commands . I have chjecked the commands.It is not working. Thankyou, Deepika [[alternative HTML version deleted]]

Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

...n main.c => main.obj call to a1, a2, b1, b2 Now, I want to move a1(), a2() from a.obj to b2.obj and on top of function b1() When I call b1() from main, it should call first a1, a2 and then function definition of b1 Can you please give me some pointers for my requirement. Thank you in advance. Deepika On Wed, Feb 10, 2016 at 2:54 PM, mats petersson <mats at planetcatfish.com> wrote: > This is the starting point: > http://llvm.org/docs/WritingAnLLVMPass.html > > I suspect you want either a "FunctionPass" or a "BlockPass". > > The question you stil hav...

Hi, Yes I am looking for IR pass that will do insert call of functions that defined in another file. Links/suggestions that guide me to start for adding IR pass will help me so much. Regards, Deepika On Wed, Feb 10, 2016 at 1:03 PM, mats petersson <mats at planetcatfish.com> wrote: > So how do you know what you want to modify (conceptually)? > > Have you got a IR pass that you are working on, or are you asking for > links/suggestions on how to start on one? > > -- >...

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100723/f6b85b59/attachment.htm

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101011/751bfe31/attachment.htm

...[asterisk-users] Failover Routing Try this - it says it is for 1.8 but might work in 1.6 = http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deepika = Nijhawan Sent: Tuesday, March 01, 2011 10:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing SIP_HEADER() gives you only access to headers of the initial INVITE = request (and not, for example, the final BYE message) How wil...

Hi, I want to call/add some functions(that defined in another file) on top of some functions, and reflect the same changes in object file. No, I am not looking for contractor. Thanks, Deepika On Tue, Feb 9, 2016 at 7:04 PM, mats petersson <mats at planetcatfish.com> wrote: > What is the condition for adding this code? > > What have you tried so far? [Or are you looking for a contractor that can > write code for you - I'm not sure this is quite the right place for...

...%2 = load i32* %isum, align 4 ret i32 %2 } With llc tool, I want to generate object file for modified llvm ir file and it should call function first "one_12"" and then function "one_11". Can someone please tell me how I can do my above requirement. Thanks in advance, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20160209/c6b38027/attachment.html>

...rom new media IP but asterisk keep sending to their old media IP that came in their 200 ok before and don't send to new one. Hence, I can hear their voice but they can't. Does anyone know how to make asterisk send RTP to new media IP that came in new INVITE within the call? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101117/12ca74df/attachment.htm

Hi All, I am having an ror application with ruby1.8.7 and rails2.3.5, the performance of my application is not good enough. Is their any plugin or gem available to improve the performance. Also I have already optimized some of my code and db queries by optimizing the mysql query and by adding indexes, but those are not gave drastic change in the performance. regards, Loganathan -- You received

It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100615/e3f6ce8e/attachment.htm

TC400 and TCE400B digium cards that do codec translation, 1 How many of these cards can be installed on one server? 2 Can we combine it with the software codec aka for example if we had two cards per server they could decompress 240 channels. However if we had another say 100 calls on top of that (340 total), could the additional calls be passed over to the cpu to be converted? Can anyone