search for: kamlesh_kmr

Displaying 12 results from an estimated 12 matches for "kamlesh_kmr".

2011 Dec 14
1
get start-time of all active calls
Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>
2011 Dec 23
1
execute command just after Dial()
Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......
2013 Feb 26
1
set time zone in sip debug logs
Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at
2014 Dec 12
1
c option doesn't work if used with q option in meetme
Hello, Asterisk version 11.13.1 I'm trying use realtime meetme application with c and q option. If both options are used together in meetme table under 'opts' field, c option (Announce user(s) count on joining a conference.) doesn't work i.e. system doesn't play user counting to caller. Is it bug or some configuration problem. Thanks, Kamlesh --------------
2013 Jul 25
2
limitation on number of contexts in extensions.conf
Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include <filename>) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL:
2012 Jun 15
1
voicemail password with phone instrument
Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- <SIP/123-00000005> Playing
2012 Aug 31
0
failed to extend from 512 to 676 message on console
Hello, Asterisk Version 1.6.2.9 on below hardware. We are using 100 Realtime SIP extensions. CPU : 1 x Intel? Core-i5 3.3 GHz. RAM : 4 GB DDR-3 SDRAM Hard Disk : 500 GB Hard Disk For last few days, getting below messages on asterisk cli. We googled to find the solution for this but could not locate the preventive steps. failed to extend from 512 to 676 failed to extend from 512 to 676 failed to
2013 Apr 17
0
failed to extend from 512 to 676 on cli
Hello, We are using around 100 real time sip peers with phpagi. On asterisk cli, getting frequent message 'failed to extend from 512 to 676'. Once we execute 'sip reload', this message disappear for some time and then comes back. Please let us know the solution for this. asterisk version 1.6.2.9mysql 5.0server: Intel(R) Core(TM) i5-2500 CPU @ 3.30GHzRAM: 4 GB Thanks,Kamlesh
2013 May 27
1
G.729 codec in pass-thru mode
Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script
2013 Aug 05
1
server for 500 concurrent SIP calls
Hi, Asterisk 1.6.2.9 PHP 5.3 Mysql 5.0 Can anyone suggest hardware specification for 500 hundred concurrent SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching. There is only one requirement is to execute one php script on call hangup (h extension) which will do some calculation and update the CDRs. Thanks, Kamlesh -------------- next part
2013 Sep 05
1
high cpu average load
Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc.... Hardware: 2 Physical processor Intel(R) Xeon(R) CPU
2012 Feb 15
1
error during dahdi installation on centos
Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root at localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root at localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all