search for: satish

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi All, I have the following section in my smb.conf file: [Satish' Hindi Alltime Hits - Gems] path = /mnt/d-drive/Music/Satish' Hindi Alltime Hits Gems browseable = yes public = yes readonly = yes guest ok = yes I am able to browse this share through win2k but not from win...

...path. I installed Xen-unstable with 2.6.30-rc3(dom0). But the SRIOV functionality isn''t working, because MSI is not supported. Please let me know, what version of Xen and dom0 supports above functionality? Is there any tool available to manage SRIOV device? Thanks in Advance. Regards, -Satish _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

...after the data they are pointing is completed/done My illustration may sound simple but my data set is extremely huge and very deep dir structures and tons of symlinks under them. Probably I can get around programmatically but checking if there is an easier option that I don't know of. satish at AA:/export/universe >--> ll total 12 drwxrwxr-x 2 satish cadence1 4096 Jun 6 16:42 10.2 lrwxrwxrwx 1 satish cadence1 4 Jun 6 16:42 new -> 10.2 drwxrwxr-x 2 satish cadence1 4096 Jun 6 16:43 9.2 drwxrwxr-x 2 satish cadence1 4096 Jun 6 16:43 8.2 -SS

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk 1.8.3.2 built by root @ shirley on a x86_64 running L...

...llowing 1. Merge the data frame so that I have a new data frame grouped by values of x2 and x3 (for example). That is, if two records have the same values of x2 and x3, they should be summed. I tried to look at merge, tapply etc. but did not see a fit with I want to do above. Thanks in advance. Satish

...e RedHat 9.1 running. Installed Samba that comes along with the OS to test it out. When I was playing with it, I deleted the smb.conf file inadvertently. I uninstalled and reinstalled the Samba utilities, but that didn't help. How do I get back the default smb.conf file? Thanks in advance. Satish Kumar

Satish: Does your app uses any AX controls?. If so, it should be registered. Partha Saradhi -----Original Message----- From: Satish A. Lele [mailto:satishl@spidersystems.co.in] Sent: Friday, July 02, 2004 11:53 AM To: wine-users@winehq.org Subject: [Wine]Active X Component Can't create object err...

...nswer() same => n,Playback(hello-world) same => n,Hangup() [from-sip] exten => _7XXX,1,Answer() same => n,Dial(SIP/${EXTEN}) same => n,Hangup() exten => 7527,1,Dial(DAHDI/G0/7527) But i am getting following error when i am calling from A to B satish-desktop*CLI> [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_...

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav

...develop a backup application for linux platform... the workflow is as follows: - freeze the file-system(ext2 or ext3) - take the bckup - release the file system. i am not able to find the relevant ioctl or api for freezing & releasing the file system... so... i need some help... thanks... Satish Singhal __________________________________________________ Do You Yahoo!? Sign up for SBC Yahoo! Dial - First Month Free http://sbc.yahoo.com

Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070919/f5e3803f...

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

...ime=0 announce-frequency = 0 announce-holdtime = no persistentmembers = yes maxlen = 0 member => Agent/1001 member => Agent/1002 ---------------------------------------------- my agent.conf [general] persistentagents=yes [agents] ackcall=no musiconhold => default agent => 1001,1234,satish agent => 1002,1234,aman ---------------------------------------------------- my dialplan ( extention.conf) exten => 9999,1,Answer exten => 9999,2,SetMusicOnHold(default) exten => 9999,3,Background(welcome) exten => 9999,4,Queue(9999) ----PGP Signature-- Satish Patel mobile:- +...

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in new stack -- Executing [s at macro-orasebcamdial:1] Dial("SIP/7527-0000000d", "iax2/orasebcam at orasebcam/7623") in new stack...