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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"

Hello, I am Ainish Dave from Ahmedabad, India. I am currently pursuing my masters level degree with background in computer science. I browsed through the list of organizations on the GSoC program page and Xapian was the one which closely matched with my interest and background. I had some queries: 1) I am interested in the project '...

Dear Members, I have been searching for a package in R which can handle multiple seasonality suggested by taylor(2003). It will be great help if anybody has used this on R before (i.e. which package). Thanks in Advance. Best Regards Atul Malik [[alternative HTML version deleted]]

Hello All, Openings for a Sr. Ruby On Rails Developers in Ahmedabad based MNC IT Company, Interested candidate can send their updated resume at shikha.gadhia-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org or call at 9824014448 for further details. Salary no Bar for Right Candidate....! Thanks Shikha -- You received this message because you are subscribed to the Google...

I have this code in a dial plan: exten => _417XX,n,GotoIf($["${CALLERID(num)}" > "SIP/41799"]?notfromlocal) exten => _417XX,n,GotoIf($["${CALLERID(num)}" < "SIP/41700"]?notfromlocal) The value of "${CALLERID(num)}" appears to be "SIP/41712-00000181" -- Executing [41720 at from-internal:5]

Hi everybody i m ashish here now i m not using this mail more so pl.mail me on abbarot at gmail.com Thank you Barot Ashish Jr.System Administrator, Lubi Electronics, Ahmedabad. __________________________________ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest

I am stumped In features.conf,I programmed this [applicationmap] Answer0 => 0,self/both,Macro,nway_start But do I pass an argument or parameter to my macro? I tried Answer0 => 0,self/both,Macro,nway_start^0 Answer0 => 0,self/both,Macro,nway_start,0 but the usuar variable ${ARG1} is empty in my dialplan. The issue is that my macro needs to know what key was pressed. -------------- next

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

...se36 clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor ds_cpl est tm2 cx16 xtpr lahf_lm bogomips : 5333.16 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: and uname-a Linux 115.108.190.84.static-ahmedabad.vsnl.net.in 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux any help appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/5de7a10c/attachment-0001.htm

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as timeout value,

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hi, Can a call to multiple devices be cancelled in all of them at same time? With next dialplan, exten => 100,1,Dial(SIP/101&SIP/102) when a call rings on 101 and 102 and one of them rejects the call "with 486 Busy here", is it possible to reject the call in the other device at same time? I read application dial options but I can't find any that can help me to achieve this

...? So appropriate clustering mechanism can be thought of according to that. Thanks & regards, Ainish On Sat, Mar 19, 2016 at 5:37 PM, James Aylett <james-xapian at tartarus.org> wrote: > On Sat, Mar 19, 2016 at 11:49:14AM +0530, Ainish Dave wrote: > > > I am Ainish Dave from Ahmedabad, India. I am currently pursuing my > masters > > level degree with background in computer science. I browsed through the > > list of organizations on the GSoC program page and Xapian was the one > which > > closely matched with my interest and background. I had some queries:...

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Seems RPM compatibility issue with OS Kernel. Check OS Kernel and download RPM (4 Nos) for same kernel. Regards, Yuvrajsinh Chauhan || Sr. DBA || CRESTEL-PSG Elitecore Technologies Pvt. Ltd. 904, Silicon Tower || Off C.G.Road Behind Pariseema Building || Ahmedabad || INDIA [GSM]: +91 9727746022 -----Original Message----- From: ocfs2-users-bounces at oss.oracle.com [mailto:ocfs2-users-bounces at oss.oracle.com] On Behalf Of ocfs2-users-request at oss.oracle.com Sent: 12 September 2012 15:53 To: ocfs2-users at oss.oracle.com Subject: Ocfs2-users Digest, Vol 1...

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

...s] Ocfs2-users Digest, Vol 105, Issue 4 Seems RPM compatibility issue with OS Kernel. Check OS Kernel and download RPM (4 Nos) for same kernel. Regards, Yuvrajsinh Chauhan || Sr. DBA || CRESTEL-PSG Elitecore Technologies Pvt. Ltd. 904, Silicon Tower || Off C.G.Road Behind Pariseema Building || Ahmedabad || INDIA [GSM]: +91 9727746022 -----Original Message----- From: ocfs2-users-bounces at oss.oracle.com [mailto:ocfs2-users-bounces at oss.oracle.com] On Behalf Of ocfs2-users-request at oss.oracle.com Sent: 12 September 2012 15:53 To: ocfs2-users at oss.oracle.com Subject: Ocfs2-users Digest, Vol 1...

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,