similar to: asterisk-users Digest, Vol 109, Issue 30

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"

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, 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:

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

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

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, 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 'Clustering of Search Results'. I have followed the

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.

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!

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 Groups "Ruby on Rails:

Hello everybody i am Ashish here. i am new to this mailing list. so dont know rules and regulation, just trying to post my problem of voicemail.conf Actuallt right now i am using Asterisk 1.2 on my LAN environment. i am able to call all my extension very nicely. Right now i am trying to deploying voicemail facility for all extensions, so if anybody is not present, then he/she can leave message,

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,

https://bugs.freedesktop.org/show_bug.cgi?id=68768 Priority: medium Bug ID: 68768 Assignee: nouveau at lists.freedesktop.org Summary: [nv50] error during boot QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: mattia.b89 at gmail.com

Hi R user,I am struggling to figure out on how I can calculate the average from the two tables in R. Any one can help me? really your help would be grateful- I am spending so much time to figure it out. It should not be so hard, I think. I have very big data but I have created a hypothetical data for simplification. for example I have : table 1 table 1: species occurance data

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>