similar to: How to get BEEP BEEP BEEP when underline sends 486 Busy Here.

I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two

Does anyone have a more clear beep tone for the voicemail? The default one seems to cut off and give the feeling that there is a problem with the vm :) If no one has this available, I may try to create a new one. Thanks. -- respectfully, Joseph - ------=============

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

Hello Everyone, We have most of the modules commented out. Can someone please let me know which modules needed to be included for Playtones? Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150509/15ea3418/attachment.html>

In order not to forget it (I have not got any answer so far) ... Uwe Ligges -------- Original Message -------- Subject: plotmath in underlined style; was: Re: [R] Underline in expression(). Date: Sun, 26 Sep 2004 18:50:02 +0200 From: Uwe Ligges <ligges@statistik.uni-dortmund.de> Organization: Fachbereich Statistik, Universitaet Dortmund To: John Janmaat <jajanmaat@netscape.net>,

Hello, I'm not understanding how to use GOSUB_RESULT in U() option from Dial app (I'm using 1.6.1) My extensions.ael is : context mylocal { 2 => { Dial(SIP/7530,,U(mynotify)); HangUp(); }; 3 => { Dial(SIP/7531); HangUp(); }; }; macro mynotify () { GOSUB_RESULT=ABORT; }; I

Dear all, I would like to underline only the colnames in a table as e.g. grid.draw(tableGrob(head(iris, 10), name="test")) I can imagine you should use grid.edit or so, bu I can't figure out how... Does anyone have a suggestion? Thanks Robbert windows xp R 2.10.1 -- View this message in context:

Hello All, Is there an analogue to \underbar or the AMS math \underline in graphical math expressions? Thanks, John. -- ===================================================================================== Dr. John Janmaat Department of Economics Acadia University Wolfville, Nova Scotia, Canada B4P 2R6 TEL: 902-585-1461 WWW: http://ace.acadiau.ca/~jjanmaat/ EMAIL: jjanmaat at acadiau.ca June

Hi File names with blanks are not very useful in Unix scripts. Our Windows users use very often (not allowed) blanks in file names. So I am looking for a possibility to change blanks to underlines. But mangled map = (*\ * *_*) does not work ?!? Best regards Horst Liesinger CAD Coordination (IT) Doppelmayr Seilbahnen GmbH

as my title said! i use tinymce pluging from: http://tinymcehammer.lanalot.com/ i have successfully loaded and use it but nderline and strikethrough didnt work for me! any suggestions ? thanks a lot philip g. -- Posted via http://www.ruby-forum.com/.

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments

Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw..... Thanks in Advance, Nick.

Hello Everyone, I am required to write a java program that will get our asterisk to: * Query the database for phone numbers * Loop through numbers and dial * Play message * Get dial pressed response - If 1 = Yes - If 2 = No - If 3 = Connect to Agent * AMD Capable * Disposition I am proficient with Java and found the Asterisk-Java API. My questions are: * What is the

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....