similar to: Issue when dialing multiple extensions using & ------Please Help

Hi, Would you please let me know the performance of asterisk realtime in case I will have millions of SIP users? Thanks, Krunal Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090318/f1e0f6c8/attachment.htm

Hi, I have a query on regression output generated by R. > result=lm( Y~X , data=trail) > summary(result) After running this 2 statements the following output is generated. Call: lm(formula = Y ~ X, data = trail) Residuals: Min 1Q Median 3Q Max -245.30 -90.77 -30.30 54.99 532.78 Coefficients: Estimate Std. Error t value Pr(>|t|)

Hi, I am trying to run many different regressions using a FOR Loop. The input data that is read into R has the following variables · Volume · Price1 · Price2 · Price3 · Price4 · Price5 · Trend · Seasonality I want to run 5 regressions, with the Volume as an dependent variable and Price, Trend & Seasonality as

Am using perl AGI to invoke the dial command thus: $AGI->exec('Dial',"$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002)"); What I expected that this will do is: 1. call the number using the string $numtodial2 - works OK 2. Set call limit to $maxcall and play a message $msgtime milliseconds before the call - works OK 3. On connect of the call send it to the macro conn

Hi, Here is the scenario: Originating local channel using AMI. On answering the channel, it will goes to a context. Which start to playback a file. & after hangup at h extension I am caliing an agi script which insert CDR into DB. Now the problem is when I script hangsup during payback CDR(billsec) returns currect result. But when it hangsup after playback cdr(billsec) returns 0 . Please

I have a simple dial plan to cascade calls when the first phone does not answer: exten => 100,1,Dial(SIP/1000,10,tr) exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr) exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr) exten => 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user

Hello all, We're trying to build a small IVR application to allow callers to use the Asterisk for outgoing calls in a 2 steps dialing mode. The context for outgoing calls is called "outgoing" (we have there an LCR and routing mechanism we want to use, depending on the destination). This is what we did, but it doesn't work: exten => _X., 13,

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at

Hi list, I''ve got a system with 3 WD and 3 seagate drives. Today I got an email that zpool status indicated one of the seagate drives as REMOVED. I''ve tried clearing the error but the pool becomes faulted again. Taken out the offending drive and plugged into a windows box with seatools install. Unfortunately seatools finds nothing wrong with the drive. Windows seems to see

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000&PJSIP/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.

On 8/7/07, raviprakash sunkara <sunkara.raviprakash.feb14 at gmail.com> wrote: > > Hello Russell, > Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using

A big G'day to everybody on the Asterisk list. I am having a lot of trouble getting the TDM2400P card working in asterisk. I will give the important details below, please let me know if I am doing anything obvious or ideas for debugging this. SUMMARY: I get the right voltages on the line with the phone on or off the hook, but no dial tone, no ringing in or ringing out. INSTALLATION AND

Hello, I want that after client and queue member call would be established, cmd queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This is my example of ael : context QUEUE { _X. => { Ringing(); Wait(4); Answer(); Queue(${Queue},wr,,,60,,,check-record); Hangup(); }; }; macro check-record() {

Preparing to use * for a 'real' installation shortly. Meanwhile, I've got a single port clone thing, 00:06.0 Communication controller: Motorola Wildcard X100P working to answer my landline and send calls to my laptop or voicemail. Sweet! Trying to "call out" from linphone, I set up this: exten => _X.,1,Dial(DAHDI/1,${EXTEN}) Both SIP client and this extension are in

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1