search for: b1000

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten => 1000,103,Hangup and.. exten => 8500,1,Wait,1 exten => 8500,2,VoiceMailMain2(${CALLERIDNUM}) exten => 8500,3,Hang...

Hi all, I'm using a certain  procedure to calculate the value of some variable(Bayes risk),B. So I got the values B1, B2, ........, B1000, each under certain input values and using a long procedure. Now, I want to put the values I got in a nummerical vector and find their minimum value. I think c( ) should work.For example if I have only 10 values I could have used c(B1,B2,B3,B4,B5,B6,B7,B8,B9,B10) But how can I do this for 1000 valu...

...onhold=default busydetect=no callprogress=no usecallerid=uk channel => 1 -------------------------------- part of extensions.conf [incoming] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,dial(SIP/5000&SIP/5001,10,tr) exten => s,3,Voicemail,u1000 exten => s,102,Voicemail,b1000 exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1}) exten => _9.,2,Congestion Vassilis

...; haven't tried this by creating a user and peer for each handset (yuck) extensions.conf entries are either one of the following (tested against both) exten => _XXXX,1,Dial(SIP/1000,30,t) exten => _XXXX,2,Voicemail(u1000) exten => _XXXX,3,Hangup exten => _XXXX,102,Voicemail(b1000) exten => _XXXX,103,Hangup exten => _XXXX,1,Dial(SIP/1000,30,t) exten => _XXXX,2,Voicemail(u1000) exten => _XXXX,3,Hangup exten => _XXXX,102,Dial(SIP/1001,30,t) exten => _XXXX,103,Voicemail(u1000) exten => _XXXX,104,Hangup exten => _XXXX,203,Dial(SIP/1002,30,t) ....... exte...

...a call-waiting feature on my line and couldnt figure out how to trigger a flash in order to go from one call to another. Solution : 1st - The inbound context (in extensions.conf of course) [pstninbound] exten=>s,1,Dial(SIP/cisco7940,40|Tt) exten=>s,2,Voicemail(u1000) exten=>s,3,Voicemail(b1000) The Tt option will allow you to transfer by hitting the # key. 2nd - The flash extension Now, somewhere in your extensions file, create a context that similar to this : exten=>604,1,Flash() exten=>604,2,Dial(SIP/cisco7940) By transfering a ZAP call to that extension, the line is flashe...

...iend secret=010101 auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no dtmfmode=inband callerid="Fred F"<200> dissallow=all Extensions.conf [default] exten => 1000,1,Dial,Zap/1|20 exten => 1000,2,Voicemail,u1000 exten => 1000,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) ; Dial Broadvoice for 30 seconds exten =&...

...simple at this point: [from-pstn] ; CALLS FROM PSTN NETWORK ; ;exten => s,1,VoicemailMain2 exten => s,1,NoOp(${CALLERID}) exten => s,2,Dial(${SIP_PHONE},15,j) exten => s,3,Voicemail(u1000) ; Gavin's voicemail, unavailable exten => s,4,Hangup exten => s,103,Voicemail(b1000) ; On the phone exten => s,104,Hangup My Zapata.conf file hasn't changed since 1.0.x and looks like the following: [channels] ; entries are inherited ; added this to see if it could be the culprit. Setting it to a known value ; such as "Test" <(800) 555-1212> does...

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

A common process when data is obtained in an Excel spreadsheet is to save the spreadsheet as a .csv file then read it into R. Experienced users might have learned to be wary of dates (as I have) but possibly have not experienced what just happened to me. I thought I might just share it with r-help as a cautionary tale. I received an Excel file giving patient details. Each patient had an ID

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

...in R that involves large data. Both time and memory issues need to be seriously considered. Below is the problem description and my tentative approach. I would appreciate if any one can share thoughts on how to solve this problem more efficiently. I have 1001 multidimensional arrays -- A, B1, ..., B1000. A takes about 500MB in memory and B_i takes 100MB. I need to run an experiment that evaluates a function f(A, B_i) for all B_i. f(A, B_i) doesn't change A, B_i during its evaluation. These evaluations are independent for all i. I also need to design various evaluation functions. Thus these kin...

Hi All, I'm using the Call Queue without the Agent Login to provide a company with a Call Queue for their tech support Staff. Basically they have several techs the work from home. I dump the calls back out the PSTN via a SIP gateway. With the Queue application is there a way to define an absolute time out so that if a caller sits in the queue for say 10 minutes with out being serviced they

...any hybrid ; [globals] CONSOLE=/dev/dsp [voicemail] exten => 7,1,Ringing exten => 7,2,Wait(2) exten => 7,3,VoicemailMain [local] include => voicemail ; SIP Phone Operator Office exten => 1000,1,Dial,SIP/opoffice|30 exten => 1000,2,Voicemail,u1000 exten => 1000,102,Voicemail,b1000 ; SIP Phone Operator Field exten => 2000,1,Dial,SIP/opfield|30 exten => 2000,2,Voicemail,u2000 exten => 2000,102,Voicemail,b2000 approach 2: ======== I moved the opoffice Snom to the office side and moved the configurations for this phone in /etc/sip.conf and /etc/extensions.conf to t...

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

...p extension and my cellphone) when asterisk receives an incomming call. EXTENSIONS.CONF: [incoming] exten => 3058472194,1,Dial(SIP/1035&SIP/17864883123@richmedium,50) exten => 3058472194,2,Wait(2) exten => 3058472194,3,voicemail(u1000) exten => 3058472194,103,voicemail(b1000) CONSOLE OUTPUT FOR THE INCOMING CALL: asterisk*CLI> -- Executing Dial("SIP/3058472194-ff33", "SIP/1035&SIP/17864883123@broadvoice|50") in new stack -- Called 1035 -- Called 17864883123@richmedium -- SIP/1035-21d1 is ringing -- SI...

...s,1)\par ;exten => 1234,2,Macro(stdexten,1234,$\{CONSOLE\})\par ;exten => 8500,1,VoicemailMain\par ;exten => 8500,2,Goto(s,6)\par \par \par [default]\par exten => 1000,1,Dial,Zap/1,20\par exten => 1000,2,Voicemail,u1000\par exten => 1000,3,Hangup\par exten => 1000,102,Voicemail,b1000\par exten => 1000,103,Hangup\par exten => _9XXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXX,2,Congestion\par \par exten => _9XXXXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXXXX,2,Congestion\par ; Extension 2000 Sipura line 1\par exten => 2000,1,Dial,sip/spa...

I have Asterisk deliver all voice mail to users as email attachments. I found by accident that there is a limit of 99 messages in your INBOX in Asterisk. The 100th attempt to record a voice mail causes the system to play your greeting and then never record the 100th message and silently disconnect the caller. So...is it safe to simply use the UNIX find command to delete any files in the

...m) exten => 74928,5,Voicemail,u999 exten => 74928,6,Hangup exten => 74928,102,Voicemail,b999 exten => 74928,103,Hangup exten => 75160,1,Answer exten => 75160,2,Dial(SIP/DavidLiu,180,tm) exten => 75160,3,Voicemail,u1000 exten => 75160,4,Hangup exten => 75160,102,Voicemail,b1000 exten => 75160,103,Hangup exten => 74573,1,Answer exten => 74573,2,Dial(SIP/TerenceParker,180,tm) exten => 74573,3,Voicemail,u1001 exten => 74573,4,Hangup exten => 74573,102,Voicemail,b1001 exten => 74573,103,Hangup -------------- next part -------------- An HTML attachment wa...