search for: b2000

...ing the output of my dotplot. The default order of the function dotchart seems to be the largest value on the bottom and the smallest on the top. I wanted it in the opposite order (which is how my data frame is currently sorted), so I used the following approach, which works orderidx <- order(b2000$RR) dotchart(b2000$RR[orderidx], labels = b2000$CNAME[orderidx],cex=0.5,xlab = "Proportion of Annual Catch, 2000") However, the default behavior of the dotplot function (within the lattice library) seems to be to order the data in reverse alphabetical order (A on the bottom, Z on the top...

...e so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has expired. My hardware is 2 X100P card exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Dial(Zap/1/5551212,10) exten => 2000,3,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup Any ideas?

...writeprotect=yes [outgoing] ignorepat => 9 exten => _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t) exten => _9NX.,2,Congestion [from-sip] exten => 2000,1,NoOp("call for "${EXTEN}) exten => 2000,2,Dial(SIP/2000,20,tr) exten => 2000,3,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,NoOp("call for "${EXTEN}) exten => 2001,2,Dial(SIP/2000,20,tr) exten => 2001,3,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) ; Call straight to ex...

...username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) -------------- next part -------------- An HTML attachment was scrubbed... UR...

...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 the Asterisk at the office side. Then I added the following line to /etc/extensions.conf to the local context: [local] include => v...

...y | app | appdata | +----+----------+-------+----------+---------------+----------------+ | 1 | from-sip | 2000 | 1 | Dial | SIP/2000,20 | | 2 | from-sip | 2000 | 2 | Voicemail | u2000 | | 3 | from-sip | 2000 | 102 | Voicemail | b2000 | | 4 | from-sip | 2000 | 103 | Hangup | | | 5 | from-sip | 2001 | 1 | Dial | SIP/2001 | | 6 | from-sip | 2001 | 2 | Voicemail | u2001 | | 7 | from-sip | 2001 | 102 | Voicemail | b2001 | | 8 | f...

...F [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. [bogon-calls] [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) Thanls Arslan, -------------- next part -------------- An HTML...

...=> 2000,2,Goto(${temp}|1) exten => 2000,102,Goto(2000|3) exten => 2000,3,DBget(dnd=DND/2000) exten => 2000,4,Goto(2000|6) exten => 2000,104,Goto(2000|5) exten => 2000,5,Dial(SIP/2000,20) exten => 2000,6,Voicemail2(u2000) exten => 2000,7,Hangup exten => 2000,106,Voicemail2(b2000) ; busy exten => 2000,107,Hangup Thanks, Brian

...nf [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 => _NXXNXXXXXX, 2, congestion() ; No answer, nothing exten => _NXXNXXXXXX, 102, busy() ; Busy -------------- next part -------------- An HTML attachment was...

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

...w=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic mailbox=2000 extensions.conf [general] static=yes writeprotect=yes [globals] PSTN-1=Zap/1 [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) include => to-pstn [to-pstn] exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN}) Can anyone help me out here? Thanks, Scott

...exten => s,102,Voicemail,b1030 exten => 8500,1,VoicemailMain [demo2-internal] include => demo2-external include => bpns-internal include => outgoing-calls [demo3-external] exten => s,1,Dial(SIP/2000,20,tr) exten => s,2,Voicemail,u2000 exten => s,102,Voicemail,b2000 exten => 8500,1,VoicemailMain [demo3-internal] include => demo3-external include => bpns-internal include => outgoing-calls It doesn't work. I have a couple asterisk guru friends who swear it should work. Here is what asterisk tells me in verbose mode: -- Star...

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Hi to all!! I'm successful to connect Asterisk to MySQL database... Can anyone learn me how to store sip user in MySQL database and how to configure voicemail?? Thanks for all!!!

...sip mailbox=2000@local dtmfmode=info callerid="Me" <2000> My extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,30) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => _9.,1,Dial(SIP/${EXTEN:1}@iconnect,60,r) exten => 2999,1,VoicemailMain(${CALLERIDNUM}) Can anyone tell me if there is some error I am overlooking. I can't see it being bandwidth or the nat gateway since the budgetone is on the same lan and uses the...

...ow = ilbc allow = gsm auth = md5 trunk = no qualify = no extension.conf [general] static = yes writeprotect = yes [bogon-calls] exten => _.,1,Congestion [from-iax] exten => 105,1,Dial(IAX2/IaxCocomall1@202.138.166.6,20) ;exten => 105,2,Voicemail(u2000) ;exten => 105,102,Voicemail(b2000) exten => 105,103,Hangup exten => 106,1,Dial(IAX2/IaxCocomall2@202.138.166.6,20) ;exten => 106,2,Voicemail(u2001) ;exten => 106,102,Voicemail(b2001) exten => 106,103,Hangup exten => 107,1,Dial(IAX2/IaxCocomall3@202.138.166.6,20) ;exten => 107,2,Voicemail(u2002) ;exten => 1...

...gsm ;allow=ulaw ;allow=alaw extensions.conf --------------- [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] ; ; Number 2000 - Dave Laptop #1 ; exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup ; ; Number 2001 - Dave Laptop #2 ; exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup Thanks & Regards Ritesh Jalan Senior Engineer - Test & Audit Net4India Ltd. 703 Bik...

...,2,Playback(silence/1) exten => 2181,3,Goto(default,main,1) exten => 2181,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => i,1,Hangup exten => T,1,Hangup [extentions] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup ;exten => 2002,1,Dial(IAX2/iaxphone,20) ;exten => 2002,2,Voicemail(u2002) ;exten => 2002,102,Voicemail(b2002) ;exten =&g...

...gsm ;allow=ulaw ;allow=alaw extensions.conf --------------- [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] ; ; Number 2000 - Dave Laptop #1 ; exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup ; ; Number 2001 - Dave Laptop #2 ; exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup

i am using the sample config files and get a dial tone. i have also gotten it to play greetings etc, but i need the phone to ring so that i am not tieing up the one phone line, please help, i know this sounds insanely stupid but i cant get it to work.