search for: b2001

...=> 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 extension 2001 exten => s,1,Answer exten => s,2,Dial(SIP/2001,20,tr) exten => s,3,Voicemail(u2001) exten => s,4,Voicemail(b2001) _________________________________________________________...

...#39;pbx_config' ] '0' => 1. Goto(2000|1) [pbx_config] '2001' => 1. Dial(SIP/2001|20|tr) [pbx_config] 2. Voicemail(u2001) [pbx_config] 102. Voicemail(b2001) [pbx_config] 103. Hangup() [pbx_config] '2002' => 1. Dial(SIP/2002|20|tr) [pbx_config] 2. Voicemail(u2001) [pbx_config] 102. Voic...

...[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... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/7a8d632b/attachment.htm

...2 | 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 | from-sip | 2001 | 103 | Hangup | | | 9 | from-sip | 2999 | 1 | VoicemailMain | ${CALLERIDNUM} | +----+----------+-------+----------+---------------+----------------+ 9 rows in set (0.00 sec) Somehow the program get the info '2001,20' s...

...ave 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 attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031115/45c2fd31/attachment.htm

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

...2011,102,Voicemail(b2011) exten => 2011,103,Hangup exten => 2012,1,Dial(SIP/2012,20) exten => 2012,2,Voicemail(u2012) exten => 2012,102,Voicemail(b2012) exten => 2012,103,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(SIP/2002,20) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup [local] ignorepat => 9 include => default include => parkedcalls include => trunklocal include => iaxtel7...

...estion exten => _7x.,1,SetCallerID(my FWD #) exten => _7x.,2,SetCIDName,"FWD.Fahad" exten => _7X.,3,Dial,Zap/1/${EXTEN:1} exten => _7x.,4,Congestion [from-sip] exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => _7X.,1,Dial,Zap/1/${EXTEN:1} [pstn_fwd_forwarding] ;exten => s,1,Dial(IAX2/my_uname:my_secret@iax2.fwdnet.net/my_fwd_number|60|tT) exten => s,1,,BackGround(welcome) exten => s,2,Hangup [sip] include=IAXFWD include=IAXFWD-IN...

...[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 => 107,102,Voicemail(b2002) exten => 107,103,Hangup TIA, Wesley

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten => s,1,Wait(30) exten => s,n,Answer ... exten => s,n,Dial(SIP/brian&SIP/joe,10,H) exten => s,n,Voicemail(u2001) exten => s,n,Hangup exten => s,103,Voicemail(u2001) exten =>

..., 2006-02-02 at 15:24 -0700, Bromont Quebec wrote: > You need to take that "Wait" and "Answer" out of there.... > > [from-pots] > exten => s,1,Dial(SIP/brian&SIP/joe,30) exten => s,2,Voicemail(u2001) > exten => s,3,Hangup exten => s,102,Voicemail(b2001) exten => > s,103,Hangup exten => h,1,Hangup exten => i,1,Hangup How does doing only that prevent Asterisk from picking up the POTS line for a period of time (like 3 or 4 rings... or 10 seconds or so to give a handset on the same POTS line an opportunity to pick it up first -- think a...

...00 - 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 Bikaji Cama Bhawan 11 Bikaji Cama Place New Delhi - 110029 Ph: +91-11-26160129 ext. 131 Cell : +91-9818616329 Web site: http://www.net4india.com ===========================...

...n => 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 => 2002,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) exten => h,1,Hangup exten => t,1,Hangup exten => i,1,Playback...

...ternally. (apologies in advance for the copious code below) Having a look, I have placed the following lines into the extensions.cfg file to allow for the extensions to work. [2001] exten => 2001,1,Dial(SIP/2001,15,t) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup [2002] exten => 2002,1,Dial(SIP/2002,15,t) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup then also in extensions.cfg, I have also set these to allow connection to voiptalk: exten => _0[1-9].,...

...and 3, makes a variable last-call2001 = whoever called, and sets the time. exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,SetGlobalVar(last-call${EXTEN}=${CALLERIDNUM}) exten => 2001,3,SetGlobalVar(2001date=${DATETIME}) exten => 2001,4,Voicemail(u2001) exten => 2001,104,Voicemail(b2001) exten => 2001,105,Hangup ; say your last call ; gets the callerid and then plays last-call'callerid' exten => 128,1,Answer exten => 128,2,Wait(1) exten => 128,3,Playback(lastmisscall) exten => 128,4,SayDigits(${last-call${CALLERIDNUM}}) exten => 128,5,Playback(lastcalla...

...00 - 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

...BLE}} or any hybrid ; [globals] CONSOLE=/dev/dsp [local] ; Operator/ Console exten => 2000,1,Dial,Zap/1|30 exten => 2000,2,Voicemail,u2000 exten => 2000,102,Voicemail,b2000 ; SIP Phones exten => 2101,1,Dial,SIP/snom1|30 exten => 2001,2,Voicemail,u2001 exten => 2001,102,Voicemail,b2001 exten => 2102,1,Dial,SIP/snom2|30 exten => 2002,2,Voicemail,u2002 exten => 2002,102,Voicemail,b2002 exten => 2103,1,Dial,SIP/snom3|30 exten => 2003,2,Voicemail,u2003 exten => 2003,102,Voicemail,b2003 exten => 2104,1,Dial,SIP/snom4|30 exten => 2004,2,Voicemail,u2004 exten...

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

...1+1=102. This +101 jump is built ; into Asterisk and does not need to be defined. ; exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup ; Now, what if the number dialed was 2001? 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(SIP/2002,20) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup exten => 2003,1,Dial(SIP/2003,20) exten => 2003,2,Voicemail(u2003) exten => 2003,102,Voicemail(b2003) exten => 2003,1...

I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which