search for: b1234

...all the incoming information but still nothing on my X-Lite. My extensions.conf: [general] static=yes writeprotect=no [globals] [sip] exten => 1,1,Dial(SIP/phone1,20,tr) exten => 2,1,Dial(SIP/phone2,20,tr) exten => 2,2,VoiceMail,u1234 exten => 2,102,VoiceMail,b1234 ;exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain,s1234 exten => 6601,1,WaitMusicOnHold(60) exten => 232999,1,Dial(SIP/phone1,30,tr) exten => 232999,2,Hangup I am behind a NATed fir...

...layback(pls-wait-connect-call) exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 1234,4,SetCIDNum(0${CALLERIDNUM}) exten => 1234,5,dial(${TRUNK}c/9871234321,20,r) exten => 1234,6,SetCIDNum(${NewCaller}) exten => 1234,7,voicemail2(u1234@default) exten => 1234,101,voicemail2(b1234@default) exten => 1234,102,hangup Mine looks like this... exten => 08700688nnn,1,Dial(SIP/operator,1,t) exten => 08700688nnn,2,playback(pls-wait-connect-call) exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM}) exten => 08700...

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

...40]: File channel.c, Line 1558 (ast_request): No channel type registered for 'Console' NOTICE[1192437440]: File app_dial.c, Line 495 (dial_exec): Unable to create channel of type 'Console' == Everyone is busy at this time -- Executing VoiceMail("SIP/-081058b8", "b1234") in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/1234/busy' -- Playing 'vm-intro' -- Playing 'beep' -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0015 -- User hung up == Parsing '/etc/asterisk/voic...

...ayed AFTER I answered the channel. This is more or less what I'm trying to do: exten => 301,1,SetLanguage(en) exten => 301,2,SetMusiconHold(default) exten => 301,3,Dial(SIP/301,5,tr) exten => 301,4,Playback(transfer) exten => 301,5,VoiceMail,u1234 exten => 301,105,VoiceMail,b1234 Is it possible, or am I asking too much from Asterisk? Thanks to all, manny

...layed AFTER I answered the channel. This is more or less what I'm trying to do: exten => 301,1,SetLanguage(en) exten => 301,2,SetMusiconHold(default) exten => 301,3,Dial(SIP/301,5,tr) exten => 301,4,Playback(transfer) exten => 301,5,VoiceMail,u1234 exten => 301,105,VoiceMail,b1234 Is it possible, or am I asking too much from Asterisk? Thanks to all, manny

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

Hi. I've recently installed Asterisk on my Linux system and added the asterisk-h323 package. I'm working with a H323 plataform with gatekeepers and gateways. I'm trying to implement a Voicemail for the endpoints, that works when the endpoinst are BUSY. I'm a newbie in asterisk so i need a little help here... 1.- I have succesfully route the BUSY calls from my endpoints to my

...(2) exten => _.,4,Playback(this-call-is-being-recorded) exten => _.,5,Goto(from-sip-post,${EXTEN},1) [from-sip-post] ; Now, send the call to where it's really supposed to go, while recording. ; exten => 1234,1,Dial(SIP/1234,40,r) exten => 1234,2,Busy exten => 1234,102,Voicemail2(b1234) exten => 5554,1,Dial(SIP/5554,40,r) exten => 5554,2,Busy exten => 5554,102,Voicemail2(b5554) ; after the call is over, make sure to clean up the files and translate them ; appropriately... exten => h,1,Macro(record-cleanup) ; --end--