search for: u1234

...te happy with how it works. Having voicemail emailed to me and notify my cell phone via SMS is a great way to impress my friends. :-) The inbound context for my X101P looks something like this: exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Answer exten => s,3,Voicemail(u1234) exten => s,4,Hangup The trouble is, some unknown caller (probably a telemarketer, but I have no CID) keeps calling and hanging up after precisely four rings, causing * to pick up the line and get a dialtone. It will happily play the outbound message, then record 90 seconds of dialtone and han...

...mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox misconfiguration. The strange thing is that both extensions and mailboxes are configured exactly the same : in extensions.conf : exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 6004,1,Voicemail(u6004) in voicemail.conf : 1234 => 4242,Test mailbox,jim@jipo.com 6004 => 4242;Other test mailbox,jim@jipo.com I don't understand why these two seemingly identical configuration yield different results. I guess that I mus...

Hello, I searched a while to find out, if there is an sftp-server implementation which provides an option similar to the defaultroot of proftpd. A typical use would be: DefaultRoot = ~ The option does the follwing: Once the use logs in, it determines the home directory of the user .ie /home/u1234 and takes this as the users root. The user cannot escape that root (he can not look at /tmp for example). Each file within his home directory is presented without the home directory prefix (for example: /index.html instead of /home/u1234/index.html). This is very different from the chroot optio...

...like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the first or the second "exten", but I always get the following message: "Unable to create channel of type 'Zap' (cause 0)" Here is the log console: -- Executing Dial("SIP/leonardo-8cc3", "Zap/g1/482343400|5|rt") in new...

...changed a bit sip.conf: [general] ; ... dtmfmode = inband disallow = all allow = ulaw allow = alaw allow = gsm run kphone, and call the 1235 extension. According to sample extensions.conf, Asterisk would dircet me right to the Voicemail: [demo] ; ... exten => 1235,1,Voicemail(u1234) ; Right to voicemail Yes, it records my message, but I couldn't stop recording by pressing any key, neither '#', not '*', not something else. So I have to wait untill it stops automatically after 10 sec. Where am I wrong? Much thanks. -- Best regards, Timur Elzhov

...##') exten => 1006,3,RemoveQueueMember(sporski|Local/${CALLERIDNUM}@agentes) exten => 1006,4, Hangup exten => 1235,1,Dial(SIP/1235,30,,http://www.teste.pt) exten => 1235,2,VoiceMail(u1235) exten => 1235,3,Hangup exten => 1234,1,Dial(SIP/1234,20) exten => 1234,2,VoiceMail(u1234) exten => 1234,3,Hangup [agentes] exten => 1234,1,Dial(local/1234) exten => 1235,1,Dial(local/1235) exten => 1236,1,Dial(local/1236) ;queue.conf [test1] music = random musiconhold=default strategy = ringall announce-holdtime = once joinempty = yes eventwhencalled = yes eventmember...

...he number from the BT100 I can see 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,Ha...

...h exten => 3,2,Goto(s,5) ; Start with the congratulations exten => 1000,1,Goto(default,s,1) exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" exten => #,2,Hangup ; Hang them up. exten => t,1,Goto(#,1) ; If they take too l...

...als Local/6001@default-1 -- 6001@default <unowned> -- __se@s 2 *CLI> show locals Local/6001@default-1 -- 6001@default <unowned> -- __se@s 2 -- Nobody picked up in 20000 ms -- Hungup 'IAX[peracles]/4' -- Executing VoiceMail("Local/6001@default-1", "u1234") in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/1234/unavail' -- Playing 'vm-intro' -- Playing 'beep' -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0012 WARNING[21525]: File app_voicemail.c, Line 631 (leave...

...; 1234,1,dial(sip/1234,20) exten => 1234,2,playback(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...

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

...the congratulations exten => 1000,1,Goto(default,s,1) exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,n,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,n,Voicemail(u1234) ; Unless busy exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" exten => #,n,Hangup...

...lt,s,1) ; ; We also create an example user, 1234, who is on the console and has ; voicemail, etc. ; exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy ; ; # for when they're done with the demo ; exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" exten => #,2,Hangup ; Hang them up....

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

...lt,s,1) ; ; We also create an example user, 1234, who is on the console and has ; voicemail, etc. ; exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy ; ; # for when they're done with the demo ; exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" exten => #,2,Hangup ; Hang them up....

...ransfer,skip) ; "Please hold while..." exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; ; (but skip if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) ; exten => 1235,1,Voicemail(u1234) ; Right to voicemail ; exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy ; ; # for when they're done with the demo ; exten => #,1,Playback(demo-thanks) ; "Thanks for trying the...

...lt,s,1) ; ; We also create an example user, 1234, who is on the console and has ; voicemail, etc. ; exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy exten => 2000,1,Dial(SIP/2000,30,tr) exten => 2001,1,Dial(SIP/2001,30,tr) exten => 2002,1,Dial(SIP/2002,30,tr) ; ; # for when they're done with the demo ;...

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

...create an example user, 1234, who is on the console and has ; voicemail, etc. ; exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234,n,Macro(stdexten,1234,${CONSOLE}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,n,Voicemail(u1234) ; Unless busy ; ; # for when they're done with the demo ; exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" exten => #,n,Ha...

...4.35.58/200@aullox) 200.64.35.58 is my ip 200 is tehe xtension and aull is the context and at veracruz i have: [aullox] exten => 200,1,Wait,2 exten => 200,2,Playback(transfer,skip) ; "Please hold while..." exten => 200,3,Dial(Phone/phone0) exten => 200,104,Voicemail(u1234) but, I get this when I call gdl->ver *CLI> NOTICE[10251]: File chan_iax.c, Line 3628 (socket_read): Rejected connect attempt from 200.67.99.127, request 'exten=200;language=en;context=aullox;username=aullox_gdl;formats=2;capability=65283;version=1;adsicpe=0' av veracruz.. and in...