search for: 500,1

I've been wondering what the difference is in the syntax of things, like Dial. Some examples show things like: exten => 500,1,Dial,SIP/${EXTEN}|10 but other examples show: exten => 500,1,Dial(SIP/${EXTEN}|10) or exten => 500,1,Dial(SIP/${EXTEN},10) Which one is correct? Or most correct? Which one is preferred, and why? I'm sure I'm not the only one with this question... :) Tim -- >>>>>&...

...customer have call 0811XXXX21, he have a answer, he have a music I don't know why the 0811XXXX20 don't have the music for wait that i am answer ... Second Question: It's possible to put into the extension, for access to the VoiceMail, the extension of the caller ? exten => 500,1,VoiceMailMain(@Home) Actually, when i call the 500, he want know my mailbox ID and after password ... if i call with the post 200, it's possible to access direclty at the password ? Thanks bye

I am trying to route incoming DID call (on a analog channel) through Asterisk to an outside (analog) line. My extensions.conf is something like the following: exten => 500,1,Dial,Zap/g1/3105551010 In this case the incoming DID call extension is 500. I am able to dial out and connect with the incoming call, however, the voice conversation is only one way. The called party is not able to hear the calling party. Does anyone have any suggestions of how to better route...

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong? extensions.conf: [office] exten => 601,1,Answer()...

Hi, I have an application, script.exe, written under mono framework and for execute them in my linux box I must write in console: mono script.exe The problem is that when I call this application in dialplan with command: exten => 500,1,System(mono script.exe) the application not run! Somebody can help me to find the problem? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/874efc9e/attachment.htm

Hi, I'm trying to write a simple perl script that will run the following: Action: Originate Channel: local/xxx@callback/r/n Exten: 1234 Context: callback Priority: 1 Extensions.conf exten => 500,1,agi,callback.pl callback perl script: use Net::Telnet (); $mgrUSERNAME='fred'; $mgrSECRET='bloggs'; $server_ip='127.0.0.1'; $tn->print("Action: originate\nExten: 1234\nContext: user\nChannel: local/xxx@user/r/n\nPriority: 1\nCallerid: 1234\n\n"); $tn-&...

...using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf //////sip.conf////////////// //////////////// [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allow=all nat=no [6000] type=peer host=dynamic context=default canreinvite=yes allow=all [1000] t...

...egan posting, so I've got a buttload of email to sift through ... I'm doing this BEFORE I flood the list with my inane questions ... But here goes: I read a reply from one guy to another about recording. The message included this context from extensions.conf: [recordings] exten => 500,1,Festival('Please record your message') exten => 500,2,Record(mymessage:gsm) exten => 500,3,Festival('You said') exten => 500,4,Playback(mymessage) exten => 500,5,Festival('Press 1 to continue or 2 to change your message') exten => 500,6,ResponseTimeout(3) S...

...e columns are in sequence with respect to the variables. I would like to sum up the two measurements for each variable and each observation (the rows) for the whole matrix. Here is an example of such a matrix with 5 variables (R, S, T, U, V) and 50 observations: set.seed(27) Mat <- matrix(rnorm(500), nrow=50, ncol=10) dimnames(Mat)[[2]] <- c("R.A","R.B","S.A","S.B","T.A","T.B","U.A","U.B","V.A","V.B") miss.ind <- rbinom(500,1,prob=0.98) Mat[!as.logical(miss.ind)] <- NA So I would lik...

...; 700,2,GotoIf($[${CONFCOUNT} <= 10]?3:100) exten => 700,3,MeetMe(600,i,1234) exten => 700,100,Playback(conf-full) [macro-voicemail] exten => s,1,Dial(${ARG1},10) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) ;So usrs can dial 500 to access their voicemail exten => 500,1,VoiceMailMain( ) But from any client i dial extension 700 to initiate teh conference i get the following error at asterisk CLI: [Mar 12 15:41:38] WARNING[2756]: pbx.c:1779 pbx_extension_helper: No application 'MeetMe' for extension (internal, 70...

Hi I've problem, i cant get tone on a FXS ZAP channel my configuration are: -- zaptel.conf -- fxoks=1 --zapata.conf -- [channels] immediate=yes context=bell signalling=fxo_ks channel=1 --extensions.conf -- [home] exten => 500,1,Dial(IAX2/guest@misery.digium.com/s@default) [bell] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,Dial(${PHONE},16,tr) ANY IDEA?

Hello all, I've been trying everything I could find, but I can't seem to get my * server connected to my SIP provider (budgetphone.nl). Here's my sip.conf: [budgetphone] port=5060 bindaddr=0.0.0.0 context=from-budgetphone register => 31307110000:secret@budgetphone.nl/500 type=friend host=budgetphone.nl fromuser=31307110000 secret=secret fromdomain=budgetphone.nl username=31307110000 And my extensions.conf: [from-budgetphone] exten => 500,1,Playback(demo-congrats) exten => 500,2,Goto(500,1) Since I don't have an ATA yet, I want to play around with this p...

...lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alexander Lopez Sent: Friday, July 08, 2005 6:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Routing DID calls to external lines Try answering the line first. Exten => 500,1,Answer() exten => 500,2,Dial,Zap/g1/3105551010 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Syed Akbar > Sent: Friday, July 08, 2005 12:01 AM > To: 'Asterisk Users Maili...

I put my voicemail groups into different contexts so that I can use Dial by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)}@default|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicemail regardless of context. #2 can I get the mail buttons to work on my polycom 501s and swissphones #3 where do I put...

Hi, Mr. Spencer written the article "Using DUNDi with a Cluster of Asterisk Servers <http://www.voip-magazine.com/content/view/3644/0/1/0/> " in the VoIP Magazine and the piece follow: [lookupdundi] exten => _X,1,Goto(${ARG1},1) switch => DUNDi/priv exten => i,1,Goto(lookupmysql,${INVALID_EXTEN},1) I didn't get understand the usage ARG1 argument in the context.

...5-6 signalling=fxo_ks ; TDM40B group=2 context=internal channel => 1-4 -------- /etc/asterisk/extensions.conf added: [incoming] exten => s,1,Dial,Zap/1 ;exten => s,1,Dial,Zap/5 [internal] exten => 34,1,Dial,Zap/1 exten => 823,1,Dial,Zap/2 exten => 400,1,Dial,Zap/3 exten => 500,1,Dial,Zap/4 exten => _9X.,1,Dial,Zap/5/${EXTEN} -------- lsmod outputs: Module Size Used by Tainted: P wcfxo 7424 0 (unused) wcfxs 15808 0 (unused) zaptel 183072 0 [wcfxo wcfxs] ppp_generic 15676 0 [z...

...xten => s,2,Answer exten => s,3,Wait(1) exten => s,4,Background(open-hiq) exten => s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID}) exten => s,6,Queue(support||||3600) exten => s,7,Voicemail(100|us) exten => 1,1,Goto(opened,s,6) exten => 500,1,Voicemail(500) thanks, Singer Wang _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

...xten => s,2,Answer exten => s,3,Wait(1) exten => s,4,Background(open-hiq) exten => s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID}) exten => s,6,Queue(support||||3600) exten => s,7,Voicemail(100|us) exten => 1,1,Goto(opened,s,6) exten => 500,1,Voicemail(500) thanks, Singer Wang

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "")...

...FAULT CONFIGURATION wrapper Mon Jan 10 13:33:57 EST 2005 : waiting server Mon Jan 10 13:33:57 2005 : Festival server started on port 1314 client(1) Mon Jan 10 13:34:23 2005 : accepted from localhost client(1) Mon Jan 10 13:34:23 2005 : disconnected extensions.conf ; Record Message exten => _*500,1,Answer exten => _*500,2,Festival('Please record your message') exten => _*500,3,Record(mymessage:gsm) exten => _*500,4,Festival('You said') exten => _*500,5,Playback(mymessage) festival.conf [general] host=asterisk port=1314 usecache=yes cachedir=/var/lib/asterisk/fes...