similar to: Busy when not registered

Folks! 1) I have modified the original sip.conf and extension.conf file instead of writing mine. This looks like a mistake. 2)I have fired off Asterisk Extensions conf with 2 extensions i.e 2000 and 2001 and made one test call. I forgot to set a time out. The calls between these two extensions were partially successful. After writing my own files, it started working. I went through following

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

I'm having the same problem here. Any solution to this problem? -Manuel (sorry for top-posting, I'm having a stupid mail client here) -----Messaggio originale----- Da: Simon Brown [mailto:Simon.Brown@otterson.com.au] Inviato: giovedì, 1. luglio 2004 02:05 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Dial plan errors I am attempting to implement the new features added

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Hi, although setting up voicemail for busy and unavailable should be easy, things aren't working the way they should in my configuration (asterisk 1.2.14 bristuffed): Here's the relevant part of the extensions.conf: exten => 56830976,1,Answer() exten => 56830976,2,Dial(SIP/hbaumgart,20,tr) exten => 56830976,3,VoiceMail,u76 exten => 56830976,4,Hangup exten =>

I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, i need some suggestion how to indicate caller that calling number is unavailable if some iax user is not registered: this is what I got in asterisk console: app_dial.c:727 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy/congested at this time why is send to busy congested but user is just unavailable !? any way to play unavailable message if user is

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

Asterisk seems to regard an unregistered phone to be busy. Is that correct? Is not an unregistered phone unavailable? It is odd to me that if someone dials an unregistered extension, then the dialplan jumps to busy and voicemail kicks in saying that the person is on the phone, when clearly they can't be if the phone hasn't registered. Any way around this?

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

Hi, The voicemail app allows you to record your own busy and unavailable messages by pressing 0 (mailbox options). But it doesn't seem to provide a way to review the message before accepting it. Therefore if you don't get the message right the first time, then you have to redial, login and try again. Is this how voicemail currently works, or is there a "review and commit" type

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

We have a customer who wants to forward an extension to their cell phone, if and only if that extension is "unavailable", or when the Dial() command times out. However, should the Dial() command return "busy" it should go to voicemail instead. As far as I know, the dialplan doesn't support this. Certainly not natively or in any particularly easy or obvious way, and I

I have found that fax detection is returning an error saying that no fax extension is present when I have defined one. The console returns this error: Aug 26 10:58:41 NOTICE[1112745536]: chan_zap.c:3989 zt_read: Fax detected, but no fax extension extensions.conf has: [default] exten => fax,1,Hangup exten => fax,2,Congestion exten => fax,102,Congestion exten => f,1,Hangup exten =>

Hi guys, What is different and the "context" to play unavail message and busy message? When a SIP connection is unregistered, voicemail will play unavail message, right? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/1a2d1c81/attachment.htm

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };