similar to: "No Mailbox" Prompt

Displaying 20 results from an estimated 20000 matches similar to: ""No Mailbox" Prompt"

2011 Apr 15
2
If voice mail not found dialplan
Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone
2006 Oct 30
1
dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
2008 Jan 26
1
CHANUNAVAIL
I've got a setup where we have 100 DID's. Our default dialplan has one line that calls a macro: exten => _22XX,1,Macro(STDEXT,${EXTEN}) The macro is pretty basic: [macro-STDEXT] exten => s,1,NoOp exten => s,2,Dial(SIP/${ARG1},15,Tt) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${ARG1}|u) exten => s-NOANSWER,n,Hangup exten =>
2007 May 24
1
vmoutcall]
--> Perhaps someone can share how? First you need to give them the option of turning the feature on and off. I do it with the following: [callback-activate] ; *********************************************** ; Callback activate/deactivate. If this function ; is enabled and there is a call file in the form ; of ${EXTEN}.call, then Asterisk will call the ; phone number contained within the
2007 Jun 16
2
MixMonitor Problem
Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>
2007 Sep 02
1
How can i send my sip channel 3 to mailbox 2? Please Help!
Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to
2007 Aug 27
3
voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when
2008 Nov 13
1
Parking help - causing Asterisk crash
Hi, I am having some trouble with parked calls timing out. In features.conf: [general] parkext => 800 ; What extension to dial to park parkpos => 801-820 ; What extensions to park calls on context=parkedcalls parkingtime=120 After the Park timesout it calls the phone that the call was parked from. If the phone is BUSY the call just get dropped. (Call waiting
2007 Oct 13
3
'Start' in extension rules
I can't seem to get the [s]tart to work in my extensions... ----- s n i p ----- [default] exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-BUSY,1,Voicemail(${EXTEN}, b) exten => 2403,1,Dial(sip/${EXTEN},20,t) exten => _X.,2,Playback(pbx-invalid) ----- s n i p ----- If I dial '2403' with is off-hook, I don't get to the voice mail, I get the playback... Setting
2008 Apr 01
4
Voicemail- Recorded Mesage Low Volume
Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on
2007 Jan 17
3
Callback/ringback
Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is
2009 Dec 12
1
Playing a message if my call lands in their voicemail
Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this
2007 Apr 02
3
SIP - Automatic Redial on No Answer
Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified
2012 Jun 17
1
Missing voicemail prompt beginning
Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started
2011 May 03
2
dial from voicemail
Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail or to just leave a message at the tone. Thanks Kelly
2006 Dec 18
3
Inform callers on recorded/monitored number.
Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best
2014 May 10
1
One mailbox for multiple extensions with individual greetings
Hi, Is there a way in Asterisk 11 to use a single voicemailbox for multiple extensions while still hearing each extension's individual greeting? Use case: someone has 2 numbers and wants all voicemail messages for both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you would hear "the person at extension 1234 is unavailable" and the message would be stored
2009 Apr 16
7
How to send "404 Not found" SIP reply?
Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404
2005 Mar 03
3
Why ${EXTEN} variable changes after Goto ?
Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN
2009 Feb 05
6
Newbie query: how to write priority n+101
Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)