Re: [asterisk-users] ResponseTimeOut()
bilal ghayyad wrote: > My Asterisk version is 1.4 and I am trying to use the > ResponseTimeOut() application to control the response > time of the Background function, but when the running > arrive for the ResponseTimeOut() then the call drop > and my debuging says: > > No Application 'ResponseTimeout' for extension > (Test_Bilal,s,3) > Spawn extension (Test_Bilal,s,3) exited non-zero on > 'Zap/1-1' > Hangup > > To what this related? There is no ResponseTimeout() in 1.4. Use Set(TIMEOUT(response)=10) core show function TIMEOUT And have a look at core show application WaitExten Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut()
On Fri, 2007-10-19 at 07:22 -0700, bilal ghayyad wrote: > My Asterisk version is 1.4 and I am trying to use the > ResponseTimeOut() application to control the response > time of the Background function, but when the running > arrive for the ResponseTimeOut() then the call drop > and my debuging says: > > No Application 'ResponseTimeout' for extension Use the TIMEOUT() function like this: exten => 123,n,Set(TIMEOUT(response)=5) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut()
ResponseTimeout was deprecated in 1.2 and removed in 1.4. Was this information not in the upgrade.txt file in 1.2 and 1.4? bilal ghayyad wrote: > Hi List; > > My Asterisk version is 1.4 and I am trying to use the > ResponseTimeOut() application to control the response > time of the Background function, but when the running > arrive for the ResponseTimeOut() then the call drop > and my debuging says: > > No Application 'ResponseTimeout' for extension > (Test_Bilal,s,3) > Spawn extension (Test_Bilal,s,3) exited non-zero on > 'Zap/1-1' > Hangup > > To what this related? > > About my extensions.conf file, I set priorityjumpin = > yes and I set autofallthrough = no (and I am sure it > is not related to the problem with ResponseTimeout > application). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResponseTimeOut()
Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup To what this related? About my extensions.conf file, I set priorityjumpin = yes and I set autofallthrough = no (and I am sure it is not related to the problem with ResponseTimeout application). Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
bilal ghayyad wrote: > Dear Phellepe; ? It's a bit uncommon to change other people's names. > It was 1.4 and I set priorityjumping and set > autofallthrough and look like fine, need to test more. Ok. So you seem to have made your decision. Although I don't understand why there's no need to do testing as you said your dialplan doesn't work. > But there are two issues related to your reply: > > 1) Why I do not have to use priority jumping You are free to use it, but it is deprecated, meaning future versions of Asterisk will not support it. Apart from that it looks ugly and makes your dialplan hard to maintain. > and what > is the alterantive better solution (where I can see > such these examples)? [macro-stdexten] in extensions.conf for example. > But by the way, I need a service provider, so what do > you know about dundi and how I can send calls for > them? Are they really a group that have routes? DUNDi is a protocol http://en.wikipedia.org/wiki/DUNDI http://www.voip-info.org/wiki-DUNDi Start a new thread please and ask the list, not me. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
bilal ghayyad wrote: > Can someone advise me why in the below context, it > does not run the Background step? > [Test_Bilal] > > include => KuwaitInternal > include => EgyptInternal > exten => 1000,1,Goto(s,1) > exten => s,1,Answer() > exten => s,2,ResponseTimeout(5) > exten => s,3,Background(WelcomeMessage) > exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) > exten => 0,2,Background(WelcomeMessage) > exten => 0,2,Playback(vm-nobodyavail) > exten => 0,3,Hangup() 1, 2, 2, 3. - That is not supposed to work. > exten => 0,102,Playback(tt-allbusy) > exten => 0,103,Hangup() Don't use priority jumping. There are many examples how to do it better. btw: Kuwait, Egypt - Are you going to become a VoIP provider ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
On Sunday 14 October 2007 17:35:04 bilal ghayyad wrote: > Can someone advise me why in the below context, it > does not run the Background step? Once I dial 1000, > then it hangup and give congestion signal? If I > comment the ResponseTimeOut, then it run the > Background but it does not wait till caller enter the > digits, once the sound file finish, then it hangup > (congestion signal), also in all the situation, it > does not go for the t extension, why? Is it because I > am originating the call from local extension (an > handset connected to FXS port) and the call should be > originated from FXO or IP Trunk, or what is the > problem exactly? > > [Test_Bilal] > > include => KuwaitInternal > include => EgyptInternal > exten => 1000,1,Goto(s,1) > exten => s,1,Answer() > exten => s,2,ResponseTimeout(5) > exten => s,3,Background(WelcomeMessage) > exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) > exten => 0,2,Background(WelcomeMessage) > exten => 0,2,Playback(vm-nobodyavail) > exten => 0,3,Hangup() > exten => 0,102,Playback(tt-allbusy) > exten => 0,103,Hangup() > exten => i,1,Playback(pbx-invalid) > exten => i,2,Goto(EgyptIncomingPSTN,s,1) > exten => t,1,Playback(vm-goodbye) > exten => t,2,Hangup() Go read the top of configs/extensions.conf.sample, specifically the part about the "autofallthrough" parameter. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut() and t extension
bilal ghayyad wrote: > Hi List; > > Can someone advise me why in the below context, it > You never told us what version you are running. If it's version 1.2, make sure you have set priorityjumping=no in your extensions.conf or use the waitexten application. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResponseTimeOut() and t extension
Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why? Is it because I am originating the call from local extension (an handset connected to FXS port) and the call should be originated from FXO or IP Trunk, or what is the problem exactly? [Test_Bilal] include => KuwaitInternal include => EgyptInternal exten => 1000,1,Goto(s,1) exten => s,1,Answer() exten => s,2,ResponseTimeout(5) exten => s,3,Background(WelcomeMessage) exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) exten => 0,2,Background(WelcomeMessage) exten => 0,2,Playback(vm-nobodyavail) exten => 0,3,Hangup() exten => 0,102,Playback(tt-allbusy) exten => 0,103,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(EgyptIncomingPSTN,s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() Any help?? Regards Bilal Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
On Sun, 2004-11-14 at 22:13 -0700, Joseph wrote: [snip] > Yes, I was looking at it already but it is available in ver. 1.0.0 and > up; I'm on 0.9 on Gentoo. Gentoo is kind of slow when it comes to > Asterisk. There is an unstable ver. 1.0.2 in unstable branch but it > doesn't compile (there is an error when compiling). FYI: the 1.0.2 version is actually stable and part of the stable branch which can be downloaded with: cvs co -r v1-0 zaptel libpri asterisk The v1-0 tag will get you the latest stable release from cvs while it is also possible to use v1-0-1 to get version 1.0.1, v1-0-2 for 1.0.2 etc. > So I will have to learn how to upgrade using CVS or wait for Gentoo > stable version. Checkout the shell script that was posted to the mailing list last week. It automates the upgrading & building process. You can find it here: http://www.szmidt.org/asterisk/asterisk-update.sh > If I use CVS I'm not sure if startup scrip will be > upgraded as well in /etc/init.d/ When I install my updated stable-cvs builds (which are rpms), it upgrades/replaces the startup scripts. Not really an issue afaik. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: This is correct. However, if you wish to have it pause between priorities for an extension change, try using the application "WaitExten". Here's a show application waitexten: -= Info about application 'WaitExten' =- Yes, I was looking at it already but it is available in ver. 1.0.0 and up; I'm on 0.9 on Gentoo. Gentoo is kind of slow when it comes to Asterisk. There is an unstable ver. 1.0.2 in unstable branch but it doesn't compile (there is an error when compiling). So I will have to learn how to upgrade using CVS or wait for Gentoo stable version. If I use CVS I'm not sure if startup scrip will be upgraded as well in /etc/init.d/ I don't see any reason the startup script would need to be updated. Apparently, however, gentoo has 1.0.1 is in the portage tree now, if you do an emerge sync, or possibly you need to look deeper. I'm not a gentoo user by any means, but this is being reported to me by a gentoo user at this very moment. -J ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
> This is correct. However, if you wish to have it pause between > priorities for an extension change, try using the application > "WaitExten". Here's a show application waitexten: > > -= Info about application 'WaitExten' =- Yes, I was looking at it already but it is available in ver. 1.0.0 and up; I'm on 0.9 on Gentoo. Gentoo is kind of slow when it comes to Asterisk. There is an unstable ver. 1.0.2 in unstable branch but it doesn't compile (there is an error when compiling). So I will have to learn how to upgrade using CVS or wait for Gentoo stable version. If I use CVS I'm not sure if startup scrip will be upgraded as well in /etc/init.d/ -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: [snip] I'm assuming you have snipped the priorities 1 and 2. exten => s,3,BackGround(welcome) exten => s,4,ResponseTimeout,15 exten => t,1,Goto(1,1) Description ResponseTimeout(seconds) Set the maximum amount of time permitted after falling through a series of priorities for a channel in which the user may begin typing an extension. If the user does not type an extension in this amount of time, control will pass to the 't' extension if it exists, and if not, the call would be terminated. If ResponseTimeout is not explicitly set in an extension, the default value of 15 seconds will be used. Thank you, it work! So it needs to be pass to: exten => t,1,Goto(1,1) I got confused by the last sentence "...and if not, the call would be terminated." This is correct. However, if you wish to have it pause between priorities for an extension change, try using the application "WaitExten". Here's a show application waitexten: -= Info about application 'WaitExten' =- [Synopsis]: Waits for some time [Description]: Wait([seconds]): Waits for the user to enter a new extension for the specified number of seconds, then returns 0. Seconds can be passed with fractions of a seconds (eg: 1.5 = 1.5 seconds) or if unspecified the default extension timeout will be used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
[snip] > I'm assuming you have snipped the priorities 1 and 2. > > exten => s,3,BackGround(welcome) > exten => s,4,ResponseTimeout,15 > exten => t,1,Goto(1,1) > > Description > ResponseTimeout(seconds) > > Set the maximum amount of time permitted after falling through a series > of priorities for a channel in which the user may begin typing an > extension. If the user does not type an extension in this amount of > time, control will pass to the 't' extension if it exists, and if not, > the call would be terminated. > > If ResponseTimeout is not explicitly set in an extension, the default > value of 15 seconds will be used. Thank you, it work! So it needs to be pass to: exten => t,1,Goto(1,1) I got confused by the last sentence "...and if not, the call would be terminated." -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: exten => s,3,BackGround(welcome) exten => s,4,ResponseTimeout,15 exten => s,5,Goto(1,1) You have misunderstood how ResponseTimeout works. It does not delay between priorities, it controls the amount of time the caller will have after the last priority is executed before Asterisk jumps to the "t" (timeout) priority. A simple fix would be: exten => s,3,ResponseTimeout(15) exten => s,4,Background(welcome) exten => t,1,Goto(1,1) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: I'm trying to implement ResponseTimeout to give a customer a few extra seconds before ringing the phone. But it doesn't work or I'm doing it the wrong way. exten => s,3,BackGround(welcome) exten => s,4,ResponseTimeout,15 exten => s,5,Goto(1,1) After playing "welcome" message it goes straight to 1,1 and ring the phone. How do I pause for 15sec and give customer some time to enter an option? I'm assuming you have snipped the priorities 1 and 2. exten => s,3,BackGround(welcome) exten => s,4,ResponseTimeout,15 exten => t,1,Goto(1,1) Description ResponseTimeout(seconds) Set the maximum amount of time permitted after falling through a series of priorities for a channel in which the user may begin typing an extension. If the user does not type an extension in this amount of time, control will pass to the 't' extension if it exists, and if not, the call would be terminated. If ResponseTimeout is not explicitly set in an extension, the default value of 15 seconds will be used. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ResponseTimeout problem
I'm trying to implement ResponseTimeout to give a customer a few extra seconds before ringing the phone. But it doesn't work or I'm doing it the wrong way. exten => s,3,BackGround(welcome) exten => s,4,ResponseTimeout,15 exten => s,5,Goto(1,1) After playing "welcome" message it goes straight to 1,1 and ring the phone. How do I pause for 15sec and give customer some time to enter an option? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ResponseTimeout, Straight to operator?
Check your config file. the 't' doesn't stand for terminate. It stands for timeout http://www.voip-info.org/wiki-Asterisk+t+extension Try adding your operator to the 't' extension instead of hanging up on them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Campbell Sent: Monday, July 26, 2004 1:03 PM To: Asterisk-Users Subject: [Asterisk-Users] ResponseTimeout, Straight to operator? Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: --- ; Wait 15 seconds for an answer (pick up the local phone) exten => s,1,Wait,2 ; Answer the phone exten => s,2,Answer ; Playback generic voice mail message exten => s,3,Background(ts_welcome_en) ; They must respond within 10 seconds ; else jump to priority t (terminate call) exten => s,4,DigitTimeout,3 exten => s,5,ResponseTimeout,10 ;Sales Voicemail ;exten => 1,1,Dial(SIP/2001,15) exten => 1,1,Dial(Zap/3,20) exten => 1,2,Playback,vm/7000/unavail exten => 1,3,Voicemail,7000 exten => 1,4,Goto,t|1 ;Support Voicemail exten => 2,1,Dial(Zap/3,20) exten => 2,2,Playback,vm/7000/unavail exten => 2,3,Voicemail,7000 exten => 2,4,Goto,t|1 ;Reception Voicemail exten => 0,1,Dial(Zap/3,20) exten => 0,2,Playback,vm/7000/unavail exten => 0,3,Voicemail,7000 exten => 0,4,Goto,t|1 ; t - terminate call exten => t,1,Playback,vm-goodbye exten => t,2,Hangup -- Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ResponseTimeout, Straight to operator?
Try This... exten => s,1,Wait,2 exten => s,2,Answer exten => s,3,DigitTimeout,3 exten => s,4,ResponseTimeout,10 exten => s,5,Background(ts_welcome_en) exten => s,6,Dial(Zap/3,20) Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout, Straight to operator?
On Mon, 2004-07-26 at 14:03, John Campbell wrote: > My client wants incoming callers who do not press a digit to go straight > to the operator. Does anyone have an idea of how this could be done? > I've looked for some examples, but I'm still not clear on it. > > Here's the relevant portion of my extensions.conf: > --- > ; Wait 15 seconds for an answer (pick up the local phone) > exten => s,1,Wait,2 > > ; Answer the phone > exten => s,2,Answer > > ; Playback generic voice mail message > exten => s,3,Background(ts_welcome_en) > > ; They must respond within 10 seconds > ; else jump to priority t (terminate call) > exten => s,4,DigitTimeout,3 > exten => s,5,ResponseTimeout,10 > > ;Sales Voicemail > ;exten => 1,1,Dial(SIP/2001,15) > exten => 1,1,Dial(Zap/3,20) > exten => 1,2,Playback,vm/7000/unavail > exten => 1,3,Voicemail,7000 > exten => 1,4,Goto,t|1 > > ;Support Voicemail > exten => 2,1,Dial(Zap/3,20) > exten => 2,2,Playback,vm/7000/unavail > exten => 2,3,Voicemail,7000 > exten => 2,4,Goto,t|1 > > ;Reception Voicemail > exten => 0,1,Dial(Zap/3,20) > exten => 0,2,Playback,vm/7000/unavail > exten => 0,3,Voicemail,7000 > exten => 0,4,Goto,t|1 > > ; t - terminate call > exten => t,1,Playback,vm-goodbye > exten => t,2,Hangup ; t - timeout exten => t,1,Dial(Zap/3/operator-ext) -- respectfully, Joseph = -= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ResponseTimeout, Straight to operator?
Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: --- ; Wait 15 seconds for an answer (pick up the local phone) exten => s,1,Wait,2 ; Answer the phone exten => s,2,Answer ; Playback generic voice mail message exten => s,3,Background(ts_welcome_en) ; They must respond within 10 seconds ; else jump to priority t (terminate call) exten => s,4,DigitTimeout,3 exten => s,5,ResponseTimeout,10 ;Sales Voicemail ;exten => 1,1,Dial(SIP/2001,15) exten => 1,1,Dial(Zap/3,20) exten => 1,2,Playback,vm/7000/unavail exten => 1,3,Voicemail,7000 exten => 1,4,Goto,t|1 ;Support Voicemail exten => 2,1,Dial(Zap/3,20) exten => 2,2,Playback,vm/7000/unavail exten => 2,3,Voicemail,7000 exten => 2,4,Goto,t|1 ;Reception Voicemail exten => 0,1,Dial(Zap/3,20) exten => 0,2,Playback,vm/7000/unavail exten => 0,3,Voicemail,7000 exten => 0,4,Goto,t|1 ; t - terminate call exten => t,1,Playback,vm-goodbye exten => t,2,Hangup -- Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
