Re: [asterisk-users] Setting up ring group
Tom Moore wrote: Hi guys, What's the best way to setup a ring group that contains 6 extensions so that when a call comes in there starts a 30 second timer and the first available device is rang instead of ringing all extensions at the same time? What I want it to do is cycle through the extensions and have the system ignore the ones that are busy and if there are not any free extensions in the ring group to have the system drop the caller to voicemail. If none of the extensions are present in the group I'd like to also drop to voicemail. Basically what I'm looking for is a multiple extensions version of the standard extension macro with multiple devices and the exten busy state ignored. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi I had the same problem. At the beginning I thought of implementing agents and queues. But that's not what I wanted. I didn't go on and look how to configure members (perhaps that would've been the better solution), maybe because I'm always thinking on how to program something and I'm not always aware that there are already solutions to many problems out there. Anyway, that's how it looks like in my extensions.conf [wait-op] ; Ask if the channel is available, if it is ; go to the next step. If it isn't go to no-op ; and skip the delay. exten = _XX,1,ChanIsAvail(SIP/${EXTEN}) exten = _XX,n,GotoIf($[ ${AVAILCHAN}= ]?no-op|s-na|1:3) ; Increment the delay by a value of five. exten = _XX,n,Set(DB(cross/delay-${key})=$[${DB(cross/delay-${key})}+5]) exten = _XX,n,Wait(${DB(cross/delay-${key})}) exten = _XX,n,Dial(SIP/${EXTEN}) [no-op] ; Do nothing exten = s,1,NoOp(Dummy) exten = s-na,1,NoOp(Channel is not available) [hotline-0] ; Define a custom name for the caller ID. ; This was an extra that I did exten = s,1,Set(CALLERID(name)=hotline ${CALLERID(name)} ${CALLERID(num)}) ; Set a key unique for each channel. So id doesn't matter how ; many calls we get, there will always exist just one key per channel ; This way we increase the delay only when we want to. exten = s,n,Set(__key=${CHANNEL}) ; Define the initial delay value on the database. That's even better than ; a global variable. One advantage, pointed out by a collegue of mine, is ; that when the process is over, you can delete the key from the DB. exten = s,n,Set(DB(cross/delay-${key})=-5) ; Set all the devices as a single variable. ; Note that all of them use the Local context exten = s,n,Set(dg0=Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Set(dg0=${dg0}Local/[EMAIL PROTECTED]) exten = s,n,Dial(${dg0}|80) ; Manage the voicemail with a macro exten = s,n,Macro(hotline-voicemail|${DIALSTATUS}|0) ; Delete the keys at hangup exten = h,1,NoOp(DB_DELETE(cross/inc-${key}) exten = h,n,Hangup [macro-hotline-voicemail] ; ${ARG1} Dialstatus ; ${ARG2} Whose voicemail? exten = s,1,Set(CHANNEL(language)=de) exten = s,n,Goto(s-${ARG1},1) exten = s-BUSY,1,Voicemail(${ARG2},b) exten = s-NOANSWER,1,Voicemail(${ARG2},u) exten = s-CONGESTION,1,Voicemail(${ARG2},b) exten = s-CHANUNAVAIL,1,Voicemail(${ARG2},u) [default] exten = 0,1,Goto(hotline-0|s|1) ... I hope it works for you :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] increase ring time out
Fidel Garcia wrote: I need to increase the ringing timeout on the AA50 appliance. How do I accomplish this? I need the phones to ring a bit more before the caller gets to the voicemail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Could you show your extensions.conf? Normally you'd do that in the Dial command: exten = _XX,1,Answer exten = _XX,n,Dial(SIP/1,20) ... Where 20 is the time you're letting the phone ring... :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incompatible voice frame panic! [SOLVED]
Vazquez David wrote: Hi all, Panic! Panic! When I get a call over mISDN to my IAX extension and try to transfer it to another IAX/SIP, I get this message: Dropping incompatible voice frame on ... of format ulaw since our native format has changed to alaw Immediately followed by one almost the same: Dropping incompatible voice frame on ... of format alaw since our native format has changed to ulaw and so on, and so forth... Any ideas??? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Phew! Solved... The only thing I changed was, in my iaxprov.conf changed codecpriority=host to codecpriority=reqonly. Now everything works smoothly... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incompatible voice frame panic!
Hi all, Panic! Panic! When I get a call over mISDN to my IAX extension and try to transfer it to another IAX/SIP, I get this message: Dropping incompatible voice frame on ... of format ulaw since our native format has changed to alaw Immediately followed by one almost the same: Dropping incompatible voice frame on ... of format alaw since our native format has changed to ulaw and so on, and so forth... Any ideas??? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX + Inidication
Hi all, I have a little problem with my IAX phones... When I pick up the headset I don't get a dialtone, and whenever I dial to a SIP phone I don't get an indication tone... Ideas? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold
Hi, I'm getting this bizarre problem. Whenever I dial (through misdn) and try to listen to my music on hold, I get this: -- Started music on hold, class 'default', on channel 'mISDN/3-u72' [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 6644 requested bytes on mISDN/3-u72 -- Stopped music on hold on mISDN/3-u72 Any idea??? Thanks :D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Vazquez David wrote: Hi, I'm getting this bizarre problem. Whenever I dial (through misdn) and try to listen to my music on hold, I get this: -- Started music on hold, class 'default', on channel 'mISDN/3-u72' [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 6644 requested bytes on mISDN/3-u72 -- Stopped music on hold on mISDN/3-u72 Any idea??? Thanks :D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Solved : I didn't have an answer statement in my extensions.conf The working context: exten = 03,1,Answer() exten = 03,2,Queue(${EXTEN}) exten = 03,3,Hangup() ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Grygoriy Dobrovolskyy wrote: To see? how? what phone do you use? Snoms imprement that, you got BLINKING and ON state BLINKING=calling or being called ON=on the phone 2008/6/24 Vazquez David [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I use Snoms. I know there's the feature. I just don't know how to use it, and there's so little documentation on the web.. Anyway, with see I meant that the secretary's phone would have one of the function keys on whenever the chef is on the phone (also when he picks it up, right before dialing). Until now I've only managed to make both phones blink on incoming calls. But that's not what I want and I could've done that with extension = 11,1,Dial(SIP/11SIP/12SIP/13...). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Gordon Henderson wrote: On Tue, 24 Jun 2008, Vazquez David wrote: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. It's like BLF because that's exactly what it's for.. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Yes. Configure hints in asterisk and BLF in the secretarys phone. I don't understand how you get notifications one way, but not the other though (unless you're doing it by not using BLF) What phones have you got? What do your hints look like in the extensions.conf file? You can't get the status of lifting the handset on a SIP phone though (well, not that I'm aware of) However, if it's an analogue phone on a TDM400 card (or equivalent, I guess) then it does work and the BLF LED on my Grandstream phone turns Red as soon as I take the analogue phone off-hook... So BLF is what you want, and optionally an analogue phone +TDM card for the boss... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My hints on extensions.conf: » [exten15] » include = numberplan-custom-1 » exten = 11,hint,SIP/12SIP/11 Where numberplan-custom-1 defines the general rules to dial any extension. And the secretary's phone (SIP/15) is subscribed to the context [exten15]. Now, I don't know if that's the right way of doing it (I'm a total n00b, you can tell, huh?). Oh, and I'm using Snoms. All Snoms. If there's not the possibility of knowing when a phone is lifted up, no problem... Thanks :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Dr. Michael J. Chudobiak wrote: I use Snoms. I know there's the feature. I just don't know how to use it, and there's so little documentation on the web.. Anyway, with see I meant that the secretary's phone would have one of the function keys on whenever the chef is on the phone (also when he picks it up, right before dialing). Until now I've only managed to make both phones blink on incoming calls. But that's not what I want and I could've done that with extension = 11,1,Dial(SIP/11SIP/12SIP/13...). This should be very easy. Use something like: exten = 602,1,Dial(SIP/boss_officeSIP/boss_home,20,trj) exten = 602,2,Voicemail([EMAIL PROTECTED]) exten = 602,102,Voicemail([EMAIL PROTECTED]) exten = 602,hint,SIP/boss_officeSIP/boss_home exten = 603,1,Dial(SIP/secretary,20,trj) exten = 603,2,Voicemail([EMAIL PROTECTED]) exten = 603,102,Voicemail([EMAIL PROTECTED]) exten = 603,hint,SIP/secretary and set snom function keys to extensions 602 and 603. (Some firmware versions say Destination instead of extension, I think.) - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks, I'll try that in a few hours... :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Grygoriy Dobrovolskyy wrote: ok, so do you configured you snom phone tu subscribe to monito extension in asterisk ? It is simple to verify: core show hints Snom's behave exactly like you want when blf enabled. 2008/6/25 Vazquez David [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Gordon Henderson wrote: On Tue, 24 Jun 2008, Vazquez David wrote: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. It's like BLF because that's exactly what it's for.. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Yes. Configure hints in asterisk and BLF in the secretarys phone. I don't understand how you get notifications one way, but not the other though (unless you're doing it by not using BLF) What phones have you got? What do your hints look like in the extensions.conf file? You can't get the status of lifting the handset on a SIP phone though (well, not that I'm aware of) However, if it's an analogue phone on a TDM400 card (or equivalent, I guess) then it does work and the BLF LED on my Grandstream phone turns Red as soon as I take the analogue phone off-hook... So BLF is what you want, and optionally an analogue phone +TDM card for the boss... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My hints on extensions.conf: » [exten15] » include = numberplan-custom-1 » exten = 11,hint,SIP/12SIP/11 Where numberplan-custom-1 defines the general rules to dial any extension. And the secretary's phone (SIP/15) is subscribed to the context [exten15]. Now, I don't know if that's the right way of doing it (I'm a total n00b, you can tell, huh?). Oh, and I'm using Snoms. All Snoms. If there's not the possibility of knowing when a phone is lifted up, no problem... Thanks :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us You mean like, from the web interface (Snom's) configure a function key to an extension? I guess so. I configured a function key to be mapped to an identity and then, instead of line, I told it to be an extension. Finally I wrote the extension I wanted to monitor... Anyways, in a couple of hours I'll try it again... Thanks for the help :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Rob Hillis wrote: Can I assume you're using Asterisk 1.4 and that you've configured your phones as peers? If this is the case, then you need to set limitonpeers to yes and call-limit to some value in sip.conf. Once this has been done, you should find that BLF behaves as you expect. Vazquez David wrote: Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4860fae540252026166755! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, I'm using 1.4. And I don't really use sip.conf, but have all my phones on users.conf. Should I put limitonpeers and call-limit on the general section of sip.conf? or on each entry in users.conf? Thanks :-D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Thankyou all! I think I've got it working :-D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
Though I wonder... The scenario is as follows: I have 4 phones with the following extensions: 11 (SIP/11) 12 (SIP/12) 13 (SIP/13) 15 (SIP/15) Whenever SIP/11 receives a call, it hints the other phones. Is it possible to pick up that call from one of them? The relevant part of my extensions.conf looks as this now: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten = s,hint,SIP/11 exten = 11,hint,SIP/12SIP/13SIP/15 exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain Thanks :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chef-secretary scenario
Hi all, I'm trying to implement such a scenario where the Chef picks up his phone and his secretary can see that he is busy. Something like blf, I guess. But so far I've only managed to notify the secretary that the chef is receiving a call. I want to do it the other way around though. I'd like for her to see in her phone, the light corresponding to the chef's extension light up whenever he uses the phone (also when he picks it up if that's possible). So she should always know when he's busy. Is there a way to do that? Thanks, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users