[Asterisk-Users] Dialplan Configuration with MYSQL
Hi, I specified within the extnum.conf package to read the configuration for the extensions.conf file from the database. When i start asterisk i see that it binds the extensions.conf file to the db!. Now i made an entry within the ast_config table of the database - when i make an reload - asterisk still loads the configuration from the extensions.conf file ! Do i have to delete the extensions.conf file - so that asterisk only reads the information from the database - or what else i'am missing ?. thanks johannes ___ 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] Dialplan transfer. (h323 transfer)
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible? Any help would be great! Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Sep 11, 2004, at 5:47 PM, Matt Hohman wrote: We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... I am sorry that's not a valid extension before i get a chance to enter anything. Any idea's? Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]___ 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] Dialplan problem - incoming calls get MOH, not ringing.
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to help. I'm sure I'm missing something spectacularly obvious, but
the wiki and googling the mailing list haven't shed any light. FWIW, my
ISDN-2e based incoming lines work just fine using similar dialplans.
Here is the excerpt from extensions.conf:
[my-sip-provider]
exten = 8441,1,answer
exten = 8441,2,Ringing
exten = 8441,3,SetCallerId( 30${CALLERIDNUM})
exten = 8441,4,SetCIDName(SIP 0${CALLERIDNUM})
exten = 8441,5,Dial(ZAP/2SIP/2010SIP/2009SIP/2011,120,trm)
exten = 8441,6,Voicemail,u1001
exten = 8441,106,Voicemail,b1001
Any ideas?
Thanks
Patrick
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Re: [Asterisk-Users] Dialplan problem - incoming calls get MOH, not ringing.
On Tue, 2004-08-17 at 15:44, Patrick Lidstone (Personal e-mail) wrote:
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to help. I'm sure I'm missing something spectacularly obvious, but
the wiki and googling the mailing list haven't shed any light. FWIW, my
ISDN-2e based incoming lines work just fine using similar dialplans.
Here is the excerpt from extensions.conf:
[my-sip-provider]
exten = 8441,1,answer
exten = 8441,2,Ringing
exten = 8441,3,SetCallerId( 30${CALLERIDNUM})
exten = 8441,4,SetCIDName(SIP 0${CALLERIDNUM})
exten = 8441,5,Dial(ZAP/2SIP/2010SIP/2009SIP/2011,120,trm)
exten = 8441,6,Voicemail,u1001
exten = 8441,106,Voicemail,b1001
You have the 'm' option enabled in your Dial() command. That will play
MOH to the calling party until the called channel answers.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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[Asterisk-Users] dialplan woes
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
main menu
Dial 1 for support
| Dial 2 for special
| Dial 3 sales
| Dial 5 For sales
|
|
|__submenu
Dial 1 for product a support
Dial 2 for pdoduct b support
Dial 3 for product c support
My problem is that if you choose option one in the second menu it loop
back to the first menu. I don't know how to handle this, and I'm sure it
can be done. here is the section of extensions.conf that deals with it
exten = ,1,Wait,2 ; Allow for PRI to grab info in facility
exten = ,2,SetCallerID(Toll Free No Cpub)
exten = ,3,BackGround(greeting)
exten = ,4,BackGround(mainmenu)
exten = ,5,Wait,5
exten = ,6,Queue(tech)
exten = 1,1,SetCallerID(Toll Free No Cpub)
exten = 1,2,AGI(openclose.agi)
exten = 1,3,GotoIf($[${STATUS} = closed]?6:4)
exten = 1,4,GotoIf($[${STATUS} = holiday]?8:10)
exten = 1,5,Goto(1,10)
exten = 1,6,BackGround(nighttime-greeting)
exten = 1,7,Goto(4,1)
exten = 1,8,BackGround(holiday-greeting)
exten = 1,9,Goto(4,1)
exten = 1,10,BackGround(tech-menu)
exten = 1,1,Playback(cpub-support)
exten = 1,2,Hangup
exten = 2,1,SetVar(QUEUE_PRIO=10)
exten = 2,2,Queue(tech)
exten = 3,1,SetVar(QUEUE_PRIO=5)
exten = 3,2,Queue(tech)
exten = 4,1,Hangup
exten = 2,1,Wait,1
exten = 2,2,SetCallerID(TimeIPS)
exten = 2,3,AGI(openclose.agi)
exten = 2,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 2,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 2,6,Goto(1,11)
exten = 2,7,BackGround(nighttime-greeting)
exten = 2,8,Goto(1,28)
exten = 2,9,BackGround(holiday-greeting)
exten = 2,10,Goto(1,28)
exten = 2,11,Dial(SIP/3085,15|m)
exten = 2,12,Dial(SIP/3082,15|m)
exten = 2,13,Dial(SIP/3006,15|m)
exten = 2,14,Dial(SIP/3007,15|m)
exten = 2,15,Background(sales-hold)
exten = 2,16,WaitMusicOnHold(60)
exten = 2,17,Dial(SIP/3085,15|m)
exten = 2,18,Dial(SIP/3082,15|m)
exten = 2,19,Dial(SIP/3006,15|m)
exten = 2,20,Dial(SIP/3007,15|m)
exten = 2,21,Background(sales-hold)
exten = 2,22,WaitMusicOnHold(60)
exten = 2,23,Dial(SIP/3085,15|m)
exten = 2,24,Dial(SIP/3082,15|m)
exten = 2,25,Dial(SIP/3006,15|m)
exten = 2,26,Dial(SIP/3007,15|m)
exten = 2,27,Voicemail(u3082)
exten = 2,28,Hangup
exten = 5,1,Wait,1
exten = 5,2,SetCallerID(Sales)
exten = 5,3,AGI(openclose.agi)
exten = 5,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 5,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 5,6,Goto(1,11)
exten = 5,7,BackGround(nighttime-greeting)
exten = 5,8,Goto(1,28)
exten = 5,9,BackGround(holiday-greeting)
exten = 5,10,Goto(1,28)
exten = 5,11,Dial(SIP/3085,15|m)
exten = 5,12,Dial(SIP/3082,15|m)
exten = 5,13,Dial(SIP/3006,15|m)
exten = 5,14,Dial(SIP/3007,15|m)
exten = 5,15,Background(sales-hold)
exten = 5,16,WaitMusicOnHold(60)
exten = 5,17,Dial(SIP/3085,15|m)
exten = 5,18,Dial(SIP/3082,15|m)
exten = 5,19,Dial(SIP/3006,15|m)
exten = 5,20,Dial(SIP/3007,15|m)
exten = 5,21,Background(sales-hold)
exten = 5,22,WaitMusicOnHold(60)
exten = 5,23,Dial(SIP/3085,15|m)
exten = 5,24,Dial(SIP/3082,15|m)
exten = 5,25,Dial(SIP/3006,15|m)
exten = 5,26,Dial(SIP/3007,15|m)
exten = 5,27,Voicemail(u3082)
exten = 5,28,Hangup
exten = 7,1,Wait,1
exten = 7,2,SetCallerID(TimeIPS)
exten = 7,3,AGI(openclose.agi)
exten = 7,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 7,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 7,6,Goto(1,11)
exten = 7,7,BackGround(nighttime-greeting)
exten = 7,8,Goto(1,28)
exten = 7,9,BackGround(holiday-greeting)
exten = 7,10,Goto(1,28)
exten = 7,11,Dial(SIP/3085,15|m)
exten = 7,12,Dial(SIP/3082,15|m)
exten = 7,13,Dial(SIP/3006,15|m)
exten = 7,14,Dial(SIP/3007,15|m)
exten = 7,15,Background(sales-hold)
exten = 7,16,WaitMusicOnHold(60)
exten = 7,17,Dial(SIP/3085,15|m)
exten = 7,18,Dial(SIP/3082,15|m)
exten = 7,19,Dial(SIP/3006,15|m)
exten = 7,20,Dial(SIP/3007,15|m)
exten = 7,21,Background(sales-hold)
exten = 7,22,WaitMusicOnHold(60)
exten = 7,23,Dial(SIP/3085,15|m)
exten = 7,24,Dial(SIP/3082,15|m)
exten = 7,25,Dial(SIP/3006,15|m)
exten = 7,26,Dial(SIP/3007,15|m)
exten = 7,27,Voicemail(u3082)
exten = 7,28,Hangup
Thanks,
Chris Locke
Systems Administrator
Stratitec INC
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Re: [Asterisk-Users] dialplan woes
I'm not sure if this is your issue or not, but it looks like ext= 1,
starts over at the bottom of the 1's. You have 1,1-10 and then 1,1 and
2 after it. I can see how asterisk might get confused if you sent your
call back to ext 1 at starting point 1 or 2.
On Tue, 2004-08-17 at 15:03, defiance wrote:
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
main menu
Dial 1 for support
| Dial 2 for special
| Dial 3 sales
| Dial 5 For sales
|
|
|__submenu
Dial 1 for product a support
Dial 2 for pdoduct b support
Dial 3 for product c support
My problem is that if you choose option one in the second menu it loop
back to the first menu. I don't know how to handle this, and I'm sure it
can be done. here is the section of extensions.conf that deals with it
exten = ,1,Wait,2 ; Allow for PRI to grab info in facility
exten = ,2,SetCallerID(Toll Free No Cpub)
exten = ,3,BackGround(greeting)
exten = ,4,BackGround(mainmenu)
exten = ,5,Wait,5
exten = ,6,Queue(tech)
exten = 1,1,SetCallerID(Toll Free No Cpub)
exten = 1,2,AGI(openclose.agi)
exten = 1,3,GotoIf($[${STATUS} = closed]?6:4)
exten = 1,4,GotoIf($[${STATUS} = holiday]?8:10)
exten = 1,5,Goto(1,10)
exten = 1,6,BackGround(nighttime-greeting)
exten = 1,7,Goto(4,1)
exten = 1,8,BackGround(holiday-greeting)
exten = 1,9,Goto(4,1)
exten = 1,10,BackGround(tech-menu)
exten = 1,1,Playback(cpub-support)
exten = 1,2,Hangup
exten = 2,1,SetVar(QUEUE_PRIO=10)
exten = 2,2,Queue(tech)
exten = 3,1,SetVar(QUEUE_PRIO=5)
exten = 3,2,Queue(tech)
exten = 4,1,Hangup
exten = 2,1,Wait,1
exten = 2,2,SetCallerID(TimeIPS)
exten = 2,3,AGI(openclose.agi)
exten = 2,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 2,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 2,6,Goto(1,11)
exten = 2,7,BackGround(nighttime-greeting)
exten = 2,8,Goto(1,28)
exten = 2,9,BackGround(holiday-greeting)
exten = 2,10,Goto(1,28)
exten = 2,11,Dial(SIP/3085,15|m)
exten = 2,12,Dial(SIP/3082,15|m)
exten = 2,13,Dial(SIP/3006,15|m)
exten = 2,14,Dial(SIP/3007,15|m)
exten = 2,15,Background(sales-hold)
exten = 2,16,WaitMusicOnHold(60)
exten = 2,17,Dial(SIP/3085,15|m)
exten = 2,18,Dial(SIP/3082,15|m)
exten = 2,19,Dial(SIP/3006,15|m)
exten = 2,20,Dial(SIP/3007,15|m)
exten = 2,21,Background(sales-hold)
exten = 2,22,WaitMusicOnHold(60)
exten = 2,23,Dial(SIP/3085,15|m)
exten = 2,24,Dial(SIP/3082,15|m)
exten = 2,25,Dial(SIP/3006,15|m)
exten = 2,26,Dial(SIP/3007,15|m)
exten = 2,27,Voicemail(u3082)
exten = 2,28,Hangup
exten = 5,1,Wait,1
exten = 5,2,SetCallerID(Sales)
exten = 5,3,AGI(openclose.agi)
exten = 5,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 5,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 5,6,Goto(1,11)
exten = 5,7,BackGround(nighttime-greeting)
exten = 5,8,Goto(1,28)
exten = 5,9,BackGround(holiday-greeting)
exten = 5,10,Goto(1,28)
exten = 5,11,Dial(SIP/3085,15|m)
exten = 5,12,Dial(SIP/3082,15|m)
exten = 5,13,Dial(SIP/3006,15|m)
exten = 5,14,Dial(SIP/3007,15|m)
exten = 5,15,Background(sales-hold)
exten = 5,16,WaitMusicOnHold(60)
exten = 5,17,Dial(SIP/3085,15|m)
exten = 5,18,Dial(SIP/3082,15|m)
exten = 5,19,Dial(SIP/3006,15|m)
exten = 5,20,Dial(SIP/3007,15|m)
exten = 5,21,Background(sales-hold)
exten = 5,22,WaitMusicOnHold(60)
exten = 5,23,Dial(SIP/3085,15|m)
exten = 5,24,Dial(SIP/3082,15|m)
exten = 5,25,Dial(SIP/3006,15|m)
exten = 5,26,Dial(SIP/3007,15|m)
exten = 5,27,Voicemail(u3082)
exten = 5,28,Hangup
exten = 7,1,Wait,1
exten = 7,2,SetCallerID(TimeIPS)
exten = 7,3,AGI(openclose.agi)
exten = 7,4,GotoIf($[${STATUS} = closed]?7:5)
exten = 7,5,GotoIf($[${STATUS} = holiday]?9:11)
exten = 7,6,Goto(1,11)
exten = 7,7,BackGround(nighttime-greeting)
exten = 7,8,Goto(1,28)
exten = 7,9,BackGround(holiday-greeting)
exten = 7,10,Goto(1,28)
exten = 7,11,Dial(SIP/3085,15|m)
exten = 7,12,Dial(SIP/3082,15|m)
exten = 7,13,Dial(SIP/3006,15|m)
exten = 7,14,Dial(SIP/3007,15|m)
exten = 7,15,Background(sales-hold)
exten = 7,16,WaitMusicOnHold(60)
exten = 7,17,Dial(SIP/3085,15|m)
exten = 7,18,Dial(SIP/3082,15|m)
exten = 7,19,Dial(SIP/3006,15|m)
exten = 7,20,Dial(SIP/3007,15|m)
exten = 7,21,Background(sales-hold)
exten = 7,22,WaitMusicOnHold(60)
exten = 7,23,Dial(SIP/3085,15|m)
exten = 7,24,Dial(SIP/3082,15|m)
exten = 7,25,Dial(SIP/3006,15|m)
exten = 7,26,Dial(SIP/3007,15|m)
exten = 7,27,Voicemail(u3082)
exten = 7,28,Hangup
Thanks,
Chris Locke
Systems Administrator
Stratitec INC
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Re: [Asterisk-Users] dialplan woes
On Tue, 2004-08-17 at 17:03, defiance wrote: I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales | | |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support My problem is that if you choose option one in the second menu it loop back to the first menu. I don't know how to handle this, and I'm sure it can be done. here is the section of extensions.conf that deals with it CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] -- Steven Critchfield [EMAIL PROTECTED] ___ 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] dialplan woes
On Tuesday 17 August 2004 18:03, defiance wrote: exten = 1,1,SetCallerID(Toll Free No Cpub) ... exten = 1,1,Playback(cpub-support) Do you see a problem? 'cos I sure do... You can use the numbers over again if you use Goto and jump to a different context. -A. ___ 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] dialplan woes
That makes sense, but how do I send it to each context? chris On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote: On Tue, 2004-08-17 at 17:03, defiance wrote: I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales | | |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support My problem is that if you choose option one in the second menu it loop back to the first menu. I don't know how to handle this, and I'm sure it can be done. here is the section of extensions.conf that deals with it CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] ___ 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] dialplan woes
CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] To elaborate, try something like this: [mainmenu] exten = s,1,Background(greeting) ; ... ; Do your closed/holiday logic here, whatever ; Dialing 1-3 takes you to different contexts which have their own logic exten = 1,Goto(support,s,1) exten = 2,Goto(special,s,1) exten = 3,Goto(sales,s,1) [support] exten = s,1,Background(product-support) ; Jump to [support-a] context which might give them a new menu exten = 1,Goto(support-a,s,1) ; Only one guy knows product B, so send to his phone exten = 2,Dial(SIP/1234) ... --Will Glynn Freedom Healthcare Group, Inc. ___ 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] dialplan woes
On Tue, 2004-08-17 at 17:33, defiance wrote: That makes sense, but how do I send it to each context? use goto(newcontext,s,1) On Tue, 2004-08-17 at 17:07, Steven Critchfield wrote: On Tue, 2004-08-17 at 17:03, defiance wrote: I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales | | |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support My problem is that if you choose option one in the second menu it loop back to the first menu. I don't know how to handle this, and I'm sure it can be done. here is the section of extensions.conf that deals with it CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] ___ 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] dialplan woes
Thanks a million man that works beautifully, and thanks for giving me an example, I am still pretty new at this so that helped alot. chris On Tue, 2004-08-17 at 17:31, William Glynn wrote: CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] To elaborate, try something like this: [mainmenu] exten = s,1,Background(greeting) ; ... ; Do your closed/holiday logic here, whatever ; Dialing 1-3 takes you to different contexts which have their own logic exten = 1,Goto(support,s,1) exten = 2,Goto(special,s,1) exten = 3,Goto(sales,s,1) [support] exten = s,1,Background(product-support) ; Jump to [support-a] context which might give them a new menu exten = 1,Goto(support-a,s,1) ; Only one guy knows product B, so send to his phone exten = 2,Dial(SIP/1234) ... --Will Glynn Freedom Healthcare Group, Inc. ___ 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] dialplan woes
* is so cool... Try doing this kind of thing with a regular PBX... Yes it will work but it'll take an hour of wading through menus and listening to the automated woman (or man if your PBX is male... lol...) list off the days of the week or your menu options... you can have a menu done in like 5 minutes... - Original Message - From: defiance [EMAIL PROTECTED] To: asterisk [EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 3:49 PM Subject: RE: [Asterisk-Users] dialplan woes Thanks a million man that works beautifully, and thanks for giving me an example, I am still pretty new at this so that helped alot. chris On Tue, 2004-08-17 at 17:31, William Glynn wrote: CONTEXTS [main context] Dial 1 for support | Dial 2 for special | Dial 3 sales | Dial 5 For sales [support context] ; don't include main context |__submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support [special context[ [sales context] [and so on.] To elaborate, try something like this: [mainmenu] exten = s,1,Background(greeting) ; ... ; Do your closed/holiday logic here, whatever ; Dialing 1-3 takes you to different contexts which have their own logic exten = 1,Goto(support,s,1) exten = 2,Goto(special,s,1) exten = 3,Goto(sales,s,1) [support] exten = s,1,Background(product-support) ; Jump to [support-a] context which might give them a new menu exten = 1,Goto(support-a,s,1) ; Only one guy knows product B, so send to his phone exten = 2,Dial(SIP/1234) ... --Will Glynn Freedom Healthcare Group, Inc. ___ 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 ___ 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] Dialplan question
Can anyone shed some light on this ??? Or is this not the right sort of question to ask? It simply doesn't work with the current software. You need to code this at the C source level. ___ 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] Dialplan question
Can anyone shed some light on this ??? Or is this not the right sort of question to ask? It simply doesn't work with the current software. You need to code this at the C source level. /me daydreams about being able to fork in the dialplan. Actually you sort of can - wonder what something like: Dial(Zap/1LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]) would do? Steve ___ 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] Dialplan question
Hi, -Original Message- /me daydreams about being able to fork in the dialplan. Actually you sort of can - wonder what something like: Dial(Zap/1LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]) would do? That works. I use it on a daily basis. Does not make your CDR any more readable though... Florian ___ 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] Dialplan question
Can anyone shed some light on this ??? Or is this not the right sort of question to ask? Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Wednesday, 4 August 2004 11:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dialplan question Does anyone know how to do the following: 1. Caller calls in 2. Asterisk answers. 3. Asterisk rings nominated extensions 4. Caller keys in certain digits while extensions are ringing 5. Caller is directed to another extension based on the digits keyed in I can achieve this if I have Asterisk play a background message after answering and before ringing the extensions (between steps 2 3). But I cannot get it to work if the extensions are rung straight away. Any help would be greatly appreciated. Simon Brown ___ 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
[Asterisk-Users] Dialplan question
Does anyone know how to do the following: 1. Caller calls in 2. Asterisk answers. 3. Asterisk rings nominated extensions 4. Caller keys in certain digits while extensions are ringing 5. Caller is directed to another extension based on the digits keyed in I can achieve this if I have Asterisk play a background message after answering and before ringing the extensions (between steps 2 3). But I cannot get it to work if the extensions are rung straight away. Any help would be greatly appreciated. Simon Brown ___ 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] dialplan help!-RESOLVED
All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: From: Ben Witso [EMAIL PROTECTED] Date: Mon Jun 21, 2004 7:28:42 PM US/Central To: Asterisk-Users [EMAIL PROTECTED] Subject: dialplan help! List-mates, I'm sorry to post such a simple question, but I can't get this to work and I'm at a loss as to why. I'm running the latest cvs head (just checked it out 1/2 hour ago). I want a call coming in on a PSTN to answer the phone, play some phrases (thank you for calling, if you know the extension you would like enter it now, otherwise please hold), then if they entered a 4 digit extension, dial it accordingly. If they don't enter anything ring a zap phone for several seconds then play a phrase and hang up. If they enter an invalid extension, play a phrase and go back to reprompt. When I call, if I do nothing it works as expected. But if I try to enter an extension, it either ignores the touch tones completely, or only sees the first digit (then acts accordingly with the invalid handling). I've moved the include around and that seems to be what makes it see a digit vs see nothing. When I call from an internal line (fromzap context) I'm able to dial the extensions just fine. Can anyone tell me why this isn't working? TIA, Ben extensions.conf: [frompstn] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,ResponseTimeout(2) exten = s,4,DigitTimeout(6) exten = s,5,Background(bgw-ThanksForCallingBgw) exten = s,6,Background(bgw-IfYouKnowExt) exten = s,7,Background(bgw-OtherwiseHold) include = internalextensions exten = t,1,Dial(ZAP/3,15) exten = t,2,Background(bgw-ThankYou) exten = t,3,Hangup exten = i,1,Background(bgw-SorryInvalidEntry) exten = i,2,Goto(s,5) exten = h,1,Hangup [fromzap] ignorepat = 9 include = internalextensions include = localcall exten = h,1,Hangup [internalextensions] exten = 4001,1,Dial(ZAP/1,30) ;exten = 4001,2,Voicemail(u4001) ;exten = 4001,102,Voicemail(b4001) exten = 4001,103,Hangup exten = 4002,1,Dial(ZAP/2,30) ;exten = 4002,2,Voicemail(u4002) ;exten = 4002,102,Voicemail(b4002) exten = 4002,103,Hangup exten = 4003,1,Dial(ZAP/3,30) ;exten = 4003,2,Voicemail(u4003) ;exten = 4003,102,Voicemail(b4003) exten = 4003,103,Hangup exten = 2050,1,SetLanguage(en) exten = 2050,2,Playback(demo-abouttotry) exten = 2050,3,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = 2050,4,Playback(demo-nogo) exten = 2050,5,Hangup PS- I have the voicemail lines commented out for extensions 4001-4003 because I haven't set them up yet. ___ 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] dialplan experts needed
Matthew, Dial works on a fall thru principle. Thus: exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) should suit your purpose (not taking into account vm), to add another exten just add it on the dial 'list': exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,Dial(SIP/3000,30) voicemail should be positioned at (exten + 101) for busy - I'd stick noop's in to allow the hangup before the next exten = 555,1,Dial(SIP/1000,30) exten = 555,2,NOOP exten = 555,3,Dial(SIP/2000,30) exten = 555,4,NOOP exten = 555,5,Dial(SIP/3000,30) exten = 555,6,NOOP exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,104,VoiceMail2(u3278) exten = 555,105,Hangup exten = 555,106,VoiceMail2(u3278) exten = 555,107,Hangup this wont allow the dial of 2000 or 3000 if 1000 is busy, it would go to vm. Ok, that's a bit of explaination, here's what you are prolly interested in exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,VoiceMail2(u3278) exten = 555,4,Hangup exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup And you probably want call waiting turned off... HTH Andy *** REPLY SEPARATOR *** On 07/06/2004 at 23:34 Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ 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
Fwd: Re: [Asterisk-Users] dialplan experts needed
Pah! my fingers are getting in the way today: exten = 555,1,Dial(SIP/1000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,203,VoiceMail2(u3278) exten = 555,204,Hangup Andy *** BEGIN FORWARDED MESSAGE *** On 07/06/2004 at 23:34 Andy Powell [EMAIL PROTECTED] wrote: From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Tue, 08 Jun 2004 14:54:33 +0200 Subject: Fwd: Re: [Asterisk-Users] dialplan experts needed Sorry misread your message, you want it to dial the next when it's BUSY... not if it's not answered.. Disregard my previous message and use... exten = 555,1,Dial(SIP/1000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup exten = 555,101,Dial(SIP/2000,30) exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,202,VoiceMail2(u3278) exten = 555,203,Hangup I've made the asumption (even if that is the 'mother of all F***ups') that if it's not answered it should just go to vm. Andy *** BEGIN FORWARDED MESSAGE *** On 07/06/2004 at 23:34 Andy Powell [EMAIL PROTECTED] wrote: From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Tue, 08 Jun 2004 14:47:32 +0200 Subject: Re: [Asterisk-Users] dialplan experts needed Matthew, Dial works on a fall thru principle. Thus: exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) should suit your purpose (not taking into account vm), to add another exten just add it on the dial 'list': exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,Dial(SIP/3000,30) voicemail should be positioned at (exten + 101) for busy - I'd stick noop's in to allow the hangup before the next exten = 555,1,Dial(SIP/1000,30) exten = 555,2,NOOP exten = 555,3,Dial(SIP/2000,30) exten = 555,4,NOOP exten = 555,5,Dial(SIP/3000,30) exten = 555,6,NOOP exten = 555,102,VoiceMail2(u3278) exten = 555,103,Hangup exten = 555,104,VoiceMail2(u3278) exten = 555,105,Hangup exten = 555,106,VoiceMail2(u3278) exten = 555,107,Hangup this wont allow the dial of 2000 or 3000 if 1000 is busy, it would go to vm. Ok, that's a bit of explaination, here's what you are prolly interested in exten = 555,1,Dial(SIP/1000,30) exten = 555,2,Dial(SIP/2000,30) exten = 555,3,VoiceMail2(u3278) exten = 555,4,Hangup exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup And you probably want call waiting turned off... HTH Andy *** REPLY SEPARATOR *** On 07/06/2004 at 23:34 Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ 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 *** END FORWARDED MESSAGE *** *** END FORWARDED MESSAGE *** ___ 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] dialplan experts needed
exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup ...should be exten = 555,1,Dial(SIP/1000,30) ; Unanswered = 2, Busy = 102 exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup exten = 555,102,Dial(SIP/2000,30) ; Unanswered = 103, Busy = 203 exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,203,Dial(SIP/3000,30) ; Unanswered = 204, Busy = 304 exten = 555,204,Voicemail2(u3278) exten = 555,205,Hangup exten = 555,304,VoiceMail2(u3278) exten = 555,305,Hangup You just have to remember that BUSY or not-registered/not-available will goto +101 in the priority of the extension. Unanswered will goto +1 in the priority of the extension. John ___ 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] dialplan experts needed
On 08/06/2004 at 11:15 John Fraizer wrote: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup ...should be That's why I follwed up with corrections Andy ___ 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] dialplan experts needed
I have the same situation - i.e. three different extensions scattered about. But I don't try them each individually. When a call comes in my asterisk attempts to ring up to four different devices at the same time. To do this using your dial plan is easy - i.e. exten = 555,1,Dial(SIP/1000SIP/2000SIP/3000,30) exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup All the phones will ring at the same time and the phone to pick up first wins. regards joe On Mon, 7 Jun 2004, Matthew Simpson wrote: In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ 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
[Asterisk-Users] dialplan experts needed
In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ 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] dialplan AGI DTMF
Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete nonsense? -- Best regards Vlad ___ 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] dialplan AGI DTMF
On Thu, 27 May 2004, Vladyslav wrote: Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete nonsense? I've been mentally scheming about a way to do this is a generalised way - but right now during conversation the only DTMF that may be detected is a * for disconnect and # to initiate a transfer. Even these are only handled if the right dial options are used. To get what you want you will need to change the source code - in res/res_parking.c, function ast_bridge_call. Steve ___ 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] Dialplan that changes tith time of day
I need to set up a dialplan that is time sensitive. That means it will change after a specific time of day. How can that be done ? Examples ? Rgds EEA _ Last ned MSN Messenger gratis http://www.msn.no/computing/messenger - Den raskeste veien mellom deg og dine venner ___ 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] Dialplan that changes tith time of day
I need to set up a dialplan that is time sensitive. That means it will change after a specific time of day. How can that be done ? This is *right* out of the handbook. Please, please *please* do some basic research before asking questions to the list... I mean come on now. Regards, Andrew ___ 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] Dialplan that changes tith time of day
On 2004 Mar 09, at 12:36, M H wrote: I need to set up a dialplan that is time sensitive. That means it will change after a specific time of day. How can that be done ? CLI show application GotoIfTime -Tilghman ___ 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] Dialplan for voice menu and two extensions
Hi, I need to set up a dialplan for a voice menu and two extensions. One extension is to a GSM phone and one to a VOIP phone e.g. a Grandstream phone that is on the local network here. Most important is the GSM phone, the VOIP phone can come later. This system is to be used for a very small business setup. We are now in the initial stages using the zaphfc driver on a BRI PCI board with the cologne chip set that is working OK with Junghanns driver (very good and kind support from Junghanns!). When customers are calling and we are out of the office they get this voice menu : Thank you for calling xyz incorporated blablabla. Then straight into the voice menu: Press 1) Leave a message on the voice mail (This should only be active between 0800 and 1600 local time) Press 2) Autometically record the callers ID or let the caller dial his number on the DTMF keypad so we can call him up after we are back in the office. (What application can do that and how do we record the numbers in a way that is easily obtainable when we get back in to the office? Maybee even a web interface is available ? ) (This should only be active between 0800 and 1600 local time) Press 3) Transfer him to the GSM cellphone on the other channel of the BRI interface. There should be some kind of security so that if the * dials out to the GSM phone thru the BRI interface, a timeout period should run and after that the line is hung up to avoid very high bills if something goes wrong. Do anyone on the list have a similar setup or a basis dialplan that we can start with to achieve this ? When I get it to work, I will be happy of course to share that dialplan back. Rgds EEA _ MSN Messenger http://www.msn.no/computing/messenger Den raskeste veien mellom deg og dine venner ___ 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] Dialplan Sanity Check please
Hi guys, I've posted my dialplan design in flowchart format and would welcome your comments for improvements. I'd like to know my logic looks ok before I go much farther or start rewriting my extensions.conf. http://www.triadinternetsystems.com/asterisk/ For the Standard Extension macro, I was trying to do the following: 1. execute telezapper if user has turned it on 2. execute Privacy Manager if user has turned it on 3. check for Do Not Disturd 4. check for Call Forward Unconditional 5. ring some phones 6. and either a. let them talk b. go to busy voicemail c. call forward on busy d. go to unavailable voicemail e. call forward on unavailable Thanks --Lance ___ 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] Dialplan question
Title: Message
Fredrik,
Your
dialplan looks correct, however you disallow 112, the emergency
number!
Does
it fail for local or interlocal calls?
I
use:
[dutchdial];;emergency (112) and other 11x
numbers;exten = _1XX,1,Dial(${ISDN}:${EXTEN})exten =
_1XX,2,Congestionexten = _01XX,1,Dial(${ISDN}:${EXTEN:1})exten =
_01XX,2,Congestion;;0900 0800 numbers;exten =
_00[89]00.,1,SetCIDNum(0206408219)exten =
_00[89]00.,2,Dial(${ISDN}:${EXTEN:1})exten =
_00[89]00.,3,Congestion;;International;exten =
_000.,1,Dial(${ISDN}:${EXTEN:1})exten =
_000.,2,Congestion;;Local (7 digits, add area code);exten =
_0XXX,1,SetCIDNum(0206408219)exten =
_0XXX,2,Dial(${ISDN}:020${EXTEN:1})exten =
_0XXX,3,Congestion;;Interlocal, 10 digits;exten =
_0XX,1,SetCIDNum(0206408219)exten =
_0XX,2,Dial(${ISDN}:${EXTEN:1})exten =
_0XX,3,Congestion
And it
works fine
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of fredrik
chabotSent: zaterdag 6 september 2003 18:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Dialplan
questionHi,Dialplan QuestionI'm in
holland and I have:[naarbuiten]ignorepat = 0;
interlocaalexten =
_00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion; locaalexten =
_0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,CongestionAnd sometimes I can get
out, most of the time however I get a busy signal halfway throu the
number.It works more often if I change Early Dial:
No Yes (use "Yes" only if proxy supports 484
response)to No. In the Budgetone 100
phone.regards,fredrik chabot---*CLI
show dialplan [ Context 'default' created by 'pbx_config' ]
Include =
'demo'
[pbx_config][ Context 'demo' created by 'pbx_config' ] '#'
= 1.
Playback(demo-thanks)
[pbx_config]
2.
Hangup()
[pbx_config] '100'
= 1.
Dial(SIP/100)
[pbx_config] '101'
= 1.
Dial(SIP/101)
[pbx_config] '190'
= 1.
Dial(Modem/g1:006400)
[pbx_config] '8500'
= 1.
VoicemailMain()
[pbx_config]
2.
Goto(s|6)
[pbx_config] 'i'
= 1.
Playback(invalid)
[pbx_config] 's'
= 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
BackGround(demo-congrats)
[pbx_config]
6.
BackGround(demo-instruct)
[pbx_config] 't'
= 1.
Goto(#|1)
[pbx_config] Include
=
'naarbuiten'
[pbx_config][ Context 'naarbuiten' created by 'pbx_config' ]
'_00[1-9]' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] '_0[1-9]XX' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] Ignore pattern =
'0'
[pbx_config][ Context 'vanbuiten' created by 'pbx_config' ]
's' = 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
Playback(tt-weasels)
[pbx_config]
6.
Dial(SIP/100|4)
[pbx_config]
7.
Dial(SIP/100SIP/101|10)
[pbx_config]
8. Dial(SIP/100SIP/101Modem/g1:0064000)
[pbx_config]___
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[Asterisk-Users] Dialplan question
Hi,
Dialplan Question
I'm in holland and I have:
[naarbuiten]
ignorepat = 0
; interlocaal
exten = _00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion
; locaal
exten = _0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,Congestion
And sometimes I can get out, most of the time however I get a busy
signal halfway throu the number.
It works more often if I change
Early Dial: No
Yes (use
"Yes" only if proxy supports 484 response)
to No. In the Budgetone 100 phone.
regards,
fredrik chabot
---
*CLI show dialplan
[ Context 'default' created by 'pbx_config' ]
Include = 'demo'
[pbx_config]
[ Context 'demo' created by 'pbx_config' ]
'#' = 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
'100' = 1. Dial(SIP/100)
[pbx_config]
'101' = 1. Dial(SIP/101)
[pbx_config]
'190' = 1. Dial(Modem/g1:006400)
[pbx_config]
'8500' = 1. VoicemailMain()
[pbx_config]
2. Goto(s|6)
[pbx_config]
'i' = 1. Playback(invalid)
[pbx_config]
's' = 1. Wait(1)
[pbx_config]
2. Answer()
[pbx_config]
3. DigitTimeout(5)
[pbx_config]
4. ResponseTimeout(10)
[pbx_config]
5. BackGround(demo-congrats)
[pbx_config]
6. BackGround(demo-instruct)
[pbx_config]
't' = 1. Goto(#|1)
[pbx_config]
Include = 'naarbuiten'
[pbx_config]
[ Context 'naarbuiten' created by 'pbx_config' ]
'_00[1-9]' = 1.
Dial(Modem/g1:${EXTEN}) [pbx_config]
2. Congestion()
[pbx_config]
'_0[1-9]XX' = 1. Dial(Modem/g1:${EXTEN})
[pbx_config]
2. Congestion()
[pbx_config]
Ignore pattern = '0'
[pbx_config]
[ Context 'vanbuiten' created by 'pbx_config' ]
's' = 1. Wait(1)
[pbx_config]
2. Answer()
[pbx_config]
3. DigitTimeout(5)
[pbx_config]
4. ResponseTimeout(10)
[pbx_config]
5. Playback(tt-weasels)
[pbx_config]
6. Dial(SIP/100|4)
[pbx_config]
7. Dial(SIP/100SIP/101|10)
[pbx_config]
8.
Dial(SIP/100SIP/101Modem/g1:0064000) [pbx_config]
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