SV: [Asterisk-Users] GotoIf problem

2005-03-12 Thread Thorben Jensen


 -Oprindelig meddelelse-
 Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] På vegne af kurt x
 Sendt: 9. marts 2005 20:57
 Til: Chris Wade
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: Re: [Asterisk-Users] GotoIf problem
 
 I,ve gotten the GotoIf statement working now.  I hard coded the value
 10 in place of the ${DIGITS} varible.  Worked like a charm.
 
 Thanks to everyone who helped.
 
 Kurt

Hi Kurt,

You are writing the ${DIGITS} variable wrong, you are missing a { 

eg.: you are writing $DIGITS} and it should be ${DIGITS}

Thorben


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SV: [Asterisk-Users] Snom phone hint exten question

2005-02-21 Thread Thorben Jensen
 
 I haven't used it in a while, but I had to put subscribecontext=sip
 for the phone's (in your case the snom) sip entry.
 
 This seems like it has been removed from the wiki.  Has it changed or
 is this incorrect?


Hi James,

I have just found out that all you need to do is make the hint in the
context where the phone registers. That means that all you need to do is put
'690,hint,SIP/bt-karen' in your [sip] context, nothing else and it should
work. Remember to take the power from the phone for a short while after you
have configured this, otherwise it won't work.

thorben

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SV: [Asterisk-Users] Snom phone hint exten question

2005-02-21 Thread Thorben Jensen
 
 Ok your example confused me a little.
 
 You put 690,hint,SIP/bt-karen
 
 From this section in my extensions from your example I should have
 
 exten = 690,hint,SIP/bt-karen
 
 exten = 691,hint,SIP/snom-james
 
 So set hint on the opposite extensions?
 
 [sip]
 
 exten = 690,hint,SIP/snom-james
 exten = 690,1,SetMusicOnHold(random)
 exten = 690,2,Dial(SIP/snom-james,30,Ttr)
 exten = 690,3,Voicemail2,u690
 exten = 690,103,Voicemail2,b690
 
 exten = 691,hint,SIP/bt-karen
 exten = 691,1,SetMusicOnHold(random)
 exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
 exten = 691,3,Voicemail,u691
 exten = 691,103,Voicemail,b691

Hi James 

I am sorry I made a typo. You need to set [sip] like this:

[sip]
exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen

exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690

exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691

That should work
thorben

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RE: [Asterisk-Users] Snom phone hint exten question

2005-02-21 Thread Thorben Jensen
 I am going to now sit in a corner and go quietly insane while playing
 the banyo with no strings.
 
 Still doesn't work, I dialed in an outside line and picked up the
 receive on extension 691, yet the light on the snom phone did not come
 on. I dialed out of extension 691 to an outside line, yet still the
 light did not come on.
 
 Snom190 has firmware 3.56m the button is set to Destination 691

Hi James,

I am using the latest CSV-HEAD of *, I do not think it works with * stable.

Thorben


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SV: [Asterisk-Users] Snom phone hint exten question

2005-02-19 Thread Thorben Jensen


 -Oprindelig meddelelse-
 Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] På vegne af James Bean
 Sendt: 19. februar 2005 08:14
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: [Asterisk-Users] Snom phone hint exten question
 
 
 Hi,
 
 I am sorry to be asking this but the wiki is down and has been for a
 couple of days and I need to get this working before Monday to get my
 live system setup.
 
 Trying to get the Snom 190's and soon to arrive 3com 3102's to use the
 function keys and for the life of me I can't work it out from the
 conversations on the archive what I am going exactly wrong here?
 
 The snom 190 with function keys is extension 690, the other extension
 (691) is just a BT102 so it doesn't have any function keys to program.
 
 When extension 691 is dialing out, or receives a call I want it to just
 tell the snom190 on ext 690 so the light shows up.
 
 (Soon as I got it going here I have a live system I will be setting it
 up on).
 
 Thank you to anyone in advance for the help.
 
 This is my extensions.conf
 
 ---
 
 [pstn]
 
 exten = s,1,SetMusicOnHold(random)
 exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
 exten = s,3,Hangup
 ;exten = s,10,VoiceMail(u100);Whatever box you want.
 
 [internal]
 
 exten = i,1,Playback(invalid)
 exten = i,2,Hangup
 exten = t,1,Hangup
 
 exten = 098,1,WaitMusicOnHold(5)
 exten = 099,1,Echo ;simple echo test when you dial 099 on your
 phone
 
 include = outgoing
 include = sip
 
 [outgoing]
 
 exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
 exten = _9X.,2,Congestion()
 exten = _9X.,3,Hangup
 
 [sip]
 
 exten = 690,2,SetMusicOnHold(random)
 exten = 690,3,Dial(SIP/snom-james,30,tr)
 exten = 690,4,voicemail2,u690
 exten = 690,102,voicemail2,b690
 
 exten = 691,hint,SIP/bt-karen
 exten = 691,1,Macro(stdexten,SIP/bt-karen)
 exten = 691,2,SetMusicOnHold(random)
 exten = 691,3,Dial(SIP/bt-karen,30,tr)
 exten = 691,4,voicemail2,u691
 exten = 691,102,voicemail,b691
 
 include = internal
 include = outgoing

Hi,

You need to 'hint' SIP/bt-karen in the pstn context:

[pstn]
exten = s,hint,SIP/bt-karen
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,3,Hangup

Thorben


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RE: [Asterisk-Users] Snom phone hint exten question

2005-02-19 Thread Thorben Jensen
 Unfortunately that did not work, I hard rebooted the snom phone, the bt102
 and the asterisk server, the light just stays off, and I tested the LED on
 the button as well just to make sure its working
 
 I also added a hint to the outgoing context so when they make an outgoing
 call, still no luck.
 
 My extensions.conf is now
 
 [pstn]
 
 exten = s,hint,SIP/bt-karen
 exten = s,1,SetMusicOnHold(random)
 exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
 exten = s,3,Hangup
 ;exten = s,5,VoiceMail(u100);Whatever box you want.
 
 [internal]
 
 exten = i,1,Playback(invalid)
 exten = i,2,Hangup
 exten = t,1,Hangup
 
 exten = 098,1,WaitMusicOnHold(5)
 exten = 099,1,Echo ;simple echo test when you dial 099 on your phone
 
 Include = outgoing
 include = sip
 
 [outgoing]
 
 exten = _9X.,hint,SIP/bt-karen
 exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
 exten = _9X.,2,Congestion()
 exten = _9X.,3,Hangup
 
 [sip]
 
 exten = 690,1,SetMusicOnHold(random)
 exten = 690,2,Dial(SIP/snom-james,30,tr)
 exten = 690,102,voicemail2,u690
 
 exten = 691,hint,SIP/bt-karen
 exten = 691,1,SetMusicOnHold(random)
 exten = 691,2,Dial(SIP/bt-karen,30,tr)
 exten = 691,102,voicemail,u691
 
 include = internal
 include = outgoing


Have you set the function key on the SNOM to 'Destination' and typed '691'
in the number?



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RE: [Asterisk-Users] Snom phone hint exten question

2005-02-19 Thread Thorben Jensen
 
 
 Have you set the function key on the SNOM to 'Destination' and typed '691'
 in the number?

I am sorry, I meant that you have to type 'bt-karen' in the number.

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SV: [Asterisk-Users] Snom phone hint exten question

2005-02-19 Thread Thorben Jensen
 Also instead of putting a whole bunch of hints in, how might I go about
 putting a cluster of SIP extensions in the hint off the PSTN situation?
 
 Could you also maybe throw me a couple of hints what the
 
 exten = 691,1,Macro(stdexten,SIP/bt-karen)
 
 Macro portion I have seen in some examples but I am not sure what it does.
 
 James

Hi James,

I think you have to put a whole bunch of hints in, as variables are not
supported with hint AFAIK.

Macro(stdexten,SIP/bt-karen) is the way to call a macro called 'stdexten' in
your dial plan.

Thorben

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RE: [Asterisk-Users] wiki down?

2005-02-19 Thread Thorben Jensen
Yes - no go from here

 -Oprindelig meddelelse-
 Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk
 Sendt: 19. februar 2005 19:14
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: [Asterisk-Users] wiki down?
 
 hi
 
 is the wiki down again?
 
 roy
 
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RE: [Asterisk-Users] Setting a Forward to an external number onyourphone

2005-02-18 Thread Thorben Jensen
[EMAIL PROTECTED] wrote:
 Hi!
 
 Maybe I have just been looking on the wrong pages but there is a
 question that is very important for me. I already studied some
 Demo-Dialplans and made some basic experiences with Asterisk.
 But what I
 need to find out is how I can handle this.
 
 I am leaving my office and I want to tell asterisk to forward
 calls now
 to my mobile phone by just hitting a key (on my IP-Phone) or
 by using a
 special key-sequence.
 
 How can this be handled because I need to change the dialplan
 based on
 some information coming from a device attached to a channel.
 When back in my office I hit the key again and the calls are
 now routed
 to my IP-Phone (or ISDN-Phone on zap-channel) again.
 
 With IP-Phones I can imagine just unregistering the phone and
 having a
 dialplan with a fallback-option or something like that. But what if I
 want to tell asterisk to forward calls from now on to a
 number I want to
 manually add just for today (hitting a key, entering the new target
 number and that's it). where can I find some information on
 how to make
 this feature available.

This is the way we do it, we have it call the local extension as well as a
mobile number, and then you choose where to pick up:

[applications]

exten = _*22*.,1,Answer()
exten = _*22*.,2,DBput(DUAL/${CALLERIDNUM}=${EXTEN:4})
exten = _*22*.,3,PlayBack(call-fwd-parallel)
exten = _*22*.,4,PlayBack(activated)
exten = _*22*.,5,Hangup

exten = **22,1,Answer()
exten = **22,2,DBdel(DUAL/${CALLERIDNUM})
exten = **22,3,PlayBack(call-fwd-parallel)
exten = **22,4,PlayBack(de-activated)
exten = **22,5,Hangup

[macro-dial]
exten = s,1,DBget(DUAL=DUAL/${ARG2})
exten = s,2,Dial(${ARG1}${OUTGOING}/${DUAL},20,wW)
exten = s,3, VoiceMail(su${ARG2})
exten = s,102,Dial(${ARG1},20,Ww)
exten = s,103,VoiceMail(su${ARG2})
exten = s,203,VoiceMail(sb${ARG2})

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[Asterisk-Users] TOUCH_MONITOR

2005-02-18 Thread Thorben Jensen








TOUCH_MONITOR is the variable to set if I need to
specify my own options for One Touch Record (filetype|filename|m).
I cannot get it to work.



Can you help?



Thanks in advance










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SV: [Asterisk-Users] Newbie question

2005-02-18 Thread Thorben Jensen
Hi Tim,

You could put he call back into the queue when the dial times out. Check for
the length of the CALLERID, if it's equal to the length of your internal
numbers then goto voicemail otherwise goto the queue.

thorben

-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Tim De Lange
Sendt: 18. februar 2005 14:25
Til: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] Newbie question

Hello!

When the oprator transfers calls to internal extensions to unavailable
or busy extensions, how can I prevent these calls from going to
voicemail, and route them back to the oprator?  But other calls, ie
internal between extensions, and calls coming in via DID should get
voicemail if extensions are busy / unavailable?

Any help be appreciated.

TIA!

Tim

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SV: [Asterisk-Users] Flash Pane - Monitor Parked Calls?

2005-02-12 Thread Thorben Jensen

Need help with how to configure for parked calls in the Flash Operator
Panel's op_buttons.cfg file ...

I've looked on the wiki, google and asternic's site and can't seem to find
how to setup op_buttons.cfg to monitor parked calls.

For example, if someone parks in 701, I'd like to see that represented on
the panel.

I've tried a number of things ... this is what I have now and it does not
work ...

   [701]
   Position=12
   Label=Park 701
   Extension=701
   Context=parkedcalls
   Icon=1

Any help would be great!

Thanks,
Bruce

[EMAIL PROTECTED]

Hi Bruce,

Try this; I took this from the sample configuration:

[PARK701]
Position=17
Icon=3
Extension=700
Label=Park 701

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