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r_o_l_a_n_d at hotmail... Guest
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Posted: Sat Jun 14, 2008 4:56 am Post subject: [asterisk-users] adding funcionatlity to asterisk?! is it po |
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hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b directed to a pool of specific sip extensions (9 extensions) where asterisk tries one after the other till lhe finds one of them thats actually on.i want to add a step for asterisk to follow which is, when a sip extension doesn't answer or its offline, instead of immediately transferring to voice mail, i want it to dial that sip holder's number so it transfers the call to his cellphone for example. and if he didn't answer his cellphone its then that i want it to direct it to voice mail.i want to add another item to the operator menu, instead of just receiving the call and telling the caller to either dial extension or 100 for operator, i want asterisk to offer the caller an additional option like for example pressing 2, would direct you to a list of key personnels with their respective extensions.please find below my extensions.conf:
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times
exten => _1XX,2,VoiceMail(${EXTEN}@default)
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it will ring 3 times
exten => _2XX,2,VoiceMail(${EXTEN}@default)
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
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joakimsen at gmail.com Guest
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Posted: Tue Jun 17, 2008 6:34 pm Post subject: [asterisk-users] adding funcionatlity to asterisk?! is it po |
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Right now the issue I see is you are using overlapping extensions....
so maybe that's not working as expected?
you have in context sipura line exten 201, exten 201 included from
context spa and also exten 2xx included from context spa.
What you want to do with sending calls elsewhere if they are not
completed look at DIALSTATUS, e,g,:
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - SIP DEVICE
; ${ARG2} - ringing seconds
; ${ARG3} - vm-box-Nr.
;
exten => s,1,Macro(docid)
exten => s,2,Dial(SIP/${ARG1},${ARG2},r) ; Ring
the ${ARG1} interface, ${ARG3} seconds maximum
exten => s,3,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Playback(silence/1)
exten => s-NOANSWER,2,Voicemail(${ARG3},u) ; If unavailable, send
to voicemail w/ unavail announce
exten => s-NOANSWER,3,Goto(s-${VMSTATUS},1)
exten => s-USEREXIT,1,Playback(cancelled)
exten => s-USEREXIT,2,Playback(goodbye)
exten => s-USEREXIT,3,Hangup
exten => s-SUCCESS,1,Playback(goodbye)
exten => s-SUCCESS,2,Hangup
exten => s-FAILED,1,Playback(sorry-youre-having-problems)
exten => s-FAILED,2,Playback(please-try-again-later)
exten => s-FAILED,3,Playback(goodbye)
exten => s-FAILED,4,Hangup
exten => o-CHANUNAVAIL,1,Goto(o-BUSY,1)
exten => s-BUSY,1,Playback(silence/1)
exten => s-BUSY,2,Voicemail(${ARG3},b) ; If busy, send to
voicemail w/ busy announce
exten => s-BUSY,3,Playback(goodbye) ; If they press #,
return to start
exten => s-BUSY,4,Hangup
exten => o,1,Goto(o-${DIALSTATUS},1)
exten => _o-.,1,Goto(o-NOANSWER,1)
exten => o-BUSY,1,Goto(s,2)
exten => o-NOANSWER,1,Playback(please-try-again)
exten => o-NOANSWER,2,GoTo(s-NOANSWER,2)
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten => a,1,Playback(this-is-the-voice-mail-system)
exten => a,2,VoicemailMain(${ARG3}) ; If they press *,
send the user into VoicemailMain
For the directory, there's a directory application built into the
voicemail system. You might want to check that out, if it fits your
needs then it's probably the simplest solution.
On Sat, Jun 14, 2008 at 5:56 AM, RoLaNd RoLaNd <r_o_l_a_n_d at hotmail.com> wrote:
Quote: | hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.
i want to add a step for asterisk to follow which is, when a sip extension
doesn't answer or its offline, instead of immediately transferring to voice
mail, i want it to dial that sip holder's number so it transfers the call to
his cellphone for example. and if he didn't answer his cellphone its then
that i want it to direct it to voice mail.
i want to add another item to the operator menu, instead of just receiving
the call and telling the caller to either dial extension or 100 for
operator, i want asterisk to offer the caller an additional option like for
example pressing 2, would direct you to a list of key personnels with their
respective extensions.
please find below my extensions.conf:
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _1XX,2,VoiceMail(${EXTEN}@default)
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _2XX,2,VoiceMail(${EXTEN}@default)
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
________________________________
Invite your mail contacts to join your friends list with Windows Live
Spaces. It's easy! Try it!
_______________________________________________
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stotaro at totarotechn... Guest
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Posted: Tue Jun 17, 2008 6:56 pm Post subject: [asterisk-users] adding funcionatlity to asterisk?! is it po |
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Didn't you ask this exact same question in the last few days?
On Sat, Jun 14, 2008 at 5:56 AM, RoLaNd RoLaNd <r_o_l_a_n_d at hotmail.com> wrote:
Quote: | hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.
i want to add a step for asterisk to follow which is, when a sip extension
doesn't answer or its offline, instead of immediately transferring to voice
mail, i want it to dial that sip holder's number so it transfers the call to
his cellphone for example. and if he didn't answer his cellphone its then
that i want it to direct it to voice mail.
i want to add another item to the operator menu, instead of just receiving
the call and telling the caller to either dial extension or 100 for
operator, i want asterisk to offer the caller an additional option like for
example pressing 2, would direct you to a list of key personnels with their
respective extensions.
please find below my extensions.conf:
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _1XX,2,VoiceMail(${EXTEN}@default)
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _2XX,2,VoiceMail(${EXTEN}@default)
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
________________________________
Invite your mail contacts to join your friends list with Windows Live
Spaces. It's easy! Try it!
_______________________________________________
-- 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
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