Re: [asterisk-users] How to configure a coverage pathfor anextension???

2009-09-17 Thread Gordon Henderson
On Wed, 16 Sep 2009, Steve Edwards wrote:

 On Wed, 16 Sep 2009, Danny Nicholas wrote:

 I'd try this:
 - exten = 4000,1,Dial(SIP/4000,20,ikKtT)
 - exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
 - exten = s-NOANSWER,2,Voicemail(4000)
 - exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
 - exten = s-BUSY,2,Voicemail(4000)
 - exten = h,1,hangup

 Don't you need a goto(s-${DIALSTATUS},1) in there somewhere?

 BTW, everybody seems to do s-${DIALSTATUS}. Why not just
 ${DIALSTATUS}?

 s- doesn't seem to add any value to me.

I suspect everyone is copying one example that was presented some years 
ago ;-)

From the more than one way to skin a cat department, I do it this way:

exten =  s,n,GotoIf($[(${DIALSTATUS} = CHANUNAVAIL) | (${DIALSTATUS} = 
CONGESTION)]?:${DIALSTATUS})

...

exten =  s,n(BUSY),Noop(We got BUSY)
exten =  s,n,Busy()
exten =  s,n(NOANSWER),Noop(We got NOANSWER)
exten =  s,n,Congestion()

etc.

So Goto'ing a priority rather than an extension.

For the original poster, the simplest/crudest way is simply:

   exten = s,n,Dial(SIP/4000,,10)
   exten = s,n,Dial(SIP/4001,,10)
   exten = s,n,Dial(SIP/4002,,10)
   exten = s,n,Dial(SIP/4003,,10)
   exten = s,n,Voicemail...

knowing that if one of the Dial's succeedes then the rest of them will not 
action and if it fails to bridge for any reason execution will just carry 
on to the next step...

Gordon

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 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] How to configure a coverage pathfor anextension???

2009-09-16 Thread Danny Nicholas
I'd try this:
- exten = 4000,1,Dial(SIP/4000,20,ikKtT)
- exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten = s-NOANSWER,2,Voicemail(4000)
- exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten = s-BUSY,2,Voicemail(4000)
- exten = h,1,hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to configure a coverage pathfor
anextension???

I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:

Extensions.conf

[local-sip]

#exten = _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten = _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten = 164,1,Dial(Dahdi/1/${EXTEN})
#exten = 0550,1,Dial(Dahdi/1/${EXTEN})
#exten = _4XXX,3,Hangup()

[incoming]

exten = 4000,1,Dial(SIP/4000,20,iKkTt) - I test this line only and it
works
exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)  When I add this line the
call arrives to the 4000
#exten = _4xxx,1,Dial(SIP/${EXTEN},10,tTr)

I dont answer the call and the Asterisk server drop the call.

[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial(DAHDI/23-1, SIP/4000,20,iKkTt)
in new stack
-- Called 4000 
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial(DAHDI/22-1, SIP/4000,20,iKkTt)
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000 
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI

What could I need to fix this???
Thanks a lot for your help.
Jhon



-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: MiƩrcoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

In regular configuration (extensions.conf) this is one way to do it:
- exten = 4000,1,Dial(SIP/4000,20,iKkTt)
- exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use

Re: [asterisk-users] How to configure a coverage pathfor anextension???

2009-09-16 Thread Steve Edwards
On Wed, 16 Sep 2009, Danny Nicholas wrote:

 I'd try this:
 - exten = 4000,1,Dial(SIP/4000,20,ikKtT)
 - exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
 - exten = s-NOANSWER,2,Voicemail(4000)
 - exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
 - exten = s-BUSY,2,Voicemail(4000)
 - exten = h,1,hangup

Don't you need a goto(s-${DIALSTATUS},1) in there somewhere?

BTW, everybody seems to do s-${DIALSTATUS}. Why not just 
${DIALSTATUS}?

s- doesn't seem to add any value to me.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 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] How to configure a coverage pathfor anextension???

2009-09-16 Thread Alex Samad
On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
 On Wed, 16 Sep 2009, Danny Nicholas wrote:
 
  I'd try this:
  - exten = 4000,1,Dial(SIP/4000,20,ikKtT)
  - exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
  - exten = s-NOANSWER,2,Voicemail(4000)
  - exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
  - exten = s-BUSY,2,Voicemail(4000)
  - exten = h,1,hangup
 
 Don't you need a goto(s-${DIALSTATUS},1) in there somewhere?

I am curios as well, what tell it to do the  jump


-- 
If at first you do succeed, try to hide your astonishment.


signature.asc
Description: Digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 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