[asterisk-users] How to configure a coverage path for anextension???

Juan Cardoza jcardoza at tpmex.com
Wed Sep 16 09:51:16 CDT 2009


It works, thanks a lot, I also change the character for comments.

 

I am familiar with that page, I had been looking for the information in that
page also in google but noting.

 

Thanks to all for your help on this, let me continue doing some tests to
complete the task to do.

Best regards

John

 

De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Ioan Indreias
Enviado el: Miércoles, 16 de Septiembre de 2009 09:24 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

 

Hi Juan,

 

1. Please use the semicolon (;) character to comment your dialplan. Your
choice (#) is intended for something else.

 

2. Probably you have to add the "j" option of Dial application (show
application Dial), like:

 

exten => 4000,1,Dial(SIP/4000,20,iKkTtj)
exten => 4000,102,Dial(SIP/4001,20,iKkTtj)

 

3. For more hints you could check voip-info
<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial>  page.

 

HTH

Ioan Indreias

www.modulo.ro

 

On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza <jcardoza at tpmex.com> wrote:

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 [4000 at 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 [4000 at 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 20000 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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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


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