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

[Ioan Indreias]

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,iKkTt*j*)
exten => 4000,102,Dial(SIP/4001,20,iKkTt*j*)

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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