Hi Juan,<div><br></div><div>1. Please use the semicolon (;) character to comment your dialplan. Your choice (#) is intended for something else.</div><div><br></div><div>2. Probably you have to add the "j" option of Dial application (show application Dial), like:</div>
<div><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></div><div><span class="Apple-style-span" style="border-collapse: collapse; ">exten => 4000,1,Dial(SIP/4000,20,iKkTt<b>j</b>)<br>exten => 4000,102,Dial(SIP/4001,20,iKkTt<b>j</b>)</span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></div><div><span class="Apple-style-span" style="border-collapse: collapse;">3. For more hints you could check <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial">voip-info</a> page.</span></div>
<div><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></div><div><span class="Apple-style-span" style="border-collapse: collapse;">HTH</span></div><div>Ioan Indreias</div><div><a href="http://www.modulo.ro">www.modulo.ro</a></div>
<div><br><br><div class="gmail_quote">On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza <span dir="ltr"><<a href="mailto:jcardoza@tpmex.com">jcardoza@tpmex.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I comment all the lines in my extensions.conf file to work only with the<br>
lines you provide me Danny:<br>
<br>
Extensions.conf<br>
<br>
[local-sip]<br>
<br>
#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)<br>
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})<br>
#exten => 164,1,Dial(Dahdi/1/${EXTEN})<br>
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})<br>
#exten => _4XXX,3,Hangup()<br>
<br>
[incoming]<br>
<br>
exten => 4000,1,Dial(SIP/4000,20,iKkTt) <----- I test this line only and it<br>
works<br>
exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) <---- When I add this line the<br>
call arrives to the 4000<br>
#exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)<br>
<br>
I dont answer the call and the Asterisk server drop the call.<br>
<br>
[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable<br>
to enable echo cancellation on channel 23 (No such device)<br>
-- Executing [4000@incoming:1] Dial("DAHDI/23-1", "SIP/4000,20,iKkTt")<br>
in new stack<br>
-- Called 4000<br>
-- SIP/4000-08a41440 is ringing<br>
-- SIP/4000-08a41440 answered DAHDI/23-1<br>
-- Accepting call from '' to '4000' on channel 0/22, span 1<br>
-- Executing [4000@incoming:1] Dial("DAHDI/22-1", "SIP/4000,20,iKkTt")<br>
in new stack<br>
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable<br>
to enable echo cancellation on channel 22 (No such device)<br>
-- Called 4000<br>
-- SIP/4000-08a359c8 is ringing<br>
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:<br>
Received SIP subscribe for peer without mailbox: 4000<br>
-- Nobody picked up in 20000 ms<br>
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'<br>
-- Hungup 'DAHDI/22-1'<br>
tp2asterisk01*CLI><br>
<br>
What could I need to fix this???<br>
Thanks a lot for your help.<br>
Jhon<br>
<br>
<br>
<br>
-----Mensaje original-----<br>
De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] En nombre de Danny Nicholas<br>
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.<br>
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
Asunto: Re: [asterisk-users] How to configure a coverage path for<br>
<div><div></div><div class="h5">anextension???<br>
<br>
In regular configuration (extensions.conf) this is one way to do it:<br>
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)<br>
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Juan Cardoza<br>
Sent: Wednesday, September 16, 2009 8:04 AM<br>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
Subject: [asterisk-users] How to configure a coverage path for<br>
anextension???<br>
<br>
I have been checking but nothing that clear my idea...<br>
<br>
I have the extension 4000 and the idea is when this extension receive a call<br>
and the extension 4000 is busy, the call from PSTN could be send to a second<br>
extension, example: 4001, this need to happen only if the first extension is<br>
busy.<br>
<br>
If not, the call need to be take by the first station.<br>
Please any one how can help me on this???<br>
<br>
Best regards<br>
Jhon<br>
<br>
<br>
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</div></div></blockquote></div><br></div>