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<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It works, thanks a lot, I also change the character for
comments.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I am familiar with that page, I had been looking for the
information in that page also in google but noting.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks to all for your help on this, let me continue doing some
tests to complete the task to do.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Best regards<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>John<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=ES style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De:</span></b><span
lang=ES style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>En nombre de </b>Ioan
Indreias<br>
<b>Enviado el:</b> Miércoles, 16 de Septiembre de 2009 09:24 a.m.<br>
<b>Para:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Asunto:</b> Re: [asterisk-users] How to configure a coverage path for
anextension???<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi Juan,<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>1. Please use the semicolon (;) character to comment your
dialplan. Your choice (#) is intended for something else.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>2. Probably you have to add the "j" option of Dial
application (show application Dial), like:<o:p></o:p></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><span class=apple-style-span>exten =>
4000,1,Dial(SIP/4000,20,iKkTt<b>j</b>)</span><br>
<span class=apple-style-span>exten => 4000,102,Dial(SIP/4001,20,iKkTt<b>j</b>)</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal><span class=apple-style-span>3. For more hints you could
check <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial">voip-info</a> page.</span><o:p></o:p></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal><span class=apple-style-span>HTH</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Ioan Indreias<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><a href="http://www.modulo.ro">www.modulo.ro</a><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p>
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<p class=MsoNormal>On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza <<a
href="mailto:jcardoza@tpmex.com">jcardoza@tpmex.com</a>> wrote:<o:p></o:p></p>
<p class=MsoNormal>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<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal>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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<br>
</span></font>
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Teleperformance values:
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