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
Re: [asterisk-users] How to configure a coverage path for 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: 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-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.
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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,iKkTt*j*)
exten = 4000,102,Dial(SIP/4001,20,iKkTt*j*)
3. For more hints you could check
voip-infohttp://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 jcard...@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 [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: 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-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
___
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Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Teleperformance values: Integrity - Respect - Professionalism - Innovation
- Commitment
The information contained in this communication is privileged and
confidential. The content is
Re: [asterisk-users] How to configure a coverage path for anextension???
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 jcard...@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 [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: 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-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:
Re: [asterisk-users] How to configure a coverage path for anextension???
it should look something like
exten = 4000,1,Dial(SIP/4000,30,t)
exten = 4000,2,Goto(4001,1)
exten = 4001,1,Dial(SIP/4001,30,t)
If 4000,1 is answered it will never reach 4000,2
if 4000 is busy or not available for another reason it wil goto 4001,1
hope this is useful
Erik de Wild
Tripple-o
Verstuurd vanaf mijn iPhone
Op 16 sep 2009 om 16:24 heeft Ioan Indreias indre...@gmail.com het
volgende geschreven:\
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 page.
HTH
Ioan Indreias
www.modulo.ro
On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza jcard...@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 [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: 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-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
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Register Now:
