Re: [asterisk-users] Exit Dial Application
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
This will execute the macro, then dial the number. You will have to take
the hangups out of callback.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fürstaller
Sent: Tuesday, April 14, 2009 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup
if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted
correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the called
party.
The optional URL will be sent to the called party if the channel supports it.
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place an
automatic
callback. If B is back and places or takes a call, the automatic callback to A
should be
started.
I've found a possibility to do this via answering the call before the dial. But
... that's
not an ideal solution. I would prefer not to answer the call in the dialplan.
Does the
option 'd' implies an answered channel? Or is this a Bug?
Chris...
This will execute the macro, then dial the number. You will have to take
the hangups out of callback.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fürstaller
Sent: Tuesday, April 14, 2009 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup
if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted
correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the called
party.
The optional URL will be sent to the called party if the channel supports
it.
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place an
automatic
callback. If B is back and places or takes a call, the automatic callback to
A should be
started.
I've found a possibility to do this via answering the call before the dial.
But ... that's
not an ideal solution. I would prefer not to answer the call in the dialplan.
Does the
option 'd' implies an answered channel? Or is this a Bug?
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
exten = 5,1,goto(callback,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins
Sent: Wednesday, April 15, 2009 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the
called party.
The optional URL will be sent to the called party if the channel supports
it.
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place
an automatic
callback. If B is back and places or takes a call, the automatic callback
to A should be
started.
I've found a possibility to do this via answering the call before the
dial. But ... that's
not an ideal solution. I would prefer not to answer the call in the
dialplan. Does the
option 'd' implies an answered channel? Or is this a Bug?
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Thanks for your replay. Jep, that would be a possibility. But then the user has
to wait until my dialtime is over. If he/she is that inpatient, then with my
solution he/she can end the dialing whenever needed. But, I'll try your
successtion, looks
interesting.
chris...
Danny Nicholas schrieb:
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
exten = 5,1,goto(callback,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins
Sent: Wednesday, April 15, 2009 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the
called party.
The optional URL will be sent to the called party if the channel supports
it.
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place
an automatic
callback. If B is back and places or takes a call, the automatic callback
to A should be
started.
I've found a possibility to do this via answering the call before the
dial. But ... that's
not an ideal solution. I would prefer not to answer the call in the
dialplan. Does the
option 'd' implies an answered channel? Or is this a Bug?
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Atis, Atis Lezdins schrieb: I think the limitation could be by analogous Zap phones, as they probably don't support sending DTMF on unanswered channel. You could try it opposite way - Dial from SIP phone to Zap. Noop, it's not a Zap problem. I tried it with two SIP phones, same behavior. Bit odd : / Regards, Atis chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: c.fuerstal...@commpany.at -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAknmDNAACgkQR0exH8dhr/ZhmQCfQ4RaMsglGxx23McMbBiflsA9 y0IAoK+EBojiyPF1qj1hhITM8vzBPVmH =LkKe -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
If you set your ARG2 to a value like 6, the phone would only ring twice before noanswer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph Fürstaller Sent: Wednesday, April 15, 2009 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Exit Dial Application -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Danny, Thanks for your replay. Jep, that would be a possibility. But then the user has to wait until my dialtime is over. If he/she is that inpatient, then with my solution he/she can end the dialing whenever needed. But, I'll try your successtion, looks interesting. chris... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 09:24:54AM -0500, Danny Nicholas wrote:
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
Anything after a '.' in a pattern match is practically ignored.
Also note that X only matches digits.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Exit Dial Application
Mea culpa. Just being a bit lazy. In real use, the _X.-noanswer would be
s-NOANSWER (at least that's how it works in MY Dialplan).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, April 15, 2009 4:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 09:24:54AM -0500, Danny Nicholas wrote:
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
Anything after a '.' in a pattern match is practically ignored.
Also note that X only matches digits.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Exit Dial Application
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial, but
I wanna do it during the dial, when user A is waiting for user B, answering the
phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
CLI core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
Regards,
Atis
On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Chris...
Atis Lezdins schrieb:
CLI core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting
for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT
variable,
if it exists.
Regards,
Atis
On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
fuch_li...@kurtkrenn.com wrote:
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup
if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Oh, sorry, missed that part :)
Try enabling full log in logger.conf, set verbosity to 3 and debug
to 1, and see what goes in it.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
No problem : ) I tried it again, here is the log output:
-- Executing [...@from-pbx:1] Set(Zap/31-1, EXITCONTEXT=callback) in
new stack
-- Executing [...@from-pbx:2] Dial(Zap/31-1, SIP/236||d) in new stack
-- Called 236
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
Nothing happens. I adopted my [callback] context:
[callback]
exten = 1,1,Verbose(hello)
exten = s,1,Verbose(s)
exten = i,1,Verbose(i)
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
But nothing happens, if I dial 1, 5, or everything else. I have no clue what's
wrong here.
chris...
Atis Lezdins schrieb:
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Oh, sorry, missed that part :)
Try enabling full log in logger.conf, set verbosity to 3 and debug
to 1, and see what goes in it.
Regards,
Atis
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 9:14 PM, Christoph Fürstaller fuch_li...@kurtkrenn.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Atis, No problem : ) I tried it again, here is the log output: -- Executing [...@from-pbx:1] Set(Zap/31-1, EXITCONTEXT=callback) in new stack -- Executing [...@from-pbx:2] Dial(Zap/31-1, SIP/236||d) in new stack -- Called 236 -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing That's CLI interface output, log should have timestamps and much more detail in it. Check /var/log/asterisk/full (assuming default install location). You'll need to enable full line in logger.conf, restart Asterisk and issue core set verbose 3 and core set debug 1 in CLI. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Atis Lezdins schrieb: That's CLI interface output, log should have timestamps and much more detail in it. Check /var/log/asterisk/full (assuming default install location). You'll need to enable full line in logger.conf, restart Asterisk and issue core set verbose 3 and core set debug 1 in CLI. Thanks for the hint. I've looked aht the full log. I've attached a snipplet from the file. But I can't see anythin which can help me. Very interesting, but not helpful for me : / Is it possible to deactivate the 'd' option? Or what else could cause my problem? Regards, Atis thanks for your help, chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: c.fuerstal...@commpany.at -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAknk7gMACgkQR0exH8dhr/Y+1QCfTM8FvjA/9Zim7m9QbdjTYbQc QGQAnR92l1smtrs8Ao8f0vlaEdHiQv3R =KE+7 -END PGP SIGNATURE- [Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing [s-d...@macro-dialone:11] Set(Zap/31-1, EXITCONTEXT=callback) in new stack [Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set [Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Set' [Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing [s-d...@macro-dialone:12] Set(Zap/31-1, orig_exten=236) in new stack [Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set [Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Dial' [Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing [s-d...@macro-dialone:13] Dial(Zap/31-1, SIP/236|30|dtT) in new stack [Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Setting NAT on RTP to On [Apr 14 22:49:25] DEBUG[7867] acl.c: # Testing 10.10.5.1 with 10.10.0.0 [Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing anything [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_DEPTH. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable orig_exten. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable EXITCONTEXT. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable calls. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable peer. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG2. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable DIALNUM. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFNA. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFBS. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFIM. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable COUNT. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG1. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_PRIORITY. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_CONTEXT. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_EXTEN. [Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable start. [Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable intern. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CALLEDTON. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ANI2. [Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable TRANSFERCAPABILITY. [Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Outgoing Call for 236 [Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Called 236 [Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel SIP/236-08219bb0 to read format slin [Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to write format slin [Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to read format g729 [Apr 14 22:49:25] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request 102: Found [Apr 14 22:49:25] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing [Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing anything [Apr 14 22:49:25] DEBUG[7867] chan_zap.c: Requested indication 3 on channel Zap/31-1 [Apr 14 22:49:26] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request 102: Found [Apr 14 22:49:26] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing [Apr 14 22:49:26] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing anything [Apr 14 22:49:27] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request 102: Found [Apr 14 22:49:27] VERBOSE[7867] logger.c: --
Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 11:11 PM, Christoph Fürstaller fuch_li...@kurtkrenn.com wrote: Thanks for the hint. I've looked aht the full log. I've attached a snipplet from the file. But I can't see anythin which can help me. Very interesting, but not helpful for me : / Is it possible to deactivate the 'd' option? Or what else could cause my problem? Ok, at first glance the app_macro looks suspicious, can You try calling dial without Macro? If unsuccessful, You could enable debug level 2, it will tell way much more of everything, including DTMF events etc. Btw, does DTMF work at all for this Zap/ line? You could verify that by using Read before Dial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Atis Lezdins schrieb: Ok, at first glance the app_macro looks suspicious, can You try calling dial without Macro? Tried it without macro - same behavior. If unsuccessful, You could enable debug level 2, it will tell way much more of everything, including DTMF events etc. Btw, does DTMF work at all for this Zap/ line? You could verify that by using Read before Dial. Called Read, entered numbers, echoed them correctly. Then I tried something ... different. I Answered the call before calling the macro. And voila it's working. Do I have to answer the channel before Dial option 'd' is working? It's a bit odd, cause the dial duration starts counting and I hear a 'beep'. That's not ideal : / I've attached a full.log. Regards, Atis chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: c.fuerstal...@commpany.at -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAknlBuMACgkQR0exH8dhr/YSYwCeOcCfSlsnQIRff3L/F5wUvHh+ wCIAnRMC+YR7n7ZGmAvPKYbwZ7V/vc0O =7cnt -END PGP SIGNATURE- [Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Answer' [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing [s-d...@macro-dialone:10] Answer(Zap/31-1, ) in new stack [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing [s-d...@macro-dialone:11] Set(Zap/31-1, _EXITCONTEXT=callback) in new stack [Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set [Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Set' [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing [s-d...@macro-dialone:12] Set(Zap/31-1, orig_exten=236) in new stack [Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set [Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Dial' [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing [s-d...@macro-dialone:13] Dial(Zap/31-1, SIP/236|30|dtT) in new stack [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Setting NAT on RTP to On [Apr 15 00:02:34] DEBUG[8816] acl.c: # Testing 10.10.5.1 with 10.10.0.0 [Apr 15 00:02:34] DEBUG[8816] rtp.c: Channel 'Zap/31-1' has no RTP, not doing anything [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_DEPTH. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable orig_exten. [Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable EXITCONTEXT. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable calls. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable peer. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG2. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable DIALNUM. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFNA. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFBS. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFIM. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable COUNT. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG1. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_PRIORITY. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_CONTEXT. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_EXTEN. [Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable start. [Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable intern. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CALLEDTON. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ANI2. [Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable TRANSFERCAPABILITY. [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Outgoing Call for 236 [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Call to peer '236' is 1 out of 10 [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our capability: 0x10e (gsm|ulaw|alaw|g729) Video flag: False [Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Called 236 [Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel SIP/236-081df8b0 to read format slin [Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to write format slin [Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to read format g729 [Apr 15 00:02:34] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '45c32fd1024523451dba563865cd0...@xxx.at' Request 102: Found [Apr 15 00:02:34] VERBOSE[8816] logger.c: -- SIP/236-081df8b0 is ringing [Apr 15 00:02:34] DEBUG[8816] rtp.c: Channel
Re: [asterisk-users] exit ChanSpy with DTMF
On 9/11/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Part of a supervisor menu I'm writing requires that I allow the supervisor to choose to ChanSpy a channel from the main menu then return back to the menu (dialplan) to choose other options when she's done. Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? In 1.4, they are stuck. -trunk has an option to allow them to escape out to a context using a DTMF digit; check the changelog in SVN for details. I'm not sure how portable it might be back to 1.4/1.2 if you want to attempt that. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit Voicemail
Voicemail in itself does not hangup, * will bring you back to the DP (to exten a). So if a user exits VM (I think they can exit by pressing # after recording) then you can drop them in a context that does what you want, you can do the same at exten a. On 12/8/05, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to have control go back to the dialplan after a call gets to voicemail? I'm looking to implement findme and campon, but I want the options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement fineme in the dial plan. The same with campon, if you got a busy voicemail you could key in *2 (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit Voicemail
I am having the same issue. There was a patch put in that is supposed to rewite a blank context to default, but it looks like in the process this patch has killed the realtime variable passed to the query. -Jon C F wrote: Voicemail in itself does not hangup, * will bring you back to the DP (to exten a). So if a user exits VM (I think they can exit by pressing # after recording) then you can drop them in a context that does what you want, you can do the same at exten a. On 12/8/05, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to have control go back to the dialplan after a call gets to voicemail? I'm looking to implement findme and campon, but I want the options to be "hidden", so if someone calling got a voicemail they could key in "*1" (or whatever) and it would go back to the dialplan so I can implement fineme in the dial plan. The same with campon, if you got a busy voicemail you could key in "*2" (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit Voicemail
Hitting 0 when asterisk announces that the call has gone to voicemail immediately goes to the 'o' reserved extension: http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMail On voicemail main/comedian mail (reading your own voicemail box), press 3 for advanced options, then press 4 to place an outgoing call. On 12/8/05, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to have control go back to the dialplan after a call gets tovoicemail? I'm looking to implement findme and campon, but I wantthe options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement finemein the dial plan. The same with campon, if you got a busy voicemail you could key in *2 (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit Voicemail to VoicemailMain?
Michael Welter wrote: I would like to call my own DID number from outside, get into voicemail, and then push '#' to exit into VoicemailMain. Is there a way to do this? This works for me, dial '89' while listening to the greeting and it hops over to voicemailmain. This would be in your incoming call context. I've not tried a exten = #,... I suppose it could work too. exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Background(/var/spool/asterisk/voicemail/default/21/unavail) exten = s,4,Voicemail2(s21) exten = s,5,Congestion exten = s,6,Hangup exten = 89,1,Answer exten = 89,2,VoiceMailMain2(s21) exten = 89,3,Wait(5) exten = 89,4,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
On Thu, Feb 26, 2004 at 11:11:05PM -0500, Alex Volkov wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Then, per the PDF manual, I did asterisk -r to connect to the server and get a console. The manual says I can type quit to disconnect from the console, leaving Asterisk running in the background. But, when I do so, I get this message: The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX. so you cannot bail out of CLI with exit -- you are in console mode. Instead, start it without -c so it respawns another service process and exits to shell, after that you can run asterisk -r and bail out with exit all you please ;-). That's basically what I did. I started Asterisk with asterisk and then ran asterisk -r to get a console. When I type exit, I get the same message as I indented above. I type help at the command line (CLI), but didn't see anything in there (except quit and exit) that would seem to be a way to get out of the CLI prompt and back to a standard command line. -gk -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
On Thu, Feb 26, 2004 at 11:01:40PM -0500, Chris Clifton wrote: Greg, There may very well be another way to detach from the console, but I start asterisk on tty5 or tty6, and leave it running there. (redhat gives you 6 console tty's by default, use [alt] + [f1,f2,f3,etc.] to switch) You can ssh into your box and do a 'asterisk -r' to connect to the console, which is nice for remote troubleshooting, etc. To exit this, simply type 'quit'. I suppose I could do something like this. I supposed I could just close the terminal window. I run Asterisk on a headless server, and ssh into it via X on my desktop (aterm terminal window). After the ssh connection is established, I can check up on Asterisk. I did this yesterday by typing asterisk -r since asterisk was already running in the background. I got a console and a CLI prompt. I diddle and did what I needed to do at the moment. And then thought, gee... I'd like to close the term window out. So, in my Linux logic, I figured it would be as simple as getting out of the Asterisk console, back to a command line, exiting superuser, exiting my ssh session and exiting my aterm windown in X here on my desktop. Typing quit (or exit) at the CLI prompt, though, returns this message: The QUIT and EXIT commands may no longer be used to shutdown the PBX. Please use STOP NOW instead, if you wish to shutdown the PBX. But, I don't want to shutdown the PBX. I just want out of the console (CLI prompt) and back to my server command line. Like I said, I could probably just close the term window and that would terminate my ssh session. But, that's not the right way to do things. I know I'm missing something - and it's probably pretty simple. But, I have no idea what it is. Thanks for the help. :-) -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
Greg Kedrovsky wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Suggest starting as 'safe_asterisk' asterisk -r exit Always works for me... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
On Fri, Feb 27, 2004 at 01:20:28PM +, Fran Boon wrote: Suggest starting as 'safe_asterisk' asterisk -r exit Thanks. Worked like a charm. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
Try typing an ! followed by the enter key at the CLI prompt amd see what happens. - Original Message - From: Fran Boon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 27, 2004 7:20 AM Subject: Re: [Asterisk-Users] exit Greg Kedrovsky wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Suggest starting as 'safe_asterisk' asterisk -r exit Always works for me... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] exit
Ed Devine wrote: Try typing an ! followed by the enter key at the CLI prompt amd see what happens. That only drops you to a prompt. It doesn't exit the console session that was active. Unless you're intending to run asterisk not as an actual background task (your session looking at the actual running console), you should be running asterisk through asterisk or safe_asterisk. You can connect to a console of a running asterisk by typing asterisk -r, from which you can exit safely by just typing exit and pressing Enter/Return. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
Just use control-c, you will be able to exist and leaving asterisk continue to run in the background. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
Re: [Asterisk-Users] exit
You must have started asterisk with asterisk -c so you cannot bail out of CLI with exit -- you are in console mode. Instead, start it without -c so it respawns another service process and exits to shell, after that you can run asterisk -r and bail out with exit all you please ;-). - Original Message - From: Greg Kedrovsky [EMAIL PROTECTED] To: asterisk-user [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:41 PM Subject: [Asterisk-Users] exit Talk about a stoopid question... How do I exit the CLI of Asterisk. Typing exit (per the pdf manual and my google results) brings up a message saying QUIT and EXIT are no longer available, that STOP NOW is used to shutdow the pbx. I do not want to shutdown the pbx. I just wanna get outta the CLI and back to my Linux command line. Gosh... I feel like a 1st grader that can't get my fly open to take a pee. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
On Saturday, 13 December, 2003 11:16, Tilghman Lesher wrote: ... Directory does not need an escape condition. If you fail to enter anything within the allotted time (see ResponseTimeout), you jump to the t extension. That makes for a rather ill solution for the poor fool (like me, often) who accidently enters the directory and starts pounding all of the usual escape keys because he is impatient. Okay, so I am a little restricted by temper... In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. We use Voicemail() out of the box in multiple production environments. Yes. Unfortunately, so have I. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
On Friday 12 December 2003 22:04, Ulexus wrote: On Thursday, 13 November, 2003 11:34, Tilghman Lesher wrote: On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) The directory is generated from the voicemail.conf, so I imagine you would also have to an entry for extension '#' to voicemail.conf as well. Don't imagine. Try it. This seems like a really cheap (if effective and expedient) way of doing it. Just a note (and I really should add this to bugs.digium.com, I suppose), both the Directory and the Voicemail2 apps have very myopic view of the rest of the dial-plan or even their current context. Namely, the lack of an escape condition for the Directory and lack of most any dial-out conditions (i.e., '0' or another extension number) in Voicemail2. Directory does not need an escape condition. If you fail to enter anything within the allotted time (see ResponseTimeout), you jump to the t extension. In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. We use Voicemail() out of the box in multiple production environments. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
The directory is generated from the voicemail.conf, so I imagine you would also have to an entry for extension '#' to voicemail.conf as well. This seems like a really cheap (if effective and expedient) way of doing it. Just a note (and I really should add this to bugs.digium.com, I suppose), both the Directory and the Voicemail2 apps have very myopic view of the rest of the dial-plan or even their current context. Namely, the lack of an escape condition for the Directory and lack of most any dial-out conditions (i.e., '0' or another extension number) in Voicemail2. In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. Luckily, this isn't too hard, since most of the important treeing is already handled with case statements. Just add the appropriate line... On Thursday, 13 November, 2003 11:34, Tilghman Lesher wrote: On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
