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darcy at Vex.Net
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 Posted: Tue Aug 30, 2016 8:56 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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|
I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
Is there any way to direct only to phones in a list that are currently
registered? I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
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asterisk at Offermans....
Guest
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| Posted: Tue Aug 30, 2016 9:35 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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|
Dear D'Arcy J.M. Cain and asterisk friends,
On Tue, Aug 30, 2016 at 09:56:05AM -0400, D'Arcy J.M. Cain wrote:
| Quote: | I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
Is there any way to direct only to phones in a list that are currently
registered? I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
|
To my opinion, you need a queue with dynamic agents.
SIP/userX subscribes and checkouts manually, or better SIP/userX adds to
the specific queue at the moment she or he registers to the system. I know
how to do the former, but I would like to know about the latter. If someone
could comment on that, I would highly appreciate.
--
Met vriendelijke groeten,
With kind regards,
Mit freundlichen Gruessen,
De jrus wah,
Will
*************************************
W.K. Offermans
Powered by ....
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\\\'',)
\/ \ ^
.\._/_)
www.FreeBSD.org
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jcolp at digium.com
Guest
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| Posted: Tue Aug 30, 2016 9:38 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
D'Arcy J.M. Cain wrote:
| Quote: | I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
Is there any way to direct only to phones in a list that are currently
registered? I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
|
The Asterisk behavior is such that even if one of those users are
unregistered the other two will continue to be called. Is something
upstream from Asterisk doing the voicemail?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
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ewieling at nyigc.com
Guest
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| Posted: Tue Aug 30, 2016 9:39 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
The dialplan below cannot go to voicemail, either something else is
going on or the dialplan below is not the actual dialplan. Calls only go
to voicemail when the dialplan runs the VoiceMail application.
Paste the Asterisk console output of a call showing the problem, maybe
someone can help.
On 08/30/2016 09:56 AM, D'Arcy J.M. Cain wrote:
| Quote: | I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
Is there any way to direct only to phones in a list that are currently
registered? I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
Cheers.
|
--
if at first you don't succeed, skydiving isn't for you
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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asterisk_list at earth...
Guest
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| Posted: Tue Aug 30, 2016 9:53 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tuesday 30 Aug 2016, D'Arcy J.M. Cain wrote:
| Quote: | I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
|
There is nothing in that dialplan snippet you provided that would send
anything to voicemail. In fact, if you used that snippet, it would give the
exact behaviour you describe yourself as wanting!
Here's a protip for you: It's *not* a massive corporate secret, which phone
will ring when somebody dials what number. Obfuscating your real dialplan
like that just makes it harder for other people to help you.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
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darcy at Vex.Net
Guest
|
| Posted: Tue Aug 30, 2016 10:32 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tue, 30 Aug 2016 10:39:14 -0400
Eric Wieling <ewieling@nyigc.com> wrote:
| Quote: | The dialplan below cannot go to voicemail, either something else is
|
Of course not. It's the individual extensions that have voice mail. I
have a similar problem when one of those destinations is a cell phone
but I know that there is no Asterisk solution for that problem. If the
cell phone answers and goes into the cell phone voice mail I am
screwed. I can't fix that unless I also run the cell phone company.
In this case the extensions are all set up in the same asterisk
server. Here is an example.
[user1](client-phone)
secret=NotTheRealSecret
callerid=John Doe <5555551212>
mailbox=user1@VoiceMail
context=pbxout-19212
And here is the extension that calls it. Sorry about the wrapping.
user1,1,Verbose(0,Entering extension user1)
same => n,GotoIf($["${DEVICE_STATE(SIP/9999906549)}" =
"UNAVAILABLE"]?DialCell)
same => n(DialWifi),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and
SoftPhone)
same => n,Dial(SIP/user1&SIP/9999906549,30)
same => n,Goto(VoiceMail)
same => n(DialCell),GotoIf($["x7165550000" = "x"]?DialDesk)
same => n,GotoIf($["${CALLERID(ani)}" = "7165550000"]?DialDesk)
same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell
"7165550000")
same => n,Dial(SIP/user1&SIP/thinktel/7165550000,30)
same => n,Goto(VoiceMail)
same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/user1,30)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Set(_ACCOUNT=user1)
same => n,VoiceMail(user1@VoiceMail,u)
same => n,Hangup()
It's a complicated extension but I believe that the following is the
essential part.
user1,1,Verbose(0,Entering extension user1)
same => n,Dial(SIP/user1,30)
same => n,VoiceMail(user1@VoiceMail,u)
same => n,Hangup()
| Quote: | going on or the dialplan below is not the actual dialplan. Calls only
go to voicemail when the dialplan runs the VoiceMail application.
|
| Quote: | Paste the Asterisk console output of a call showing the problem,
maybe someone can help.
|
Here is what the log shows. I can't put the unregistered user back at
the moment. Perhaps I can do it overnight when no one is going to the
building.
[Aug 23 10:20:55] WARNING[-1][C-0001fee7] app_dial.c: Unable to create
channel of type 'SIP' (cause 20 - Subscriber absent)
I am assuming that the voice mail is for the absent (unregistered) user.
| Quote: | | Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
|
|
Other than sanitizing the buzzer phone number and the user names this
is the full extension for the buzzer.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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darcy at Vex.Net
Guest
|
| Posted: Tue Aug 30, 2016 10:39 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tue, 30 Aug 2016 15:53:04 +0100
A J Stiles <asterisk_list@earthshod.co.uk> wrote:
| Quote: | | Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
|
|
| Quote: | Here's a protip for you: It's *not* a massive corporate secret,
|
I will give you the benefit of the doubt and assume that you are not
trying to patronize me.
| Quote: | which phone will ring when somebody dials what number. Obfuscating
your real dialplan like that just makes it harder for other people to
help you.
|
All I did was sanitize the buzzer phone number and the names of the
users. Other than that the above is the complete dialplan. See my
other message for details of those extensions. I didn't think that
there was anything relevant in that but I could be wrong.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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ewieling at nyigc.com
Guest
|
| Posted: Tue Aug 30, 2016 10:42 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On 08/30/2016 11:31 AM, D'Arcy J.M. Cain wrote:
| Quote: | Paste the Asterisk console output of a call showing the problem,
Here is what the log shows. I can't put the unregistered user back at
the moment. Perhaps I can do it overnight when no one is going to the
building.
[Aug 23 10:20:55] WARNING[-1][C-0001fee7] app_dial.c: Unable to create
channel of type 'SIP' (cause 20 - Subscriber absent)
I am assuming that the voice mail is for the absent (unregistered) user.
|
You should set your verbosity to 3 and then reproduce the call and paste
the CLI output. It is pointless to troubleshoot without the CLI
output. The warning is expected when the phone is offline and does not
cause the issue you are experiencing. It almost looks like Local/
channels are involved, but we'll know more once we see the CLI output.
--
_____________________________________________________________________
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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admin at tootai.net
Guest
|
| Posted: Tue Aug 30, 2016 10:56 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
Le 30/08/2016 à 15:56, D'Arcy J.M. Cain a écrit :
| Quote: | I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
Is there any way to direct only to phones in a list that are currently
registered? I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.
|
Something like
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
same => n,ExecIf($["${DEVICE_STATE(SIP/user2)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user2)
same => n,ExecIf($["${DEVICE_STATE(SIP/user3)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user3)
same => n,Dial(${toRing:1}) ;to remove the first &
would do the work
--
Daniel
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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darcy at Vex.Net
Guest
|
| Posted: Tue Aug 30, 2016 11:01 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tue, 30 Aug 2016 11:42:04 -0400
Eric Wieling <ewieling@nyigc.com> wrote:
| Quote: | | Quote: | [Aug 23 10:20:55] WARNING[-1][C-0001fee7] app_dial.c: Unable to
create channel of type 'SIP' (cause 20 - Subscriber absent)
I am assuming that the voice mail is for the absent (unregistered)
user.
|
You should set your verbosity to 3 and then reproduce the call and
paste the CLI output. It is pointless to troubleshoot without the
|
I can't do it during the day but I will try overnight. Can't be
tonight as I have a gig (120 Diner at 120 Church if you happen to be in
Toronto) but I will try later this week.
| Quote: | CLI output. The warning is expected when the phone is offline and
does not cause the issue you are experiencing. It almost looks like
Local/ channels are involved, but we'll know more once we see the CLI
|
What do you mean by "Local channels?" They are local to the server.
Do you mean that I might be in the wrong context?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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darcy at Vex.Net
Guest
|
| Posted: Tue Aug 30, 2016 11:06 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
same => n,ExecIf($["${DEVICE_STATE(SIP/user2)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user2)
same => n,ExecIf($["${DEVICE_STATE(SIP/user3)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user3)
same => n,Dial(${toRing:1}) ;to remove the first &
would do the work
|
That looks good and is easy to add and delete from the list. I will
give this a try one night this week. Not sure what that last line
would do if all of the phones are off but if they are the buzzer won't
be answered anyway.
Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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admin at tootai.net
Guest
|
| Posted: Tue Aug 30, 2016 11:30 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
Le 30/08/2016 à 18:05, D'Arcy J.M. Cain a écrit :
| Quote: | On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
same => n,ExecIf($["${DEVICE_STATE(SIP/user2)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user2)
same => n,ExecIf($["${DEVICE_STATE(SIP/user3)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user3)
same => n,Dial(${toRing:1}) ;to remove the first &
would do the work
|
That looks good and is easy to add and delete from the list. I will
give this a try one night this week. Not sure what that last line
would do if all of the phones are off but if they are the buzzer won't
be answered anyway.
|
Don't execute the Dial cmd if ${toRing} is empty 
...
same => n,ExecIf($["x${toRing}" != "x"]?Dial(${toRing:1})) ;to remove
the first &
--
Daniel
--
_____________________________________________________________________
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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darcy at Vex.Net
Guest
|
| Posted: Wed Aug 31, 2016 8:57 pm Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | Something like
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
|
Failed. I checked the online docs and the syntax seems to be correct
but I get this:
[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 5555551111, 3)
Is there a module that I need to load?
In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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admin at tootai.net
Guest
|
| Posted: Thu Sep 01, 2016 4:03 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
Le 01/09/2016 à 03:57, D'Arcy J.M. Cain a écrit :
| Quote: | On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | Something like
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
|
Failed. I checked the online docs and the syntax seems to be correct
|
No. The trailing ) is missing
| Quote: | but I get this:
[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 5555551111, 3)
Is there a module that I need to load?
In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0.
|
What's the output of CLI command "core show application ExecIf" ?
--
Daniel
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darcy at Vex.Net
Guest
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| Posted: Thu Sep 01, 2016 5:55 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
|
|
On Thu, 1 Sep 2016 06:22:18 -0400
Mark Wiater <mark.wiater@greybeam.com> wrote:
| Quote: | On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
| Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
| Quote: | same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
|
Failed. I checked the online docs and the syntax seems to be
correct but I get this:
[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 5555551111, 3)
|
Set is a function, not an application. ExecIF executes an application.
|
I wondered about that but the docs are not very explicit on the subject
and I thought that the other poster was suggesting something that he
had used/tested.
| Quote: | I'm a bit confused by this whole topic. The dialplan snippet in the
original email
| Quote: | exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
|
should have rung the phones forever as long as one phone was active
and not forwarding or DNDing.
|
So does the Dial command go directly to the registered device or does
it use the extension? I was assuming that it was going to the
extension's voice mail if it wasn't there but that's in the extension
dialplan and I suspect that the extension is irrelevant and only the
SIP registration matters. That would be a good thing since many
extensions also ring the user's cell phone and that would be annoying
if they were at home when someone came to the office door.
Perhaps my problem was that one of the users was removed from sip.conf
but their phone was still in the above plan. For example, user2 leaves
the company, is removed from sip.conf but we forgot to remove him from
the door buzzer extension. That might give me the behaviour that I was
seeing.
I think I have enough information now to analyze this problem if it
happens again. Thanks for the help.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
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