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darcy at Vex.Net
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 Posted: Thu Sep 01, 2016 6:02 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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On Thu, 1 Sep 2016 11:02:57 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | | Quote: | [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.
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What's the output of CLI command "core show application ExecIf" ?
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It looks like this doesn't matter any more but I do wonder why I don't
have that command.
# asterisk -x "core show application ExecIf"
Your application(s) is (are) not registered
Command 'core show application ExecIf' failed.
What module am I missing?
--
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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tony at softins.co.uk
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| Posted: Thu Sep 01, 2016 8:50 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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In article <20160901070151.10ae4f26@imp>,
D'Arcy J.M. Cain <darcy@Vex.Net> wrote:
| Quote: | On Thu, 1 Sep 2016 11:02:57 +0200
Administrator TOOTAI <admin@tootai.net> wrote:
| Quote: | | Quote: | [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" ?
|
It looks like this doesn't matter any more but I do wonder why I don't
have that command.
# asterisk -x "core show application ExecIf"
Your application(s) is (are) not registered
Command 'core show application ExecIf' failed.
What module am I missing?
|
The ExecIf command is provided in the module app_exec, which is usually
located at /usr/lib/asterisk/modules/app_exec.so
Maybe you had turned off app_exec in the menuconfigi when building, or maybe your
modules.conf has a noload => app_exec.so
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
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darcy at Vex.Net
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asterisk at mothersell...
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| Posted: Thu Sep 01, 2016 11:05 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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What about doing this?
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,Set(toRing=${toRing}${IF($["${DEVICE_STATE(SIP/user1)}" = "NOT IN USE"]?&SIP/user1:)})
same => n,Set(toRing=${toRing}${IF($["${DEVICE_STATE(SIP/user2)}" = "NOT IN USE"]?&SIP/user2:)})
same => n,Set(toRing=${toRing}${IF($["${DEVICE_STATE(SIP/user3)}" = "NOT IN USE"]?&SIP/user3:)})
same => n,Dial(${toRing:1}) ;to remove the first &
Chad
| Quote: | On Sep 1, 2016, at 11:27 AM, D'Arcy J.M. Cain <darcy@Vex.Net> wrote:
On Thu, 1 Sep 2016 13:49:57 +0000 (UTC)
tony@softins.co.uk (Tony Mountifield) wrote:
| Quote: | | Quote: | What module am I missing?
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The ExecIf command is provided in the module app_exec, which is
usually located at /usr/lib/asterisk/modules/app_exec.so
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Yes, I see it.
| Quote: | Maybe you had turned off app_exec in the menuconfigi when building,
or maybe your modules.conf has a noload => app_exec.so
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The distributed modules.conf does not appear to mention it at all. I
don't need it now after all so I won't add it in until I can evaluate
the security issues that it might bring with it. It's hard to find
documentation for it other than the actual source code.
http://doxygen.asterisk.org/trunk/dc/d73/app__exec_8c-source.html
--
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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admin at tootai.net
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| Posted: Thu Sep 01, 2016 11:16 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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Le 01/09/2016 à 17:27, D'Arcy J.M. Cain a écrit :
| Quote: | On Thu, 1 Sep 2016 13:49:57 +0000 (UTC)
tony@softins.co.uk (Tony Mountifield) wrote:
| Quote: | | Quote: | What module am I missing?
|
The ExecIf command is provided in the module app_exec, which is
usually located at /usr/lib/asterisk/modules/app_exec.so
|
Yes, I see it.
| Quote: | Maybe you had turned off app_exec in the menuconfigi when building,
or maybe your modules.conf has a noload => app_exec.so
|
The distributed modules.conf does not appear to mention it at all. I
don't need it now after all so I won't add it in until I can evaluate
the security issues that it might bring with it. It's hard to find
documentation for it other than the actual source code.
http://doxygen.asterisk.org/trunk/dc/d73/app__exec_8c-source.html
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You can do it with GotoIf
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,GotoIf($["${DEVICE_STATE(SIP/user1)}" != "NOT IN USE"]?User2)
same => n,Set(toRing=${toRing}&SIP/user1)
same => n(User2),GotoIf($["${DEVICE_STATE(SIP/user2)}" != "NOT IN
USE"]?User3)
same => n,Set(toRing=${toRing}&SIP/user3)
same => n(User3),GotoIf($["${DEVICE_STATE(SIP/user3)}" != "NOT IN
USE"]?Call)
same => n,Set(toRing=${toRing}&SIP/user3)
same => n(Call),GotoIf($["x${toRing}" = "x"]?NoPhoneToCall)
same => n,Dial(${toRing:1}) ;to remove the first &
--
Daniel
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ewieling at nyigc.com
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| Posted: Thu Sep 01, 2016 11:54 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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The dial application dials devices not extensions. The only way to
"dial" an extension from the dialplan is to use chan_local.
On 09/01/2016 06:55 AM, D'Arcy J.M. Cain wrote:
| Quote: | 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.
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dplatt at radagast.org
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| Posted: Thu Sep 01, 2016 12:36 pm Post subject: [asterisk-users] Multiple phones when one is unregistered |
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| Quote: | So does the Dial command go directly to the registered device or does
it use the extension?
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If you've given the Dial() command the SIP/user1 format, it will attempt
to dial directly to the SIP device/phone/endpoint you specify. If you
specify SIP/user1&SIP/user2&... it attempts to dial directly to all of
them simultaneously, and the first one which picks up, gets the call
(the dialouts to the others are dropped when the first one answers).
To the best of my knowledge there is *no* automatic fallback to the
Asterisk voicemail which might be associated with one or more of
these SIP users.
The usual way that you'd get to Asterisk voicemail, is if your
dialplan catches the error which would result from Dial() if none
of the users is available and answers, and explicitly calls
the Voicemail() app.
Things can become more complicated in a couple of situations:
(1) If one of the SIP users you specify isn't actually a SIP
endpoint device, but is a SIP identity on another system (PBX
or VoIP provider or etc.), then you really don't have any control
over how that endpoint would handle situations where the called
user isn't available. The endpoint might answer with *its*
voicemail, immediately.
(2) If you were to dial a Local/ destination rather than a SIP/
destination, then that dialing operation *is* run back through
your dialplan, and it might divert the call to voicemail instantly.
The easiest solution to each of these is "Don't do that". Don't
multi-dial to anything other than SIP (or IAX) endpoints which are
real, physical devices that either ring (if they're connected) or
fail to respond or reject the call (if they aren't available). Don't
multi-dial to any SIP device which implements its own internal
"voicemail" feature (e.g. has an answering machine attached).
I do what you're thinking of all the time. On my Asterisk
setup, one incoming PSTN number goes to an extension which
multi-dials about half-a-dozen of my SIP softphones. No matter
which tablet or PC I happen to be using, if I'm running the
SIP softphone app, it'll ring.
The only time the call fails from this dial is if none of
the SIP devices answer. I could route to Asterisk voicemail in
this case, but I don't bother - Asterisk simply rejects the call
with a no-answer or not-available status, the VoIP provider fails
the call, and Google Voice (which is where the original number is
anchored) sends the call to its own voicemail system and I get an
email.
The only down-side to this is that the Asterisk log gets a bunch
of "SIP call failed" status messages each time this happens - one for
each dialed SIP user that wasn't "on the net" at the time. This isn't
a problem for me in practice.
| Quote: | 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.
|
Correct.
--
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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andrew.ruthven at cata...
Guest
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| Posted: Sat Sep 03, 2016 2:51 am Post subject: [asterisk-users] Multiple phones when one is unregistered |
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On Thu, 2016-09-01 at 10:36 -0700, Dave Platt wrote:
| Quote: | Things can become more complicated in a couple of situations:
(1) If one of the SIP users you specify isn't actually a SIP
    endpoint device, but is a SIP identity on another system (PBX
    or VoIP provider or etc.), then you really don't have any control
    over how that endpoint would handle situations where the called
    user isn't available.  The endpoint might answer with *its*
    voicemail, immediately.
(2) If you were to dial a Local/ destination rather than a SIP/
    destination, then that dialing operation *is* run back through
    your dialplan, and it might divert the call to voicemail
instantly.
|
Another option is what I've had happen recently. I have my main number
dial all the phones in my house, including an old Cisco 7905 that on
busy or no answer would send back a 302 redirect to extension 8000 -
VoiceMail. To make matters worse inbound callers would be dumped into
VoiceMail as though they'd entered it from internally, rather than
external.
While I tried various different ways on the Cisco to stop that
behaviour, I found the only solution was to tell the Dial() command to
ignore the 302 by adding the i flag. Problem solved.
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP, ITCP
At work: andrew.ruthven@catalyst.net.nz
At home: andrew@etc.gen.nz
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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