Re: [asterisk-users] Multiple phones when one is unregistered
Le 01/09/2016 à 03:57, D'Arcy J.M. Cain a écrit :
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI wrote:
Something like
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
Failed. I checked the online docs and the syntax seems to be correct
No. The trailing ) is missing
but I get this:
[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 55, 3)
Is there a module that I need to load?
In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0.
What's the output of CLI command "core show application ExecIf" ?
--
Daniel
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Re: [asterisk-users] Multiple phones when one is unregistered
On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${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, 55, 3)
Set is a function, not an application. ExecIF executes an application.
I'm a bit confused by this whole topic. The dialplan snippet in the
original email
exten => 55,1,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.
Mark
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Re: [asterisk-users] Multiple phones when one is unregistered
On Thu, 1 Sep 2016 06:22:18 -0400
Mark Wiater wrote:
> On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
> > exten => 55,1,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, 55, 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.
> I'm a bit confused by this whole topic. The dialplan snippet in the
> original email
>
> > exten => 55,1,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:da...@vex.net
VoIP: sip:da...@vex.net
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Re: [asterisk-users] Multiple phones when one is unregistered
On Thu, 1 Sep 2016 11:02:57 +0200 Administrator TOOTAI wrote: > > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application > > 'ExecIf' for extension (unauthenticated, 55, 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? -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
On 9/1/2016 6:55 AM, D'Arcy J.M. Cain wrote: So does the Dial command go directly to the registered device or does it use the extension? Yes, that's why you provide the technology part (SIP/, IAX/, DAHDI/) 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. If the target device has a forward configured, I believe Asterisk will dutifully ring the forward number like it was told, but could be wrong. Otherwise, it'll only go to voicemail, in asterisk, if you tell it to in the dialplan. Could it be going to voicemail on a forward somewhere? Perhaps my problem was that one of the users was removed from sip.conf but their phone was still in the above plan. In my experience, that too should not be a problem. I've got the same dial command structure, ringing multiple SIP/extensions in one dial command. I have some instances where the SIP entry in sip.conf no longer exists, and some instances where the device is not registered, and asterisk just displays a warning and continues on. 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 don't think so. At least I don't see that. Mark -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
In article <20160901070151.10ae4f26@imp>, D'Arcy J.M. Cain wrote: > On Thu, 1 Sep 2016 11:02:57 +0200 > Administrator TOOTAI wrote: > > > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application > > > 'ExecIf' for extension (unauthenticated, 55, 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: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
On Thu, 1 Sep 2016 13:49:57 + (UTC) t...@softins.co.uk (Tony Mountifield) wrote: > > 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. > 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 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
What about doing this?
exten => 55,1,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
> On Sep 1, 2016, at 11:27 AM, D'Arcy J.M. Cain wrote:
>
> On Thu, 1 Sep 2016 13:49:57 + (UTC)
> t...@softins.co.uk (Tony Mountifield) wrote:
>>> 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.
>
>> 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
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:da...@vex.net
> VoIP: sip:da...@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
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Multiple phones when one is unregistered
Le 01/09/2016 à 17:27, D'Arcy J.M. Cain a écrit :
On Thu, 1 Sep 2016 13:49:57 + (UTC)
t...@softins.co.uk (Tony Mountifield) wrote:
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.
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
You can do it with GotoIf
exten => 55,1,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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Re: [asterisk-users] Multiple phones when one is unregistered
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: 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. -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.11.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: --- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) Bugs fixed in this release: --- * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) * ASTERISK-26227 - sqlalchemy error due to long identifier name (Reported by Mark Michelson) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26214 - Allow arbitrary time for fax detection to end on a channel (Reported by Richard Mudgett) * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst) * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel executing Playback (Reported by Richard Mudgett) * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and DTD in docs. (Reported by Alexander Traud) * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional code. (Reported by Corey Farrell) * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even on lost packets. (Reported by Alexander Traud) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime performance - remove unneeded check on endpoint's contacts. (Reported by Alexei Gradinari) * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by Richard Miller) * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported by Corey Farrell) * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb (Reported by Corey Farrell) * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme) * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group (Reported by Corey Farrell) * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) * ASTERISK-26160 - pjsip: Updated->Reachable during qualify (Reported by Matt Jordan) * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when building menuselect (Reported by Jeffrey Walton) * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and bugs (Reported by Alexei Gradinari) * ASTERISK-26177 - func_odbc: Database handle is kept when it should be released (Reported by Leandro Dardini) * ASTERISK-26184 - chan_sip: Reference leaks in error paths. (Reported by Corey Farrell) * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged during duplicate replacement (Reported by Corey Farrell) * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for reuse (Reported by Scott Griepentrog) * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-26172 - res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows. (Reported by Alexei Gradinari) * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered (Reported by Dmitriy Serov) * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer) * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by Alexei Gradinari) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13 (Reported by Daniel Denson) * ASTERISK-26326 - Crash when dialing MulticastRTP channel (Reported by George Joseph) Improvements made in this release: --- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell) * ASTERISK-22131 - Update the make dependencies script to pull, build, and install the correct pjproject (Reported by Matt Jordan) * ASTERISK-25471 -
Re: [asterisk-users] Multiple phones when one is unregistered
> So does the Dial command go directly to the registered device or does > it use the extension? 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. > 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. -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 - Security Fix Mode
Hello Everyone, As many of you are already aware, we are rapidly approaching the time when the Asterisk 11 branch will go into what is known as security fix only mode. Up to this point, bug fixes have been included and merged into the 11 branch. For Asterisk 11, this new phase of life shall begin October 25th of this year. This means that from a development perspective, the Asterisk development team will not be putting effort into bug fixes for the 11 branch after the 25th of October. Security related patches will be merged for another year before completely putting the branch to rest. During the course of that year, releases will be made as needed and as security related patches are merged. For any questions, you can either reply to this message or look at the Asterisk Versioning Policy wiki page [1] as it explains most of this process in greater detail. Thanks so much again for all of your support and effort. Asterisk would not be what it is if it were not for all the great contributors that are and have been involved. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP audio question
I have a device I wish to use that when activated will output RTP audio 8kHz, 16-bit linear RTP audio. Can asterisk use that as a source and dial a phone so the user would hear that audio? Thanks, Jerry -- _ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
