Nitesh Sharma
2014-Oct-31 04:26 UTC
[asterisk-users] asterisk-users Digest, Vol 123, Issue 38
Hi I am new to mailing list ,please correct me if the way of posting is not
correct
Relpying to :
Re: make asterisk do something when an outgoing call is
picked up (lee)
For making asterisk do something on outgoing call Dial application is
itself used
Like for Playing an announcement to the caller on pick up the is an option
A(x) where x is the file to play to the called party.
Also you can call an macro from within the Dial application , so you can
perform IVR to the caller using the macro
On the asterisk cli type core show application Dial and then look for
the A(x) option and Macro Option to know in detail
I have used both A(x ) option and Macro Option ..so sure about them .
On Fri, Oct 31, 2014 at 7:00 AM, <asterisk-users-request at
lists.digium.com>
wrote:
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>
> Today's Topics:
>
> 1. Register multiple phones to a single AOR with PJSIP
> (Carlos Chavez)
> 2. Re: make asterisk do something when an outgoing call is
> picked up (lee)
> 3. PlayTones not working (Henry Fernandes)
> 4. Re: Register multiple phones to a single AOR with PJSIP
> (Scott Griepentrog)
> 5. Re: ${HASH(SIP_CAUSE,<channel-name>)} (Paul Belanger)
> 6. Re: make asterisk do something when an outgoing call is
> picked up (John Kiniston)
> 7. Re: AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben
> Klang) (Paul Albrecht)
> 8. Re: AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben
> Klang) (Paul Albrecht)
> 9. Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate
> AMI/AGI (Ben Klang) (Ben Klang)
> 10. MWI publish VIA pjsip for non sip channels (Matt Hoskins)
> 11. Re: MWI publish VIA pjsip for non sip channels (Joshua Colp)
> 12. Re: MWI publish VIA pjsip for non sip channels (Matt Hoskins)
> 13. Re: MWI publish VIA pjsip for non sip channels (Joshua Colp)
> 14. Re: MWI publish VIA pjsip for non sip channels (Matt Hoskins)
> 15. Re: Register multiple phones to a single AOR with PJSIP
> (Matthew Jordan)
> 16. Paul Albrecht (Matthew Jordan)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 30 Oct 2014 13:18:25 -0600
> From: Carlos Chavez <cursor at telecomabmex.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Register multiple phones to a single AOR
> with PJSIP
> Message-ID: <54528F01.4080700 at telecomabmex.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> I just finished installing Asterisk 13 on our test server and I can
> now use PJSIP to register phones and make and receive calls. The only
> problem I am having is that when I register multiple phones to a single
> account only one of them rings. The AOR for the account has maxcontacts
> at 3.
>
> If I do a pjsip show endpoints I can see two "Contact"
entries
> which I take to mean that both phones have registered:
>
> Endpoint: 101 Not in
> use 0 of inf
> InAuth: 101/101
> Aor: 101 3
> Contact: 101/sip:101 at 192.168.2.193:5063 Avail
178.681
> Contact: 101/sip:101 at 192.168.2.197:58086;transport=UDP;r
> Avail 4.198
> Transport: transport-udp udp 0 0 0.0.0.0:5060
>
> I have tried with several phones and have rebooted the Asterisk
> server and phones several times just to make sure configs are loaded
> properly but I cannot get Asterisk to ring multiple phones at once. I
> used
> https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime to
> configure this instance of Asterisk. Am I missing some setting to allow
> Asterisk to ring all phones registered to a single AOR?
>
> --
> Telecomunicaciones Abiertas de M?xico S.A. de C.V.
> Carlos Ch?vez
> +52 (55)9116-91161
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 30 Oct 2014 20:21:15 +0100
> From: lee <lee at yagibdah.de>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] make asterisk do something when an
> outgoing call is picked up
> Message-ID: <87k33h8itw.fsf at gulltop.yagibdah.de>
> Content-Type: text/plain; charset=utf-8
>
> Thorsten G?llner <tg at ovm-group.com> writes:
>
> > Am 26.10.2014 00:43, schrieb lee:
> >> Hi,
> >>
> >> how can I make asterisk do something when an outgoing call is
picked up?
> >>
> >>
> >> The background is that I would like to record incoming and
outgoing
> >> phone calls. In order to do this, I need to play an announcement
> >> telling the person calling or being called that the call will be
> >> recorded.
> >>
> >
> > Maybe this will do a good job for recording all calls:
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
> >
> > And playing an announcement, when a call is picked, should be done
> > within your dialplan with this function:
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+Playback
>
> Thank you --- I'm not sure what to do with these. I've been able
to use
> Playback to play an announcement, and ChanSpy just looks complicated.
>
> What if I press a button on the phone while a call is going on? Can I
> somehow make it so that recording starts when I do that?
>
>
> --
> Again we must be afraid of speaking of daemons for fear that daemons
> might swallow us. Finally, this fear has become reasonable.
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 30 Oct 2014 13:40:20 -0600
> From: Henry Fernandes <henry at usinternet.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] PlayTones not working
> Message-ID: <D077F044.EFF5%henry at usinternet.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I?m trying to use Playtones to have a tone played periodically throughout
> phone calls. Unfortunately, I can?t seem to get PlayTones to work. I
> never
> hear the audio tones.
>
> Here is the output on the Asterisk console.
> -- Executing [19525553312 at proxy-dial:2]
> PlayTones("SIP/testphone-00000032",
> "1400/500,2000/5000") in new stack
>
> [2014-10-30 14:28:31] WARNING[23154]: translate.c:206 framein: no samples
> for ulawtolin
>
> -- Executing [1952553312 at proxy-dial:3]
Dial("SIP/testphone-00000032",
> "SIP/19525553312 at proxy01,,gU(record_call_id)") in new stack
>
>
>
> I?ve checked the debug log and I can?t see any related errors or warning
> beyond the one above.
>
> -H
>
>
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> ------------------------------
>
> Message: 4
> Date: Thu, 30 Oct 2014 14:47:44 -0500
> From: Scott Griepentrog <sgriepentrog at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Register multiple phones to a single AOR
> with PJSIP
> Message-ID:
> <CACrpESYeXvxzF> W-CAoa7d2pEZrtqXnK9Yc3T6ixG9EV1U2msA at
mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> ?You need to change your dialplan to use the PJSIP_DIAL_CONTACTS function
> like this:
>
> exten => _X.,1,Dial(${PJSIP_DIAL_CONTACTS(200)},30)?
>
> It expands to the list of contacts, separated by &, so that the
contacts
> are dialed at the same time.
>
> The documentation page you reference should be updated to include that
> detail.
>
>
> On Thu, Oct 30, 2014 at 2:18 PM, Carlos Chavez <cursor at
telecomabmex.com>
> wrote:
>
> > I just finished installing Asterisk 13 on our test server and I
can
> > now use PJSIP to register phones and make and receive calls. The only
> > problem I am having is that when I register multiple phones to a
single
> > account only one of them rings. The AOR for the account has
maxcontacts
> at
> > 3.
> >
> > If I do a pjsip show endpoints I can see two "Contact"
entries which
> I
> > take to mean that both phones have registered:
> >
> > Endpoint: 101 Not in
> > use 0 of inf
> > InAuth: 101/101
> > Aor: 101 3
> > Contact: 101/sip:101 at 192.168.2.193:5063 Avail
178.681
> > Contact: 101/sip:101 at 192.168.2.197:58086;transport=UDP;r
Avail
> > 4.198
> > Transport: transport-udp udp 0 0 0.0.0.0:5060
> >
> > I have tried with several phones and have rebooted the Asterisk
> server
> > and phones several times just to make sure configs are loaded properly
> but
> > I cannot get Asterisk to ring multiple phones at once. I used
> > https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
to
> > configure this instance of Asterisk. Am I missing some setting to
allow
> > Asterisk to ring all phones registered to a single AOR?
> >
> > --
> > Telecomunicaciones Abiertas de M?xico S.A. de C.V.
> > Carlos Ch?vez
> > +52 (55)9116-91161
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> [image: Digium logo]
> Scott Griepentrog
> Digium, Inc ? Software Developer
> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
> Check us out at: http://digium.com ? http://asterisk.org
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> ------------------------------
>
> Message: 5
> Date: Thu, 30 Oct 2014 16:07:56 -0400
> From: Paul Belanger <paul.belanger at polybeacon.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}
> Message-ID:
> <CALLKq0TN0hkCgbs3TX3CJo> ysY+m+9O0zonEEpfdQwdujMY9dQ at
mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens <jonas.kellens at
telenet.be>
> wrote:
> > Hello,
> >
> > I read on the wiki :
> >
> > Asterisk 1.8 will allow to read SIP response codes in the dialplan via
> > ${HASH(SIP_CAUSE,<channel-name>)}. Additionally make sure
you're using
> the
> > destination channel, not the source channel.
> >
> > But when I use this in my dialplan, this 'variable' is empty.
> >
> > Dialplan :
> >
> > exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
> > exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})
> >
> > CLI :
> >
> > [Oct 30 14:48:03] -- Executing [h at pbx-routing:5]
> > NoOp("SIP/SipAT01-00000015", "sip cause = ") in
new stack
> > [Oct 30 14:48:03] -- Executing [h at pbx-routing:6]
> > NoOp("SIP/SipAT01-00000015", "sip cause = ") in
new stack
> >
> >
> > Can anyone tell me how this should be used ?
> >
> sip.conf: storesipcause=yes
>
>
> --
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> Github: https://github.com/pabelanger | Twitter:
> https://twitter.com/pabelanger
>
>
>
> ------------------------------
>
> Message: 6
> Date: Thu, 30 Oct 2014 13:58:40 -0700
> From: John Kiniston <johnkiniston at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] make asterisk do something when an
> outgoing call is picked up
> Message-ID:
> <CAFJQOGc93qSdXmvese2y+iNsJQbJgrEXRbNCe1d> mf_TXuzACg at
mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Lee I recommend you use the MixMonitor application along with a combination
> of Playback to play your message to the Calling party and the A argument to
> Dial to play a file to the called party.
>
> So for your outbound calls:
>
> exten => _NXXXXXX,1,NoOP()
> same => n,MixMonitor(recording-${CDR(UNIQUEID)}.wav)
> same =>
> n,Playback(this-call-may-be-monitored-or-recorded,noanswer)
> same =>
> n,Dial(SIP/${EXTEN},A(this-call-may-be-monitored-or-recorded))
>
> While your inbound calls would look like
>
> exten => s,1,NoOP()
> same => n,Answer()
> same => n,MixMonitor(recording-${CDR(UNIQUEID)}.wav)
> same => n,Playback(this-call-may-be-monitored-or-recorded)
> same =>
> n,Dial(SIP/1001,Playback(this-call-may-be-monitored-or-recorded,))
>
> On Thu, Oct 30, 2014 at 12:21 PM, lee <lee at yagibdah.de> wrote:
>
> > Thorsten G?llner <tg at ovm-group.com> writes:
> >
> > > Am 26.10.2014 00:43, schrieb lee:
> > >> Hi,
> > >>
> > >> how can I make asterisk do something when an outgoing call is
picked
> up?
> > >>
> > >>
> > >> The background is that I would like to record incoming and
outgoing
> > >> phone calls. In order to do this, I need to play an
announcement
> > >> telling the person calling or being called that the call will
be
> > >> recorded.
> > >>
> > >
> > > Maybe this will do a good job for recording all calls:
> > > http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
> > >
> > > And playing an announcement, when a call is picked, should be
done
> > > within your dialplan with this function:
> > > http://www.voip-info.org/wiki/view/Asterisk+cmd+Playback
> >
> > Thank you --- I'm not sure what to do with these. I've been
able to use
> > Playback to play an announcement, and ChanSpy just looks complicated.
> >
> > What if I press a button on the phone while a call is going on? Can I
> > somehow make it so that recording starts when I do that?
> >
> >
> > --
> > Again we must be afraid of speaking of daemons for fear that daemons
> > might swallow us. Finally, this fear has become reasonable.
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
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> ------------------------------
>
> Message: 7
> Date: Thu, 30 Oct 2014 15:57:43 -0500
> From: Paul Albrecht <palbrecht at glccom.com>
> To: Asterisk Developers Mailing List <asterisk-dev at
lists.digium.com>,
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate
> AMI/AGI (Ben Klang)
> Message-ID: <8E422546-5698-4A85-90E0-32E46FE6EEDF at glccom.com>
> Content-Type: text/plain; charset="windows-1252"
>
>
> On Oct 29, 2014, at 2:45 PM, Ben Klang <bklang at mojolingo.com>
wrote:
>
> >
> >> On 10/28/2014 06:03 PM, Ben Langfeld wrote:
> >>> On 28 October 2014 19:47, Derek Andrew <Derek.Andrew at
usask.ca> wrote:
> >>> What is the alternative to the dial plan? Is everyone talking
about
> getting rid of the statements like:
> >>> exten => s,1,
> >>>
> >>> what is the alternative?
> >>>
> >>> Remote applications based on APIs like ARI. This is the start
of the
> discussion, and please remember that nothing has been decided or even
> presented as a robust plan yet. This is brain-storming.
> >>>
> >>> Additionally, note that the original proposal was to deprecate
AMI/AGI
> in favour of ARI once it is feature complete with those protocols; an
> entirely lesser change than the removal of the dialplan in its entirety.
> >>>
> >
> > Since this thread has my name on it, I guess it?s past time that I
> explain my motivation for making the suggestion, and try to restore some of
> the context that was present in the discussion at AstriDevCon.
> >
> > Before I jump into the details of my proposal, I?d like to clarify
> terms...
> >
>
> It?s intellectually dishonest to redefine the terms of an argument to
> presuppose your own conclusion. If you don?t intend to use the term
> ?deprecate? as it is commonly understood by software developers and users
> than you should avoid the use of the term ?deprecate? so that others
> clearly understand your argument. If you really mean ?deprecate? as
> commonly understood by software developers and users then you should be
> prepared to defend that proposition.
>
> > Now, on to what I originally proposed...
> >
>
> It?s clear from the title of the agenda item what was proposed. You
> proposed deprecating AMI/AGI and that entails deprecating the dial plan.
> The fact that deprecating the dial plan is now on the table is a direct
> consequence of your proposal. This is reflected in both comments made at
> AstiCon and Matt?s summary of Astricon on the development list. You can?t
> have it both ways. You want to deprecate dial plan or not. Which is it?
>
> > It is my opinion that while AGI and AMI are probably individually
> fixable, doing so would cause backward-incompatible changes?
>
> Deprecating the dial plan and AGI/AMI is incompatible going forward. What
> is supposed to happen? Are users supposed to throw away there applications
> whenever ARI/Stasis is feature complete? Is ARI/Stasis really any easier to
> use than the dial plan? Are we all supposed to use Adhearsion?
>
>
>
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> ------------------------------
>
> Message: 8
> Date: Thu, 30 Oct 2014 15:59:35 -0500
> From: Paul Albrecht <palbrecht at glccom.com>
> To: Matthew Jordan <mjordan at digium.com>, Asterisk Developers
Mailing
> List <asterisk-dev at lists.digium.com>,
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate
> AMI/AGI(Ben Klang)
> Message-ID: <66661B22-2F91-49F8-8CAA-8383C443BE21 at glccom.com>
> Content-Type: text/plain; charset="windows-1252"
>
>
> On Oct 29, 2014, at 4:26 PM, Matthew Jordan <mjordan at digium.com>
wrote:
>
> > On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht <palbrecht at
glccom.com>
> wrote:
> >>
> >> On Oct 28, 2014, at 5:03 PM, Ben Langfeld <ben at
langfeld.me> wrote:
> >>
> >> On 28 October 2014 19:47, Derek Andrew <Derek.Andrew at
usask.ca> wrote:
> >>>
> >>> What is the alternative to the dial plan? Is everyone talking
about
> >>> getting rid of the statements like:
> >>> exten => s,1,
> >>>
> >>> what is the alternative?
> >>
> >>
> >> Remote applications based on APIs like ARI. This is the start of
the
> >> discussion, and please remember that nothing has been decided or
even
> >> presented as a robust plan yet. This is brain-storming.
> >>
> >>
> >> We?re not at the start of the ?discussion? to deprecate the dial
plan.
> The
> >> start of the ?discussion? began when some developers decided to
try
> standing
> >> Asterisk on its head by adding ?asynchronous AGI.? Evidently,
that was
> good
> >> so then they continued the ?discussion? by adding ARI/Stasis. Now
the
> >> ?discussion? is in full career as ARI/Stasis has metastasized
beyond its
> >> original scope to encompass all of Asterisk. None of said
?discussion?
> ever
> >> happened on the lists nor was the broader Asterisk community
involved
> as far
> >> as I can determine. A parallel ?discussion? was started by a shill
at
> >> AstiCon this year to begin to get the ?vast unwashed? onboard with
> >> ARI/Stasis, that is, so that Matt could come back from AstiCon
claiming
> that
> >> the broader Asterisk community is in agreement that ARI/Stasis is
the
> future
> >> of Asterisk and that the dial plan can be deprecated. The
inevitable
> result
> >> of these parallel paths is a completely predictable train wreck
when the
> >> developers designing features that users don?t want crash into
users who
> >> have been using Asterisk as originally designed.
> >>
> >> Additionally, note that the original proposal was to deprecate
AMI/AGI
> in
> >> favour of ARI once it is feature complete with those protocols; an
> entirely
> >> lesser change than the removal of the dialplan in its entirety.
> >>
> >>
> >> So you're saying that deprecating the dial plan is not on the
table? How
> >> then do you explain statements like this: "Leif: we're in
a transition,
> >> moving from dialplan model to external control model. Probably
need
> >> external application to be built for us to move completely away
from
> >> AMI/AGI.? or this "Paul: take away apps, and whatever is in
the core is
> >> what we should care about.?
> >>
> >
> > Paul:
> >
> > This is a notice that you are in violation of the Asterisk community
> > code of conduct:
> >
> >
>
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Community+Code+of+Conduct
> >
> > You have repeatedly insulted members of the Asterisk community using
> > derogatory language that is inappropriate for this mailing list. This
> > creates a hostile atmosphere that makes it difficult for productive
> > communication to occur, which is the lifeblood of this open source
> > project. Members of an open source community should not feel like they
> > are under attack merely for expressing an opinion. While we value the
> > opinions you bring to the discussion, your tone and choice of language
> > is completely inappropriate and will not be tolerated.
> >
> > If you continue to use inflammatory language and rhetoric, you will be
> > banned from participation in the Asterisk project.
> >
> > Matt
> >
> > --
> > Matthew Jordan
> > Digium, Inc. | Engineering Manager
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> > Check us out at: http://digium.com & http://asterisk.org
>
>
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 30 Oct 2014 17:20:54 -0400
> From: Ben Klang <bklang at mojolingo.com>
> To: Asterisk Developers Mailing List <asterisk-dev at
lists.digium.com>
> Cc: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda
> item Deprecate AMI/AGI (Ben Klang)
> Message-ID: <0D421FE1-918C-4FFC-B397-567FEA12EBF4 at mojolingo.com>
> Content-Type: text/plain; charset="windows-1252"
>
> Il giorno Oct 30, 2014, alle ore 4:57 PM, Paul Albrecht <
> palbrecht at glccom.com> ha scritto:
> >
> > On Oct 29, 2014, at 2:45 PM, Ben Klang <bklang at mojolingo.com
<mailto:
> bklang at mojolingo.com>> wrote:
> >
> >>
> >>> On 10/28/2014 06:03 PM, Ben Langfeld wrote:
> >>>> On 28 October 2014 19:47, Derek Andrew <Derek.Andrew at
usask.ca
> <mailto:Derek.Andrew at usask.ca>> wrote:
> >>>> What is the alternative to the dial plan? Is everyone
talking about
> getting rid of the statements like:
> >>>> exten => s,1,
> >>>>
> >>>> what is the alternative?
> >>>>
> >>>> Remote applications based on APIs like ARI. This is the
start of the
> discussion, and please remember that nothing has been decided or even
> presented as a robust plan yet. This is brain-storming.
> >>>>
> >>>> Additionally, note that the original proposal was to
deprecate
> AMI/AGI in favour of ARI once it is feature complete with those protocols;
> an entirely lesser change than the removal of the dialplan in its entirety.
> >>>>
> >>
> >> Since this thread has my name on it, I guess it?s past time that I
> explain my motivation for making the suggestion, and try to restore some of
> the context that was present in the discussion at AstriDevCon.
> >>
> >> Before I jump into the details of my proposal, I?d like to clarify
> terms...
> >>
> >
> > It?s intellectually dishonest to redefine the terms of an argument to
> presuppose your own conclusion. If you don?t intend to use the term
> ?deprecate? as it is commonly understood by software developers and users
> than you should avoid the use of the term ?deprecate? so that others
> clearly understand your argument. If you really mean ?deprecate? as
> commonly understood by software developers and users then you should be
> prepared to defend that proposition.
>
> I had thought that the term ?deprecate? was already understood to be the
> definition I gave, but earlier posts on the mailing list seemed to indicate
> confusion. My definition mirrors the Wikipedia definition:
> https://en.wikipedia.org/wiki/Deprecation <
> https://en.wikipedia.org/wiki/Deprecation>. Perhaps I just should have
> linked to that originally, as their explanation is even better than my own.
>
> In any event, what we are talking about is the deprecation as I defined
> it. If you prefer another word for it, I?m fine with that too. I just want
> to be clear that my proposal is to discourage use of AMI/AGI in new
> projects, but not to immediately remove it.
>
> >
> >> Now, on to what I originally proposed...
> >>
> >
> > It?s clear from the title of the agenda item what was proposed. You
> proposed deprecating AMI/AGI and that entails deprecating the dial plan.
> The fact that deprecating the dial plan is now on the table is a direct
> consequence of your proposal. This is reflected in both comments made at
> AstiCon and Matt?s summary of Astricon on the development list. You can?t
> have it both ways. You want to deprecate dial plan or not. Which is it?
>
> Actually, AMI/AGI and Dialplan are separate. You can disable AMI and you
> can unload res_agi.so. Dialplan/extensions.conf continue to work just
> fine. Certainly AMI/AGI make use of Dialplan, but deprecating AMI/AGI
> doesn?t mean you have to deprecate Dialplan.
>
> >
> >> It is my opinion that while AGI and AMI are probably individually
> fixable, doing so would cause backward-incompatible changes?
> >
> > Deprecating the dial plan and AGI/AMI is incompatible going forward.
> What is supposed to happen? Are users supposed to throw away there
> applications whenever ARI/Stasis is feature complete? Is ARI/Stasis really
> any easier to use than the dial plan? Are we all supposed to use
Adhearsion?
> >
>
> You?re certainly welcome to use Adhearsion :) For what it?s worth,
> Adhearsion will continue to support AMI/AGI because we have to until ARI is
> feature-complete. For Adhearsion users, the transition to ARI should be
> seamless because that?s one of the things that the framework promises: to
> paper over the idiosyncrasies of the underlying protocols.
>
> If you don?t want to use Adhearsion, I?d recommend you look at ARI for
> developing new projects. There are libraries in many languages that make
> it easy to use. It?s got a great start and will only improve as people
> continue to use it and develop additional features. Today, it is not yet a
> replacement for AMI/AGI, but I?m very optimistic that it will be in the
> near future.
>
> I suspect that I?m not convincing to you, and that you want to continue
> using AMI/AGI. That?s fine, I?m not telling you to throw out any code. I
> think Asterisk?s historical policy toward backward compatibility and
> removing features speaks for itself. Rather than continue to debate the
> semantics of my proposal, I?d like to continue the discussion on how we can
> improve ARI and improve the state of the world for all Asterisk developers
> in the years to come.
>
> /BAK/
> --
> Ben Klang
> Principal/Technology Strategist, Mojo Lingo
> bklang at mojolingo.com <mailto:bklang at mojolingo.com>
> +1.404.475.4841
>
> Mojo Lingo -- Voice applications that work like magic
> http://mojolingo.com <http://mojolingo.com/>
> Twitter: @MojoLingo
>
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
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> ------------------------------
>
> Message: 10
> Date: Thu, 30 Oct 2014 17:01:42 -0500 (CDT)
> From: Matt Hoskins <matt.hoskins at npgco.com>
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] MWI publish VIA pjsip for non sip channels
> Message-ID: <9a3428ea.000019bc.00000006 at IT10015vm.npgco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Before I go down a rabbit hole, does the mwi publish/subscription work for
> non SIP phones?
>
> For instance, I have a single voicemail server, connected to multiple
> asterisk boxes via SIP. On each of those servers, there are a mix of SIP
> and SCCP phones attached. Currently, I'm using res_xmpp to distribute
mwi
> from the voicemail server to the endpoint servers. Would this type of
> setup work with PJSIP? The net effect here is that I want to get away
> from res_xmpp, if possible.
>
> Matt Hoskins | NPG Corp | Systems Architect
>
>
>
> ------------------------------
>
> Message: 11
> Date: Thu, 30 Oct 2014 19:08:52 -0300
> From: Joshua Colp <jcolp at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
> channels
> Message-ID: <5452B6F4.3040302 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Matt Hoskins wrote:
> > Before I go down a rabbit hole, does the mwi publish/subscription work
> for
> > non SIP phones?
>
> Yes. SIP is simply used as the transport mechanism. It works pretty much
> the same as res_xmpp except without needing an XMPP server.
>
> Cheers,
>
> --
> 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
>
>
>
> ------------------------------
>
> Message: 12
> Date: Thu, 30 Oct 2014 17:16:30 -0500 (CDT)
> From: Matt Hoskins <matt.hoskins at npgco.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
> channels
> Message-ID: <00d324ca.000019bc.00000008 at IT10015vm.npgco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Of course, I left out a detail that may (or may not change) the answer.
> I'm using the external chan-sccp-b sccp module, not the chan_skinny
> bundled with asterisk.
>
> Matt Hoskins | NPG Corp | Systems Architect
>
> 816.749.2815 (Internal: ext. 10015)
>
>
>
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
Colp
> Sent: Thursday, October 30, 2014 5:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
>
> Matt Hoskins wrote:
> > Before I go down a rabbit hole, does the mwi publish/subscription work
> > for non SIP phones?
>
> Yes. SIP is simply used as the transport mechanism. It works pretty much
> the same as res_xmpp except without needing an XMPP server.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29
> t&r=YmFzZQ%3D%3D &
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5
> vcmc%3D&r=YmFzZQ%3D%3D
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hcGktZGlnaXR
> hbC5jb20%3D&r=YmFzZQ%3D%3D -- New to Asterisk? Join us for a live
> introductory webinar every Thurs:
>
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5
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> --
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>
> ------------------------------
>
> Message: 13
> Date: Thu, 30 Oct 2014 19:18:38 -0300
> From: Joshua Colp <jcolp at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
> channels
> Message-ID: <5452B93E.5050107 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Matt Hoskins wrote:
> > Of course, I left out a detail that may (or may not change) the
answer.
> > I'm using the external chan-sccp-b sccp module, not the
chan_skinny
> > bundled with asterisk.
>
> Still doesn't matter. Provided it works with res_xmpp it'll work
with
> the new SIP method.
>
> Cheers,
>
> --
> 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
>
>
>
> ------------------------------
>
> Message: 14
> Date: Thu, 30 Oct 2014 17:20:59 -0500 (CDT)
> From: Matt Hoskins <matt.hoskins at npgco.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
> channels
> Message-ID: <97ddc0e6.000019bc.0000000b at IT10015vm.npgco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Awesome - Thanks for the quick replies. I'm sure I could have
> tried-and-see but with going from Asterisk 11 to 13, there'd be so many
> things changing - it helps to know from the outset.
>
> Thanks again.
>
> Matt Hoskins | NPG Corp | Systems Architect
>
>
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
Colp
> Sent: Thursday, October 30, 2014 5:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip channels
>
> Matt Hoskins wrote:
> > Of course, I left out a detail that may (or may not change) the
answer.
> > I'm using the external chan-sccp-b sccp module, not the
chan_skinny
> > bundled with asterisk.
>
> Still doesn't matter. Provided it works with res_xmpp it'll work
with the
> new SIP method.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29
> t&r=YmFzZQ%3D%3D &
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5
> vcmc%3D&r=YmFzZQ%3D%3D
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hcGktZGlnaXR
> hbC5jb20%3D&r=YmFzZQ%3D%3D -- New to Asterisk? Join us for a live
> introductory webinar every Thurs:
>
> http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5
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> asterisk-users mailing list
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>
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>
> ------------------------------
>
> Message: 15
> Date: Thu, 30 Oct 2014 19:35:25 -0500
> From: Matthew Jordan <mjordan at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Register multiple phones to a single AOR
> with PJSIP
> Message-ID:
> <
> CAN2PU+4s8ya-KBwnNOTuYJNzsJJ6g0rUGQXede+syKsO2_J3Jw at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Thu, Oct 30, 2014 at 2:47 PM, Scott Griepentrog <
> sgriepentrog at digium.com>
> wrote:
>
> > ?You need to change your dialplan to use the PJSIP_DIAL_CONTACTS
function
> > like this:
> >
> > exten => _X.,1,Dial(${PJSIP_DIAL_CONTACTS(200)},30)?
> >
> > It expands to the list of contacts, separated by &, so that the
contacts
> > are dialed at the same time.
> >
> > The documentation page you reference should be updated to include that
> > detail.
> >
>
> How about this page instead:
>
> https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
>
> Matt
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
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> ------------------------------
>
> Message: 16
> Date: Thu, 30 Oct 2014 20:32:05 -0500
> From: Matthew Jordan <mjordan at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Paul Albrecht
> Message-ID:
> <
> CAN2PU+6daVuN0mMQv6Q--BU64wbUCAZ7FpvQzg1kAsa7Td6y1Q at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> Open source projects survive on freedom of communication. Such
> projects are diminished when a community member can no longer
> participate, as the project no longer benefits from their opinions and
> insight. However, one of the few things worse than this loss of
> participation is to have a hostile environment where people are afraid
> to voice opinions. If we cannot discuss ideas ? even radical ones ?
> openly and freely without fear of recrimination, then we are dead as
> an open source project.
>
> In the Asterisk Developer Community, we often have disagreements about
> technical decisions and the direction of the project. Sometimes those
> disagreements are quite passionate. That's a good thing. We are all
> only human, and sometimes we all make mistakes. The only way we can
> keep the project moving forward in the best manner possible is if we
> allow for disagreements and conversation.
>
> However, there is an acceptable way to disagree with each other, and
> an unacceptable way. Repeatedly denigrating others in the community,
> refusing to listen to their opinions and explanations, and continuing
> to attack those who disagree with you creates a hostile environment
> where productive conversation is impossible. Paul Albrecht repeatedly
> chose to communicate in this fashion and refused to change his
> behaviour.
>
> In light of his recent e-mails, which came after I privately warned
> Paul that he was in violation of the community code of conduct [1], I
> felt Paul had no desire to change his rhetoric or his language and
> have thus removed him from the Asterisk project e-mail lists and other
> project resources.
>
> This was not a decision taken lightly. This is the first time I've had
> to do this as the lead of the Asterisk project, and I sincerely hope
> it is the last.
>
> I'm sure this decision will not sit easily with everyone. I understand
> that, and my desire is not to create a place where passionate opinions
> cannot be expressed. What I do hope, however, is that we can have a
> community where we all have a basic level of respect for one another,
> such that when we do disagree, we can do so without resorting to
> insults and derogatory comments.
>
> To quote Jeff Atwood [2]:
>
> ?At the risk of sounding aspirational, here's one thing I know to be
> true, and I advise every community to take to heart: I expect you to
> act like a group of friends who care about each other, no matter how
> dumb some of us might be, no matter what political opinions some of us
> hold, no matter what things some of us like or dislike.?
>
> Hopefully, we can move past this as a community and continue to
> support and improve the Asterisk project.
>
> Matt
>
> [1]
>
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Community+Code+of+Conduct
> [2] http://blog.codinghorror.com/what-if-we-could-weaponize-empathy/
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>
>
> ------------------------------
>
> _______________________________________________
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>
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> End of asterisk-users Digest, Vol 123, Issue 38
> ***********************************************
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Apparently Analagous Threads
- Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?
- Asterisk 13 - sorcery realtime for pjsip publish objects
- Asterisk 13 - sorcery realtime for pjsip publish objects
- Asterisk 13 - sorcery realtime for pjsip publish objects
- Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?
Wisdom of the Ancients
