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[asterisk-users] asterisk-users Digest, Vol 123, Issue 38

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ntshshrm13 at gmail.com
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Posted: Fri Oct 31, 2014 8:55 am Post subject: [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@lists.digium.com (asterisk-users-request@lists.digium.com)> wrote:

Quote:

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When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."

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@telecomabmex.com (cursor@telecomabmex.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: [asterisk-users] Register multiple phones to a single AOR
with PJSIP
Message-ID: <54528F01.4080700@telecomabmex.com (54528F01.4080700@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@192.168.2.193:5063 Avail 178.681
Contact: 101/sip:101@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@yagibdah.de (lee@yagibdah.de)>
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] make asterisk do something when an
outgoing call is picked up
Message-ID: <87k33h8itw.fsf@gulltop.yagibdah.de (87k33h8itw.fsf@gulltop.yagibdah.de)>
Content-Type: text/plain; charset=utf-8

Thorsten G?llner <tg@ovm-group.com (tg@ovm-group.com)> writes:

Quote:

Am 26.10.2014 00:43, schrieb lee:

Quote:

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@usinternet.com (henry@usinternet.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: [asterisk-users] PlayTones not working
Message-ID: <D077F044.EFF5%henry@usinternet.com ([email]D077F044.EFF5%25henry@usinternet.com[/email])>
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@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@proxy-dial:3] Dial("SIP/testphone-00000032",
"SIP/19525553312@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@digium.com (sgriepentrog@digium.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] Register multiple phones to a single AOR
with PJSIP
Message-ID:
<CACrpESYeXvxzF=W-CAoa7d2pEZrtqXnK9Yc3T6ixG9EV1U2msA@mail.gmail.com (W-CAoa7d2pEZrtqXnK9Yc3T6ixG9EV1U2msA@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@telecomabmex.com (cursor@telecomabmex.com)>
wrote:

Quote:

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@192.168.2.193:5063 Avail 178.681
Contact: 101/sip:101@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:
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To UNSUBSCRIBE or update options visit:
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--
[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@polybeacon.com (paul.belanger@polybeacon.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}
Message-ID:
<CALLKq0TN0hkCgbs3TX3CJo=ysY+m+9O0zonEEpfdQwdujMY9dQ@mail.gmail.com ([email]ysY%2Bm%2B9O0zonEEpfdQwdujMY9dQ@mail.gmail.com[/email])>
Content-Type: text/plain; charset=UTF-8

On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

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@pbx-routing:5]
NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack
[Oct 30 14:48:03] -- Executing [h@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@polybeacon.com (paul.belanger@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@gmail.com (johnkiniston@gmail.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] make asterisk do something when an
outgoing call is picked up
Message-ID:
<CAFJQOGc93qSdXmvese2y+iNsJQbJgrEXRbNCe1d=mf_TXuzACg@mail.gmail.com (mf_TXuzACg@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@yagibdah.de (lee@yagibdah.de)> wrote:

Quote:

Thorsten G?llner <tg@ovm-group.com (tg@ovm-group.com)> writes:

Quote:

Am 26.10.2014 00:43, schrieb lee:

Quote:

--
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@glccom.com (palbrecht@glccom.com)>
To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com (asterisk-dev@lists.digium.com)>,
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate
AMI/AGI (Ben Klang)
Message-ID: <8E422546-5698-4A85-90E0-32E46FE6EEDF@glccom.com (8E422546-5698-4A85-90E0-32E46FE6EEDF@glccom.com)>
Content-Type: text/plain; charset="windows-1252"

On Oct 29, 2014, at 2:45 PM, Ben Klang <bklang@mojolingo.com (bklang@mojolingo.com)> wrote:

Quote:

On 10/28/2014 06:03 PM, Ben Langfeld wrote:

Quote:

On 28 October 2014 19:47, Derek Andrew <Derek.Andrew@usask.ca (Derek.Andrew@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.

Quote:

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?

Quote:

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@glccom.com (palbrecht@glccom.com)>
To: Matthew Jordan <mjordan@digium.com (mjordan@digium.com)>, Asterisk Developers Mailing
List <asterisk-dev@lists.digium.com (asterisk-dev@lists.digium.com)>,
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate
AMI/AGI(Ben Klang)
Message-ID: <66661B22-2F91-49F8-8CAA-8383C443BE21@glccom.com (66661B22-2F91-49F8-8CAA-8383C443BE21@glccom.com)>
Content-Type: text/plain; charset="windows-1252"

On Oct 29, 2014, at 4:26 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote:

Quote:

On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht <palbrecht@glccom.com (palbrecht@glccom.com)> wrote:

Quote:

On Oct 28, 2014, at 5:03 PM, Ben Langfeld <ben@langfeld.me (ben@langfeld.me)> wrote:

On 28 October 2014 19:47, Derek Andrew <Derek.Andrew@usask.ca (Derek.Andrew@usask.ca)> wrote:

Quote:

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@mojolingo.com (bklang@mojolingo.com)>
To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com (asterisk-dev@lists.digium.com)>
Cc: asterisk-users@lists.digium.com (asterisk-users@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@mojolingo.com (0D421FE1-918C-4FFC-B397-567FEA12EBF4@mojolingo.com)>
Content-Type: text/plain; charset="windows-1252"

Il giorno Oct 30, 2014, alle ore 4:57 PM, Paul Albrecht <palbrecht@glccom.com (palbrecht@glccom.com)> ha scritto:

Quote:

On Oct 29, 2014, at 2:45 PM, Ben Klang <bklang@mojolingo.com (bklang@mojolingo.com) <mailto:bklang@mojolingo.com (bklang@mojolingo.com)>> wrote:

Quote:

On 10/28/2014 06:03 PM, Ben Langfeld wrote:

Quote:

On 28 October 2014 19:47, Derek Andrew <Derek.Andrew@usask.ca (Derek.Andrew@usask.ca) <mailto:Derek.Andrew@usask.ca (Derek.Andrew@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.

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.

Quote:

Now, on to what I originally proposed...

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.

Quote:

It is my opinion that while AGI and AMI are probably individually fixable, doing so would cause backward-incompatible changes?

You?re certainly welcome to use Adhearsion Smile

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@mojolingo.com (bklang@mojolingo.com) <mailto:bklang@mojolingo.com (bklang@mojolingo.com)>
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com <http://mojolingo.com/>
Twitter: @MojoLingo

Quote:

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Message: 10
Date: Thu, 30 Oct 2014 17:01:42 -0500 (CDT)
From: Matt Hoskins <matt.hoskins@npgco.com (matt.hoskins@npgco.com)>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: [asterisk-users] MWI publish VIA pjsip for non sip channels
Message-ID: <9a3428ea.000019bc.00000006@IT10015vm.npgco.com (9a3428ea.000019bc.00000006@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@digium.com (jcolp@digium.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
channels
Message-ID: <5452B6F4.3040302@digium.com (5452B6F4.3040302@digium.com)>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Matt Hoskins wrote:

Quote:

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@npgco.com (matt.hoskins@npgco.com)>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
channels
Message-ID: <00d324ca.000019bc.00000008@IT10015vm.npgco.com (00d324ca.000019bc.00000008@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@lists.digium.com (asterisk-users-bounces@lists.digium.com)
[mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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:

Quote:

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

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introductory webinar every Thurs:

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Message: 13
Date: Thu, 30 Oct 2014 19:18:38 -0300
From: Joshua Colp <jcolp@digium.com (jcolp@digium.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
channels
Message-ID: <5452B93E.5050107@digium.com (5452B93E.5050107@digium.com)>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Matt Hoskins wrote:

Quote:

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@npgco.com (matt.hoskins@npgco.com)>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] MWI publish VIA pjsip for non sip
channels
Message-ID: <97ddc0e6.000019bc.0000000b@IT10015vm.npgco.com (97ddc0e6.000019bc.0000000b@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@lists.digium.com (asterisk-users-bounces@lists.digium.com)
[mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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:

Quote:

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

--
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-- Bandwidth and Colocation Provided by
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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
vcmcvaGVsbG8%3D&r=YmFzZQ%3D%3D

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Message: 15
Date: Thu, 30 Oct 2014 19:35:25 -0500
From: Matthew Jordan <mjordan@digium.com (mjordan@digium.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: Re: [asterisk-users] Register multiple phones to a single AOR
with PJSIP
Message-ID:
<CAN2PU+4s8ya-KBwnNOTuYJNzsJJ6g0rUGQXede+syKsO2_J3Jw@mail.gmail.com ([email]CAN2PU%2B4s8ya-KBwnNOTuYJNzsJJ6g0rUGQXede%2BsyKsO2_J3Jw@mail.gmail.com[/email])>
Content-Type: text/plain; charset="utf-8"

On Thu, Oct 30, 2014 at 2:47 PM, Scott Griepentrog <sgriepentrog@digium.com (sgriepentrog@digium.com)>
wrote:

Quote:

?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@digium.com (mjordan@digium.com)>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>
Subject: [asterisk-users] Paul Albrecht
Message-ID:
<CAN2PU+6daVuN0mMQv6Q--BU64wbUCAZ7FpvQzg1kAsa7Td6y1Q@mail.gmail.com ([email]CAN2PU%2B6daVuN0mMQv6Q--BU64wbUCAZ7FpvQzg1kAsa7Td6y1Q@mail.gmail.com[/email])>
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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