[asterisk-users] Paul Albrecht

2014-10-30 Thread Matthew Jordan 
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/

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Re: [asterisk-users] Register multiple phones to a single AOR with PJSIP

2014-10-30 Thread Matthew Jordan 
On Thu, Oct 30, 2014 at 2:47 PM, Scott Griepentrog 
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

-- 
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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins 
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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,

--
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Digium, Inc. | Senior Software Developer
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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Joshua Colp 

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

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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins 
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
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Re: [asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Joshua Colp 

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,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] MWI publish VIA pjsip for non sip channels

2014-10-30 Thread Matt Hoskins 
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

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Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

2014-10-30 Thread Ben Klang 
Il giorno Oct 30, 2014, alle ore 4:57 PM, Paul Albrecht  
ha scritto:
> 
> On Oct 29, 2014, at 2:45 PM, Ben Klang  > wrote:
> 
>> 
>>> On 10/28/2014 06:03 PM, Ben Langfeld wrote:
 On 28 October 2014 19:47, Derek Andrew >>> > 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 
.  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
bkl...@mojolingo.com 
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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

2014-10-30 Thread Paul Albrecht 

On Oct 29, 2014, at 4:26 PM, Matthew Jordan  wrote:

> On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht  wrote:
>> 
>> On Oct 28, 2014, at 5:03 PM, Ben Langfeld  wrote:
>> 
>> On 28 October 2014 19:47, Derek Andrew  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


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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

2014-10-30 Thread Paul Albrecht 

On Oct 29, 2014, at 2:45 PM, Ben Klang  wrote:

> 
>> On 10/28/2014 06:03 PM, Ben Langfeld wrote:
>>> On 28 October 2014 19:47, Derek Andrew  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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Re: [asterisk-users] make asterisk do something when an outgoing call is picked up

2014-10-30 Thread John Kiniston 
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 => _NXX,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  wrote:

> Thorsten Göllner  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.
>
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Re: [asterisk-users] ${HASH(SIP_CAUSE,)}

2014-10-30 Thread Paul Belanger 
On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens  wrote:
> Hello,
>
> I read on the wiki :
>
> Asterisk 1.8 will allow to read SIP response codes in the dialplan via
> ${HASH(SIP_CAUSE,)}. 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/SipAT01-0015", "sip cause = ") in new stack
> [Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
> NoOp("SIP/SipAT01-0015", "sip cause = ") in new stack
>
>
> Can anyone tell me how this should be used ?
>
sip.conf: storesipcause=yes


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Re: [asterisk-users] Register multiple phones to a single AOR with PJSIP

2014-10-30 Thread Scott Griepentrog 
​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 
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
> use0 of inf
>  InAuth:  101/101
> Aor:  1013
>   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 México S.A. de C.V.
> Carlos Chávez
> +52 (55)9116-91161
>
>
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[asterisk-users] PlayTones not working

2014-10-30 Thread Henry Fernandes 
I¹m trying to use Playtones to have a tone played periodically throughout
phone calls.  Unfortunately, I can¹t 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-0032",
"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-0032",
"SIP/19525553312@proxy01,,gU(record_call_id)") in new stack



I¹ve checked the debug log and I can¹t see any related errors or warning
beyond the one above.

-H


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Re: [asterisk-users] make asterisk do something when an outgoing call is picked up

2014-10-30 Thread lee 
Thorsten Göllner  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.

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[asterisk-users] Register multiple phones to a single AOR with PJSIP

2014-10-30 Thread Carlos Chavez 
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 
use0 of inf

 InAuth:  101/101
Aor:  1013
  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?


--
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Carlos Chávez
+52 (55)9116-91161


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Re: [asterisk-users] ${HASH(SIP_CAUSE,)}

2014-10-30 Thread Gareth Blades 


On 30/10/14 13:52, Jonas Kellens wrote:

Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via 
*${HASH(SIP_CAUSE,)}*. 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/SipAT01-0015", "sip cause = ") in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6] 
NoOp("SIP/SipAT01-0015", "sip cause = ") in new stack





Take a look at my blog entry about it :-

http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html
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[asterisk-users] ${HASH(SIP_CAUSE,)}

2014-10-30 Thread Jonas Kellens 

Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via 
*${HASH(SIP_CAUSE,)}*. 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/SipAT01-0015", "sip cause = ") in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6] 
NoOp("SIP/SipAT01-0015", "sip cause = ") in new stack



Can anyone tell me how this should be used ?


Kind regards,

Jonas.
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[asterisk-users] Asterisk registration with Dialogic HMP.

2014-10-30 Thread Shahnaz Ali 
Hi,

I am very new to Asterisk world.  This is what we are trying to achieve.

We have Asterisk Server in our lab with Ver. 11.2.1 - FreePBX. 

We have Dialogic HMP application, which we want to get registered with 
Asterisk. After registration we would like to use HMP as incoming or outgoing 
IVR. 

I need to know what has to be done at Asterisk side to make this happen.

We created normal extension, and tried to register extension through HMP and 
its failing. Asterisk is replying back with unauthorised status. The way SIP 
clients connect to Asterisk, is not the way HMP should register to Asterisk? 

BR,
Shahnaz Ali.




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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-30 Thread Yaron Nachum 
Hello everyone,
I have opened a ticket number - ASTERISK-24471
.

I have attached the backtrace of the core file. The backtrace was taken on
the server running 12.6.1.

If you need any information please get back to me.

Thank you.
Yaron.

On Thu, Oct 30, 2014 at 8:43 AM, Yaron Nachum 
wrote:

> No,
> I went over all my scripts.
>
> Thanks for the help.
>
> Yaron
>
> On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
>> On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum 
>> wrote:
>> > Mathew,
>> > When I run 'ps -ef|grep asterisk' the following processes are displayed:
>> > root  6861 1  0 Aug27 ?00:00:00 /bin/sh
>> > /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
>> > /ivr/app/asterisk/etc/asterisk/asterisk.conf
>> > asterisk  8062  6861  3 Oct27 ?00:44:56
>> > /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
>> > /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
>> > root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
>> > asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] 
>> > asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] 
>> >
>> > also when I run top the same amount of zombie processes are displayed:
>> > Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
>> >
>> > Regarding the AGI - we are using AGI in order to run php scripts for
>> > external logic. I have printed the PIDs of the php scripts and none of
>> them
>> > are related to the PID's of those zombie processes.
>> > Do you have any idea how to find out what are these processes?
>> > Yaron.
>> >
>> Are you doing anything like:
>>
>> # asterisk -rx 'core show channels'
>>
>> via an external process?
>>
>> --
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>> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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