Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Joshua C. Colp]

On Sat, Jul 3, 2021 at 3:37 PM Jonas Kellens 
wrote:

> Hello Joshua
>
>
> could it be a bug ?
>
> I am using asterisk-certified-13.21-cert6
>

There's been no other reports of issues, but it could be. The 13 branch,
however, only receives bug fixes. Additionally the chan_sip module is
community supported meaning if there is a bug then there is no timeframe on
when it would get looked at, or if it would. You'd also have to provide a
complete SIP trace and debug log.

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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Jonas Kellens]

Hello Joshua


could it be a bug ?

I am using asterisk-certified-13.21-cert6



Kind regards.

J.


Op 01-07-21 om 20:20 schreef Joshua C. Colp:
On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens > wrote:


Hello Joshua

this is the SIP REGISTER at 11:10:45

REGISTER sip:tstv7.domain.tld SIP/2.0
Via: SIP/2.0/UDP
192.168.1.18:5060;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport
From: 
;tag=3630891428
To: 

Call-ID: 3270725701@192_168_1_18
CSeq: 452 REGISTER
Contact: 

Authorization: Digest username="testacc7700105",
realm="tstv7.domain.tld", algorithm=MD5,
uri="sip:tstv7.domain.tld", nonce="42a70292",
response="e2945dacd2d95b47a4801b2471070702"
Max-Forwards: 70
User-Agent: C610 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE,
NOTIFY
Content-Length: 0

I see here "180".


I also see in the SIP debug traffic that a SIP REGISTER occurs
every 180 seconds, which is what is set in the SIP client.


This "Expired" notice occured only once at 11:20:55. How come this
happens only once ?

And why should there *ever* be an "Expired" if there is a SIP
REGISTER every 180 seconds ?!


If what you are saying is correct, then I do not know. The chan_sip 
module decided that it should be expired. Why that is, I do not know.


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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www.asterisk.org 


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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Joshua C. Colp]

On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens 
wrote:

> Hello Joshua
>
> this is the SIP REGISTER at 11:10:45
>
> REGISTER sip:tstv7.domain.tld SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.18:5060
> ;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport
> From: 
> ;tag=3630891428
> To: 
> 
> Call-ID: 3270725701@192_168_1_18
> CSeq: 452 REGISTER
> Contact: 
> 
> Authorization: Digest username="testacc7700105", realm="tstv7.domain.tld",
> algorithm=MD5, uri="sip:tstv7.domain.tld", nonce="42a70292",
> response="e2945dacd2d95b47a4801b2471070702"
> Max-Forwards: 70
> User-Agent: C610 IP/42.075.00.000.000
> Expires: 180
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
> I see here "180".
>
>
> I also see in the SIP debug traffic that a SIP REGISTER occurs every 180
> seconds, which is what is set in the SIP client.
>
>
> This "Expired" notice occured only once at 11:20:55. How come this happens
> only once ?
>
> And why should there *ever* be an "Expired" if there is a SIP REGISTER
> every 180 seconds ?!
>

If what you are saying is correct, then I do not know. The chan_sip module
decided that it should be expired. Why that is, I do not know.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Jonas Kellens]

Hello Joshua

this is the SIP REGISTER at 11:10:45

REGISTER sip:tstv7.domain.tld SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.18:5060;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport

From: ;tag=3630891428
To: 
Call-ID: 3270725701@192_168_1_18
CSeq: 452 REGISTER
Contact: 
Authorization: Digest username="testacc7700105", 
realm="tstv7.domain.tld", algorithm=MD5, uri="sip:tstv7.domain.tld", 
nonce="42a70292", response="e2945dacd2d95b47a4801b2471070702"

Max-Forwards: 70
User-Agent: C610 IP/42.075.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

I see here "180".


I also see in the SIP debug traffic that a SIP REGISTER occurs every 180 
seconds, which is what is set in the SIP client.



This "Expired" notice occured only once at 11:20:55. How come this 
happens only once ?


And why should there *ever* be an "Expired" if there is a SIP REGISTER 
every 180 seconds ?!




Kind regards.



Op 01-07-21 om 17:41 schreef Joshua C. Colp:
On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens 
mailto:jonas.kell...@telenet.be>> wrote:


Hello Joshua


these are the 2 previous events on the Manager interface :

2021-06-30 11:10:45
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Registered
    [6] => Address: my.lo.cal.ip:55014
)



2021-06-30 11:10:48
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Reachable
    [6] =>
)


So there is a re-register at 11:10:45


How do you explain the "Expired" 10 minutes later ??


Without the actual SIP REGISTER traffic to show how long the 
registration was for, I can't really say anything further.


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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www.asterisk.org 


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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Joshua C. Colp]

On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens 
wrote:

> Hello Joshua
>
>
> these are the 2 previous events on the Manager interface :
>
> 2021-06-30 11:10:45
> Array
> (
> [0] => Event: PeerStatus
> [1] => Privilege: system,all
> [2] => SystemName: tstv7
> [3] => ChannelType: SIP
> [4] => Peer: SIP/testacc7700921
> [5] => PeerStatus: Registered
> [6] => Address: my.lo.cal.ip:55014
> )
>
>
>
> 2021-06-30 11:10:48
> Array
> (
> [0] => Event: PeerStatus
> [1] => Privilege: system,all
> [2] => SystemName: tstv7
> [3] => ChannelType: SIP
> [4] => Peer: SIP/testacc7700921
> [5] => PeerStatus: Reachable
> [6] =>
> )
>
>
> So there is a re-register at 11:10:45
>
>
> How do you explain the "Expired" 10 minutes later ??
>

Without the actual SIP REGISTER traffic to show how long the registration
was for, I can't really say anything further.

-- 
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Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Jonas Kellens]

Hello Joshua


these are the 2 previous events on the Manager interface :

2021-06-30 11:10:45
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Registered
    [6] => Address: my.lo.cal.ip:55014
)



2021-06-30 11:10:48
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Reachable
    [6] =>
)


So there is a re-register at 11:10:45


How do you explain the "Expired" 10 minutes later ??





Op 30-06-21 om 20:32 schreef Joshua C. Colp:
On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens 
mailto:jonas.kell...@telenet.be>> wrote:


Hello


I see the following event from the Asterisk Manager :

2021-06-30 11:20:55
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Unregistered
    [6] => Cause: Expired
)


The cause is in this message, the registration expired. A 
re-registration did not occur before the registration expiration so it 
expired and was unregistered.


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 





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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Joshua C. Colp]

On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens 
wrote:

> Hello
>
>
> I see the following event from the Asterisk Manager :
>
> 2021-06-30 11:20:55
> Array
> (
> [0] => Event: PeerStatus
> [1] => Privilege: system,all
> [2] => SystemName: tstv7
> [3] => ChannelType: SIP
> [4] => Peer: SIP/testacc7700921
> [5] => PeerStatus: Unregistered
> [6] => Cause: Expired
> )
>
>
The cause is in this message, the registration expired. A re-registration
did not occur before the registration expiration so it expired and was
unregistered.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Asterisk manager : core show hints

[Joshua C. Colp]

On Thu, Aug 22, 2019, at 5:33 AM, Jonas Kellens wrote:
> Hello 
> 
> I see on the CLI :
> 
> tst*CLI> core show hints
>  -= Registered Asterisk Dial Plan Hints =-
>  50@blf : SIP/testacc7 State:Idle Watchers 3
>  6001@blf : Custom:q-6001 State:Idle Watchers 1
>  5@blf : SIP/testacc6 State:Unavailable Watchers 1
> 
> 
> 
> Is there a way to get this info through the manager API ?

There is an ExtensionStateList action[1]. You can also get individual extension 
state, or device state, or you could use the AMI action which allows you to 
execute a CLI command. I'd suggest looking through the manager action 
documentation to find the action that does precisely what you want if 
ExtensionStateList isn't it.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+ManagerAction_ExtensionStateList

-- 
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Digium - A Sangoma Company | 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] Asterisk Manager Interface AMI over HTTP.

[Антон Сацкий]

Check http.conf
On Aug 31, 2015 5:31 PM, "Aziz TestAccount"  wrote:

> Hi All!
>
> I'm using Asterisk 11.6-cert11 and trying to set the AMI over HTTP ,
> without success. I always get the Error :
>
> ---
>
> Asterisk Call Manager/1.3
> Response: Error
> Message: Missing action in request
> ---
>
>
> I made the following configuration in manager.conf :
>
> [general]
> enabled = yes
> webenabled = yes
> enablestatic=yes
> port = 
> bindaddr = 0.0.0.0
>
> [admin]
> secret = admin1234
> deny=0.0.0.0/0.0.0.0
> permit=192.168.1.0/255.255.255.0 
> 
> read = system,call,log,verbose,command,agent,user,originate
> write = system,call,log,verbose,command,agent,user,originate
>
>
> And I'm trying to access via HTTP using the following link : 
> http://192.168.1.134:/manager?action=login=aziz=aziz1234/
>
>
> Could anyone tell me if I'm missing something.
>
> Thanks in advance.
>
>
>
>
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Re: [asterisk-users] Asterisk Manager Interface (AMI)

[Sam Govind]

Hey,

I think I remember the same post before. previously I heard someone telling
to use vicidial or some other thing  like that.But I don't think that those
are totally AMI based call-generators.

What I'd recently done is make a php page which connects to Asterisk's AMI
port. I send page request with destination number as parameter and depending
upon the HTTP arguments it send an ORIGINATE event to Asterisk with the
destination number to be dialled out via DAHDI(PRI) and once the call is
answered bridge it to a local dial plan extension which in term played a
sound-file/message to the connecting number.

So whenever I want Asterisk to initiate a call I send a HTTP request to my
Web-Server(hosting Asterisk) a call originated and played a message. You can
choose your design and directly connect to AMI and keep on sending ORIGINATE
events until you've all 200 channels occupied.

Hope it will help.

On Tue, Sep 13, 2011 at 6:26 AM, Kaushal Shriyan
kaushalshri...@gmail.comwrote:

 Hi,

 I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels
 (25 channels per PRI). Can someone please help me understand using
 Asterisk Manager Interface (AMI) available in Asterisk to dial out 200
 numbers and run a campaign for 200 numbers concurrently and play a mp3
 file ?

 Please suggest/guide.

 Regards,

 Kaushal

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Re: [asterisk-users] Asterisk Manager Problem

[Danny Nicholas]

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deric Page
Sent: Thursday, July 15, 2010 2:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Manager Problem

 

I am originating a call to a Local channel using an Originate Action:

Action: Originate

Channel: Local/d...@outdial

Context: outdial

Exten: answer

Priority: 1

Timeout: 45000

ActionID: some_id

In my dialplan, I have this:

[outdial]

exten = dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT})

exten = dial,n,NoOp(Dial Status = ${DIALSTATUS})

exten = dial,n,Agi(agi://localhost/Outdial.agi, ${DIAL_STRING})

exten = dial,n,Hangup()

exten = answer,1,NoOp(Dial Status = ${DIALSTATUS})

exten = answer,n,Playback(${GREETING_NAME})

exten = answer,n,WaitForSilence(2000)

exten = answer,n,Agi(agi://localhost/Outdial.agi)

exten = answer,n,Hangup()

Everything seems to work fine so long as the Dial command executes
successfully.  For example, if someone picks up the other end after the Dial
command completes, processing jumps to the answer extension as expected.
If no one ever answers and the Dial command times out, processing continues
on with the next priority of the dial extension, again as expected.

However, if someone sets a bad DIAL_STRING (such as using a channel that
doesn't exist), things start to behave oddly.  It continues on in the next
priority of the dial extension (as I would expect).  However, as soon as
it executes the call to the AGI script, it also starts processing the
answer extension at the same time.  As a result, I end up with two calls
into my AGI script.  Unfortunately, I don't know what I'm doing wrong here.

Thanks,

Deric Page

-- 

-- probably not the answer, but you could add these lines to the bottom of
[outdial]

Exten = *,1,hangup

Exten = t,1,hangup

Exten = i,1,hangup

This would hang-up any call that had an invalid number or did not dial
correctly.

deric.p...@nisc.coop

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Re: [asterisk-users] Asterisk Manager Interface (AMI) proxy recommendation

[Tiago Geada]

Hi!

You can just add several users to manager.conf or you can use AstManProxy...

On 21 March 2010 20:27, Leo Burd l...@media.mit.edu wrote:

 Hello there,

 I'm new to Asterisk and I'm trying to figure out a way to make the
 Asterisk Manager Interface (AMI) accessible to multiple users at the
 same time.  Would anyone recommend an AMI proxy that could be accessed
 from PHP code?

 Thanks in advance,

 Leo





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Re: [asterisk-users] Asterisk Manager Problem

[Andrei Verovski (aka MacGuru)]

A small clarification - a package I'm referring to is called Asterisk GUI, not 
Asterisk Manager. Sorry for mistype.

On Friday 25 September 2009 01:00:53 pm andreil1 wrote:
 Hi!

 I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
 repository), configured http.conf and manager.conf according to the
 manual.

 However, whenever I try to connect to Asterisk manager via web browser
 (http://192.168.0.1: , where xxx port defined in asterisk -
 http.conf), I've got this error:

 Not Implemented
 Attempt to use unimplemented / unsupported method
 Asterisk Server

 Downgrading Asterisk to version 1.6, or even installing on another
 SuSE box did not help.

 Anyone have any idea what is wrong?

 Many thanks in advance for any suggestion(s)

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Re: [asterisk-users] Asterisk manager login with java not working

[Philipp Kempgen]

Gopalakrishnan A.N schrieb:
 I am logging into asterisk manager thru a Java program but not able to
login, if i use PHP I am able to login. I have attached my java code with
this mail. Can someone step me up to go ahead

What does the manager interface respond?
What does the CLI say?


Philipp Kempgen
-- 
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Asterisk manager login with java not working

[Sebastian]

I would recommend you to use Asterisk-Java library has support for manager,
agi, etc.

http://asterisk-java.org/




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: lunes, 08 de junio de 2009 07:47 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk manager login with java not working

Gopalakrishnan A.N schrieb:
 I am logging into asterisk manager thru a Java program but not able to
login, if i use PHP I am able to login. I have attached my java code
with
this mail. Can someone step me up to go ahead

What does the manager interface respond?
What does the CLI say?


Philipp Kempgen
-- 
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -  http://www.amoocon.de
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Re: [asterisk-users] Asterisk Manager API Action Originate

[Matthew Nicholson]

The response code corresponds directly to the control frame types in
frame.h.  The code will be different depending on the technology used to
dial.  The 0 code is some sort of failure not involving any other
control frames.  Usually timeout is 3 (RINGING).  I have provided a list
taken from frame.h (1.4 svn r194356).

0 = FAILURE
1 = HANGUP (rarely seen on PRI in my experience)
3 = RINGING (i.e. timeout)
4 = ANSWER
5 = BUSY (that is where your 5 errors are coming from)
8 = CONGESTION


On Tue, 2009-05-12 at 11:48 -0700, Nicholas Blasgen wrote:
 Matt  Others,
 
 So to continue the issue, here's what I've learned.
 
 Tested on Asterisk:
 
 1.4.24.1
 SVN 193870
 SVN 191778
 
 So I think that covers most everything.  What I've learned is that any
 Timeout sends back a response code of ZERO instead of what I would
 have expected, ONE.  Anyone offer any other suggestions to try?
 
 My way to test this was to make a simple script to perform an AMI
 Originate call with a 4 second timeout.  I then have a standard tool
 to display all AMI Events.  On every system I tried I would get
 Response of Failure and Error Code of ZERO.
 
 On Tue, May 12, 2009 at 10:13 AM, Nicholas Blasgen
 nicho...@refractivedialer.com wrote:
 Matt,
 
 Oh, I thought it was Asterisk 1.4.23 like I wrote in my first
 email, but turns out to be Asterisk SVN-branch-1.4-r191778.
 
 But yes, I am talking about originateresponse.  I'm going to
 do some more debugging today to see if I can get the more
 information about the issue.  When I either Originate from the
 CLI or from AMI, I don't get anything on the console for
 either the errors or the initial connection.  I've had a lot
 of issues trying to debug Originate as a result.  And no CDR
 logs are being recorded.
 
 
 
 On Tue, May 12, 2009 at 5:36 AM, Matt Riddell
 li...@venturevoip.com wrote:
 On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote:
  Has anyone else had issues with Originate returning
 the wrong error
  code?  According to the docs, the following errors
 are supposed to be
  returned:
 
  0 = no such extension or number
  1 = no answer
  4 = answered
  8 = congested or not available
 
 
 Are you referring to the originateresponse event?
 
 Which version of Asterisk?
 
 
 
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-- 
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Re: [asterisk-users] Asterisk Manager API Action Originate

[Matthew Nicholson]

On Mon, 2009-05-11 at 20:44 -0700, Nicholas Blasgen wrote:
 (I would show you my CDR but it seems Originate doesn't log in the CDR
 like every other call for some reason).

Try testing with unanswered=yes in your CDR.  If that still does not
fix the problem, this is probably a bug. Test with the latest SVN (a
similar bug was recently fixed), and if you still see the problem, open
a new issue.
-- 
Matthew Nicholson
Digium, Inc. | Software Developer


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Re: [asterisk-users] Asterisk Manager API Action Originate

[Matt Riddell]

On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote:
 Has anyone else had issues with Originate returning the wrong error
 code?  According to the docs, the following errors are supposed to be
 returned:

 0 = no such extension or number
 1 = no answer
 4 = answered
 8 = congested or not available

Are you referring to the originateresponse event?

Which version of Asterisk?

-- 
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] Asterisk Manager API Action Originate

[Nicholas Blasgen]

Matt,

Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but
turns out to be Asterisk SVN-branch-1.4-r191778.

But yes, I am talking about originateresponse.  I'm going to do some more
debugging today to see if I can get the more information about the issue.
When I either Originate from the CLI or from AMI, I don't get anything on
the console for either the errors or the initial connection.  I've had a lot
of issues trying to debug Originate as a result.  And no CDR logs are being
recorded.

On Tue, May 12, 2009 at 5:36 AM, Matt Riddell li...@venturevoip.com wrote:

 On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote:
  Has anyone else had issues with Originate returning the wrong error
  code?  According to the docs, the following errors are supposed to be
  returned:
 
  0 = no such extension or number
  1 = no answer
  4 = answered
  8 = congested or not available

 Are you referring to the originateresponse event?

 Which version of Asterisk?

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Re: [asterisk-users] Asterisk Manager API Action Originate

[Nicholas Blasgen]

Matt  Others,

So to continue the issue, here's what I've learned.

Tested on Asterisk:

1.4.24.1
SVN 193870
SVN 191778

So I think that covers most everything.  What I've learned is that any
Timeout sends back a response code of ZERO instead of what I would have
expected, ONE.  Anyone offer any other suggestions to try?

My way to test this was to make a simple script to perform an AMI Originate
call with a 4 second timeout.  I then have a standard tool to display all
AMI Events.  On every system I tried I would get Response of Failure and
Error Code of ZERO.

On Tue, May 12, 2009 at 10:13 AM, Nicholas Blasgen 
nicho...@refractivedialer.com wrote:

 Matt,

 Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but
 turns out to be Asterisk SVN-branch-1.4-r191778.

 But yes, I am talking about originateresponse.  I'm going to do some more
 debugging today to see if I can get the more information about the issue.
 When I either Originate from the CLI or from AMI, I don't get anything on
 the console for either the errors or the initial connection.  I've had a lot
 of issues trying to debug Originate as a result.  And no CDR logs are being
 recorded.


 On Tue, May 12, 2009 at 5:36 AM, Matt Riddell li...@venturevoip.comwrote:

 On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote:
  Has anyone else had issues with Originate returning the wrong error
  code?  According to the docs, the following errors are supposed to be
  returned:
 
  0 = no such extension or number
  1 = no answer
  4 = answered
  8 = congested or not available

 Are you referring to the originateresponse event?

 Which version of Asterisk?



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Re: [asterisk-users] Asterisk Manager Interface Status Bug and Re: Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)

[Jared Smith]

On Wed, 2008-04-16 at 07:55 -0500, Nestor A. Diaz wrote:
 I frequently use the status function but it doesn't work as expected, i 
 use a program to parse the output of the status command but it don't 
 behave as expected, because i always wait for the latest package: 
 StatusComplete, and this package never arrives in the stream  
 sometimes StatusComplete is shown, sometimes not, but when it decide 
 not to show StatusComplete, asterisk really don't show 
 StatusComplete,  so sad  for me...

This sounds like it could be a bug... please open a ticket on the bug
tracker (http://bugs.digium.com/) so that the developers can keep track
of it and make sure it gets fixed.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Lee Jenkins]

Bill Andersen wrote:
 Has anyone tried to used VB6 to communicate with the Asterisk Manager?
 
 If so, would you be willing to share some basic code showing your
 approach to getting connected and parsing results?
 
 I've got a Telnet control that is allowing me to connect, authenticate
 and see the flow of status, etc., but I'm sure there is a better way
 to do this without using Telnet (maybe not?).  Any suggestions?
 
 I want to write a presence monitor (a virtual sidecar if you will)
 
 Bill
 

As Razza said, you can just use the winsock control included with VB.  The 
protocol is very simple, basically just name/value pairs delimited by #13#10 
(CRLF) with an extra CRLF at the end to denote termination of the packet.

Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=123432|CALLERID(name)=Automated Call
Async: true
extra CRLF == extra CRLF here.

So, like this:

1. Send your properly formatted packet to AMI .

2. Read incoming response terminated by double #13#10.

3. Parse values as you are comfortable with.

I am in the process of writing a similar product for one of our customers. 
Well, a re-write to add features and make it cross platform.  Here's a 
screenshot running on Linux/GTK:
http://leebo.dreamhosters.com/images/guiApp.png

A couple of side notes from what I've learned myself and read on this mailing 
list or through the wiki:

1. Packet Volume
The volume of messages that you can get from the AMI is impressive.  I've 
tested 
on our Asterisk system which has only 2 pots lines and two sip trunks with 10 
desktop phones and the amount of messages can be staggering!

Use a proxy for AMI if you have any decent phone traffic.  AstManProxy is VERY 
propular.  I wrote one as well, but its still beta and I think there's another 
one out there somewhere.  Usually with these proxy servers you can filter out 
unwanted/extraneous events to reduce the amount of messages your app has to 
contend with.

2. Make good use of Observer/Mediator pattern to distribute events to different 
parts of your GUI.  Monolithic loops to write everything out on a timer's event 
or after a Sleep() for instance, is not a good way to go in my experience.

3. Check the source for manager interface for changes between Asterisk 1.2 and 
1.4 (and 1.6?) if you're using 1.2 or plan to.  I believe the latest version of 
AMI is 1.1 (someone can correct me here).  A few label names for some of the 
AMI 
packets have been changed and a couple events (like LINK event) have been 
changed drastically.

I originally wrote against the 1.2 Manager interface only to find that I had to 
refactor some code and write descendant classes to handle the slight 
differences 
between the two versions' events.  I could have saved myself some work had I 
thought to look for the changes.  I think this link is up to date:
http://svn.digium.com/view/asterisk/trunk/doc/manager_1_1.txt?revision=98152view=markup

Happy coding.


-- 
Warm Regards,

Lee

Everything I needed to learn in life, I learned selling encyclopedias door to 
door.

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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Bill Andersen]

 I don't know if it would be of any use to you but we have some C# code
 that handles the basics of communicating the the Asterisk Manager
 Interface. It doesn't do anything fancy just sends single commands and
 checks the responses. We don't use it for monitoring.
 
 Regards,
 
 Greyman.

Thanks for the offer, I think I've got it figured out using winsock.

Thanks again.

Bill


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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Razza]

On 13/02/2008, Bill Andersen [EMAIL PROTECTED] wrote:
 Has anyone tried to used VB6 to communicate with the Asterisk Manager?

 If so, would you be willing to share some basic code showing your
 approach to getting connected and parsing results?
 Bill

I wrote some very very basic stuff ages ago using standard
mswinsck.ocx, will dig it out.

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Re: [asterisk-users] Asterisk Manager and Visual Basic

[Grey Man]

- Original Message 

 From: Bill Andersen [EMAIL PROTECTED]

 To: asterisk-users@lists.digium.com

 Sent: Wednesday, 13 February, 2008 8:31:01 PM

 Subject: [asterisk-users] Asterisk Manager and Visual Basic



 Has anyone tried to used VB6 to communicate with the Asterisk Manager?

 If so, would you be willing to share some basic code showing your
 approach to getting connected and parsing results?

 I've got a Telnet control that is allowing me to connect, authenticate
 and see the flow of status, etc., but I'm sure there is a better way
 to do this without using Telnet (maybe not?).  Any suggestions?



Hi Bill,

I don't know if it would be of any use to you but we have some C# code that 
handles the basics of communicating the the Asterisk Manager Interface. It 
doesn't do anything fancy just sends single commands and checks the responses. 
We don't use it for monitoring.

Regards,

Greyman.






  Get the name you always wanted with the new y7mail email address.
www.yahoo7.com.au/y7mail



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Re: [asterisk-users] asterisk manager and perl

[Tilghman Lesher]

On Monday 19 November 2007 16:48, cfh wrote:
 I m trying to use perl script to generate call with a server asterik .

perl -MCPAN -e'install Asterisk::Perl'

-- 
Tilghman

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Re: [asterisk-users] Asterisk Manager API - Originate command

[Atis]

On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote:
 Just ran into some issue with the originate AMI command. It seems that there
 is a limit of around 120 calls I can place with the originate command
 simutanously. By that I mean sending Asterisk a lot of originate command
 very fast. Anyone know if there is a limitation? Thnx.

What did you mean by simultaneously? Opening 120 manager
connections, and originating call at exactly the same time? I doubt..
So, probably there is some interval - within second/minute, etc.. And
how many manager connections do you use? Maybe asterisk have some
limit of them. Also - i think, there is some limit of asterisk
accepting commands sequentially from one connection.

Btw, what is your CPU load, when creating those 120 calls instantly?

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org

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Re: [asterisk-users] Asterisk Manager API - Originate command

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
 Hi all, 
  
 Just ran into some issue with the originate AMI command. It seems that
 there is a limit of around 120 calls I can place with the originate
 command simutanously. By that I mean sending Asterisk a lot of originate
 command very fast. Anyone know if there is a limitation? Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c
T3+G284pc4LV/JMlj13v8gU=
=oaJj
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Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Just to clear things up. It was one TCP connection to the manager
interface and the originate commands are send in a batch. I was able to
get away with 80 calls in a batch. Anything more than that is not good. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Sent: Monday, September 10, 2007 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote:
 Just ran into some issue with the originate AMI command. It seems that

 there is a limit of around 120 calls I can place with the originate 
 command simutanously. By that I mean sending Asterisk a lot of 
 originate command very fast. Anyone know if there is a limitation?
Thnx.

What did you mean by simultaneously? Opening 120 manager connections,
and originating call at exactly the same time? I doubt..
So, probably there is some interval - within second/minute, etc.. And
how many manager connections do you use? Maybe asterisk have some limit
of them. Also - i think, there is some limit of asterisk accepting
commands sequentially from one connection.

Btw, what is your CPU load, when creating those 120 calls instantly?

Regards,
Atis


--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835
[Toll free, USA] ?BEST? - www.BEST.eu.org

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Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Thanks for sharing your experience. I will play around with the Asteirsk
server tomorrow again. I took a look at it just before I left the
office. It has loads of crap. It's got all those non-essential things
and X windows running. Also, I can probably be able to get away with
starting a call every 30-50ms. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, September 10, 2007 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
 Hi all,
  
 Just ran into some issue with the originate AMI command. It seems that

 there is a limit of around 120 calls I can place with the originate 
 command simutanously. By that I mean sending Asterisk a lot of 
 originate command very fast. Anyone know if there is a limitation?
Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c
T3+G284pc4LV/JMlj13v8gU=
=oaJj
-END PGP SIGNATURE-

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Re: [asterisk-users] Asterisk Manager API - Originate command

[Wai Wu]

Just checked. I do have Async set to yes.  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, September 10, 2007 7:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

Thanks for sharing your experience. I will play around with the Asteirsk
server tomorrow again. I took a look at it just before I left the
office. It has loads of crap. It's got all those non-essential things
and X windows running. Also, I can probably be able to get away with
starting a call every 30-50ms. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, September 10, 2007 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Wai Wu wrote:
 Hi all,
  
 Just ran into some issue with the originate AMI command. It seems that

 there is a limit of around 120 calls I can place with the originate 
 command simutanously. By that I mean sending Asterisk a lot of 
 originate command very fast. Anyone know if there is a limitation?
Thnx.

First off, you should be using Async: true

Secondly, you shouldn't really be doing 120 simultaneous calls.

If your server can take say 300 concurrent calls, you will probably need
to start those up with about 30ms between them.

If you really need to start 120 calls all at the identical time, you
probably want to be looking at clustering Asterisk servers.

In SmoothTorque we set a minimum value for delay between calls, then
have a funnel which accepts calls from predictive campaigns.

The funnel knows about the connections to the Asterisk servers, and
distributes the calls in a round robin fashion (assuming all servers are
up).

Each server has a queue which allows calls sent to that server to back
up, and if a queue gets too full calls won't be sent to that server.

If, after stopping the sending of calls to a server, the queue does not
empty out, the server is placed into an inactive state, and a server
marked as standby is moved into the active state.

Any calls which remain in the queue are redistributed to other servers.

Hope that helps!

- --
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension

[Atis]

On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote:
 Hi Atis,

 Is your code open source, or are you willing to share your PHP code
 snippets with me? And thanks for the information on Asterisk's
 stability. Do you think there is an issue in the implementation or
 just network/traffic issues?

 Thanks for your time.

Hi,

Sorry, but i can't share - it's company's property, and you wouldn't
want it, because it includes a bunch of other things - our own
libraries, customer recognition, etc, etc..

However, for your purpose - the code for such program would be
trivial. All you need is Stomp library for php, and then just convert
all data to some format that your program will recognize (i use XML).

Also, you might take a look on AJAM -
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29

Regards,
Atis

-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
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Re: [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension

[Atis]

On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote:
 Hi Everyone,

 I am writing an open source application that brings desktops widgets
 to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I
 am trying to get my head around the Asterisk Manager Interface.

 I had been using the Event: NewCallerid to detect a new call which my
 Asterisk server doesn't seem to send to the socket anymore, because of
 which I have reverted to using Event: Newexten.

 Which is the most efficient way of monitoring if a new phone call is
 coming my way? Also my application will only monitor a single
 extension, should I filter the requests on the client side or can a
 manager interface user be restricted to a single extensions events.

I don't know about manager, but i've done the same using PHP script
that executes from dialplan before dial + ActiveMQ (message queue) +
custom app. I just didn't wanted to do filtering with manager, and so
on.. Additionally, from my experience, creating a bunch of manager
connections isn't quite good for asterisk stability..

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
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Re: [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension

[Devraj Mukherjee]

Hi Atis,

Is your code open source, or are you willing to share your PHP code
snippets with me? And thanks for the information on Asterisk's
stability. Do you think there is an issue in the implementation or
just network/traffic issues?

Thanks for your time.

On 9/4/07, Atis [EMAIL PROTECTED] wrote:
 On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote:
  Hi Everyone,
 
  I am writing an open source application that brings desktops widgets
  to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I
  am trying to get my head around the Asterisk Manager Interface.
 
  I had been using the Event: NewCallerid to detect a new call which my
  Asterisk server doesn't seem to send to the socket anymore, because of
  which I have reverted to using Event: Newexten.
 
  Which is the most efficient way of monitoring if a new phone call is
  coming my way? Also my application will only monitor a single
  extension, should I filter the requests on the client side or can a
  manager interface user be restricted to a single extensions events.

 I don't know about manager, but i've done the same using PHP script
 that executes from dialplan before dial + ActiveMQ (message queue) +
 custom app. I just didn't wanted to do filtering with manager, and so
 on.. Additionally, from my experience, creating a bunch of manager
 connections isn't quite good for asterisk stability..

 Regards,
 Atis


 --
 Atis Lezdins,
 IT Responsible of BEST Riga,
 [EMAIL PROTECTED]
 ICQ: 142239285
 Skype: atis.lezdins
 Cell Phone: +371 28806004 [Tele2, Latvia]
 Work phone: +1 800 7502835 [Toll free, USA]
 ?BEST? - www.BEST.eu.org

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Incredibles)

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Re: [asterisk-users] Asterisk Manager Proxy - Still required?

[BJ Weschke]

On 8/17/07, Andrew Ruthven [EMAIL PROTECTED] wrote:
 Hi,

 With more recent version of v1.2 and with v1.4 are things like the
 AstManProxy still recommended if you want to have a bunch of
 applications talking directly to Asterisk?


 If you're looking to have a number of clients monitor events, etc,
I'd say that having a proxy in between is still a good thing. The
performance of the manager itself is greatly improved since before 1.2
but there are still ongoing (albeit random, sporadic) issues with cpu
race and huge memory allocations that still need to get resolved.

-- 
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http://www.btwtech.com/

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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Thomas Kenyon]

Anselm Martin Hoffmeister wrote:
 
 I did something similar using multiple records in a row.
 Something like
 
 exten = 931,1,Answer()
 exten = 931,2,Wait(2)
 exten = 931,3,Set(E=1000)
 exten = 931,4,Playback(beep)
 exten = 931,5,Set(E=$[${E} + 1])
 exten = 931,6,Record(/tmp/asterisk-recording-${E:1})
 exten = 931,7,Playback(/tmp/asterisk-recording-${E:1})
 exten = 931,8,Wait(2)
 exten = 931,9,Goto(4)
 
 This will loop: beep, record until # pressed, replay, wait, beep...
 The files will be written with ascending numbers starting 001. Move
 them to another place before doing the next recording session.
 
Couldn't you use %d instead of settup up variable E?
ie.

exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Playback(beep)
exten = 931,4,Record(/tmp/asterisk-recording-%d.wav)
exten = 931,5,Playback(${RECORDED_FILE})
exten = 931,6,Goto(2)

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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Peder @ NetworkOblivion]

FYI to anybody who cares, here is what I did:

1.  Create web page where you enter a file name and a number to call
2.  Insert the file name into the *DB via Asterisk Manager
3.  Through Asterisk Manager create a call file from a recording 
extension to the phone number entered in #1
4.  The recording extension answers, plays a beep, records the call to 
the file name that it pulls from the *DB
5.  It plays the recording back and then hangs up

It works perfectly.  Not quite what I planned, but it does work.

Doug Lytle wrote:
 Peder @ NetworkOblivion wrote:
 That's great, now say you have 5 or 6 AA's and each one has 10 different 
 parts that you want to record (thank you for calling...  for Steve 
 press
 
 This is what I do.  I found it some place on the wiki, it lets you 
 record many prompts. 
 
 exten = 4850,1,Goto(recordings,s,1)
 
 ; **
 ; Welcome to the Audio prompt recording menu
 ; **
 
 exten = s,1,Playback(local/extension-recording-menu)
 
 ; 
 ; Please record your message, when
 ; completed press the # key
 ; 
 
 exten = s,2,Playback(local/please-record-msg)
 exten = s,3,Record(mymessage:gsm)
 
 ; 
 ; You said
 ; 
 
 exten = s,4,Playback(local/you-said)
 exten = s,5,Playback(mymessage)
 
 ; ***
 ; Press 1 to continue or 2 to change your message
 ; ***
 
 exten = s,6,Background(local/press1-or-2)
 exten = s,7,Set(TIMEOUT(response)=2)
 exten = s,8,Set(TIMEOUT(digit)=2)
 
 exten = 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm 
 /var/lib/asterisk/sounds/local/`date +%s`.gsm)
 
 ; 
 ; Thank you, your recording has been saved
 ; 
 
 exten = 1,2,Playback(local/recording-saved)
 
 ; *
 ; Press 3 to record another message, or 4 to hangup
 ; *
 
 exten = 1,3,Background(local/press3-torecord-4tohang)
 
 exten = 2,1,Goto(recordings,s,2)
 exten = 3,1,Goto(recordings,s,2)
 
 exten = 4,1,Playback(vm-goodbye)
 exten = 4,2,Hangup()
 
 exten = t,1,Playback(local/sorry-didnot-getthat)
 exten = t,2,Goto(recordings,s,6)
 
 exten = i,1,Playback(local/sorry-invalid-choice)
 exten = i,2,Goto(recordings,s,2)
 
 
 Doug
 


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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Anselm Martin Hoffmeister]

Am Freitag, den 10.08.2007, 11:26 -0500 schrieb Peder @ NetworkOblivion:
 That's great, now say you have 5 or 6 AA's and each one has 10 different 
 parts that you want to record (thank you for calling...  for Steve 
 press 1 for dave press 2).  Rather than having to record a long 
 message, I want to break it into pieces so that if dave leaves, we can 
 just record that one chunk rather than the whole thing.  I would need 
 lots of extensions pre-setup for each chunk.  Not very efficient.
 
 Gordon Henderson wrote:
  On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
  
  I am trying to use Asterisk Manager via php to record auto attendant
  greetings and I just can't figure out how to do it.  I've got the php
  page working and I can click to call between two phones.  However if I
  click to call just a single phone and then try to enable monitor, when
  I pick up the ringing phone, it just hangs up and doesn't record
  anything.  I'm sure I just don't know the appropriate syntax.  Has
  anybody done something like this?  I can do the php stuff, I just need
  the Asterisk Manager syntax.

I did something similar using multiple records in a row.
Something like

exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten = 931,6,Record(/tmp/asterisk-recording-${E:1})
exten = 931,7,Playback(/tmp/asterisk-recording-${E:1})
exten = 931,8,Wait(2)
exten = 931,9,Goto(4)

This will loop: beep, record until # pressed, replay, wait, beep...
The files will be written with ascending numbers starting 001. Move
them to another place before doing the next recording session.

HTH
Anselm


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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Doug Lytle]

Peder @ NetworkOblivion wrote:
 That's great, now say you have 5 or 6 AA's and each one has 10 different 
 parts that you want to record (thank you for calling...  for Steve 
 press

This is what I do.  I found it some place on the wiki, it lets you 
record many prompts. 

exten = 4850,1,Goto(recordings,s,1)

; **
; Welcome to the Audio prompt recording menu
; **

exten = s,1,Playback(local/extension-recording-menu)

; 
; Please record your message, when
; completed press the # key
; 

exten = s,2,Playback(local/please-record-msg)
exten = s,3,Record(mymessage:gsm)

; 
; You said
; 

exten = s,4,Playback(local/you-said)
exten = s,5,Playback(mymessage)

; ***
; Press 1 to continue or 2 to change your message
; ***

exten = s,6,Background(local/press1-or-2)
exten = s,7,Set(TIMEOUT(response)=2)
exten = s,8,Set(TIMEOUT(digit)=2)

exten = 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm 
/var/lib/asterisk/sounds/local/`date +%s`.gsm)

; 
; Thank you, your recording has been saved
; 

exten = 1,2,Playback(local/recording-saved)

; *
; Press 3 to record another message, or 4 to hangup
; *

exten = 1,3,Background(local/press3-torecord-4tohang)

exten = 2,1,Goto(recordings,s,2)
exten = 3,1,Goto(recordings,s,2)

exten = 4,1,Playback(vm-goodbye)
exten = 4,2,Hangup()

exten = t,1,Playback(local/sorry-didnot-getthat)
exten = t,2,Goto(recordings,s,6)

exten = i,1,Playback(local/sorry-invalid-choice)
exten = i,2,Goto(recordings,s,2)


Doug

-- 
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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Gordon Henderson]

On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:

 I am trying to use Asterisk Manager via php to record auto attendant
 greetings and I just can't figure out how to do it.  I've got the php
 page working and I can click to call between two phones.  However if I
 click to call just a single phone and then try to enable monitor, when
 I pick up the ringing phone, it just hangs up and doesn't record
 anything.  I'm sure I just don't know the appropriate syntax.  Has
 anybody done something like this?  I can do the php stuff, I just need
 the Asterisk Manager syntax.

 I want to call a phone, when they pick up, it starts recording to a file
 and when they hang up, it closes the file.

 Any help would be appreciated.

I can't help but think you're making life hard for yourself.

Why not do it by dialling a code on the telephone and having the dialplan 
Record() what's being spoken rather than go to the bother of writing PHP 
to call asterisk via the monitor interface...

But I don't know the whole story of your implementation!

I record prompts like this:

; Record Intro message.

exten = 771,1,Answer()
exten = 771,2,Wait(1)
exten = 771,3,Playback(beep)
exten = 771,4,Record(/var/spool/app/introMessage:wav)
exten = 771,5,Playback(/var/spool/app/introMessage)
exten = 771,6,Hangup()

Gordon

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Re: [asterisk-users] Asterisk Manager to Record Greetings

[David Bandel]

On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
 I am trying to use Asterisk Manager via php to record auto attendant
 greetings and I just can't figure out how to do it.  I've got the php
 page working and I can click to call between two phones.  However if I
 click to call just a single phone and then try to enable monitor, when
 I pick up the ringing phone, it just hangs up and doesn't record
 anything.  I'm sure I just don't know the appropriate syntax.  Has
 anybody done something like this?  I can do the php stuff, I just need
 the Asterisk Manager syntax.

 I want to call a phone, when they pick up, it starts recording to a file
 and when they hang up, it closes the file.

 Any help would be appreciated.


Peder,

Don't know how to do this from php and the asterisk manager, but the
following will do what you want:

[record]
exten=_9003.,1,answer()
exten=_9003.,n,Record(${EXTEN:4}:ulaw||60|t)
exten=_9003.,n,Playback(${EXTEN:4})
exten=_9003.,n,Hangup()

just include the above context into the appropriate context for a
phone you want to record from.  Dial 90031234 (for example).  You will
hear a beep and can record whatever you want. For accounting for
example, then press the * key.  You will hear a playback.

The recording will be found in /var/lib/asterisk/sounds and called
1234 (if you dialed 90031234 and your language is en).  If you don't
like it, hit redial and do it over.  If you like it, just move the
file to another name or use a different last 4 digits.

Ciao,

David A. Bandel
-- 
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- Nemesis Air Racing Team motto

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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Peder @ NetworkOblivion]

That's great, now say you have 5 or 6 AA's and each one has 10 different 
parts that you want to record (thank you for calling...  for Steve 
press 1 for dave press 2).  Rather than having to record a long 
message, I want to break it into pieces so that if dave leaves, we can 
just record that one chunk rather than the whole thing.  I would need 
lots of extensions pre-setup for each chunk.  Not very efficient.

Gordon Henderson wrote:
 On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
 
 I am trying to use Asterisk Manager via php to record auto attendant
 greetings and I just can't figure out how to do it.  I've got the php
 page working and I can click to call between two phones.  However if I
 click to call just a single phone and then try to enable monitor, when
 I pick up the ringing phone, it just hangs up and doesn't record
 anything.  I'm sure I just don't know the appropriate syntax.  Has
 anybody done something like this?  I can do the php stuff, I just need
 the Asterisk Manager syntax.

 I want to call a phone, when they pick up, it starts recording to a file
 and when they hang up, it closes the file.

 Any help would be appreciated.
 
 I can't help but think you're making life hard for yourself.
 
 Why not do it by dialling a code on the telephone and having the dialplan 
 Record() what's being spoken rather than go to the bother of writing PHP 
 to call asterisk via the monitor interface...
 
 But I don't know the whole story of your implementation!
 
 I record prompts like this:
 
 ; Record Intro message.
 
 exten = 771,1,Answer()
 exten = 771,2,Wait(1)
 exten = 771,3,Playback(beep)
 exten = 771,4,Record(/var/spool/app/introMessage:wav)
 exten = 771,5,Playback(/var/spool/app/introMessage)
 exten = 771,6,Hangup()
 
 Gordon
 
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Re: [asterisk-users] Asterisk Manager to Record Greetings

[James FitzGibbon]

On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:

 That's great, now say you have 5 or 6 AA's and each one has 10 different
 parts that you want to record (thank you for calling...  for Steve
 press 1 for dave press 2).  Rather than having to record a long
 message, I want to break it into pieces so that if dave leaves, we can
 just record that one chunk rather than the whole thing.  I would need
 lots of extensions pre-setup for each chunk.  Not very efficient.


You could front it with something other than extensions.  I tag all my
recordings with a 4 digit number, but I use an AGI script to manage them.
The AGI (written in Perl) authenticates the user, then lets them punch in
the announcement number.  A database lookup translates the announcement
number to the pathname of the the file, then the user gets the choice to
listen to the existing recording or re-record it.

Adding new announcements is a simple SQL insert.

TMTOWTDI

-- 
j.
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Re: [asterisk-users] Asterisk Manager to Record Greetings

[Anthony Francis]

Gordon Henderson wrote:
 On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:

   
 I am trying to use Asterisk Manager via php to record auto attendant
 greetings and I just can't figure out how to do it. 
 

 I can't help but think you're making life hard for yourself.

 Why not do it by dialling a code on the telephone and having the dialplan 
 Record() what's being spoken rather than go to the bother of writing PHP 
 to call asterisk via the monitor interface...

   
One idea is to just have PHP allow them to upload their pre-recorded 
greetings and then have a menu where they can select which greeting they 
are playing and when. Maybe you cna even get a browser plugin app to 
allow them to record it in realtime, and also give them the option to 
dial in and record the aa using a file code as the others suggested.

Anthony

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Re: [asterisk-users] Asterisk Manager

[Anthony Francis]

Arun Kumar wrote:
 Hi

 this is my code for * manager:

 $oSocket = fsockopen($strHost, 5038, 
 $errnum, $errdesc) or die(Connection to host failed);
 fputs($oSocket, Action: login\r\n);
 fputs($oSocket, Username: $strUser\r\n);
 fputs($oSocket, Secret: 
 $strSecret\r\n\r\n);
 fputs($oSocket, Action: Originate\r\n);
 fputs($oSocket, Channel: 
 $strChannel\r\n);
 fputs($oSocket, WaitTime: 
 $strWaitTime\r\n);
 fputs($oSocket, CallerId: 
 $strCallerId\r\n);
 fputs($oSocket, Context: 
 $strContext\r\n);
 fputs($oSocket, Exten: $strExten\r\n);
 fputs($oSocket, Priority: 
 $strPriority\r\n\r\n);
 fputs($oSocket, Action: Logoff\r\n\r\n);

 when call gets answered it goes to my specified exten can I also 
 handle if my call is not got answered b'coz of some reason.

 that is when get ans goto exten= 101 if call is not got and goto exten=102

 please help.

 thanks

 arun

 

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Hi,

What language is this? At any rate to determine if the call is not 
got answered you need to watch the events using the calls uniqueid so 
you can determine what happens with it.

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Re: [asterisk-users] Asterisk Manager

[Mojo with Horan Company, LLC]

(Using things like fgets and strcmp and substr and sprintf and such.) 
I've found that reading everything asterisk's got for me in one go and 
then using explode function with '\r\n' delimiter seemed to work well 
for me.

Anthony Francis wrote:
 Arun Kumar wrote:
 Hi

 this is my code for * manager:

 $oSocket = fsockopen($strHost, 5038, 
 $errnum, $errdesc) or die(Connection to host failed);
 fputs($oSocket, Action: login\r\n);
 fputs($oSocket, Username: $strUser\r\n);
 fputs($oSocket, Secret: 
 $strSecret\r\n\r\n);
 fputs($oSocket, Action: Originate\r\n);
 fputs($oSocket, Channel: 
 $strChannel\r\n);
 fputs($oSocket, WaitTime: 
 $strWaitTime\r\n);
 fputs($oSocket, CallerId: 
 $strCallerId\r\n);
 fputs($oSocket, Context: 
 $strContext\r\n);
 fputs($oSocket, Exten: $strExten\r\n);
 fputs($oSocket, Priority: 
 $strPriority\r\n\r\n);
 fputs($oSocket, Action: Logoff\r\n\r\n);

 when call gets answered it goes to my specified exten can I also 
 handle if my call is not got answered b'coz of some reason.

 that is when get ans goto exten= 101 if call is not got and goto exten=102

 please help.

 thanks

 arun

 

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 Hi,
 
 What language is this? At any rate to determine if the call is not 
 got answered you need to watch the events using the calls uniqueid so 
 you can determine what happens with it.
 
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 asterisk-users mailing list
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RE: [asterisk-users] asterisk manager interface stability

[Dean Collins]

Do you have any specific experience with astmanproxy?

Can anyone give me an idea on number of simultaneous connections this
can legitimately handle with ease?

This has been around for a while by the looks of it but I haven't heard
about it before.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Martin Smith
 Sent: Wednesday, 16 May 2007 1:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] asterisk manager interface stability
 
 You guys all sound like you're talking about AstManProxy.
 
 See:
 http://www.voip-info.org/tiki-index.php?page=AstManProxy
 
 
 I'm not saying it is the solution to your problem per se, but I can't
 help but think of it when I read the descriptions of what people want
 (you even use the word proxy!). Figured I'd send this out in case
 someone hadn't seen it.
 
 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Lee Jenkins
  Subject: Re: [asterisk-users] asterisk manager interface stability
 
   The new Asterisk Manager web API in 1.4 is a good step
  where sending of
   Actions does not require an actual Telnet conneciton to the
  AMI, but I
   think to be able to handle larger numbers of concurrent
  connections that
   a separate send-only and a separate receive-only type of
  interface be
   built where Asterisk would just output all AMI data to a single
   server-like application that would then broadcast it to all
  connected
   clients. This would remove the burden of so many connections going
   directly into Asterisk and would allow for much larger scaling of
   AMI-type applications that require real-time output of AMI events.
  
 
  I definitely agree here personally.  Clients could connect to this
  proxy and subscribe to only the events that are interesting
  or applicable.
 
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Re: [asterisk-users] asterisk manager interface stability

[Lee Jenkins]

Matt Florell wrote:

On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:

Matt Florell wrote:
 The issue has more to do with the sheer amount of data passed to the
 client from within the Asterisk application when you have 50-100+
 clients connected to the AMI on full output mode. Running a system with
 FreePBX/Trixbox especially generates vast amounts of output that has to
 be generated on every AMI connection for every client. This is not
 trivial and can result in lockups very easily, although this has gotten
 much better since the early 1.0 versions.

 The new Asterisk Manager web API in 1.4 is a good step where sending of
 Actions does not require an actual Telnet conneciton to the AMI, but I
 think to be able to handle larger numbers of concurrent connections 
that

 a separate send-only and a separate receive-only type of interface be
 built where Asterisk would just output all AMI data to a single
 server-like application that would then broadcast it to all connected
 clients. This would remove the burden of so many connections going
 directly into Asterisk and would allow for much larger scaling of
 AMI-type applications that require real-time output of AMI events.


I definitely agree here personally.  Clients could connect to this
proxy and subscribe to only the events that are interesting or 
applicable.


 As for how to go about doing this, I can't help you there. I did 
build a

 very specialized version of something like this 4 years ago for the
 astGUIclient project called the Asterisk Central Queue System(ACQS) It
 is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is
 limited in what it does, but it does scale much better than using 
direct

 AMI connections.

I've been considering writing something like this for a project that I'm
thinking about doing that would require potentially high number of
concurrent clients to consume AMI services.

 From your experience, does the software that you wrote require
significant CPU to cache and then doll out the kind of volume of
messages that AMI can send?


One of the great parts about removing the broadcasting of AMI events
outside of the Asterisk process is that the broadcast server process
can exist on a separate physical server removing any kind of overhead
on the Asterisk server.

In my experience doing the proxy on the same machine uses less CPU
resources than the same number of AMI connected clients, and doesn't
have any of the deadlock issues that can happen with a lot of direct
AMI connections.

For my application(ACQS) I use MySQL as a storage engine for all of
the recent events received and sent so that they can be independantly
queried by any client apps that need to see them.

MATT---



Neat.  So the clients use a polling model?  Individual clients then 
query only for events that are interesting?

--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk manager interface stability

[Matt Florell]

On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:

Matt Florell wrote:
 On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
 Matt Florell wrote:
  The issue has more to do with the sheer amount of data passed to the
  client from within the Asterisk application when you have 50-100+
  clients connected to the AMI on full output mode. Running a system with
  FreePBX/Trixbox especially generates vast amounts of output that has to
  be generated on every AMI connection for every client. This is not
  trivial and can result in lockups very easily, although this has gotten
  much better since the early 1.0 versions.
 
  The new Asterisk Manager web API in 1.4 is a good step where sending of
  Actions does not require an actual Telnet conneciton to the AMI, but I
  think to be able to handle larger numbers of concurrent connections
 that
  a separate send-only and a separate receive-only type of interface be
  built where Asterisk would just output all AMI data to a single
  server-like application that would then broadcast it to all connected
  clients. This would remove the burden of so many connections going
  directly into Asterisk and would allow for much larger scaling of
  AMI-type applications that require real-time output of AMI events.
 

 I definitely agree here personally.  Clients could connect to this
 proxy and subscribe to only the events that are interesting or
 applicable.

  As for how to go about doing this, I can't help you there. I did
 build a
  very specialized version of something like this 4 years ago for the
  astGUIclient project called the Asterisk Central Queue System(ACQS) It
  is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is
  limited in what it does, but it does scale much better than using
 direct
  AMI connections.

 I've been considering writing something like this for a project that I'm
 thinking about doing that would require potentially high number of
 concurrent clients to consume AMI services.

  From your experience, does the software that you wrote require
 significant CPU to cache and then doll out the kind of volume of
 messages that AMI can send?

 One of the great parts about removing the broadcasting of AMI events
 outside of the Asterisk process is that the broadcast server process
 can exist on a separate physical server removing any kind of overhead
 on the Asterisk server.

 In my experience doing the proxy on the same machine uses less CPU
 resources than the same number of AMI connected clients, and doesn't
 have any of the deadlock issues that can happen with a lot of direct
 AMI connections.

 For my application(ACQS) I use MySQL as a storage engine for all of
 the recent events received and sent so that they can be independantly
 queried by any client apps that need to see them.

 MATT---


Neat.  So the clients use a polling model?  Individual clients then
query only for events that are interesting?

Warm Regards,

Lee


Yes, the clients only connect to the MySQL database and can query the
events as they need to for their display.

MATT---





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RE: [asterisk-users] asterisk manager interface stability

[Dean Collins]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt Florell
 Sent: Friday, 18 May 2007 4:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] asterisk manager interface stability
  
 
  Neat.  So the clients use a polling model?  Individual clients then
  query only for events that are interesting?
 
  Warm Regards,
 
  Lee
 
 Yes, the clients only connect to the MySQL database and can query the
 events as they need to for their display.
 
 MATT---
 
 
 
 
  ___


 Wouldn't that make it highly inefficient? Is there no two way
dialog or am I missing something?

Basically if I have 100 end user clients that needed real time
interaction with astproxy are you saying that each client would need to
poll once per second (eg 100 polls per second) in order to see if
something happened that second that was relevant to them?)


 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
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Re: [asterisk-users] asterisk manager interface stability

[Matt Florell]

On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote:



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt Florell
 Sent: Friday, 18 May 2007 4:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] asterisk manager interface stability
  
 
  Neat.  So the clients use a polling model?  Individual clients then
  query only for events that are interesting?
 
  Warm Regards,
 
  Lee

 Yes, the clients only connect to the MySQL database and can query the
 events as they need to for their display.

 MATT---



 
  ___


 Wouldn't that make it highly inefficient? Is there no two way
dialog or am I missing something?


It is inefficient, but it is non-blocking which the AMI is not. having
a separate listen and separate send processes removes the problems
with AMI locking up on high volume Asterisk systems.


Basically if I have 100 end user clients that needed real time
interaction with astproxy are you saying that each client would need to
poll once per second (eg 100 polls per second) in order to see if
something happened that second that was relevant to them?)



Not a problem for MySQL, that's what it does best. The astguiclient
application can do 20+ queries per second per client depending on the
application. I currently have one company with over 200 client
applications(AJAX) sending 3000-4000 queries per second to the MySQL
server, and it handles it just fine. On the client side, the load is
not very high either, even on a PIII 700MHz machine.


MATT---



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
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RE: [asterisk-users] asterisk manager interface stability

[Dean Collins]

Sweet thanks Matt.

 

If there are any developers in Manhattan (or very nearby) who have
experience with Astproxy and are looking for sweat equity ownership in a
new Asterisk application get in touch. Also looking for someone with ROR
UI skills but I might already have that role filled.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph

  http://click.mexuar.com/webuser/click/7/userurl/Cognation  
http://click.mexuar.com/webuser/nojs/7/userurl/Cognation 
 

 

 -Original Message-

 From: [EMAIL PROTECTED] [mailto:asterisk-users-

 [EMAIL PROTECTED] On Behalf Of Matt Florell

 Sent: Friday, 18 May 2007 6:46 PM

 To: Asterisk Users Mailing List - Non-Commercial Discussion

 Subject: Re: [asterisk-users] asterisk manager interface stability

 

 On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote:

 

 

   -Original Message-

   From: [EMAIL PROTECTED]
[mailto:asterisk-users-

   [EMAIL PROTECTED] On Behalf Of Matt Florell

   Sent: Friday, 18 May 2007 4:22 PM

   To: Asterisk Users Mailing List - Non-Commercial Discussion

   Subject: Re: [asterisk-users] asterisk manager interface stability



   

Neat.  So the clients use a polling model?  Individual clients
then

query only for events that are interesting?

   

Warm Regards,

   

Lee

  

   Yes, the clients only connect to the MySQL database and can query
the

   events as they need to for their display.

  

   MATT---

  

  

  

   

___

 

 

   Wouldn't that make it highly inefficient? Is there no two way

  dialog or am I missing something?

 

 It is inefficient, but it is non-blocking which the AMI is not. having

 a separate listen and separate send processes removes the problems

 with AMI locking up on high volume Asterisk systems.

 

  Basically if I have 100 end user clients that needed real time

  interaction with astproxy are you saying that each client would need
to

  poll once per second (eg 100 polls per second) in order to see if

  something happened that second that was relevant to them?)

 

 

 Not a problem for MySQL, that's what it does best. The astguiclient

 application can do 20+ queries per second per client depending on the

 application. I currently have one company with over 200 client

 applications(AJAX) sending 3000-4000 queries per second to the MySQL

 server, and it handles it just fine. On the client side, the load is

 not very high either, even on a PIII 700MHz machine.

 

 

 MATT---

 

 

  Regards,

 

  Dean Collins

  Cognation Pty Ltd

  [EMAIL PROTECTED]

  +1-212-203-4357 Ph

  +1-917-207-3420 Mb

  +61-2-9016-5642 (Sydney in-dial).

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http://lists.digium.com/mailman/listinfo/asterisk-users

 

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Re: [asterisk-users] asterisk manager interface stability

[Lee Jenkins]

Matt Florell wrote:

On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:

Matt Florell wrote:
 On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
 Matt Florell wrote:
  The issue has more to do with the sheer amount of data passed to the
  client from within the Asterisk application when you have 50-100+
  clients connected to the AMI on full output mode. Running a 
system with
  FreePBX/Trixbox especially generates vast amounts of output that 
has to

  be generated on every AMI connection for every client. This is not
  trivial and can result in lockups very easily, although this has 
gotten

  much better since the early 1.0 versions.
 
  The new Asterisk Manager web API in 1.4 is a good step where 
sending of
  Actions does not require an actual Telnet conneciton to the AMI, 
but I

  think to be able to handle larger numbers of concurrent connections
 that
  a separate send-only and a separate receive-only type of 
interface be

  built where Asterisk would just output all AMI data to a single
  server-like application that would then broadcast it to all 
connected

  clients. This would remove the burden of so many connections going
  directly into Asterisk and would allow for much larger scaling of
  AMI-type applications that require real-time output of AMI events.
 

 I definitely agree here personally.  Clients could connect to this
 proxy and subscribe to only the events that are interesting or
 applicable.

  As for how to go about doing this, I can't help you there. I did
 build a
  very specialized version of something like this 4 years ago for the
  astGUIclient project called the Asterisk Central Queue 
System(ACQS) It

  is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is
  limited in what it does, but it does scale much better than using
 direct
  AMI connections.

 I've been considering writing something like this for a project 
that I'm

 thinking about doing that would require potentially high number of
 concurrent clients to consume AMI services.

  From your experience, does the software that you wrote require
 significant CPU to cache and then doll out the kind of volume of
 messages that AMI can send?

 One of the great parts about removing the broadcasting of AMI events
 outside of the Asterisk process is that the broadcast server process
 can exist on a separate physical server removing any kind of overhead
 on the Asterisk server.

 In my experience doing the proxy on the same machine uses less CPU
 resources than the same number of AMI connected clients, and doesn't
 have any of the deadlock issues that can happen with a lot of direct
 AMI connections.

 For my application(ACQS) I use MySQL as a storage engine for all of
 the recent events received and sent so that they can be independantly
 queried by any client apps that need to see them.

 MATT---


Neat.  So the clients use a polling model?  Individual clients then
query only for events that are interesting?

Warm Regards,

Lee


Yes, the clients only connect to the MySQL database and can query the
events as they need to for their display.

MATT---




Cool.  I hadn't thought of doing it that way.  My idea was to somehow 
keep an in memory cache for client connections.  As events were received 
from the AMI, poll a hash table in memory and forward the event to 
client connections who have registered interest in that event.


--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk manager interface stability

[Lee Jenkins]

Matt Florell wrote:

On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote:



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt Florell
 Sent: Friday, 18 May 2007 4:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] asterisk manager interface stability
  
 
  Neat.  So the clients use a polling model?  Individual clients then
  query only for events that are interesting?
 
  Warm Regards,
 
  Lee

 Yes, the clients only connect to the MySQL database and can query the
 events as they need to for their display.

 MATT---



 
  ___


 Wouldn't that make it highly inefficient? Is there no two way
dialog or am I missing something?


It is inefficient, but it is non-blocking which the AMI is not. having
a separate listen and separate send processes removes the problems
with AMI locking up on high volume Asterisk systems.


Basically if I have 100 end user clients that needed real time
interaction with astproxy are you saying that each client would need to
poll once per second (eg 100 polls per second) in order to see if
something happened that second that was relevant to them?)



Not a problem for MySQL, that's what it does best. The astguiclient
application can do 20+ queries per second per client depending on the
application. I currently have one company with over 200 client
applications(AJAX) sending 3000-4000 queries per second to the MySQL
server, and it handles it just fine. On the client side, the load is
not very high either, even on a PIII 700MHz machine.




Nice.  And using a DB to cache events no doubt simplifies the mechanics 
of the application making it easier to develop and maintain.


--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk manager interface stability

[Lee Jenkins]

Dean Collins wrote:



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, 18 May 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk manager interface stability

Neat.  So the clients use a polling model?  Individual clients then
query only for events that are interesting?

Warm Regards,

Lee

Yes, the clients only connect to the MySQL database and can query the
events as they need to for their display.

MATT---




___



 Wouldn't that make it highly inefficient? Is there no two way
dialog or am I missing something?

Basically if I have 100 end user clients that needed real time
interaction with astproxy are you saying that each client would need to
poll once per second (eg 100 polls per second) in order to see if
something happened that second that was relevant to them?)




Although I would lean toward an in-memory cache/handling of events, you 
could have a another app or thread pool that queries the database on 
behalf of the clients and notifies clients accordingly, which might 
negate the need of clients to poll the database and reduce network traffic.


--

Warm Regards,

Lee



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RE: [asterisk-users] asterisk manager interface stability

[Dean Collins]

If it's not stable what needs to be done to improve this? What are the
issues? What are the alternatives (eg is Adhearsion an alternative here)

 

I am about to start looking into a project that requires every user to
have AMI access so looking to fund development in this space. 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph

  http://click.mexuar.com/webuser/click/7/userurl/Cognation  
http://click.mexuar.com/webuser/nojs/7/userurl/Cognation 
 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Wednesday, 16 May 2007 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk manager interface stability

 

There are many past posts stating that AMI is not stable when multiple
client computers are allowed to connect, particularly when connections
are dropped.

 

Has much progress been made on this? Is it more stable now than in the
past?

 

As of what versions were these issues improved?

 

Is it feasible to connect a large number of windows computers directly
to AMI for the purpose of initiating calls from software?

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Re: [asterisk-users] asterisk manager interface stability

[Matt Florell]

The issue has more to do with the sheer amount of data passed to the client
from within the Asterisk application when you have 50-100+ clients connected
to the AMI on full output mode. Running a system with FreePBX/Trixbox
especially generates vast amounts of output that has to be generated on
every AMI connection for every client. This is not trivial and can result in
lockups very easily, although this has gotten much better since the early
1.0 versions.

The new Asterisk Manager web API in 1.4 is a good step where sending of
Actions does not require an actual Telnet conneciton to the AMI, but I think
to be able to handle larger numbers of concurrent connections that a
separate send-only and a separate receive-only type of interface be built
where Asterisk would just output all AMI data to a single server-like
application that would then broadcast it to all connected clients. This
would remove the burden of so many connections going directly into Asterisk
and would allow for much larger scaling of AMI-type applications that
require real-time output of AMI events.

As for how to go about doing this, I can't help you there. I did build a
very specialized version of something like this 4 years ago for the
astGUIclient project called the Asterisk Central Queue System(ACQS) It is
based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in
what it does, but it does scale much better than using direct AMI
connections.
http://astguiclient.sourceforge.net/acqs.html

MATT---


On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote:


If it's not stable what needs to be done to improve this? What are the
issues? What are the alternatives (eg is Adhearsion an alternative here)



I am about to start looking into a project that requires every user to
have AMI access so looking to fund development in this space.





Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph

[image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognation



--

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Damon Estep
*Sent:* Wednesday, 16 May 2007 7:32 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] asterisk manager interface stability



There are many past posts stating that AMI is not stable when multiple
client computers are allowed to connect, particularly when connections are
dropped.



Has much progress been made on this? Is it more stable now than in the
past?



As of what versions were these issues improved?



Is it feasible to connect a large number of windows computers directly to
AMI for the purpose of initiating calls from software?

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Re: [asterisk-users] asterisk manager interface stability

[Bryan Laird]

why not do it via an snmp interface?

If you spend the time building an solid snmp base you would open up  
for an easier world of custom gui's as well as possibly some cleaner  
ties into an nms infrastructure.



On May 16, 2007, at 10:14 AM, Matt Florell wrote:

The issue has more to do with the sheer amount of data passed to  
the client from within the Asterisk application when you have 50-100 
+ clients connected to the AMI on full output mode. Running a  
system with FreePBX/Trixbox especially generates vast amounts of  
output that has to be generated on every AMI connection for every  
client. This is not trivial and can result in lockups very easily,  
although this has gotten much better since the early 1.0 versions.


The new Asterisk Manager web API in 1.4 is a good step where  
sending of Actions does not require an actual Telnet conneciton to  
the AMI, but I think to be able to handle larger numbers of  
concurrent connections that a separate send-only and a separate  
receive-only type of interface be built where Asterisk would just  
output all AMI data to a single server-like application that would  
then broadcast it to all connected clients. This would remove the  
burden of so many connections going directly into Asterisk and  
would allow for much larger scaling of AMI-type applications that  
require real-time output of AMI events.


As for how to go about doing this, I can't help you there. I did  
build a very specialized version of something like this 4 years ago  
for the astGUIclient project called the Asterisk Central Queue  
System(ACQS) It is based on 1.0 Asterisk but it still works with  
1.2 and 1.4. It is limited in what it does, but it does scale much  
better than using direct AMI connections.

http://astguiclient.sourceforge.net/acqs.html

MATT---


On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote:
If it's not stable what needs to be done to improve this? What are  
the issues? What are the alternatives (eg is Adhearsion an  
alternative here)



I am about to start looking into a project that requires every user  
to have AMI access so looking to fund development in this space.




Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph





From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] ] On Behalf Of Damon Estep

Sent: Wednesday, 16 May 2007 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk manager interface stability


There are many past posts stating that AMI is not stable when  
multiple client computers are allowed to connect, particularly when  
connections are dropped.



Has much progress been made on this? Is it more stable now than in  
the past?



As of what versions were these issues improved?


Is it feasible to connect a large number of windows computers  
directly to AMI for the purpose of initiating calls from software?



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-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird, Sr. Manager CM Operations
Phone: 703-944-9909
   -+-
Cablemodems are the gateway to the Internet.
The Internet is a gateway to some things that are  better left un-seen.


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Re: [asterisk-users] asterisk manager interface stability

[Jon Pounder]

Quoting Bryan Laird [EMAIL PROTECTED]:


why not do it via an snmp interface?

If you spend the time building an solid snmp base you would open up
for an easier world of custom gui's as well as possibly some cleaner
ties into an nms infrastructure.


you have my vote on that implementation method.

snmp really is simple, but it seems to be a neglected protocol that  
has been around for a long time.






On May 16, 2007, at 10:14 AM, Matt Florell wrote:

The issue has more to do with the sheer amount of data passed to
the client from within the Asterisk application when you have   
50-100 + clients connected to the AMI on full output mode. Running   
a  system with FreePBX/Trixbox especially generates vast amounts of  
  output that has to be generated on every AMI connection for every  
  client. This is not trivial and can result in lockups very  
easily,   although this has gotten much better since the early 1.0  
versions.


The new Asterisk Manager web API in 1.4 is a good step where
sending of Actions does not require an actual Telnet conneciton to   
 the AMI, but I think to be able to handle larger numbers of
concurrent connections that a separate send-only and a separate
receive-only type of interface be built where Asterisk would just
output all AMI data to a single server-like application that would   
 then broadcast it to all connected clients. This would remove the   
 burden of so many connections going directly into Asterisk and
would allow for much larger scaling of AMI-type applications that
require real-time output of AMI events.


As for how to go about doing this, I can't help you there. I did
build a very specialized version of something like this 4 years ago  
  for the astGUIclient project called the Asterisk Central Queue
System(ACQS) It is based on 1.0 Asterisk but it still works with
1.2 and 1.4. It is limited in what it does, but it does scale much   
 better than using direct AMI connections.

http://astguiclient.sourceforge.net/acqs.html

MATT---


On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote:
If it's not stable what needs to be done to improve this? What are   
 the issues? What are the alternatives (eg is Adhearsion an
alternative here)



I am about to start looking into a project that requires every user  
  to have AMI access so looking to fund development in this space.




Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph





From: [EMAIL PROTECTED] [mailto:asterisk-   
[EMAIL PROTECTED] ] On Behalf Of Damon Estep

Sent: Wednesday, 16 May 2007 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk manager interface stability


There are many past posts stating that AMI is not stable when
multiple client computers are allowed to connect, particularly when  
  connections are dropped.



Has much progress been made on this? Is it more stable now than in   
 the past?



As of what versions were these issues improved?


Is it feasible to connect a large number of windows computers
directly to AMI for the purpose of initiating calls from software?



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 http://lists.digium.com/mailman/listinfo/asterisk-users



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-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird, Sr. Manager CM Operations
Phone: 703-944-9909
   -+-
Cablemodems are the gateway to the Internet.
The Internet is a gateway to some things that are  better left un-seen.




Jon Pounder

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Re: [asterisk-users] asterisk manager interface stability

[Lee Jenkins]

Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the 
client from within the Asterisk application when you have 50-100+ 
clients connected to the AMI on full output mode. Running a system with 
FreePBX/Trixbox especially generates vast amounts of output that has to 
be generated on every AMI connection for every client. This is not 
trivial and can result in lockups very easily, although this has gotten 
much better since the early 1.0 versions.


The new Asterisk Manager web API in 1.4 is a good step where sending of 
Actions does not require an actual Telnet conneciton to the AMI, but I 
think to be able to handle larger numbers of concurrent connections that 
a separate send-only and a separate receive-only type of interface be 
built where Asterisk would just output all AMI data to a single 
server-like application that would then broadcast it to all connected 
clients. This would remove the burden of so many connections going 
directly into Asterisk and would allow for much larger scaling of 
AMI-type applications that require real-time output of AMI events.




I definitely agree here personally.  Clients could connect to this 
proxy and subscribe to only the events that are interesting or applicable.


As for how to go about doing this, I can't help you there. I did build a 
very specialized version of something like this 4 years ago for the 
astGUIclient project called the Asterisk Central Queue System(ACQS) It 
is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is 
limited in what it does, but it does scale much better than using direct 
AMI connections.


I've been considering writing something like this for a project that I'm 
thinking about doing that would require potentially high number of 
concurrent clients to consume AMI services.


From your experience, does the software that you wrote require 
significant CPU to cache and then doll out the kind of volume of 
messages that AMI can send?


If I end up writing something myself, I'll release it as OS...
--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk manager interface stability

[Earl Terwilliger]

On Wednesday 16 May 2007 11:43, Lee Jenkins wrote:
 Matt Florell wrote:
  The issue has more to do with the sheer amount of data passed to the
  client from within the Asterisk application when you have 50-100+
  clients connected to the AMI on full output mode. Running a system with
  FreePBX/Trixbox especially generates vast amounts of output that has to
  be generated on every AMI connection for every client. This is not
  trivial and can result in lockups very easily, although this has gotten
  much better since the early 1.0 versions.
 
  The new Asterisk Manager web API in 1.4 is a good step where sending of
  Actions does not require an actual Telnet conneciton to the AMI, but I
  think to be able to handle larger numbers of concurrent connections that
  a separate send-only and a separate receive-only type of interface be
  built where Asterisk would just output all AMI data to a single
  server-like application that would then broadcast it to all connected
  clients. This would remove the burden of so many connections going
  directly into Asterisk and would allow for much larger scaling of
  AMI-type applications that require real-time output of AMI events.

 I definitely agree here personally.  Clients could connect to this
 proxy and subscribe to only the events that are interesting or
 applicable.

  As for how to go about doing this, I can't help you there. I did build a
  very specialized version of something like this 4 years ago for the
  astGUIclient project called the Asterisk Central Queue System(ACQS) It
  is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is
  limited in what it does, but it does scale much better than using direct
  AMI connections.

 I've been considering writing something like this for a project that I'm
 thinking about doing that would require potentially high number of
 concurrent clients to consume AMI services.

  From your experience, does the software that you wrote require
 significant CPU to cache and then doll out the kind of volume of
 messages that AMI can send?

 If I end up writing something myself, I'll release it as OS...

You might be interested in a python server script I wrote (called ProxyMan) 
that does this kind of thing. It is part of my EventMonitor package but runs 
fine on its own.

#A multi-threaded server which connects to an Asterisk Manager
#and logs all events
#
#Connects to the Asterisk Manager and listens for all events
#Optionally listens on socket and accepts client connections
# proxies all client commands to the Asterisk Manager Interface
# sends all data received from the manager to all connected clients
#Optionally prints data as received (also in optional hex dump format)
#Optionally logs all data to a MySQL database table


 You can get it here:

http://www.micpc.com/eventmonitor

earl
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RE: [asterisk-users] asterisk manager interface stability

[Martin Smith]

You guys all sound like you're talking about AstManProxy.

See:
http://www.voip-info.org/tiki-index.php?page=AstManProxy


I'm not saying it is the solution to your problem per se, but I can't
help but think of it when I read the descriptions of what people want
(you even use the word proxy!). Figured I'd send this out in case
someone hadn't seen it.

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Lee Jenkins
 Subject: Re: [asterisk-users] asterisk manager interface stability
 
  The new Asterisk Manager web API in 1.4 is a good step 
 where sending of 
  Actions does not require an actual Telnet conneciton to the 
 AMI, but I 
  think to be able to handle larger numbers of concurrent 
 connections that 
  a separate send-only and a separate receive-only type of 
 interface be 
  built where Asterisk would just output all AMI data to a single 
  server-like application that would then broadcast it to all 
 connected 
  clients. This would remove the burden of so many connections going 
  directly into Asterisk and would allow for much larger scaling of 
  AMI-type applications that require real-time output of AMI events.
 
 
 I definitely agree here personally.  Clients could connect to this 
 proxy and subscribe to only the events that are interesting 
 or applicable.
 
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Re: [asterisk-users] asterisk manager interface stability

[Matt Florell]

On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:

Matt Florell wrote:
 The issue has more to do with the sheer amount of data passed to the
 client from within the Asterisk application when you have 50-100+
 clients connected to the AMI on full output mode. Running a system with
 FreePBX/Trixbox especially generates vast amounts of output that has to
 be generated on every AMI connection for every client. This is not
 trivial and can result in lockups very easily, although this has gotten
 much better since the early 1.0 versions.

 The new Asterisk Manager web API in 1.4 is a good step where sending of
 Actions does not require an actual Telnet conneciton to the AMI, but I
 think to be able to handle larger numbers of concurrent connections that
 a separate send-only and a separate receive-only type of interface be
 built where Asterisk would just output all AMI data to a single
 server-like application that would then broadcast it to all connected
 clients. This would remove the burden of so many connections going
 directly into Asterisk and would allow for much larger scaling of
 AMI-type applications that require real-time output of AMI events.


I definitely agree here personally.  Clients could connect to this
proxy and subscribe to only the events that are interesting or applicable.

 As for how to go about doing this, I can't help you there. I did build a
 very specialized version of something like this 4 years ago for the
 astGUIclient project called the Asterisk Central Queue System(ACQS) It
 is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is
 limited in what it does, but it does scale much better than using direct
 AMI connections.

I've been considering writing something like this for a project that I'm
thinking about doing that would require potentially high number of
concurrent clients to consume AMI services.

 From your experience, does the software that you wrote require
significant CPU to cache and then doll out the kind of volume of
messages that AMI can send?


One of the great parts about removing the broadcasting of AMI events
outside of the Asterisk process is that the broadcast server process
can exist on a separate physical server removing any kind of overhead
on the Asterisk server.

In my experience doing the proxy on the same machine uses less CPU
resources than the same number of AMI connected clients, and doesn't
have any of the deadlock issues that can happen with a lot of direct
AMI connections.

For my application(ACQS) I use MySQL as a storage engine for all of
the recent events received and sent so that they can be independantly
queried by any client apps that need to see them.

MATT---


If I end up writing something myself, I'll release it as OS...
--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk manager interface stability

[Matt Florell]

How does the Events cache in AstManProxy work?(is there a cache?)

MATT---

On 5/16/07, Martin Smith [EMAIL PROTECTED] wrote:

You guys all sound like you're talking about AstManProxy.

See:
http://www.voip-info.org/tiki-index.php?page=AstManProxy


I'm not saying it is the solution to your problem per se, but I can't
help but think of it when I read the descriptions of what people want
(you even use the word proxy!). Figured I'd send this out in case
someone hadn't seen it.

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Lee Jenkins
 Subject: Re: [asterisk-users] asterisk manager interface stability

  The new Asterisk Manager web API in 1.4 is a good step
 where sending of
  Actions does not require an actual Telnet conneciton to the
 AMI, but I
  think to be able to handle larger numbers of concurrent
 connections that
  a separate send-only and a separate receive-only type of
 interface be
  built where Asterisk would just output all AMI data to a single
  server-like application that would then broadcast it to all
 connected
  clients. This would remove the burden of so many connections going
  directly into Asterisk and would allow for much larger scaling of
  AMI-type applications that require real-time output of AMI events.
 

 I definitely agree here personally.  Clients could connect to this
 proxy and subscribe to only the events that are interesting
 or applicable.

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Re: [asterisk-users] Asterisk Manager and Ruby

[Alan Ferrency]

(Sorry for the way-late response to this short thread...)

We use rami in production on an Asterisk 1.2.3 server, and have had
basically zero problems at least since 1.2.3 was released.

rami and ruby's built in RPC provide a very easy to use proxy, if you
have multiple clients which all need access to the AMI as we do.

But yeah, I'd expect Asterisk has diverged a lot since rami was last
updated. I did a round of refactoring at the time we were initially
developing our screen pop app, but none of it has had to change in
over a year.

Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]


On Wed, 1 Nov 2006, snacktime wrote:

 On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote:
  Hi,
 
  Any one using Rubi asterisk manager interface
  http://rubyforge.org/projects/rami/ ?
 
  How stable/usable it is?

 It probably hasn't seen much use.  I created that back when I was just
 learning ruby, so it could probably use some refactoring as well.
 And If anything has changed in the asterisk manager protocol that
 would be an issue also.  I created it against the beta version at the
 time, can't remember what that was.

 Chris
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Re: [asterisk-users] Asterisk Manager Interface: Auto-answer of'Originate' command

[Moises Silva]

On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:

So, to be clear, I have to do this in the application using the
management interface (which I don't happen to control) rather than in
the Asterisk dialplan (which I do)?

Yes, because the channel you originate does not exists in the PBX yet,
until answered. So, Im not sure how are you originating the channel,
but the interface originating the channel, should allow you to do this
type of things.



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Moises Silva
 Sent: 11 January 2007 17:40
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Manager Interface:
 Auto-answer of'Originate' command

 Please read the voip-info.org documentation regarding
 Originate action, there you will find how to set variables on
 the originate channel.

 On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
  Does anyone know of a way to make an originate request
 coming over the
  management interface (e.g. AstTapi click-to-dial) include
 the relevant
  Alert-Info SIP headers to enable the originating phone to
 auto-answer?
 
  I've tried setting up a custom context (see below), but the
 dial plan
  is only entered AFTER the originating call is answered, so the SIP
  header is added to the terminating call leg, not the
 originating call leg.
 
  [click-to-call-custom]
  exten = _X.,1,NoOp(Click to Call)
  exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
  exten = _X.,3,Goto(from-internal,${EXTEN},1)
 
  __
  Steve Langstaff
 
  Citel.
  The VoIP Migration Company.(tm)
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Re: [asterisk-users] Asterisk Manager Interface: Auto-answer of'Originate' command

[Steve Davies]

How about:

Action: originate
Channel: Local/[EMAIL PROTECTED]
etc

Then in extensions.conf

[indirect]
exten = _X.,1,NoOp(Click to Call)
exten = _X.,n,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _X.,n,Goto(from-internal,${EXTEN},1)

Get the idea? Does that help?

Cheers,
Steve

On 1/12/07, Moises Silva [EMAIL PROTECTED] wrote:

On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
 So, to be clear, I have to do this in the application using the
 management interface (which I don't happen to control) rather than in
 the Asterisk dialplan (which I do)?
Yes, because the channel you originate does not exists in the PBX yet,
until answered. So, Im not sure how are you originating the channel,
but the interface originating the channel, should allow you to do this
type of things.


  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Moises Silva
  Sent: 11 January 2007 17:40
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Asterisk Manager Interface:
  Auto-answer of'Originate' command
 
  Please read the voip-info.org documentation regarding
  Originate action, there you will find how to set variables on
  the originate channel.
 
  On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
   Does anyone know of a way to make an originate request
  coming over the
   management interface (e.g. AstTapi click-to-dial) include
  the relevant
   Alert-Info SIP headers to enable the originating phone to
  auto-answer?
  
   I've tried setting up a custom context (see below), but the
  dial plan
   is only entered AFTER the originating call is answered, so the SIP
   header is added to the terminating call leg, not the
  originating call leg.
  
   [click-to-call-custom]
   exten = _X.,1,NoOp(Click to Call)
   exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
   exten = _X.,3,Goto(from-internal,${EXTEN},1)
  
   __
   Steve Langstaff
  

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RE: [asterisk-users] Asterisk Manager Interface: Auto-answerof'Originate' command

[Steve Langstaff]

That's better - the originating phone now auto-answers. Thanks.

Unfortunately the terminating phone also auto-answers, so I guess I've
got to find out how to not inherit the ALERT_INFO variable across the
channels (I've tried ALERT_INFO without the '_' prefix, but then the
originating phone does not get the variable.


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steve Davies
 Sent: 12 January 2007 15:57
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Manager Interface: 
 Auto-answerof'Originate' command
 
 How about:
 
 Action: originate
 Channel: Local/[EMAIL PROTECTED]
 etc
 
 Then in extensions.conf
 
 [indirect]
 exten = _X.,1,NoOp(Click to Call)
 exten = _X.,n,SetVar(_ALERT_INFO=info=alert-autoanswer)
 exten = _X.,n,Goto(from-internal,${EXTEN},1)
 
 Get the idea? Does that help?
 
 Cheers,
 Steve
 
 On 1/12/07, Moises Silva [EMAIL PROTECTED] wrote:
  On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
   So, to be clear, I have to do this in the application using the 
   management interface (which I don't happen to control) 
 rather than 
   in the Asterisk dialplan (which I do)?
  Yes, because the channel you originate does not exists in 
 the PBX yet, 
  until answered. So, Im not sure how are you originating the 
 channel, 
  but the interface originating the channel, should allow you 
 to do this 
  type of things.
 
  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Moises Silva
Sent: 11 January 2007 17:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager Interface:
Auto-answer of'Originate' command
   
Please read the voip-info.org documentation regarding Originate 
action, there you will find how to set variables on the 
 originate 
channel.
   
On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
 Does anyone know of a way to make an originate request
coming over the
 management interface (e.g. AstTapi click-to-dial) include
the relevant
 Alert-Info SIP headers to enable the originating phone to
auto-answer?

 I've tried setting up a custom context (see below), but the
dial plan
 is only entered AFTER the originating call is 
 answered, so the 
 SIP header is added to the terminating call leg, not the
originating call leg.

 [click-to-call-custom]
 exten = _X.,1,NoOp(Click to Call) exten = 
 _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
 exten = _X.,3,Goto(from-internal,${EXTEN},1)

 __
 Steve Langstaff

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RE: [asterisk-users] Asterisk Manager Interface: Auto-answerof'Originate' command

[Steve Langstaff]

[Replying to my own post, but it's good news!]

Yipee - sorted.

I needed 2 contexts:

[click-to-call-originate-custom]
exten = _X.,1,NoOp(Click to Call Originator)
exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _X.,3,Goto(from-internal,${EXTEN},1)

[click-to-call-target-custom]
exten = _X.,1,NoOp(Click to Call Target)
exten = _X.,2,Goto(from-internal,${EXTEN},1)

In AstTapi I needed to configure:

User Channel: Local/[EMAIL PROTECTED]
Dial By Context: checked
Context: click-to-call-target-custom

Thanks Steve.

 -Original Message-
 From: Steve Langstaff 
 Sent: 12 January 2007 16:43
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [asterisk-users] Asterisk Manager Interface: 
 Auto-answerof'Originate' command
 
 That's better - the originating phone now auto-answers. Thanks.
 
 Unfortunately the terminating phone also auto-answers, so I 
 guess I've got to find out how to not inherit the ALERT_INFO 
 variable across the channels (I've tried ALERT_INFO without 
 the '_' prefix, but then the originating phone does not get 
 the variable.
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Steve 
  Davies
  Sent: 12 January 2007 15:57
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Asterisk Manager Interface: 
  Auto-answerof'Originate' command
  
  How about:
  
  Action: originate
  Channel: Local/[EMAIL PROTECTED]
  etc
  
  Then in extensions.conf
  
  [indirect]
  exten = _X.,1,NoOp(Click to Call)
  exten = _X.,n,SetVar(_ALERT_INFO=info=alert-autoanswer)
  exten = _X.,n,Goto(from-internal,${EXTEN},1)
  
  Get the idea? Does that help?
  
  Cheers,
  Steve
  
  On 1/12/07, Moises Silva [EMAIL PROTECTED] wrote:
   On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
So, to be clear, I have to do this in the application using the 
management interface (which I don't happen to control)
  rather than
in the Asterisk dialplan (which I do)?
   Yes, because the channel you originate does not exists in
  the PBX yet,
   until answered. So, Im not sure how are you originating the
  channel,
   but the interface originating the channel, should allow you
  to do this
   type of things.
  
   
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Moises Silva
 Sent: 11 January 2007 17:40
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Manager Interface:
 Auto-answer of'Originate' command

 Please read the voip-info.org documentation regarding 
 Originate 
 action, there you will find how to set variables on the
  originate
 channel.

 On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
  Does anyone know of a way to make an originate request
 coming over the
  management interface (e.g. AstTapi click-to-dial) include
 the relevant
  Alert-Info SIP headers to enable the originating phone to
 auto-answer?
 
  I've tried setting up a custom context (see below), but the
 dial plan
  is only entered AFTER the originating call is
  answered, so the
  SIP header is added to the terminating call leg, not the
 originating call leg.
 
  [click-to-call-custom]
  exten = _X.,1,NoOp(Click to Call) exten =
  _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
  exten = _X.,3,Goto(from-internal,${EXTEN},1)
 
  __ Steve Langstaff
 
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Re: [asterisk-users] Asterisk Manager Interface: Auto-answer of 'Originate' command

[Moises Silva]

Please read the voip-info.org documentation regarding Originate
action, there you will find how to set variables on the originate
channel.

On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:

Does anyone know of a way to make an originate request coming over the
management interface (e.g. AstTapi click-to-dial) include the relevant
Alert-Info SIP headers to enable the originating phone to auto-answer?

I've tried setting up a custom context (see below), but the dial plan is
only entered AFTER the originating call is answered, so the SIP header
is added to the terminating call leg, not the originating call leg.

[click-to-call-custom]
exten = _X.,1,NoOp(Click to Call)
exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _X.,3,Goto(from-internal,${EXTEN},1)

__
Steve Langstaff

Citel.
The VoIP Migration Company.(tm)
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RE: [asterisk-users] Asterisk Manager Interface: Auto-answer of'Originate' command

[Steve Langstaff]

So, to be clear, I have to do this in the application using the
management interface (which I don't happen to control) rather than in
the Asterisk dialplan (which I do)?

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Moises Silva
 Sent: 11 January 2007 17:40
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Manager Interface: 
 Auto-answer of'Originate' command
 
 Please read the voip-info.org documentation regarding 
 Originate action, there you will find how to set variables on 
 the originate channel.
 
 On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
  Does anyone know of a way to make an originate request 
 coming over the 
  management interface (e.g. AstTapi click-to-dial) include 
 the relevant 
  Alert-Info SIP headers to enable the originating phone to 
 auto-answer?
 
  I've tried setting up a custom context (see below), but the 
 dial plan 
  is only entered AFTER the originating call is answered, so the SIP 
  header is added to the terminating call leg, not the 
 originating call leg.
 
  [click-to-call-custom]
  exten = _X.,1,NoOp(Click to Call)
  exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
  exten = _X.,3,Goto(from-internal,${EXTEN},1)
 
  __
  Steve Langstaff
 
  Citel.
  The VoIP Migration Company.(tm)
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 http://www.gnu.org;
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RE: [asterisk-users] Asterisk Manager

[Jonathan k. Creasy]

  CLIPPED
 I would have some kind of user 1010 (the actual extension and username
 too)
 Let's say that in manager.conf i would have again some user 1010 but i
 would like that this user can only see the events associated to the
 extension 1010 ...
  CLIPPED

I am pretty sure that using the proxy, astmanproxy, you can achieve this
goal. It is recommended to use the proxy so that there is only one
connection to the server and all the other applications will connect to
the proxy. 

-Jonathan
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Re: [asterisk-users] Asterisk Manager

[Daniel Gradecak]

Hello Jonathan, thank you for answering ...

I read about astmanproxy but it cannot help me. I am using asterisk-java 
all my application is written in java too. I already have a kind of 
proxy ad I am not doing
several connection to the asterisk manager. I am afraid this is not 
helping me much. Anyway, I have done this in my proxy but i thought i 
could avoid things like that in my code...


I did not test the asterisk manager contexts and dial plan, so I wonder 
if I make a call via astman from 1010 to a GSM and that 1010 is in a 
context that is not allowing calls to GSM
would astman execute it anyway or would it look also in the 1010 
context? I am asking that because my system guys are not available until 
friday ...


Jonathan k. Creasy wrote:

 CLIPPED
I would have some kind of user 1010 (the actual extension and username
too)
Let's say that in manager.conf i would have again some user 1010 but i
would like that this user can only see the events associated to the
extension 1010 ...
 CLIPPED



I am pretty sure that using the proxy, astmanproxy, you can achieve this
goal. It is recommended to use the proxy so that there is only one
connection to the server and all the other applications will connect to
the proxy. 


-Jonathan
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RE: [asterisk-users] Asterisk Manager

[Jonathan k. Creasy]

Not meaning to argue with you but the proxy replaces the manager
interface so it could most likely be a seamless replacement to your
application. It was for all but one of my applications and the problem
there was in the way I parsed the startup string. 


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Daniel Gradecak
 Sent: Tuesday, December 12, 2006 1:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Manager
 
 Hello Jonathan, thank you for answering ...
 
 I read about astmanproxy but it cannot help me. I am using
asterisk-java
 all my application is written in java too. I already have a kind of
 proxy ad I am not doing
 several connection to the asterisk manager. I am afraid this is not
 helping me much. Anyway, I have done this in my proxy but i thought
i
 could avoid things like that in my code...
 
 I did not test the asterisk manager contexts and dial plan, so I
wonder
 if I make a call via astman from 1010 to a GSM and that 1010 is in a
 context that is not allowing calls to GSM
 would astman execute it anyway or would it look also in the 1010
 context? I am asking that because my system guys are not available
until
 friday ...
 
 Jonathan k. Creasy wrote:
   CLIPPED
  I would have some kind of user 1010 (the actual extension and
username
  too)
  Let's say that in manager.conf i would have again some user 1010
but i
  would like that this user can only see the events associated to the
  extension 1010 ...
   CLIPPED
 
 
  I am pretty sure that using the proxy, astmanproxy, you can achieve
this
  goal. It is recommended to use the proxy so that there is only one
  connection to the server and all the other applications will connect
to
  the proxy.
 
  -Jonathan
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Re: [asterisk-users] Asterisk Manager

[Tim Panton]

On 12 Dec 2006, at 16:27, Daniel Gradecak wrote:


Hello,

I am not an asterisk expert but i am developing a web application that
is using asterisk. I would like to know if it is possible to  
configure a

Manager to only monitor a special
extension, and of course how to do that.

The application is written in java and is using asterisk-java.  
Right now

i have one manager that i am connected to and i receive all the events
but i would like to have some kind of administrator
and user. The administrator manager can receive all events but the
normal user (agent) should only receive the events that are associated
to its extension.

I would have some kind of user 1010 (the actual extension and  
username too)

Let's say that in manager.conf i would have again some user 1010 but i
would like that this user can only see the events associated to the
extension 1010 ...

Does it makes any sens, and how to do that?


The manager doesn't have any filters - per-se.
You would need to add a layer in your asterisk-java program that  
filtered

the channels/extensions you were interested in.
The easiest thing might be to have your manager layer put the
events into a lightweight (in memory?) database, then use some standard
JDBC/servlets (or whatever) to query those events using the
channel current user's as a key.

Now, depending on what you are trying to do, there may be other ways
to get there

Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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RE: [asterisk-users] Asterisk manager

[Ed Nuñez]

Your line number nine should also specify a file name to monitor to and the 
format, like this

exten = 9,2,Monitor(from-${CALLEDID}-at-${TIMESTAMP},wav)

or better yet, use MixMon instead, because this will merge the two files into 
just one.  (both sides of the call)

Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Tuesday, December 12, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk manager

Hi

i am trying to record a call with

exten = 9,1,Answer
exten = 9,2,Monitor
exten = 9,3,Dial(SIP/200)

This will record the call, but asterisk generates 2 files in
/var/spool/asterisk/monitor/

-in.wav
-out.wav

Can i have only one file?
Can i customize the path where to save the files?

Thanks
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Re: [asterisk-users] Asterisk manager

[Pavel Jezek]

 -= Info about application 'MixMonitor' =-

[Synopsis]
Record a call and mix the audio during the recording

[Description]
 MixMonitor(file.ext[|options[|command]])

Records the audio on the current channel to the specified file.
If the filename is an absolute path, uses that path, otherwise
creates the file in the configured monitoring directory from
asterisk.conf.




nik600 wrote:

Hi

i am trying to record a call with

exten = 9,1,Answer
exten = 9,2,Monitor
exten = 9,3,Dial(SIP/200)

This will record the call, but asterisk generates 2 files in
/var/spool/asterisk/monitor/

-in.wav
-out.wav

Can i have only one file?
Can i customize the path where to save the files?

Thanks
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RE: [asterisk-users] asterisk manager originate command

[Michael Collins]

 I want to know how to get the uniqueid or a call started from asterisk
 manager using Originate command.

Are you wanting the uniqueid for the call right after it is started,
i.e., while it is still in progress?  What is in your Dial command?

-MC
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Re: [asterisk-users] asterisk manager originate command

[Rodrigo Gonzalez]

There is no dial command, I'm sending originate action from asterisk 
manager.


Michael Collins wrote:

I want to know how to get the uniqueid or a call started from asterisk
manager using Originate command.



Are you wanting the uniqueid for the call right after it is started,
i.e., while it is still in progress?  What is in your Dial command?

-MC
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RE: [asterisk-users] asterisk manager originate command

[Michael Collins]

 There is no dial command, I'm sending originate action from asterisk
 manager.

Oops, I didn't ask my question correctly.  You're right, it isn't a
dial command.  What I wanted to know was the contents of your
originate action, e.g.:

Channel= 'zap/g0/' . $dialed_num

(From one of my Perl scripts using
POE::Component::Client::Asterisk::Manager)

Thanks,
MC
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Re: [asterisk-users] asterisk manager originate command

[Rodrigo Gonzalez]

My code is using phpagi-asmanagerbut what is sent is...

Action: Originate
Channel: SIP/802
Context: from-internal
Exten:  number to dial 
Priority: 1
Callerid: 802



Michael Collins wrote:

There is no dial command, I'm sending originate action from asterisk
manager.



Oops, I didn't ask my question correctly.  You're right, it isn't a
dial command.  What I wanted to know was the contents of your
originate action, e.g.:

Channel= 'zap/g0/' . $dialed_num

(From one of my Perl scripts using
POE::Component::Client::Asterisk::Manager)

Thanks,
MC
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Re: [asterisk-users] asterisk manager originate command

[Moises Silva]

currently thats not possible unless you speciffy the async flag, in
that case  Event: OriginateSuccess or Event: OriginateFailed event
will be launched with the uniqueid

Regards

On 12/4/06, Rodrigo Gonzalez [EMAIL PROTECTED] wrote:

My code is using phpagi-asmanagerbut what is sent is...

Action: Originate
Channel: SIP/802
Context: from-internal
Exten:  number to dial 
Priority: 1
Callerid: 802



Michael Collins wrote:
 There is no dial command, I'm sending originate action from asterisk
 manager.


 Oops, I didn't ask my question correctly.  You're right, it isn't a
 dial command.  What I wanted to know was the contents of your
 originate action, e.g.:

 Channel= 'zap/g0/' . $dialed_num

 (From one of my Perl scripts using
 POE::Component::Client::Asterisk::Manager)

 Thanks,
 MC
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Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?

[Marco Mouta]

take a look at Flash Operator Panel, as far as i know they use AMI , and
they also provide real time channel status.

On 11/18/06, Michael Collins [EMAIL PROTECTED] wrote:


 I'm interested in knowing if anyone else has worked around this issue:



I have an application that needs to check the status of the calls going
through Asterisk about every 5 seconds or so.  I don't want to do asterisk
–rx 'show channels verbose' at the Linux command line 12 times per minute
so I am looking at the AMI.  I see that there isn't a manager command for
'show channels.'  Has anyone come up with an equivalent of 'show channels'
using the extant manager commands?  If so, could you post how you did it?



Thanks!

-MC

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Com os melhores cumprimentos,

Marco Mouta
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Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?

[Nico Busch]

As far as I know, you can do something like this, i.e. in PHP:
fputs($socket, Action: Command\r\n);
fputs($socket, Command: show channels concise\r\n);

I hope that helps, but perhaps there is a better possibility.

NB


Michael Collins schrieb:


I’m interested in knowing if anyone else has worked around this issue:

I have an application that needs to check the status of the calls 
going through Asterisk about every 5 seconds or so. I don’t want to do 
“asterisk –rx ‘show channels verbose’” at the Linux command line 12 
times per minute so I am looking at the AMI. I see that there isn’t a 
manager command for ‘show channels.’ Has anyone come up with an 
equivalent of ‘show channels’ using the extant manager commands? If 
so, could you post how you did it?


Thanks!

-MC



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Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?

[Earl Terwilliger]

On Saturday 18 November 2006 13:28, Michael Collins wrote:
 I'm interested in knowing if anyone else has worked around this issue:



 I have an application that needs to check the status of the calls going
 through Asterisk about every 5 seconds or so.  I don't want to do
 asterisk -rx 'show channels verbose' at the Linux command line 12
 times per minute so I am looking at the AMI.  I see that there isn't a
 manager command for 'show channels.'  Has anyone come up with an
 equivalent of 'show channels' using the extant manager commands?  If so,
 could you post how you did it?



 Thanks!

 -MC

An approach i take with my event monitor is to have a manager application 
register for all events, then when something in asterisk happens you get an  
event record. No need to 'poll' to check. Then you can easily determine what 
is going on. 

http://micpc.com/eventmonitor uses the event records to display the state of 
phones, channels, etc. 'realtime' with AJAX without polling.


earl


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Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?

[Richard Lyman]

Michael Collins wrote:


I’m interested in knowing if anyone else has worked around this issue:

I have an application that needs to check the status of the calls 
going through Asterisk about every 5 seconds or so. I don’t want to do 
“asterisk –rx ‘show channels verbose’” at the Linux command line 12 
times per minute so I am looking at the AMI. I see that there isn’t a 
manager command for ‘show channels.’ Has anyone come up with an 
equivalent of ‘show channels’ using the extant manager commands? If 
so, could you post how you did it?


Thanks!

-MC


you need to look again
maybe read, http://www.dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF

action: command
command: show channels

Response: Follows
Privilege: Command
Channel Location State Application(Data)
0 active channels
0 active calls
--END COMMAND--


goodluck

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RE: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?

[Michael Collins]

 you need to look again
 maybe read, http://www.dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF

Mea culpa!

I did not realize it was so easy!  I totally missed the command action
when looking over the docs.  Thanks for pointing out the mistake and
thanks too for the link.  It has helped a lot.

-MC
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RE: [asterisk-users] Asterisk Manager and Ruby

[Dominique Dartois]

I use it to learn ruby + Asterisk Manager.
A script is run every 10 seconds to create an HTML page to display the
codecs used by active channels. The script also makes rddtool generate a
graph.
This is not a production Asterisk. The script is runing for about a month.
(It's a quick and dirty written script).

Dominique.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de snacktime
Envoyé : jeudi 2 novembre 2006 08:38
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk Manager and Ruby

On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote:
 Hi,

 Any one using Rubi asterisk manager interface 
 http://rubyforge.org/projects/rami/ ?

 How stable/usable it is?

It probably hasn't seen much use.  I created that back when I was just
learning ruby, so it could probably use some refactoring as well.
And If anything has changed in the asterisk manager protocol that would be
an issue also.  I created it against the beta version at the time, can't
remember what that was.

Chris
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RE: [asterisk-users] Asterisk manager

[Ed Nuñez]

Sorry, but I failed to mention that I am running Asterisk BE B 1-1



I am trying to send commands to Asterisk manager via a telnet session.  I am 
able to lo in and receive event logs from AMI, but when I try to issue commands 
I get an invalid/unknown command error.  Here are some of the commands I am 
trying to send.

Asterisk Call Manager/1.0
Action: login
Username: xxx
Secret: x

Response: Success
Message: Authentication accepted

ACTION: Originate 
Channel: Local/1656
Exten: 1710
Priority: 1 
Context: it

Response: Error
Message: Invalid/unknown command



ACTION: Command 
command: show dialplan 

Response: Error
Message: Invalid/unknown command



Action: Originate 
Channel: Zap/g1/17329250730 
Context: default 
Exten: 1656
Priority: 1 
Callerid: 3125551212 

Response: Error
Message: Invalid/unknown command


Here is how my manager.conf file looks

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects = yes

[ami]
secret = XXX
permit=0.0.0.0/0.0.0.0
;deny=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user


Any help would be greatly appresiated


Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730


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Re: [asterisk-users] Asterisk manager

[Dovid B]

I remember when I started working with the manager (I was using VB) that you 
have to send a string of characters that you are going to the next line. I 
am not sure if this will solve the issue. (sorry if my response dosent make 
much sense - a bit on the tired side)



- Original Message - 
From: Ed Nuñez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, November 01, 2006 6:50 PM
Subject: RE: [asterisk-users] Asterisk manager


Sorry, but I failed to mention that I am running Asterisk BE B 1-1



I am trying to send commands to Asterisk manager via a telnet session.  I am 
able to lo in and receive event logs from AMI, but when I try to issue 
commands I get an invalid/unknown command error.  Here are some of the 
commands I am trying to send.


Asterisk Call Manager/1.0
Action: login
Username: xxx
Secret: x

Response: Success
Message: Authentication accepted

ACTION: Originate
Channel: Local/1656
Exten: 1710
Priority: 1
Context: it

Response: Error
Message: Invalid/unknown command



ACTION: Command
command: show dialplan

Response: Error
Message: Invalid/unknown command



Action: Originate
Channel: Zap/g1/17329250730
Context: default
Exten: 1656
Priority: 1
Callerid: 3125551212

Response: Error
Message: Invalid/unknown command


Here is how my manager.conf file looks

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects = yes

[ami]
secret = XXX
permit=0.0.0.0/0.0.0.0
;deny=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user


Any help would be greatly appresiated


Ed Nuñez
IT/Telecom Engineer

4037 Metric Drive
Winter Park, FL

(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730


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Re: [asterisk-users] Asterisk Manager and Ruby

[snacktime]

On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote:

Hi,

Any one using Rubi asterisk manager interface
http://rubyforge.org/projects/rami/ ?

How stable/usable it is?


It probably hasn't seen much use.  I created that back when I was just
learning ruby, so it could probably use some refactoring as well.
And If anything has changed in the asterisk manager protocol that
would be an issue also.  I created it against the beta version at the
time, can't remember what that was.

Chris
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Re: [asterisk-users] Asterisk Manager

[MapsAir]

Yes, I did check it. The asterisk.st1 is 
running at /var/run folder, I tried to change to /var/run/asterisk/asterisk.ct1, 
then I will have the error "Unable to connect to remote asterisk (does 
/var/run/asterisk/asterisk.ctl exist?" 

I thing it have something with the 
permission. How can I check what user running under the php? How can 
I specify the user for PHP? What user should I specify?

My 
/etc/asterisk/manager.conf has 

[admin]secret = 
passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = 
system,call,log,verbose,command,agent,userwrite = 
system,call,log,verbose,command,agent,user




  - Original Message - 
  From: 
  Lacy Moore - 
  Aspendora 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, October 25, 2006 11:34 
  AM
  Subject: Re: [asterisk-users] Asterisk 
  Manager
  
  
  

Asterisk is current running with the a file in 
/var/run/asterisk.ctl for the user asterisk. I have set asterisk to be 
the owner of the folder /var/run, and start asterisk with user is 
asterisk. HTTPD is run under asterisk user. My manager.conf has 
an entry.
  
  Check to make sure the file is actually /var/run/asterisk.ctl and not 
  /var/run/asterisk/asterisk.ctl.
  
  

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Re: [asterisk-users] Asterisk Manager

[Lacy Moore - Aspendora]

Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My 
manager.conf has an entry.

Check to make sure the file is actually /var/run/asterisk.ctl and not /var/run/asterisk/asterisk.ctl.
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Re: [asterisk-users] Asterisk manager

[Michel Vaillancourt]

Voipers Portugal wrote:
 I know that, that is why I asked if there was any tool that would do
 something like that, but by acessing the Manager API?
 
 Anyone?
 

Our interface uses ARI and MySQL.  There is no reason that you could 
not manage a secure box with the interface app, with MySQL replication of the 
master table out to a slave-only table on the exposed machine.  That way, 
there is really nothing on the exposed machine to compromise.  Go the extra 
step and SSL the replication channel and the only thing you'd have to have in 
the clear would be the SIP connections themselves.

-- 
--Michel Vaillancourt
Senior Telephony Engineer
Neoxo Inc  (www.neoxo.com)
+1 514 395 1106 ext 117
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RE: [asterisk-users] Asterisk manager

[Bill Gibbs]

www.freepbx.org



Bill











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voipers Portugal
Sent: Tuesday, October 03, 2006
8:22 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
manager





Dear all,

Do you know any tool that can administrate Asterisk remotely? I only need basic
functionalities like adding new extensions, queus and basic configuration. The
problem is that I can't install that in the same machine as Asterisk (since it
is running in open wrt). 

Can anyone help me out?

Jose Simoes 






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Re: [asterisk-users] Asterisk manager

[Voipers Portugal]

But that cannot be done remotly, can it? I can't load any more programs in my Asterisk machine. Does it support that?On 10/3/06, Bill Gibbs 
[EMAIL PROTECTED] wrote:
















www.freepbx.org




Bill











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Voipers Portugal
Sent: Tuesday, October 03, 2006
8:22 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
manager





Dear all,

Do you know any tool that can administrate Asterisk remotely? I only need basic
functionalities like adding new extensions, queus and basic configuration. The
problem is that I can't install that in the same machine as Asterisk (since it
is running in open wrt). 

Can anyone help me out?

Jose Simoes 







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RE: [asterisk-users] Asterisk manager

[Dean Collins]

No you cant install freepbx on a openwrt



The only choice you have is to ssh in
using putty or similar to add extensions and make conf mods.











Cheers,



Dean













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voipers Portugal
Sent: Tuesday, 3 October 2006 8:35
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Asterisk manager





But that cannot be done
remotly, can it? I can't load any more programs in my Asterisk machine. Does it
support that?



On 10/3/06, Bill
Gibbs 
[EMAIL PROTECTED] wrote:





www.freepbx.org




Bill











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Voipers Portugal
Sent: Tuesday, October 03, 2006
8:22 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
manager







Dear all,

Do you know any tool that can administrate Asterisk remotely? I only need basic
functionalities like adding new extensions, queus and basic configuration. The
problem is that I can't install that in the same machine as Asterisk (since it
is running in open wrt). 

Can anyone help me out?

Jose Simoes 










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 http://lists.digium.com/mailman/listinfo/asterisk-users 












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Re: [asterisk-users] Asterisk manager

[Voipers Portugal]

I know that, that is why I asked if there was any tool that would do something like that, but by acessing the Manager API?Anyone?On 10/3/06, Dean Collins
 [EMAIL PROTECTED] wrote:















No you cant install freepbx on a openwrt



The only choice you have is to ssh in
using putty or similar to add extensions and make conf mods.











Cheers,



Dean













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Voipers Portugal
Sent: Tuesday, 3 October 2006 8:35
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Asterisk manager





But that cannot be done
remotly, can it? I can't load any more programs in my Asterisk machine. Does it
support that?



On 10/3/06, Bill
Gibbs 
[EMAIL PROTECTED] wrote:





www.freepbx.org




Bill











From: 
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Voipers Portugal
Sent: Tuesday, October 03, 2006
8:22 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
manager







Dear all,

Do you know any tool that can administrate Asterisk remotely? I only need basic
functionalities like adding new extensions, queus and basic configuration. The
problem is that I can't install that in the same machine as Asterisk (since it
is running in open wrt). 

Can anyone help me out?

Jose Simoes 










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 http://lists.digium.com/mailman/listinfo/asterisk-users 













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Re: [asterisk-users] Asterisk Manager Interface Question

[Moises Silva]

100ms

On 8/28/06, Roi Stork [EMAIL PROTECTED] wrote:

The version that I'm using is 1.2.7.1.
What is the default value of writetimeout in manager.conf?



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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] Asterisk Manager Vb 6

[Dovid Bender]

I have basic code that see's what comes in but 
nothing much more. I was going to make it that when calls come in VB looks what 
comes up etc. and work from there. If you want it let me know.

  - Original Message - 
  From: 
  hernany.ce 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, July 31, 2006 2:49 PM
  Subject: [asterisk-users] Asterisk 
  Manager Vb 6
  
  
  
  Hi 
  everybody
  
  
  Does anyone here have 
  a sample code written in Visual Basic 6 that could share with me 
  ??
  I have to design a 
  Monitor Interface and I don´t know how to start 
  it.
  
  Thanks in 
  advance.
  
  Hernany 
  Oliveira
  
  
  

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