[asterisk-users] Experiences with Polycom-Kirk 6000 and DECT/GAP handsets

[Olivier]

Hi,

I would be curious to learn about experiences with  Polycom-Kirk 6000
and DECT/GAP handsets (mostly Gigaset handsets of all kinds) in a
multi-cells environment.

More precisely, what about roaming and handover ?

Regards

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Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

[SamyGo]

>
> Actually, if OP upgrades to Asterisk 10 they will get video conferencing
> with app_confbridge.


I think I'm not as much updated then and definitely am going to test this
application. Paul have you ever seen this application in action ! this is
going to be great then - built-in Video conference app.

On Tue, Apr 10, 2012 at 12:46 AM, Paul Belanger wrote:

> On 12-04-09 05:58 AM, SamyGo wrote:
>
>> Hi,
>>
>> Actually asterisk don't provide video conference. In simple terms, the
>> setting which zohair told just enables two end points to use video codecs
>> and establish a one-one video session for video capable phones.
>>
>> For making your asterisk do a video conferencing you may need to look into
>> Vmukti project 
>> http://sourceforge.net/**projects/vmukti/
>>
>> Please explain your requirements so anyone can help you in better way.
>>
>>  Actually, if OP upgrades to Asterisk 10 they will get video conferencing
> with app_confbridge.
>
> -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger |
> IRC: pabelanger (Freenode) Check us out at: http://digium.com &
> http://asterisk.org
>
>
>
>
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[asterisk-users] How to disable CDR adaptative logging on asterisk 1.8 ?

[Olivier]

Hi,

On a 1.8.10 system, I've got (with cdr show status) :

Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   No

* Registered Backends
  ---
csv
cdr-custom
Adaptive ODBC

The strange thing is that cdr.conf is almost empty (ie only a [csv]
section is uncommented and I've got no cdr_adaptative_odbc.conf file
(I renamed the one I add to  cdr_adaptative_odbc.conf.old).

How can I turn CDR Adaptive ODBC off ?

Regards

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[asterisk-users] Strange Asterisk port behavior

[Mikhail Lischuk]

 

Greetings! 

Recently I had to change the port Asterisk listens to
(non-standard, to hide from bruteforce attacks), but at the same time I
wanted to not break the system for all current users. So I needed some
way to listen to two ports for some time. 

I did some research in the
Internet and found the only one solution - via iptables REDIRECT 

For
some reason it was not working for me, and I found many discussions
saying that lots of people can't get it working either. 

Despite the
statistics for rule say that there are packets processed by the rule,
they did not reach the Asterisk. Moreover, the statistic is kind of
strange - only 8 packets per hour... is way too few for system with 100
active users, I guess. 

AND here starts the strange thing. Despite
statistics saying that so few packets are redirected to the new port,
almost all peers went up - with the new port. 

Then i get to tcpdump...
And I see some weird stuff: 

(A.A.A.A is client and B.B.B.B is
asterisk) 

14:41:38.506577 IP A.A.A.A.53082 > B.B.B.B.1: UDP,
length 512
14:41:38.506806 IP B.B.B.B > A.A.A.A: ICMP B.B.B.B udp port
1 unreachable, length 548 

Here ^^, some client trying to access
the old port, and getting Port Unreachable reply. But here:


14:41:49.396724 IP A.A.A.A.65027 > B.B.B.B.1: UDP, length
673
14:41:49.397742 IP B.B.B.B.1 > A.A.A.A.65027: UDP, length
555
14:41:49.397819 IP B.B.B.B.1 > A.A.A.A.65027: UDP, length 560


some other client accessing the very same port, and Asterisk accepts
request! Despite having another port in sip.conf, and netstat showing
that no process is listening to the 1 port. 

tcpdump'ing with port
filter shows that Asterisk has lots of active conversations on both
ports - the old one and the new one. 

Would you kindly share some holy
wisdom and explain me how can Asterisk listen to both ports
simultaneously, despite all configs? 

And, sorry for long post.
Couldn't make it shorter. 

-- 
With Best Regards
Mikhail Lischuk

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Re: [asterisk-users] Google TTS - Asterisk

[Danny Nicholas]

You could possibly process the hindi fonts as Unicode.  As for the Google 
Speech and TTS, it “should” all be covered under open source agreements.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram
Sent: Saturday, April 07, 2012 12:52 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google TTS - Asterisk

 

Hi

 

I was trying out Lefteris’ Asterisk google TTS and its working great with 
English..I have a question here – since it support hindi,tamil and other 
languages as well.. I would like to experiment it but am struck at one place :

 

   ;;Play message in Spanish

exten => 1234,n,agi(googletts.agi,"Esta es una simple prueba en español.",es)

;;Play message in Greek

exten => 1234,n,agi(googletts.agi,"Αυτό είναι ένα απλό τέστ στα ελληνικά.",el)

 

now in the above text – if I try to paste hindi fonts in “vi editor” or copy 
the hindi text from wordpad to vi editor – the entire text gets garbled.. and 
therefore call disconnects when google tries to parse it. Can someone please 
let me know how to handle this ?

 

Also is it a legitimate way of using Google speech recognition and TTS in this 
way ? i.e no license issues would arise?

 

thanks

Sriram

 

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[asterisk-users] MessageSend, SIP, and call files

[Roger Burton West]

As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.

After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and something like this almost works:

Channel: Local/8902
Application: MessageSend
Set: MESSAGE(body)=messagebody
Data: sip:glowworm
Data: sip:glowworm

but (a) I need that reserved local number to let the call work at all
(the number just does an Answer(), Wait(10), Hangup) and (b) I can't
seem to set the sender's name. That ought to be the second Data
parameter; actually the second one seems to determine where the message
goes, and whatever I set the first one to the sender name always comes
up as "asterisk". (Specifically, in the packet capture, I have

From: "asterisk" 

.) Now, I _can_ achieve the desired result, but only by having _another_
local number that does

exten => 8901,n,SET(MESSAGE(body)=${msg_out_body})
exten => 8901,n,MessageSend(${msg_out_to},${msg_out_from})

and setting up the callfile with:

Extension: 8901
Set: msg_out_to=glowworm
Set: msg_out_from=

at which point the message will appear to originate from FROM (note that
if I put a display name component in the msg_out_from it gets ignored -
but that is the terminals' peculiarity). But that's ugly. Has anyone got
this working with a relatively straight callfile setup?

While I'm writing, does Asterisk 10 have any way to send a SIP message
that isn't text/plain?

Roger

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[asterisk-users] Recent FreePBX vulnerability attacks

[Jared Geiger]

We saw some activity related to this FreePBX unpatched vulnerability
this past weekend on some hosted PBXes.

 http://seclists.org/fulldisclosure/2012/Mar/234

Usually we see the typical SIP Vicious attacks, but this one is much
more involved and dangerous.

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Re: [asterisk-users] MessageSend, SIP, and call files

[Danny Nicholas]

This is what "core show applications" in 10.1.3 shows
 SendDTMF: Sends arbitrary DTMF digits
   SendFAX: Sends a specified TIFF/F file as a FAX.
 SendImage: Sends an image file.
  SendText: Send a Text Message.
   SendURL: Send a URL.

You are using sendtext - you might want to use sendurl instead.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, April 10, 2012 9:05 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MessageSend, SIP, and call files

As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.

After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and something like this almost works:

Channel: Local/8902
Application: MessageSend
Set: MESSAGE(body)=messagebody
Data: sip:glowworm
Data: sip:glowworm

but (a) I need that reserved local number to let the call work at all (the
number just does an Answer(), Wait(10), Hangup) and (b) I can't seem to set
the sender's name. That ought to be the second Data parameter; actually the
second one seems to determine where the message goes, and whatever I set the
first one to the sender name always comes up as "asterisk". (Specifically,
in the packet capture, I have

From: "asterisk" 

.) Now, I _can_ achieve the desired result, but only by having _another_
local number that does

exten => 8901,n,SET(MESSAGE(body)=${msg_out_body})
exten => 8901,n,MessageSend(${msg_out_to},${msg_out_from})

and setting up the callfile with:

Extension: 8901
Set: msg_out_to=glowworm
Set: msg_out_from=

at which point the message will appear to originate from FROM (note that if
I put a display name component in the msg_out_from it gets ignored - but
that is the terminals' peculiarity). But that's ugly. Has anyone got this
working with a relatively straight callfile setup?

While I'm writing, does Asterisk 10 have any way to send a SIP message that
isn't text/plain?

Roger

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Re: [asterisk-users] Call Deflection with DAHDISendCallreroutingFacility

[Richard Mudgett]

> I want to use Call Deflection with DAHDISendCallreroutingFacility
> Application.
> I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
> my dialplan is like this:

You should always specify the switchtype and signaling parameters for
ISDN issues as well.  In this case it is not critical to determine
what is happening.

> 
> [Call-Deflection]
> exten => 66,n,Proceeding()
> exten => 66,1,wait(5)
> exten => 66,n,DAHDISendCallreroutingFacility(88050048,8262000,cfb)
> exten => 66,n,wait(5)
> exten => 66,n,Hangup()
> 
> after Executing
> DAHDISendCallreroutingFacility("DAHDI/i1/2188602827-3",
> "88050048,8262000,cfb")
> in new stack == Spawn extension (Call-Deflection, 66, 3) exited
> non-zero on 'DAHDI/i1/2188602827-3'
> 
> Asterisk exit immediately and last wait(5) won't Execute.
> 
> I used another PRI Analyzer and this is message sequence:
> 
> Asterisk <--setup-- Local exchange
> Asterisk --proceeding--> Local exchange
> Asterisk --facility--> Local exchange
> Asterisk --Disconnect(Subscriber Absent)--> Local exchange
> Asterisk <--Release-- Local exchange
> Asterisk --Release complete--> Local exchange
> 
> from the Analyzer report Asterisk send Disconnect immediately after
> Facility message
> (don't wait for response from Local exchange).
> please help me solve this problem

Asterisk does not care about the response from the switch in this case
so it does not wait for the defined response before hanging up the call.
DAHDISendCallreroutingFacility always returns nonzero to hangup the call.
I think it needs to have a built in wait(5) after sending the request
before returning to accommodate switches like yours that need time to
process the request.

Please file a bug report on this so it does not get lost.
https://issues.asterisk.org/jira
Thanks.

Richard

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[asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
the agent's phone starts ringing.

Strangely, I can't find anything real useful on this after searching
Google, this list, various Asterisk forums etc.

Is this supported? If not, is there some other maybe not so supported way
to accomplish this?

I get how I can just fire an AGI from the dial plan but once I leave
control to the queue, I can't really do that, I don't think.

Thanks in advance for any help!

--Todd
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Danny Nicholas]

Put your Queue command In a macro like this

[agi-and-queue]

Exten => s,1,Verbose(start AGI then do queue)

Exten => s,n,AGI(queproc.sh)

Exten => s,n,queue(myqueue)

 

You will need to put nohup into the AGI so it can run whether the line gets
picked up or not.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, April 10, 2012 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Run AGI while agent ringing instead of only when
connected

 

What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when the
agent's phone starts ringing.

 

Strangely, I can't find anything real useful on this after searching Google,
this list, various Asterisk forums etc.

 

Is this supported? If not, is there some other maybe not so supported way to
accomplish this?

 

I get how I can just fire an AGI from the dial plan but once I leave control
to the queue, I can't really do that, I don't think.

 

Thanks in advance for any help!

 

--Todd

 

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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Carlos Chavez]

On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the
> agent answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run
> when the agent's phone starts ringing.
> 
> 
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
> 
> 
> Is this supported? If not, is there some other maybe not so supported
> way to accomplish this?
> 
> 
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
> 
> 
The only way I really see to do that is to monitor events via AMI so
you can trigger the AGI when the phone starts to ring.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] MessageSend, SIP, and call files

[Roger Burton West]

On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote:
>This is what "core show applications" in 10.1.3 shows
> SendDTMF: Sends arbitrary DTMF digits
>   SendFAX: Sends a specified TIFF/F file as a FAX.
> SendImage: Sends an image file.
>  SendText: Send a Text Message.
>   SendURL: Send a URL.
>You are using sendtext - you might want to use sendurl instead.

Those are all about sending data in an existing channel, though -
the trick is that I don't _have_ a channel, which is presumably why
MessageSend exists. Is there a way to set up a channel without ringing
the phone?


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Re: [asterisk-users] MessageSend, SIP, and call files

[Danny Nicholas]

Not that I'm aware of;  however you can call and have the phone auto-answer
just to take the message - it's a SIP header tweak that has been discussed
here.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, April 10, 2012 3:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MessageSend, SIP, and call files

On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote:
>This is what "core show applications" in 10.1.3 shows
> SendDTMF: Sends arbitrary DTMF digits
>   SendFAX: Sends a specified TIFF/F file as a FAX.
> SendImage: Sends an image file.
>  SendText: Send a Text Message.
>   SendURL: Send a URL.
>You are using sendtext - you might want to use sendurl instead.

Those are all about sending data in an existing channel, though - the trick
is that I don't _have_ a channel, which is presumably why MessageSend
exists. Is there a way to set up a channel without ringing the phone?


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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Danny Nicholas]

You have read this thread?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, April 10, 2012 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Run AGI while agent ringing instead of only when
connected

 

What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when the
agent's phone starts ringing.

 

Strangely, I can't find anything real useful on this after searching Google,
this list, various Asterisk forums etc.

 

Is this supported? If not, is there some other maybe not so supported way to
accomplish this?

 

I get how I can just fire an AGI from the dial plan but once I leave control
to the queue, I can't really do that, I don't think.

 

Thanks in advance for any help!

 

--Todd

 

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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Thanks Danny.

What I am trying to do is send a popup screen to the agent. I am doing this
now using Flash Operator Panel but I am trying to get away from that. I
need to know the agent the call is being sent to when calling the AGI. So,
right now I am getting all that but only when the call is connected.
Running the AGI first will just allow me to popup a window for all agents
instead of the one that is currently ringing. This is why I need it to be
fired as part of the queue, it seems only the Queue (and the AMI) know
which agent is ringing.

--Todd


On Tue, Apr 10, 2012 at 3:21 PM, Danny Nicholas  wrote:

> Put your Queue command In a macro like this
>
> [agi-and-queue]
>
> Exten => s,1,Verbose(start AGI then do queue)
>
> Exten => s,n,AGI(queproc.sh)
>
> Exten => s,n,queue(myqueue)
>
> ** **
>
> You will need to put nohup into the AGI so it can run whether the line
> gets picked up or not.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
> ** **
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
> ** **
>
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
>
> ** **
>
> Is this supported? If not, is there some other maybe not so supported way
> to accomplish this?
>
> ** **
>
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
>
> ** **
>
> Thanks in advance for any help!
>
> ** **
>
> --Todd
>
> ** **
>
> --
> _
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

I was trying to leave the AMI out of this because I am unsure if I can
monitor it in real time without building an external listener which Flash
Operator Panel does for me right now.

When I went down this road, I thought it would be a piece of cake to just
fire an AGI, well it is until you get queues involved.

--Todd

On Tue, Apr 10, 2012 at 3:34 PM, Carlos Chavez wrote:

> On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
> > What I am trying to accomplish is to run an AGI script each time an
> > agent's line starts ringing. I currently have the AGI firing when the
> > agent answers the call using the Queue command, something like
> > queue(MyQueue,MyAgi.php). Works great but I need the AGI to run
> > when the agent's phone starts ringing.
> >
> >
> > Strangely, I can't find anything real useful on this after searching
> > Google, this list, various Asterisk forums etc.
> >
> >
> > Is this supported? If not, is there some other maybe not so supported
> > way to accomplish this?
> >
> >
> > I get how I can just fire an AGI from the dial plan but once I leave
> > control to the queue, I can't really do that, I don't think.
> >
> >
> The only way I really see to do that is to monitor events via AMI
> so
> you can trigger the AGI when the phone starts to ring.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
this for a number of years off and on, I never post unless I have dug hard,
searching all the Asterisk resources I know of.

This is where I got most of my info but the solutions mentioned on that
page require the call to be "Connected" to the agent before the AGI fires.
Once the agent is connected, I can get all sorts of info from Channel Vars.
Still, once the agent is connected, it's sort of too late, I need the AGI
to fire will the agent is ringing.

Thanks for your help so far.

On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas  wrote:

> You have read this thread?
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
> ** **
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
> ** **
>
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
>
> ** **
>
> Is this supported? If not, is there some other maybe not so supported way
> to accomplish this?
>
> ** **
>
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
>
> ** **
>
> Thanks in advance for any help!
>
> ** **
>
> --Todd
>
> ** **
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Danny Nicholas]

Were this my task, I would do a PERL/C daemon to run the AGI.  This is how I
do it in PERL

   my $astman = new Asterisk::Manager;

   $astman->user('user');

   $astman->secret('secret');

   my $man_addr='127.0.0.1';



   my $man_ok=1;

   open (my $man_in, "/etc/asterisk/manager.conf") or
$man_ok=undef;

   if ($man_ok) {

  while (<$man_in>) {

 if ($_ =~ /^bindaddr/) {

(undef,$man_addr) = split /\=/, $_;

}

 }

  close $man_in;

  }

   $man_addr =~ s/\s//g;



   ( $man_addr )=( $man_addr =~ /(.*)/ );



   $astman->host($man_addr);

   $astman->connect || die "Could not connect to " .
$astman->host . "!\n";

 

   my %resp = $astman->sendcommand(  Action => 'Originate',

   Channel =>
$extval,

   Variable =>
"ARG1=$fileval",

   Exten => $extval,

   Context =>
'playit',

   priority => 1,

   Number => 5551212

   );



   sleep 2;

   %resp = $astman->sendcommand(  Action => 'Logoff');

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, April 10, 2012 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Run AGI while agent ringing instead of only
when connected

 

Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
this for a number of years off and on, I never post unless I have dug hard,
searching all the Asterisk resources I know of.

 

This is where I got most of my info but the solutions mentioned on that page
require the call to be "Connected" to the agent before the AGI fires. Once
the agent is connected, I can get all sorts of info from Channel Vars.
Still, once the agent is connected, it's sort of too late, I need the AGI to
fire will the agent is ringing.

 

Thanks for your help so far.

On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas  wrote:

You have read this thread?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, April 10, 2012 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Run AGI while agent ringing instead of only when
connected

 

What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when the
agent's phone starts ringing.

 

Strangely, I can't find anything real useful on this after searching Google,
this list, various Asterisk forums etc.

 

Is this supported? If not, is there some other maybe not so supported way to
accomplish this?

 

I get how I can just fire an AGI from the dial plan but once I leave control
to the queue, I can't really do that, I don't think.

 

Thanks in advance for any help!

 

--Todd

 


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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Todd Routhier]

Thanks again Danny, Perl was the first thing I tinkered with back in the
90's but haven't messed with it for years.

Looking over what you sent, I get about 90% of what's going on there. With
a little searching and brushing up on my Perl, I think I will be able to
make this work.

This is a good solution and, if I can get this to work, I won't even need
the AGI. I can basically just hit what I need using CURL within the Perl
script (I think).

All the AGI was going to do for me is hit a URL with some parameters out on
the Internet. So, pretty sure I can do all that within the Perl Script and
leave AGI out of it completely.

--Todd

On Tue, Apr 10, 2012 at 4:02 PM, Danny Nicholas  wrote:

> Were this my task, I would do a PERL/C daemon to run the AGI.  This is how
> I do it in PERL
>
>my $astman = new Asterisk::Manager;
>
>$astman->user('user');
>
>$astman->secret('secret');
>
>my $man_addr='127.0.0.1';
>
> 
>
>my $man_ok=1;
>
>open (my $man_in, "/etc/asterisk/manager.conf") or
> $man_ok=undef;
>
>if ($man_ok) {
>
>   while (<$man_in>) {
>
>  if ($_ =~ /^bindaddr/) {
>
> (undef,$man_addr) = split /\=/, $_;
>
> }
>
>  }
>
>   close $man_in;
>
>   }
>
>$man_addr =~ s/\s//g;
>
> 
>
>( $man_addr )=( $man_addr =~ /(.*)/ );
>
> 
>
>$astman->host($man_addr);
>
>$astman->connect || die "Could not connect to " .
> $astman->host . "!\n";
>
> ** **
>
>my %resp = $astman->sendcommand(  Action => 'Originate',***
> *
>
>Channel =>
> $extval,
>
>Variable =>
> "ARG1=$fileval",
>
>Exten =>
> $extval,
>
>Context =>
> 'playit',
>
>priority => 1,*
> ***
>
>Number =>
> 5551212
>
>);
>
> 
>
>sleep 2;
>
>%resp = $astman->sendcommand(  Action => 'Logoff');
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:55 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Run AGI while agent ringing instead of
> only when connected
>
> ** **
>
> Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been at
> this for a number of years off and on, I never post unless I have dug hard,
> searching all the Asterisk resources I know of.
>
> ** **
>
> This is where I got most of my info but the solutions mentioned on that
> page require the call to be "Connected" to the agent before the AGI fires.
> Once the agent is connected, I can get all sorts of info from Channel Vars.
> Still, once the agent is connected, it's sort of too late, I need the AGI
> to fire will the agent is ringing.
>
> ** **
>
> Thanks for your help so far.
>
> On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas  wrote:
> 
>
> You have read this thread?
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
> *Sent:* Tuesday, April 10, 2012 3:15 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
> when connected
>
>  
>
> What I am trying to accomplish is to run an AGI script each time an
> agent's line starts ringing. I currently have the AGI firing when the agent
> answers the call using the Queue command, something like
> queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
> the agent's phone starts ringing.
>
>  
>
> Strangely, I can't find anything real useful on this after searching
> Google, this list, various Asterisk forums etc.
>
>  
>
> Is this supported? If not, is there some other maybe not so supported way
> to accomplish this?
>
>  
>
> I get how I can just fire an AGI from the dial plan but once I leave
> control to the queue, I can't really do that, I don't think.
>
>  
>
> Thanks in advance for any help!
>
>  
>
> --Todd
>
>  
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

[lists65]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: Monday, April 09, 2012 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
list...@gmail.com
>Sent: Monday, April 09, 2012 8:34 PM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

>I am not a programmer and I have learned so much from examples and the
list.
>Perhaps someone could tell me what I am doing wrong in my example below:

>I am getting the caller ID and caller name from my local POTS line and I
want to add it into a sql table.  I am trying with the following code but
the data never gets put into the table.

>Can anyone correct my syntax and tell me what I am doing wrong?


>[callerinfo]
>exten => s,1,MYSQL(Connect connid localhost myuser mypassword cnam) exten
=> s,n,MYSQL(Query resultid ${connid} INSERT INTO `calleridcapture`
>(`number`,`name`) VALUES (${CALLERID(num)},${CALLERID(name)})
>exten => s,n,MYSQL(Clear ${resultid})
>exten => s,n,MYSQL(Disconnect ${connid}) exten => s,n,NoOp(Callerid Name
${CALLERID(name)}) exten => s,n,NoOp(Callerid Number  ${CALLERID(num)})


>The NoOP does show the correct CALLERID name & number when I test it.  The
information just doesn't go into my calleridcapture table in the cnam
database.

>Thanks very much for your help
>Again I am not a programmer and I am sure my syntax is wrong.

>This is Asterisk 1.8.10.0
>

As the previous two posters alluded, you need to encapsulate your values in
quotes.  I think you can get by without the backticks, not 100% sure as I've
converted from MYSQL to func_odbc.  If you're not going to go with Steve's
recommendation of AGI, I would highly recommend  switching from func_mysql
to func_odbc; func_odbc is much more straightforward in my opinion, and you
definitely get much better error messages within the CLI as you're watching
your code execute.  ofps.oreilly.com/titles/9780596517342/asterisk-DB.html
is a good resource for setting up odbc.

Noah


Thanks for your responses.

Well, the AGI piece sounds good, but again I am not a programmer but I
certainly will try and try to find some code that I could piece together and
try to make it work.

I will check out Oreilly's stuff, they always seem to have good books when
you are trying to learn something new.

Thanks again, I really appreciate your resonse.



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Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

[lists65]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, April 09, 2012 9:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

On Mon, 9 Apr 2012, list...@gmail.com wrote:

> I am getting the caller ID and caller name from my local POTS line and 
> I want to add it into a sql table.  I am trying with the following 
> code but the data never gets put into the table.
>
> Can anyone correct my syntax and tell me what I am doing wrong?
>
> [callerinfo]
> exten => s,1,MYSQL(Connect connid localhost myuser mypassword cnam) 
> exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO 
> `calleridcapture`
> (`number`,`name`) VALUES (${CALLERID(num)},${CALLERID(name)})
> exten => s,n,MYSQL(Clear ${resultid})
> exten => s,n,MYSQL(Disconnect ${connid}) exten => s,n,NoOp(Callerid 
> Name  ${CALLERID(name)}) exten => s,n,NoOp(Callerid Number  
> ${CALLERID(num)})
>
> The NoOP does show the correct CALLERID name & number when I test it. 
> The information just doesn't go into my calleridcapture table in the 
> cnam database.

I'm just a 1.2 Luddite, but I am a reasonably skilled c programmer. I've
never used the mysql() application because it seems ugly, limited and
'hackish' to me.

If it were me, I'd code it up as an AGI in a 'real' language where you have
access to 'real' error codes and messages and you don't need a bunch of
quoting hocus-pocus. (Supposedly, the quoting nonsense has gotten better
since 1.2.)

You say you're not a programmer so that may not be an option for you -- but
you got this far :)

The first thing I'd do (aside from using verbose() instead of noop()) would
be to display the result from each step. If the connection is failing,
looking farther is pointless.

Don't you need to put single or double quotes around your individual values?
Change 'mysql' to 'echo' on your 'select' line and see if the statement is
valid at the MySQL command line.

Then, I'd drop the backticks in the 'hail-mary' hope that they are confusing
mysql() somehow.

Then, I'd crack open another beer and reach for a book on c or PHP or Perl.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Thanks for your help and comments.  Again I am not a programmer but you did
give me some direction and I will see if I can find some examples that will
work for my needs.  I have seen AGI before and you are right it does seem to
work very well.

Thanks for your help and comments.  I hope to figure this out.  The SQL is
new to me but I really like the database stuff.





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Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

[lists65]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, April 09, 2012 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

list...@gmail.com wrote:
> exten =>  s,n,MYSQL(Query resultid ${connid} INSERT INTO 
> `calleridcapture`
> (`number`,`name`) VALUES (${CALLERID(num)},${CALLERID(name)})


Here is an example of one of my inserts:

exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO Indianapolis set
phone="${CALLERID(number)}" \, flag="YES" \, note="Blacklisted by
Tele-Torture - ${TODAY}")

Doug

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Thank you Doug.  I did see some examples with the "\" but I was confused
because I thought you didn't have to do that with 1.8 (maybe I misread that
some where along the line).  I will look at your example and see if I can
adapt what I am doing to make it work for me.

I appreciate your help and comments as well.  That's what is so good about
this list, seems like there is always someone that can help.  I really like
this stuff, I wished that I had got into programming a long time ago so I
could start to understand syntax better.  I missed out when I didn't pick up
learning how to code.




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Re: [asterisk-users] MYSQL INSERT QUESTION IN DIALPLAN

[Doug Lytle]

list...@gmail.com wrote:

Thank you Doug.  I did see some examples with the "\" but I was confused
because I thought you didn't have to do that with 1.8 (maybe I misread that
some where along the line).


You are probably correct, but I'm still on 1.4.x.  Glad you got it working!

Doug


--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] Call Deflection with DAHDISendCallreroutingFacility

[Mehdi Shirazi]

>> I want to use Call Deflection with DAHDISendCallreroutingFacility
>> Application.
>> I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
>> my dialplan is like this:

>You should always specify the switchtype and signaling parameters for
>ISDN issues as well.  In this case it is not critical to determine
>what is happening.
 
I used one E1 with switchtype=euroisdn signalling=pri_cpe 
connected to C&C08 Huawei Local eXchange.All supplementary services are enabled 
in LX.


>> 
>> [Call-Deflection]
>> exten => 66,n,Proceeding()
>> exten => 66,1,wait(5)
>> exten => 66,n,DAHDISendCallreroutingFacility(88050048,8262000,cfb)
>> exten => 66,n,wait(5)
>> exten => 66,n,Hangup()
>> 
>> after Executing
>> DAHDISendCallreroutingFacility("DAHDI/i1/2188602827-3",
>> "88050048,8262000,cfb")
>> in new stack == Spawn extension (Call-Deflection, 66, 3) exited
>> non-zero on 'DAHDI/i1/2188602827-3'
>> 
>> Asterisk exit immediately and last wait(5) won't Execute.
>> 
>> I used another PRI Analyzer and this is message sequence:
>> 
>> Asterisk <--setup-- Local exchange
>> Asterisk --proceeding--> Local exchange
>> Asterisk --facility--> Local exchange
>> Asterisk --Disconnect(Subscriber Absent)--> Local exchange
>> Asterisk <--Release-- Local exchange
>> Asterisk --Release complete--> Local exchange
>> 
>> from the Analyzer report Asterisk send Disconnect immediately after
>> Facility message
>> (don't wait for response from Local exchange).
>> please help me solve this problem

>Asterisk does not care about the response from the switch in this case
>so it does not wait for the defined response before hanging up the call.
>DAHDISendCallreroutingFacility always returns nonzero to hangup the call.
>I think it needs to have a built in wait(5) after sending the request
>before returning to accommodate switches like yours that need time to
>process the request.

How about Disconnect message after facility message ? I changed parameter: 
|cfb|cfnr|unknown but always Asterisk send Disconnect message with 
"Subscriber Absent" cause. DAHDISendCallreroutingFacility sends Disconnect 
message?
maybe this cause is why LX don't do anything. 

>Please file a bug report on this so it does not get lost.
>https://issues.asterisk.org/jira
>Thanks.
ok I will do but I am not sure about add always "Cause value=Subscriber Absent" 
to bug report or not.

>Richard

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Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

[p070075 Muhammad Atif Ramzan]

Thanks SamyGo
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Re: [asterisk-users] Run AGI while agent ringing instead of only when connected

[Satish Barot]

I would implement it in a different way.
As you seem to be a seasoned player just a hint here.
How about adding local channels as queue members and executing agi in local
channel context before actual dial()?
Only limitation is, AGI will get executed for each dial irrespective of
whether an extension rings or not. But at least you can identify which
extension is being dialed.
See 'Using Local Channels' on
http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html

--Satish Barot

On Wed, Apr 11, 2012 at 2:45 AM, Todd Routhier  wrote:

> Thanks again Danny, Perl was the first thing I tinkered with back in the
> 90's but haven't messed with it for years.
>
> Looking over what you sent, I get about 90% of what's going on there. With
> a little searching and brushing up on my Perl, I think I will be able to
> make this work.
>
> This is a good solution and, if I can get this to work, I won't even need
> the AGI. I can basically just hit what I need using CURL within the Perl
> script (I think).
>
> All the AGI was going to do for me is hit a URL with some parameters out
> on the Internet. So, pretty sure I can do all that within the Perl Script
> and leave AGI out of it completely.
>
> --Todd
>
>
> On Tue, Apr 10, 2012 at 4:02 PM, Danny Nicholas  wrote:
>
>> Were this my task, I would do a PERL/C daemon to run the AGI.  This is
>> how I do it in PERL
>>
>>my $astman = new Asterisk::Manager;
>>
>>$astman->user('user');
>>
>>$astman->secret('secret');
>>
>>my $man_addr='127.0.0.1';
>>
>> 
>>
>>my $man_ok=1;
>>
>>open (my $man_in, "/etc/asterisk/manager.conf") or
>> $man_ok=undef;
>>
>>if ($man_ok) {
>>
>>   while (<$man_in>) {
>>
>>  if ($_ =~ /^bindaddr/) {
>>
>> (undef,$man_addr) = split /\=/, $_;
>>
>> }
>>
>>  }
>>
>>   close $man_in;
>>
>>   }
>>
>>$man_addr =~ s/\s//g;
>>
>> 
>>
>>( $man_addr )=( $man_addr =~ /(.*)/ );
>>
>> 
>>
>>$astman->host($man_addr);
>>
>>$astman->connect || die "Could not connect to " .
>> $astman->host . "!\n";
>>
>> ** **
>>
>>my %resp = $astman->sendcommand(  Action => 'Originate',**
>> **
>>
>>Channel =>
>> $extval,
>>
>>Variable =>
>> "ARG1=$fileval",
>>
>>Exten =>
>> $extval,
>>
>>Context =>
>> 'playit',
>>
>>priority => 1,
>> 
>>
>>Number =>
>> 5551212
>>
>>);
>>
>> 
>>
>>sleep 2;
>>
>>%resp = $astman->sendcommand(  Action => 'Logoff');
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
>> *Sent:* Tuesday, April 10, 2012 3:55 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Run AGI while agent ringing instead of
>> only when connected
>>
>> ** **
>>
>> Yes Sir.. Studied it pretty hard, did I miss a solution? Trust me, been
>> at this for a number of years off and on, I never post unless I have dug
>> hard, searching all the Asterisk resources I know of.
>>
>> ** **
>>
>> This is where I got most of my info but the solutions mentioned on that
>> page require the call to be "Connected" to the agent before the AGI fires.
>> Once the agent is connected, I can get all sorts of info from Channel Vars.
>> Still, once the agent is connected, it's sort of too late, I need the AGI
>> to fire will the agent is ringing.
>>
>> ** **
>>
>> Thanks for your help so far.
>>
>> On Tue, Apr 10, 2012 at 3:42 PM, Danny Nicholas 
>> wrote:
>>
>> You have read this thread?
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
>>
>>  
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
>> *Sent:* Tuesday, April 10, 2012 3:15 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] Run AGI while agent ringing instead of only
>> when connected
>>
>>  
>>
>> What I am trying to accomplish is to run an AGI script each time an
>> agent's line starts ringing. I currently have the AGI firing when the agent
>> answers the call using the Queue command, something like
>> queue(MyQueue,MyAgi.php). Works gre

[asterisk-users] OT - How to localize Freepbx 2.10 or 2.9 ?

[Olivier]

Hi,

May I ask this off-topic question ?

I've got an Asterisk 1.8/Freepbx 2.10 install on a Squeeze server.
No matter which language is selected with the top right corner
scrolling list, the GUI remains in english.
In my system, there are plenty of amp.po localized files, here and there.

Any hint on how to solve this .

Regards

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