Re: [asterisk-users] Asterisk Tech Tips: Cookin' with Asterisk

[virendra bhati]

Hi Steve Sokol,

Thanks for inform. I will join it.

On Wed, Apr 13, 2011 at 9:46 PM, Steve Sokol  wrote:

> Greetings Asterisk Users,
>
> I'm happy to announce that Russell Bryant and Leif Madsen have volunteered
> to host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon
> central time.  Russell and Leif are project leaders and have collaborated on
> two Asterisk books:  *Asterisk: The Definitive Guide* and *Asterisk
> Cookbook*, both published by O'Reilly & Associates.  *Asterisk: The
> Definitive Guide* is the Asterisk bible -- a must-read for anyone learning
> to implement an Asterisk-based system.  *Asterisk Cookbook* is a
> collection of recipes, simple code solutions you can put to work
> immediately, along with a detailed discussion that offers insight into why
> and how the recipe works.
>
> Leif will be presenting "Hot-Desking With The Asterisk Database".  Learn
> how to utilize the Asterisk database and clever usage of the Asterisk
> Dialplan to implement hot desking.
>
> Russell will present "Debugging The Asterisk Dialplan". Learn how to
> utilize the Asterisk database and other dialplan functionality to build a
> flexible dialplan debugging system.
>
> Following the presentations we will open the floor for any Asterisk
> questions.  Join us: register now! 
>
> Thanks,
>
> -S
>
> Steve Sokol
> Digium, Inc.
>
>
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>



-- 
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Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] How to know extensions status ???

[virendra bhati]

How to use telnet ? I never work on that . please guide me...

On Wed, Apr 13, 2011 at 6:48 PM, Danny Nicholas  wrote:

>--
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
> *Sent:* Wednesday, April 13, 2011 4:28 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] How to know extensions status ???
>
>
>
> Hi,
>
> How to know the all SIP extensions status with AMI's  ExtensionState ?
>
> What is the value should I pass in
>
> Context: <> ??
> which will be define at context here ? shell I use sip.conf's  context for
> that extension or any other?
>
> extension : <> ??
> extension will be SIP/100 or just 100 ??
>
>
> Please guide me ...
>
> -
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
>
>  *[Danny Nicholas] *
>
> *My guess would be just 100 (extension: 100) but you can verify it in a
> telnet session.*
>
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Thanks and regards

 Virendra Bhati
+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] Processing sip messages

[Godson Gera]

Though you can modify SIP headers, there is no straight forward way of
processing SIP messages as Asterisk is made to abstract away that protocol
layer. However it generally is possible to track the Ringing and Answer
events in AMI which in turn the reflect the 180/183 and 200 SIP messages.
However, to be more precise. you can enable sip debug (which will make
asterisk to print all the SIP messages ) and process the logs or do IPC
between your program and "asterisk -r "

The easy way is to use SER family tools like kamailio , OpenSIPS for tasks
like these.



On Thu, Apr 14, 2011 at 10:52 AM, Nasir Iqbal wrote:

> Is there an way (asteisk command / AMI / Agi ) to process incoming SIP
> messages like ( 100 trying , 183 session progress , 200 Ack) ,
>
> I am intersted to findout delay between 183 and 200 message
>
> Regards
> Nasir Iqbal
>
>
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Godson Gera
Asterisk Consultant 
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[asterisk-users] Processing sip messages

[Nasir Iqbal]

Is there an way (asteisk command / AMI / Agi ) to process incoming SIP
messages like ( 100 trying , 183 session progress , 200 Ack) ,

I am intersted to findout delay between 183 and 200 message

Regards
Nasir Iqbal
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Re: [asterisk-users] Asterisk port 5000 open

[David White]

http://www.google.com/search?q=port+5000+asterisk



answer is in the first hit :)



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Thomas
Sent: Wednesday, April 13, 2011 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk port 5000 open



Hi,



I have been trying to find out what module is causing asterisk to open port 5000



I have already disabled some ( sccp, mgcp, iax and other modules ) since I only 
want sip port opened




/etc/asterisk# netstat -aln --programs  | grep asterisk

tcp0  X.X.X.X:5060 0.0.0.0:*   LISTEN  
22523/asterisk

udp0 X.X.X.X:5000  0.0.0.0:*   
22523/asterisk

udp0 X.X.X.X:5060 0.0.0.0:*   
22523/asterisk



I have port 5000 blocked with IP tables, but would like better to understand 
what is it for.



Not sure if there's a list of known ports used by asterisk.



Thanks,



--
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[asterisk-users] Asterisk port 5000 open

[Robert Thomas]

Hi,

I have been trying to find out what module is causing asterisk to open port
5000

I have already disabled some ( sccp, mgcp, iax and other modules ) since I
only want sip port opened


/etc/asterisk# netstat -aln --programs  | grep asterisk
tcp0  X.X.X.X:5060 0.0.0.0:*   LISTEN
 22523/asterisk
udp0 X.X.X.X:5000  0.0.0.0:*
22523/asterisk
udp0 X.X.X.X:5060 0.0.0.0:*
22523/asterisk

I have port 5000 blocked with IP tables, but would like better to understand
what is it for.

Not sure if there's a list of known ports used by asterisk.

Thanks,

-- 
Robert
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Re: [asterisk-users] T38 fax detection using g729

[Niccolò Belli]

Il 13/04/2011 19:54, Larry Moore ha scritto:
> That is because the remote endpoint, eutelia, will need to detect the
> Fax Tones and send the T.38 ReINVITE to you, they may not have T.38
> enabled.

Uhm... it's very unlikely.

> As a suggestion you could configure your incoming calls from eutelia to
> go directly to the fax receive function whilst having the g729 codec
> enabled, I expect you will then see T.38 re-invite come from Asterisk.

See attachment from bug https://issues.asterisk.org/view.php?id=19100
Doesn't Eutelia send a T.38 re-invite there?

Anyway I will check again making a packet dump to be sure.

> What is in your configuration for 159?

[159]
language=it
type=friend
qualify=yes ;ping per controllo stato
dtmfmode = rfc2833
context=phones-sip
host=dynamic
disallow=all
allow=g729
secret=mysecret

Cheers,
Darkbasic

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Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

[Satish Patel]

As far as you are not upgrading kernel you are good with Dahdi.

Kernel upgrade require to install dahadi

--
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On Apr 13, 2011, at 4:04 PM, Shaun Ruffell  wrote:


On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:

Centos 5.6 came out. Any one tried to update to the 5.6 yet?

I am running Asterisk 1.8 and is there any risk to upgrade to  
Centos 5.6?


I'm not sure about Asterisk in general, but if you use DAHDI, please  
be sure

to install version 2.4.1.2.

http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Satish Patel]

I'd say boot with ubuntu livecd and run lspci command.

Make sure your card back side LED working.

--
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On Apr 13, 2011, at 2:32 PM, Kaushal Shriyan  
 wrote:





On Wed, Apr 13, 2011 at 9:23 PM, Tim Nelson   
wrote:


On Wed, Apr 13, 2011 at 8:07 PM, satish patel  
 wrote:

Try dmesg command

root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module  3.5.19.0 (c)  
1994-2010 Sangoma Technologies Inc
[ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c)  
1994-2010 Sangoma Technologies Inc
[ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0  
(c) 1994-2010 Sangoma Technologies Inc
[ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010  
Sangoma Technologies Inc
[ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006  
Sangoma Technologies Inc.



Hi Satish


dmesg | grep -i Sangoma does not show anything


If lspci isn't showing the device, then your board is not  
recognizing it. Try another slot, and make sure the card is seated  
100% tightly into the slot.


And, of course, make sure the system is powered off while doing  
this. 99.999% of people already know this, but I've been bitten by  
the other 0.001% that don't. :-)


If you're still not showing it, a call to Sangoma support is  
probably in order. They are top notch.


--Tim

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Hi

I get  anaconda.log:23:07:59 DEBUG   : ignoring driverless device  
Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card
Linux asterisk 2.6.18-194.11.1.el5 #1 SMP Tue Aug 10 19:05:06 EDT  
2010 x86_64 x86_64 x86_64 GNU/Linux

lspci | grep sangoma does not return anything

Please suggest further.

Thanks

Kaushal
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Re: [asterisk-users] Asterisk as a Condo door opener/intercom

[Kyle Kienapfel]

On Wed, Apr 13, 2011 at 11:34 AM, Andreas Sikkema  wrote:
> On 4/12/11 1:21 AM, Don Kelly wrote:
>> Continuing top posting...
>>
>> The same argument could be made for any commercial solution. Why use
>> Asterisk when we could throw $4,000 at our problem for a commercial
>> solution?
>>
>> I'd like to have a solution that would have the features you suggest for
>> $400.
>
> What part of the system isn't working? The "route calls to the
> appartment" part? That could be replaced by Asterisk with enough
> (analogue?) ports to serve the front door and appartments using existing
> wiring.
>
> If the door part also needs replacing because it is proprietary to the
> old system, you could use a SIP dooropener/intercom, but these are
> generally expensive, starting around EUR800/$800? or so and probably
> need an expension for apartment buttons. And you'd need to run new
> wiring to the door and perhaps change the lock.
>
> And then there's the apartments, it could get *very* expensive when you
> need to replace wiring and the "phones" in each apartment to something
> VoIP like.
>
>
> --
> Andreas Sikkema
>
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>
My apartment building switched from something that tapped into phone
lines to having the panel outside take a 3 digit extension, and then
it does a phone call. Needs a bit of administation to associate the
extensions to the right number to call, but then the call can go to a
cellphone or a voip provider... I like being able to buzz myself in
with my laptop :)

Outdoor intercom SIP phone + server + door striker and hopefully less
than $2000 in time to tie it all together.

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Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

[Danny Nicholas]

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jian Gao
Sent: Wednesday, April 13, 2011 3:43 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

 

I just upgraded a non critical server.

Before upgrade: Centos 5.5(i386) + Dahdi 2.4.1 + Asterisk 1.8.3

First upgrade to Centos 5.6. After reboot, Asterisk is OK but Dahdi
failed!!!

Then I upgrade Dahdi to 2.4.1.2, also upgraded Asterisk to 1.8.3.2. After
reboot, Dahdi came back.

:)

Jian  


On 11-04-13 01:04 PM, Shaun Ruffell wrote: 

On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:

Centos 5.6 came out. Any one tried to update to the 5.6 yet?
 
I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?

 
I'm not sure about Asterisk in general, but if you use DAHDI, please be sure
to install version 2.4.1.2.
 
http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html
 
[Danny Nicholas] 

Wish you had tackled just the DAHDI so we would know if that would keep the
1.6'ers alive.

 
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Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

[Jian Gao]

I just upgraded a non critical server.

Before upgrade: Centos 5.5(i386) + Dahdi 2.4.1 + Asterisk 1.8.3

First upgrade to Centos 5.6. After reboot, Asterisk is OK but Dahdi 
failed!!!


Then I upgrade Dahdi to 2.4.1.2, also upgraded Asterisk to 1.8.3.2. 
After reboot, Dahdi came back.


:)
*
Jian *

On 11-04-13 01:04 PM, Shaun Ruffell wrote:

On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:

Centos 5.6 came out. Any one tried to update to the 5.6 yet?

I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?

I'm not sure about Asterisk in general, but if you use DAHDI, please be sure
to install version 2.4.1.2.

http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html

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[asterisk-users] Applet based softphone for Asterisk

[ABBAS SHAKEEL]

Hello

Can some body let me know any softphone that is developed using java & can
support at least sip protocol. Must be open source and ready to be used. I
am trying to accomplish is to integrate it with an applet. some thing like
click to call on web page.

Sorry if this is not correct place for this question.

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Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

[Shaun Ruffell]

On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:
> Centos 5.6 came out. Any one tried to update to the 5.6 yet?
>
> I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?

I'm not sure about Asterisk in general, but if you use DAHDI, please be sure
to install version 2.4.1.2.

http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html

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

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[asterisk-users] Safe to upgrade to Centos 5.6 now ???

[Jian Gao]

Centos 5.6 came out. Any one tried to update to the 5.6 yet?

I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?

--
*Jian*

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Re: [asterisk-users] Realtime SIP & peer status

[Rob Coward]

  

Rather than add extra overhead to your dialplan and the asterisk
server, why not make use of the AMI and have a background process
listening for the various events and updating your database accordingly
? 

See
http://www.voip-info.org/wiki/view/asterisk+manager+events#ExtensionStatusEvent
and
http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/lib/Asterisk/AMI.pm#Events


Regards, 

Rob 

On Wed, 13 Apr 2011 10:15:30 +0200, Jonas Kellens
wrote: 

> Hello,
> 
> I'm using SIP realtime with MySQL DB.
> 
> Is it
possible to get the status of the SIP peer (free / calling) from this
realtime DB ?
> 
> If not, is there another way to obtain the call state
of a SIP peer ?
> 
> Kind regards,
> Jonas.
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Re: [asterisk-users] Asterisk as a Condo door opener/intercom

[Andreas Sikkema]

On 4/12/11 1:21 AM, Don Kelly wrote:
> Continuing top posting...
> 
> The same argument could be made for any commercial solution. Why use
> Asterisk when we could throw $4,000 at our problem for a commercial
> solution?
> 
> I'd like to have a solution that would have the features you suggest for
> $400.

What part of the system isn't working? The "route calls to the
appartment" part? That could be replaced by Asterisk with enough
(analogue?) ports to serve the front door and appartments using existing
wiring.

If the door part also needs replacing because it is proprietary to the
old system, you could use a SIP dooropener/intercom, but these are
generally expensive, starting around EUR800/$800? or so and probably
need an expension for apartment buttons. And you'd need to run new
wiring to the door and perhaps change the lock.

And then there's the apartments, it could get *very* expensive when you
need to replace wiring and the "phones" in each apartment to something
VoIP like.


-- 
Andreas Sikkema

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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Kaushal Shriyan]

On Wed, Apr 13, 2011 at 9:23 PM, Tim Nelson  wrote:

>
> On Wed, Apr 13, 2011 at 8:07 PM, satish patel wrote:
>
>>  Try dmesg command
>>
>> root@:~# dmesg | grep -i Sangoma
>> [ 2303.473601] WANPIPE(tm) Hardware Support Module  3.5.19.0 (c) 1994-2010
>> Sangoma Technologies Inc
>> [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010
>> Sangoma Technologies Inc
>> [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c)
>> 1994-2010 Sangoma Technologies Inc
>> [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010
>> Sangoma Technologies Inc
>> [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma
>> Technologies Inc.
>>
>>
> Hi Satish
>
>
> dmesg | grep -i Sangoma does not show anything
>
>
> If lspci isn't showing the device, then your board is not recognizing it.
> Try another slot, and make sure the card is seated 100% tightly into the
> slot.
>
> And, of course, make sure the system is powered off while doing this.
> 99.999% of people already know this, but I've been bitten by the other
> 0.001% that don't. :-)
>
> If you're still not showing it, a call to Sangoma support is probably in
> order. They are top notch.
>
> --Tim
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>

Hi

I get  anaconda.log:23:07:59 DEBUG   : ignoring driverless device Sangoma
Technologies Corp. A200/Remora FXO/FXS Analog AFT card
Linux asterisk 2.6.18-194.11.1.el5 #1 SMP Tue Aug 10 19:05:06 EDT 2010
x86_64 x86_64 x86_64 GNU/Linux
lspci | grep sangoma does not return anything

Please suggest further.

Thanks

Kaushal
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Re: [asterisk-users] AGI and forking

[Tilghman Lesher]

On Wednesday 13 April 2011 08:08:03 A J Stiles wrote:
> Hi.  I just want to make sure I understand this before doing something
> that might break things spectacularly for our users and customers  :)
> 
> We are using Asterisk 1.6.2.9 and my programming language of choice is
> Perl.
> 
> I want, when a call comes in on someone's DDI number  (which the person
> who dialled it can only possibly have obtained by dialling 1471 after
> we called them),  to be able to look up the caller's details from one
> of our databases (where the number ought to be stored, because we
> already dialled it).
> 
> Now, this search is going to take some time; so I'd like for the AGI
> script to fork a clone of itself, so the parent process can exit and
> the dialplan continue on to ring the person's phone, while the database
> lookup is done in the background  (the script doesn't need to have any
> further contact with Asterisk -- it will initiate any necessary future
> communication via other channels).
> 
> 
> Is this the sort of thing I need?
> 
> ##  begin code snippet  ##
> 
> #!/usr/bin/perl -w
> use strict;
> use Asterisk::AGI;
> 
> my $AGI = new Asterisk::AGI;
> my %params = $AGI->ReadParse();
> 
> $SIG{CHLD} = "IGNORE";
> 
> if (my $child_pid = fork) {
> #  This is executed in the parent process
> exit;
> }
> elsif (defined $child_pid) {
> #  This is executed in the child process
> 
> close STDIN;
> close STDOUT;
> close STDERR;
> 
> #  Load some more modules and do some stuff
> #   that will take a long time
> 
> exit;
> }
> else {
> die "Could not fork: $!";
> };
> 
> ##  end code snippet  ##
> 
> Am I right in thinking I shouldn't have to worry about zombie processes,
> because the parent exits before the child and the init in modern Linux
> distros is smart enough to deal with orphaned processes itself?

Almost.  You should also set a new session ID to ensure that the child gets
a new processgroup.  Otherwise, on some systems, it will still wait for the
child to also exit).  In Perl, this is accessible from the POSIX module,
function setsid().

-- 
Tilghman

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Re: [asterisk-users] [OT] Yealink Phones

[Russell Brown]

Quoth Andrew Thomas:-

>Have you seen the 'Action URL' bit yet?  Makes everything almost
>key-system like ;)

I saw it in the DSS key settings but havn't worked out anything useful
to do with it yet?

What are you using it for (and how?)?


-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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Re: [asterisk-users] T38 fax detection using g729

[Larry Moore]

On 13/04/2011 10:14 PM, Niccolò Belli wrote:

Hi, I continue the discussion from
https://issues.asterisk.org/view.php?id=19103

If T.38 reinvite detection should still work, why it doesn't?
If I use faxdetect = t38 it does never detect the fax, even using alaw.



That is because the remote endpoint, eutelia, will need to detect the 
Fax Tones and send the T.38 ReINVITE to you, they may not have T.38 enabled.


It would appear in your case your Asterisk installation will need to 
detect the Fax Tones so as to make the decision the incoming call is a 
fax and then switch to the fax extension. For Asterisk to detect the Fax 
Tones you will need to set faxdetect to either yes or cng, you will also 
require using alaw or ulaw codec.


As a suggestion you could configure your incoming calls from eutelia to 
go directly to the fax receive function whilst having the g729 codec 
enabled, I expect you will then see T.38 re-invite come from Asterisk.


What is in your configuration for 159?

Larry.

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[asterisk-users] Fw: SIP Trunk send DID or DNIS information

[A S]

Hello,


Some background before i ask the question.

I am attempting to implement a SIP trunk between an askerisk an a Mitel 5000 
system.  The mitel is giving me 404 errors when I send a call over to it even 
though the call desination is valid. The mitel also reports an error saying 
cp_dest_id is NULL.  (Call Processor destination ID is what I assume that 
means).  The Mitel has an incomming routing mechanism that says this in the 
help:

"the system will look for the dialed number provided by DID (DDI in Europe) 
or DNIS."  In reference to incomming trunks.
 
So, I am thinking there is no DID or DNIS information being send accross.  I 
caputed packets and I did not see a DID or DNIS field in the headers.  I am 
esentially trying to make the asterisk system look like a SIP provider 
(bandwidth.com for instance) with DIDs.  I can call from the Mitel to the 
Asterisk but i just cannot get the call to come back the other direction. 
 
What does it take to send information like this (DID and or DNIS) over a SIP 
trunk?  Can it be done and what should the headers, etc actually look like when 
looking at the packets? I basically want the asterisk to talk like a SIP DID 
and 
Trunk service provider.
 
-Jason--
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[asterisk-users] Asterisk Tech Tips: Cookin' with Asterisk

[Steve Sokol]

Greetings Asterisk Users, 


I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to 
host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon 
central time. Russell and Leif are project leaders and have collaborated on two 
Asterisk books: Asterisk: The Definitive Guide and Asterisk Cookbook , both 
published by O'Reilly & Associates. Asterisk: The Definitive Guide is the 
Asterisk bible -- a must-read for anyone learning to implement an 
Asterisk-based system. Asterisk Cookbook is a collection of recipes, simple 
code solutions you can put to work immediately, along with a detailed 
discussion that offers insight into why and how the recipe works. 


Leif will be presenting "Hot-Desking With The Asterisk Database". Learn how to 
utilize the Asterisk database and clever usage of the Asterisk Dialplan to 
implement hot desking. 


Russell will present "Debugging The Asterisk Dialplan". Learn how to utilize 
the Asterisk database and other dialplan functionality to build a flexible 
dialplan debugging system. 


Following the presentations we will open the floor for any Asterisk questions. 
Join us: register now! 


Thanks, 


-S 


Steve Sokol 
Digium, Inc. 

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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Tim Nelson]

On Wed, Apr 13, 2011 at 8:07 PM, satish patel < satish...@hotmail.com > wrote: 



Try dmesg command 

root@:~# dmesg | grep -i Sangoma 
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010 
Sangoma Technologies Inc 
[ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 
Sangoma Technologies Inc 
[ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) 
1994-2010 Sangoma Technologies Inc 
[ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma 
Technologies Inc 
[ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma 
Technologies Inc. 



Hi Satish 



dmesg | grep -i Sangoma does not show anything 



If lspci isn't showing the device, then your board is not recognizing it. Try 
another slot, and make sure the card is seated 100% tightly into the slot. 


And, of course, make sure the system is powered off while doing this. 99.999% 
of people already know this, but I've been bitten by the other 0.001% that 
don't. :-) 


If you're still not showing it, a call to Sangoma support is probably in order. 
They are top notch. 


--Tim --
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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Jim Dickenson]

Do you have the Sangoma wanpipe software installed?
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Apr 13, 2011, at 7:37 AM, satish patel wrote:

> Try dmesg command 
> 
> root@:~# dmesg | grep -i Sangoma
> [ 2303.473601] WANPIPE(tm) Hardware Support Module  3.5.19.0 (c) 1994-2010 
> Sangoma Technologies Inc
> [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 
> Sangoma Technologies Inc
> [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) 
> 1994-2010 Sangoma Technologies Inc
> [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma 
> Technologies Inc
> [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma 
> Technologies Inc.
> 
> 
> From: kaushalshri...@gmail.com
> Date: Wed, 13 Apr 2011 19:38:06 +0530
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo 
> Cancellation ( PCI Express ) Card
> 
> Hi,
> 
> I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI 
> Express ) Card installed on the box. Its not detected. Details are as below :-
> 
> [root@asterisk ~]# lspci
> 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01)
> 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
> 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
> 00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
> 00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller
> 00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
> 00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
> 00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller
> 00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10)
> 00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller
> 00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge
> 00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge
> 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 
> Audio Controller (rev 01)
> 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] 
> HyperTransport Technology Configuration
> 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] 
> Address Map
> 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM 
> Controller
> 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] 
> Miscellaneous Control
> 01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 
> 200G Series]
> 01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480)
> 02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI 
> Bridge (rev aa)
> 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit 
> Ethernet PCI Express (rev 20)
> 05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet 
> Adapter (rev 11)
> [root@asterisk ~]# cat /etc/redhat-release
> CentOS release 5.5 (Final)
> [root@asterisk ~]# asterisk -v
> Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others.
> Created by Mark Spencer 
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
> details.
> This is free software, with components licensed under the GNU General Public
> License version 2 and other licenses; you are welcome to redistribute it under
> certain conditions. Type 'core show license' for details.
> =
> Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk 
> -r' to connect.
> [root@asterisk ~]# 
> 
> Please suggest/guide
> 
> Thanks
> 
> Kaushal 
> 
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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Kaushal Shriyan]

On Wed, Apr 13, 2011 at 8:07 PM, satish patel  wrote:

>  Try dmesg command
>
> root@:~# dmesg | grep -i Sangoma
> [ 2303.473601] WANPIPE(tm) Hardware Support Module  3.5.19.0 (c) 1994-2010
> Sangoma Technologies Inc
> [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010
> Sangoma Technologies Inc
> [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c)
> 1994-2010 Sangoma Technologies Inc
> [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma
> Technologies Inc
> [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma
> Technologies Inc.
>
>
Hi Satish


dmesg | grep -i Sangoma does not show anything
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Re: [asterisk-users] Asterisk thread limit

[satish patel]

Oops!  Asterisk open 419 total thread and stop accepting connection. Can we 
control number of thread to open or limit ?

root@:~# ps -C asterisk -L -o pid,tid,pcpu,state,nlwp,args | wc -l
419


> Date: Wed, 13 Apr 2011 10:58:31 -0400
> From: lath...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk thread limit
> 
> On Wed, Apr 13, 2011 at 10:50 AM, satish patel  wrote:
> > Hi Guys!
> >
> > I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it
> > could handle in production so following is my senario.
> >
> > [sipp_client]---[Asterisk][sipp_server]
> >
> > sipp_client
> > ./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l
> > 1000 -r 250 -rp 5000 -m 1000
> >
> > sipp_server
> > ./sipp -sn uas -i 172.30.245.208
> >
> >
> > In above if i set -r 250 -rp 5000 calls per sec. in this case my asterisk
> > stopped accepting calls at  382 active calls and sipp client through error
> > "1302704824.872674: Can create thread to send RTP packets. (But asterisk is
> > still live to accept calls)
> > "
> >
> > I have ulimit is set to unlimited so just wondering is there any asterisk
> > number of thread limitation which we can set to go beyond this boundary?
> >
> > -S
> 
> Memory limit or load limit might cause this also.
> 
> -- 
> ~~~ Andrew "lathama" Latham lath...@gmail.com ~~~
> 
> --
> _
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Re: [asterisk-users] Asterisk thread limit

[Andrew Latham]

On Wed, Apr 13, 2011 at 10:50 AM, satish patel  wrote:
> Hi Guys!
>
> I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it
> could handle in production so following is my senario.
>
> [sipp_client]---[Asterisk][sipp_server]
>
> sipp_client
> ./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l
> 1000 -r 250 -rp 5000 -m 1000
>
> sipp_server
> ./sipp -sn uas -i 172.30.245.208
>
>
> In above if i set -r 250 -rp 5000 calls per sec. in this case my asterisk
> stopped accepting calls at  382 active calls and sipp client through error
> "1302704824.872674: Can create thread to send RTP packets. (But asterisk is
> still live to accept calls)
> "
>
> I have ulimit is set to unlimited so just wondering is there any asterisk
> number of thread limitation which we can set to go beyond this boundary?
>
> -S

Memory limit or load limit might cause this also.

-- 
~~~ Andrew "lathama" Latham lath...@gmail.com ~~~

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[asterisk-users] Asterisk thread limit

[satish patel]

Hi Guys!

I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could 
handle in production so following is my senario. 

[sipp_client]---[Asterisk][sipp_server]

sipp_client
./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 
-r 250 -rp 5000 -m 1000

sipp_server
./sipp -sn uas -i 172.30.245.208


In above if i set -r 250 -rp 5000 calls per sec. in this case my asterisk 
stopped accepting calls at  382 active calls and sipp client through error 
"1302704824.872674: Can create thread to send RTP packets. (But asterisk is 
still live to accept calls) 
" 

I have ulimit is set to unlimited so just wondering is there any asterisk 
number of thread limitation which we can set to go beyond this boundary?

-S 
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Re: [asterisk-users] Problems With DAHDI on Ubuntu

[satish patel]

make sure following init script is on at start up. 

/etc/init.d/dahdi 

Run lsmod command to make sure driver is loaded. 

-S


> Date: Wed, 13 Apr 2011 09:35:52 -0500
> From: sruff...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problems With DAHDI on Ubuntu
> 
> On Wed, Apr 13, 2011 at 09:48:47AM -0400, Shawn L wrote:
> > I have 2 separate Asterisk servers that are both exibiting this
> > problem.  1 has a 4 port FXO digium card, the other an 8 port.
> > 
> > For some reason when the machine reboots, the dahdi drivers are not
> > properly loaded.  Then asterisk ends up starting without dahdi
> > support.  I've tried everything that I can think of, even to the point
> > of running dahdi_cfg in the asterisk startup script before asterisk
> > itself is started, but it doesn't help.
> > 
> > To fix it, all i have to do is login and run
> > dahdi_cfg
> > /etc/init.d/asterisk restart
> > 
> > but that's a pain to have to do after every reboot.  I've never had
> > this problem with asterisk systems in the past, but now it's happening
> > on the last 2 servers we've setup.  Does anyone have any ideas?  I
> > can't understand why explicitly calling  dahdi_cfg or the
> > dahdi_startup script before starting asterisk isn't working.
> 
> Is there any output from either /etc/init.d/dahdi start or dmesg when
> this happens?
> 
> Normally the process is this (just in case this helps highlight
> something that might not be standard on your system):
> 
> 1) The system boots up, udev runs and *does not* load any drivers
> because all the drivers have been blacklisted in
> /etc/modprobe.d/dahdi.blacklist.conf
> 
> 2) The init scripts are run (distribution specific how you set there,
> chkconfig/update-rc.d/etc...) and /etc/init.d/dahdi looks in
> /etc/dahdi/modules to determine which board specific modules are loaded.
> 
> 3) After all the modules are loaded /etc/init.d/dahdi runs dahdi_cfg to
> parse /etc/dahdi/system.conf and configure the spans and channels
> appropriately.
> 
> In your email you say that the drivers are not properly loaded but to
> fix it neither "dahdi_cfg" or "/etc/init.d/asterisk restart" actually
> loads the drivers.
> 
> Cheers,
> Shaun
> 
> -- 
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[satish patel]

Try dmesg command 

root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module  3.5.19.0 (c) 1994-2010 
Sangoma Technologies Inc
[ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 
Sangoma Technologies Inc
[ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) 
1994-2010 Sangoma Technologies Inc
[ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma 
Technologies Inc
[ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma 
Technologies Inc.


From: kaushalshri...@gmail.com
Date: Wed, 13 Apr 2011 19:38:06 +0530
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo 
Cancellation ( PCI Express ) Card

Hi,

I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI 
Express ) Card installed on the box. Its not detected. Details are as below :-

[root@asterisk ~]# lspci
00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01)


00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller


00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller


00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10)
00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller
00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge
00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge


00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 Audio 
Controller (rev 01)
00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] 
HyperTransport Technology Configuration


00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address 
Map
00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM 
Controller
00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] 
Miscellaneous Control


01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 
200G Series]
01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480)
02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI 
Bridge (rev aa)


04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit 
Ethernet PCI Express (rev 20)
05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet 
Adapter (rev 11)
[root@asterisk ~]# cat /etc/redhat-release


CentOS release 5.5 (Final)
[root@asterisk ~]# asterisk -v
Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer 


Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under


certain conditions. Type 'core show license' for details.
=
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' 
to connect.


[root@asterisk ~]# 

Please suggest/guide

Thanks

Kaushal 


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Re: [asterisk-users] Problems With DAHDI on Ubuntu

[Shaun Ruffell]

On Wed, Apr 13, 2011 at 09:48:47AM -0400, Shawn L wrote:
> I have 2 separate Asterisk servers that are both exibiting this
> problem.  1 has a 4 port FXO digium card, the other an 8 port.
> 
> For some reason when the machine reboots, the dahdi drivers are not
> properly loaded.  Then asterisk ends up starting without dahdi
> support.  I've tried everything that I can think of, even to the point
> of running dahdi_cfg in the asterisk startup script before asterisk
> itself is started, but it doesn't help.
> 
> To fix it, all i have to do is login and run
> dahdi_cfg
> /etc/init.d/asterisk restart
> 
> but that's a pain to have to do after every reboot.  I've never had
> this problem with asterisk systems in the past, but now it's happening
> on the last 2 servers we've setup.  Does anyone have any ideas?  I
> can't understand why explicitly calling  dahdi_cfg or the
> dahdi_startup script before starting asterisk isn't working.

Is there any output from either /etc/init.d/dahdi start or dmesg when
this happens?

Normally the process is this (just in case this helps highlight
something that might not be standard on your system):

1) The system boots up, udev runs and *does not* load any drivers
because all the drivers have been blacklisted in
/etc/modprobe.d/dahdi.blacklist.conf

2) The init scripts are run (distribution specific how you set there,
chkconfig/update-rc.d/etc...) and /etc/init.d/dahdi looks in
/etc/dahdi/modules to determine which board specific modules are loaded.

3) After all the modules are loaded /etc/init.d/dahdi runs dahdi_cfg to
parse /etc/dahdi/system.conf and configure the spans and channels
appropriately.

In your email you say that the drivers are not properly loaded but to
fix it neither "dahdi_cfg" or "/etc/init.d/asterisk restart" actually
loads the drivers.

Cheers,
Shaun

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

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[asterisk-users] T38 fax detection using g729

[Niccolò Belli]

Hi, I continue the discussion from
https://issues.asterisk.org/view.php?id=19103

If T.38 reinvite detection should still work, why it doesn't?
If I use faxdetect = t38 it does never detect the fax, even using alaw.

Cheers,
Darkbasic

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[asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card

[Kaushal Shriyan]

Hi,

I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI
Express ) Card installed on the box. *Its not detected.* Details are as
below :-

[root@asterisk ~]# lspci
00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01)
00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge
00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller
00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller
00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller
00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10)
00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller
00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge
00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge
00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97
Audio Controller (rev 01)
00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
HyperTransport Technology Configuration
00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
Address Map
00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM
Controller
00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
Miscellaneous Control
01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress
200G Series]
01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series
(RS480)
02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI
Bridge (rev aa)
04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit
Ethernet PCI Express (rev 20)
05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet
Adapter (rev 11)
[root@asterisk ~]# cat /etc/redhat-release
CentOS release 5.5 (Final)
[root@asterisk ~]# asterisk -v
Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
-r' to connect.
[root@asterisk ~]#

Please suggest/guide

Thanks

Kaushal
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Re: [asterisk-users] Problems With DAHDI on Ubuntu

[Matt Riddell]

On 14/04/11 1:48 AM, Shawn L wrote:

I have 2 separate Asterisk servers that are both exibiting this
problem.  1 has a 4 port
FXO digium card, the other an 8 port.

For some reason when the machine reboots, the dahdi drivers are not
properly loaded.  Then asterisk
To fix it, all i have to do is login and run
dahdi_cfg
/etc/init.d/asterisk restart

but that's a pain to have to do after every reboot.  I've never had this


Don't know why it's happening, but add those lines to /etc/rc.local as a 
quick hack in the interim until you find out what's causing it.


--
Cheers,

Matt Riddell
___

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/cc.php (Call Centre Solutions)

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Re: [asterisk-users] Realtime SIP & peer status

[Steve Edwards]

On Wed, 13 Apr 2011, Andrew Thomas wrote:


Fair enough.  Then if this is really what you want I guess an AGI is the
best way to go.

As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].


I do almost exactly the same for PRIs with AGIs written in C. Each call 
may execute dozens of AGIs over a 10 minute or so call. 5 AGIs get 
executed before the caller hears the first prompt and it is almost 
instantaneous.


This is on a 6 year old Xeon (3.40GHz) server handling about 100 
concurrent calls.


You can execute XXX AGIs written in C in the time it takes to load the 
interpreter (Perl or PHP) for your script.


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

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[asterisk-users] Problems With DAHDI on Ubuntu

[Shawn L]

I have 2 separate Asterisk servers that are both exibiting this problem.  1
has a 4 port
FXO digium card, the other an 8 port.

For some reason when the machine reboots, the dahdi drivers are not properly
loaded.  Then asterisk
ends up starting without dahdi support.  I've tried everything that I can
think of, even to the point
of running dahdi_cfg in the asterisk startup script before asterisk itself
is started, but it doesn't help.

To fix it, all i have to do is login and run
dahdi_cfg
/etc/init.d/asterisk restart

but that's a pain to have to do after every reboot.  I've never had this
problem with asterisk systems in
the past, but now it's happening on the last 2 servers we've setup.  Does
anyone have any ideas?  I
can't understand why explicitly calling  dahdi_cfg or the dahdi_startup
script before starting asterisk
isn't working.

Thanks
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Re: [asterisk-users] AGI and forking

[Steve Edwards]

On Wed, 13 Apr 2011, A J Stiles wrote:

I want, when a call comes in on someone's DDI number (which the person 
who dialled it can only possibly have obtained by dialling 1471 after we 
called them), to be able to look up the caller's details from one of our 
databases (where the number ought to be stored, because we already 
dialled it).


Now, this search is going to take some time; so I'd like for the AGI 
script to fork a clone of itself, so the parent process can exit and the 
dialplan continue on to ring the person's phone, while the database 
lookup is done in the background (the script doesn't need to have any 
further contact with Asterisk -- it will initiate any necessary future 
communication via other channels).


I solved a similar problem with a multi-threaded AGI.

I created a thread to play 'Please wait while we verify your card details' 
while the main program did the database look ups and sent the auth out to 
our card processor.


By the time the prompt finished, I had the response back from the card 
processor so the 'wait' appeared to the caller to be instantaneous.


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

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Re: [asterisk-users] AGI and forking

[Thorsten Göllner]

Am 13.04.2011 15:08, schrieb A J Stiles:

Hi.  I just want to make sure I understand this before doing something that
might break things spectacularly for our users and customers  :)

We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.

I want, when a call comes in on someone's DDI number  (which the person who
dialled it can only possibly have obtained by dialling 1471 after we called
them),  to be able to look up the caller's details from one of our databases
(where the number ought to be stored, because we already dialled it).

Now, this search is going to take some time; so I'd like for the AGI script to
fork a clone of itself, so the parent process can exit and the dialplan
continue on to ring the person's phone, while the database lookup is done in
the background  (the script doesn't need to have any further contact with
Asterisk -- it will initiate any necessary future communication via other
channels).


Is this the sort of thing I need?

##  begin code snippet  ##

#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;

my $AGI = new Asterisk::AGI;
my %params = $AGI->ReadParse();

$SIG{CHLD} = "IGNORE";

if (my $child_pid = fork) {
 #  This is executed in the parent process
 exit;
}
elsif (defined $child_pid) {
 #  This is executed in the child process

 close STDIN;
 close STDOUT;
 close STDERR;

 #  Load some more modules and do some stuff
 #   that will take a long time

 exit;
}
else {
 die "Could not fork: $!";
};

##  end code snippet  ##

Am I right in thinking I shouldn't have to worry about zombie processes,
because the parent exits before the child and the init in modern Linux
distros is smart enough to deal with orphaned processes itself?

It should work - I think. BUT I am not really sure what will happen, if 
the child process exits. The child works with a copy of all asterisk 
ressources given to it, when forking. So when the child dies, perhaps 
asterisk will do a hangup or continues in the dialplan for this process. 
I think, that could cause some unwanted results.


You should try to write a daemon process which handles the database 
lookups or whataver while being totally independent of the atserisk 
process. This is a little bit more overhead for the communication 
between the astersk process (the call itself) and your lookup-daemon. 
But it would be more stable - for my point of view.


-Thorsten-

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Re: [asterisk-users] How to know extensions status ???

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, April 13, 2011 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know extensions status ???

 

Hi,

How to know the all SIP extensions status with AMI's  ExtensionState ?

What is the value should I pass in 

Context: <> ?? 
which will be define at context here ? shell I use sip.conf's  context for
that extension or any other?  

extension : <> ??
extension will be SIP/100 or just 100 ??


Please guide me ...
 
-
Thanks and regards

 Virendra Bhati
+91-9172341457



[Danny Nicholas] 

My guess would be just 100 (extension: 100) but you can verify it in a
telnet session.

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[asterisk-users] AGI and forking

[A J Stiles]

Hi.  I just want to make sure I understand this before doing something that 
might break things spectacularly for our users and customers  :)

We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.

I want, when a call comes in on someone's DDI number  (which the person who 
dialled it can only possibly have obtained by dialling 1471 after we called 
them),  to be able to look up the caller's details from one of our databases  
(where the number ought to be stored, because we already dialled it).

Now, this search is going to take some time; so I'd like for the AGI script to 
fork a clone of itself, so the parent process can exit and the dialplan 
continue on to ring the person's phone, while the database lookup is done in 
the background  (the script doesn't need to have any further contact with 
Asterisk -- it will initiate any necessary future communication via other 
channels).


Is this the sort of thing I need?

##  begin code snippet  ##

#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;

my $AGI = new Asterisk::AGI;
my %params = $AGI->ReadParse();

$SIG{CHLD} = "IGNORE";

if (my $child_pid = fork) {
#  This is executed in the parent process
exit;
}
elsif (defined $child_pid) {
#  This is executed in the child process

close STDIN;
close STDOUT;
close STDERR;

#  Load some more modules and do some stuff
#   that will take a long time

exit;
}
else {
die "Could not fork: $!";
};

##  end code snippet  ##

Am I right in thinking I shouldn't have to worry about zombie processes, 
because the parent exits before the child and the init in modern Linux 
distros is smart enough to deal with orphaned processes itself?

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] send voicemail to multiple emails

[vip killa]

I understand now the need for "externnotify" to run on vm check is to update
MWI. But I agree with M Hulber, an extra variable would be nice to tell the
script why "externnotify" was called...

On Wed, Apr 13, 2011 at 6:32 AM, M Hulber  wrote:

>  Instead of picking from multiple scripts, send the action to the script in
> a variable like the dial status
>
>
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[asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom

[David - asterisk list]

Asterisk as a phone system makes perfect sense in a condo. You can get

all the DID's you want and eliminate costs for the owners. You can offer
standard FXO for people who don't care and IP sets for people who want
to "upgrade" to feature sets.

Your door openner is a piece of cake.
1.  Create an option in your dialplan only in the "from-access-door"
context that reads DTMF from the called station only.
2. Use this to access an external program to turn on a serial port line
for 10 seconds.
3. This line drives a solid state relay (~$30) so you won't blow the
sink current on the PC port that drives a standard door lock.

A commercial door strike is about $100. The program to run the port is
childs play. Here is a test prog I used for turning on a power hungry
last printer. Change the comments and the sleep time and you're done.

  /*
   * lpon Lineprinter ON
   *  *** test program only **
   *
   *  (c) David Cook, 1994
   *
   *  Set signlal lines on serial port to turn on 5vdc
   *  signal. Used for solid-state relay (low current
   *  draw on RS232C port) to switch high voltage/high
   *  current load for printer.
   *
   *  Part of an intelligent print spooler to only power
   *  on/off high draw printer when required.
   *
   * Usage:   lpon  
   *  For example, lpon /dev/cua4 4 to set bit 3 on
   *  port /dev/cua4.
   *  "4" = 0100 or bit 3 which is DTR
   *  "2" = 0010 or bit 2 which is RTS
   *  "6" = 0110 or both DRT&  RTS
   */
  #include
  #include
  #include
  #include
  #include
  #include
  #include
  #include
  #include

  #include "lpswitch.h"

  /* Main program. */
  int main(int argc, char **argv)
  {
struct termios port_config;
int fd;
int set_bits = 2;

/* Open monitor device. */
if ((fd = open(SWDEV, O_RDWR | O_NDELAY))<  0) {
  fprintf(stderr, "lpswtich: %s: %d\n", SWDEV, strerror(errno));
  exit(1);}

cfmakeraw(&port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW,&port_config );
ioctl(fd, TIOCMSET,&set_bits );

/* wait for printer to warm up */
sleep(45);

/* not say "ready" and release the printer */
set_bits = 6;

cfmakeraw(&port_config );
port_config.c_iflag=port_config.c_iflag|IXON;
port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
tcsetattr( fd, TCSANOW,&port_config );
ioctl(fd, TIOCMSET,&set_bits );

close(fd);
}



On 12/04/2011 8:16 AM, asterisk-users-requ...@lists.digium.com wrote:

 Message: 3
 Date: Mon, 11 Apr 2011 18:21:39 -0500
 From: "Don Kelly"
 Subject: Re: [asterisk-users] Asterisk as a Condo door opener/intercom
 To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

 Message-ID:<8E20A6A94C9548C8A0E27B502B18F200@DonPC>
 Content-Type: text/plain;  charset="us-ascii"

 Continuing top posting...

 The same argument could be made for any commercial solution. Why use
 Asterisk when we could throw $4,000 at our problem for a commercial
 solution?

 I'd like to have a solution that would have the features you suggest for
 $400.

 --Don


 On Behalf Of C F
 Sent: Monday, April 11, 2011 11:43 AM

 Search the lists. Some hints:
 Viking electronics makes a door box that connects to any analog line
 (IIRC e-20).
 They also make a DTMF keypad that integrates in series with any analog
 line. They might also make a door box with a DTMF keypad on it.
 Sandman makes a relay that will get energized when there is a ring on
 the line which could be used to unlock the door.

 However, why would you use asterisk? Using asterisk for the sole
 purpose of MDU entry system is like using windows for asterisk, it
 works but why?
 Go for the commercial solutions, it comes with a geziilion options for
 your setup one of them the ability of chosing an apartment, another
 add key fobs, another one is the ability of using a code for the
 residence (not guests) to unlock the door. Also the interface with
 asterisk you will have to build one from scratch. The commercial
 solutions have em built in.

 On 4/10/11, Bruce B   wrote:

 >   Hi Everyone,
 >
 >   Looking to replace a condo intercom system. Apparently the current one

 taps

 >   into the lines and dials phone numbers but needs to be changed as it's
 >   faulty.
 >
 >   I will probably still use the same analogue dialing and back it up with a
 >   VoIP line and use the current cabling that is in place. But as for as the
 >   door opening function goes, I am not sure how to interface and how open
 >   these modules are usually built.
 >
 >   I would appreciate it if someone with experience can throw in some

 pointers

 >   as to what I might be facing and what challenges I have to solve to

 replace

 >   this with a nice Asterisk system.
 >
 >   Thanks,
 >



--

Re: [asterisk-users] [OT] Yealink Phones

[--[ UxBoD ]--]

- Original Message -
> 
> I've just started deploying these (well the T28P model) after years
> of
> Snom issues and they look pretty good (although the documentation is
> execrable; if you thought the Snom stuff was obtuse Yealink have got
> them knocked into a cocked hat!).
> 
> Anyway, for provisioning I use HTTP with a DHCP entry like:-
> 
>   #
> #   Yealink Phones
> #
> group {
> #
> # The phone should pickup the
> # model config file (y0.cfg for the
> # T28P) first and then the MAC.cfg file
> #
>   # Yes tftp-server-name to set the DHCP option 
> but
>   # the http:// tells the phone to get it's files 
> via
>   # http.
>   option tftp-server-name 
> "http://192.168.1.13/yealink";;
> #
> host yealinkT28P {
> hardware ethernet 00:15:65:1b:d9:12;
> fixed-address 192.168.1.33;
> option host-name "yealinkT28P";
> }
> }
> 
> As the comments say, the phone's first pick up the model dependant
> config file (y0.cfg for the T28P model) and then the
> MAC.cfg file.
> 
> This is nice as you have one model.cfg file for the site-wide config
> and
> then fine tune specific phones (setup different BLF keys and,
> obviously,
> SIP logins for each device) in the MAC.cfg files.
> 
> In the y0.cfg file I have:-
> 
>   #
>   #   Auto Provision
>   [ autoprovision ]
>   path = /config/Setting/autop.cfg
>   server_address = http://192.168.1.13/yealink
>   [ autop_mode ]
>   path = /config/Setting/autop.cfg
>   # Mode 7 = at Power On and Weekly
>   mode = 7
>   #   Sunday between 0100 and 0500
>   schedule_dayofweek=0
>   schedule_time = 01:00
>   schedule_time_end = 05:00
>   #
> 
> 
> Re non-web based access.  Obviously the config files are on your
> DHCP/Apache/Asterisk server so you can edit them however you like.
> 
> You can also enable telnet access to the phones with a 'hidden'
> config option of:-
> 
>   #
>   [ telnet ]
>   path=/config/Network/Network.cfg
>   telnet_enable=1
>   #
> 
> but the login/password are the admin defaults so a bit of a security
> hole there.  Not really found much useful telnetting into the phone
> but
> I've not played around with it much.
> 
> One other useful tip:  If you play around in the web interface, set
> the
> phone up and then export the config, you end up with a config.bin
> file
> which is just tar of the config files.  A quick diff and you can
> easily
> find out what you need to tweak in your Autoprovision config files.
> 
> Hope that helps.
> 
> PS - anyone else with useful Yealink tips?
> 

We are looking to switch to Yealink from SNOM and that last tip for saving the 
configuration is one I have recently asked them about. All sounds very 
promising and we hope to get some eval units soon :)
-- 
Thanks, Phil

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[asterisk-users] accessing currents calls from outside asterisk

[Albert]

Hi,

I am working on integration of 2 systems: asterisk and messaging
platform. What I need is to access somehow information about current
calls. Should I do it over AMI ?

I need to be able to perform those 2 actions:
- How can I obtain msisdns of current calls ?
- How to hangup one of current calls ?

Thanks for your help guys!

Regards,
Albert
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Re: [asterisk-users] send voicemail to multiple emails

[M Hulber]

Instead of picking from multiple scripts, send the action to the script 
in a variable like the dial status


On 4/12/11 4:58 PM, Warren Selby wrote:

Sorry for the top post, on my phone...

It makes sense for someone who has written a custom visual voicemail 
application to be able know when the vm has been checked...and that's 
just one example off the top of my head.


What I think would really be nice is that instead of removing features 
from externnotify, it could be enhanced in a way that you can tell it 
to run one script on a new vm, and a different one on a checked vm, 
and yet another one on a deleted vm, etc. Obviously all of the bases 
would have to be covered (move vm, listen to vm but not delete, skip 
vm, new vm, transferred vm, etc).


If I had the programming chops, I would take a crack at it myself.

Thanks,
--Warren Selby, dCAP

On Apr 12, 2011, at 3:28 PM, vip killa > wrote:


Honestly, I don't understand why "externnotify" should run when 
someone checks their voicemail... the change i made, makes sense so 
maybe that should be contributed to the asterisk source.


On Tue, Apr 12, 2011 at 4:26 PM, Steve Edwards @sedwards.com > wrote:


On Tue, 12 Apr 2011, vip killa wrote:

i ended up modify app_voicemail.c to not run
"externnotify" when someone checks their voicemail, it
now only runs when a new message is left... then
externnotify will send out the emails and other
notifications...


On Tue, 12 Apr 2011, Steve Edwards wrote:

I would suggest that enhancing mailcmd to include relevant
channel variables (like VM_*) in the child environment or on
the command line would be more useful than changing the
behavior of externnotify.


The 'difference' between enhancing mailcmd and changing
externotify is that the enhancement can be contributed and
accepted into the source so everybody can use it and the Asterisk
developers will maintain it for you while your change will always
be yours alone to remember to re-apply and update with each
revision of Asterisk.


-- 
Thanks in advance,

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

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Re: [asterisk-users] Variable inheritance with dialplan command Originate

[Naomi Rosenberg]

Hi

> I believe I made one mistake in my example, I don't use a call to Queue
> in my local channel without a partner channel (the customer). I'll
> revisit this later today when I have some time, I'll be glad to help you
> if I can recall the right solution :)

That would explain it. I wonder if we can get it working. Originate turned out 
to have its own issues (mysteriously hanging itself up after 30 seconds) and 
preesently my workaround involves an AGI script that writes and copies a call 
file. Ouch!

Naomi



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Re: [asterisk-users] [OT] Yealink Phones

[Andrew Thomas]

Hi Russell,

Have you seen the 'Action URL' bit yet?  Makes everything almost
key-system like ;)

BTW - one downfall of the Yealink is that it can't send different DND
commands to different accounts (it sends the one command to all
accounts). Not very useful if providers use different commands for DND
(like they tend to).  I know Yealink are working on this though - as I
am one of the 'beta' testers.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell
Brown
Sent: 13 April 2011 10:02
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [OT] Yealink Phones



I've just started deploying these (well the T28P model) after years of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).

Anyway, for provisioning I use HTTP with a DHCP entry like:-

#
#   Yealink Phones
#
group {
#
# The phone should pickup the
# model config file (y0.cfg for the
# T28P) first and then the MAC.cfg file
#
# Yes tftp-server-name to set the DHCP
option but
# the http:// tells the phone to get
it's files via
# http.
option tftp-server-name
"http://192.168.1.13/yealink";;
#
host yealinkT28P {
hardware ethernet 00:15:65:1b:d9:12;
fixed-address 192.168.1.33;
option host-name "yealinkT28P";
}
}

As the comments say, the phone's first pick up the model dependant
config file (y0.cfg for the T28P model) and then the MAC.cfg
file.

This is nice as you have one model.cfg file for the site-wide config and
then fine tune specific phones (setup different BLF keys and, obviously,
SIP logins for each device) in the MAC.cfg files.

In the y0.cfg file I have:-

#
#   Auto Provision
[ autoprovision ]
path = /config/Setting/autop.cfg
server_address = http://192.168.1.13/yealink
[ autop_mode ]
path = /config/Setting/autop.cfg
# Mode 7 = at Power On and Weekly
mode = 7
#   Sunday between 0100 and 0500
schedule_dayofweek=0
schedule_time = 01:00
schedule_time_end = 05:00
#


Re non-web based access.  Obviously the config files are on your
DHCP/Apache/Asterisk server so you can edit them however you like.

You can also enable telnet access to the phones with a 'hidden' config
option of:-

#
[ telnet ]
path=/config/Network/Network.cfg
telnet_enable=1
#

but the login/password are the admin defaults so a bit of a security
hole there.  Not really found much useful telnetting into the phone but
I've not played around with it much.

One other useful tip:  If you play around in the web interface, set the
phone up and then export the config, you end up with a config.bin file
which is just tar of the config files.  A quick diff and you can easily
find out what you need to tweak in your Autoprovision config files.

Hope that helps.

PS - anyone else with useful Yealink tips?

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimis

Re: [asterisk-users] Realtime SIP & peer status

[Jonas Kellens]

On 04/13/2011 11:46 AM, Andrew Thomas wrote:

Fair enough.  Then if this is really what you want I guess an AGI is the
best way to go.

As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].

Platform : CentOS 5.5
Asterisk : 1.6.2.16.1
Concurrent connections : 25
RAM : 512MB
CPU : 2GHz


Kind regards,
Jonas.

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Re: [asterisk-users] Realtime SIP & peer status

[Andrew Thomas]

Fair enough.  Then if this is really what you want I guess an AGI is the
best way to go.

As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2011 10:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP & peer status


On 04/13/2011 11:28 AM, Andrew Thomas wrote:
> Maybe I should have asked 'why do you want to put the status in to a 
> mySQL database'?
>
> BTW - extensions.conf has mySQL functions built in - so no external 
> script is actually needed.

Well, I read out this information in a website which serves as a 
comprehensible GUI.

I know I can use mysql-functions in the dialplan, but when I need to 
write something on answering, then I need the AGI-option of the 
Dial()-command.


Kind regards,
Jonas.

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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Realtime SIP & peer status

[Andrew Thomas]

http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL

And yes, I meant "Asterisk has mySQL commands built in [that can be
accessed via. extensions.conf]".  Sorry if I mislead.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq
Malik
Sent: 13 April 2011 10:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP & peer status


On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
> BTW - extensions.conf has mySQL functions built in - so no external 
> script is actually needed.
> 
>   
Could you point me in the right direction for that?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Realtime SIP & peer status

[Ishfaq Malik]

On Wed, 2011-04-13 at 10:32 +0100, Ishfaq Malik wrote:
> On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
> > BTW - extensions.conf has mySQL functions built in - so no external
> > script is actually needed.
> > 
> > 
> Could you point me in the right direction for that?
> 
Ignore that, I just realised what you meant...
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Realtime SIP & peer status

[Ishfaq Malik]

On Wed, 2011-04-13 at 11:24 +0200, Jonas Kellens wrote:
> On 04/13/2011 11:20 AM, Ishfaq Malik wrote:
> > On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
> >
> >> On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
> >>  
> >>> On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
> >>>
> >>>
>  Hello,
> 
>  I'm using SIP realtime with MySQL DB.
> 
>  Is it possible to get the status of the SIP peer (free / calling) from
>  this realtime DB ?
> 
>  If not, is there another way to obtain the call state of a SIP peer ?
> 
>   
> >>> You could use core show channels in the console/via AMI to determine if
> >>> any extensions are on a call or even making a call.
> >>>
> >>>
> >>>
> >> If this information is not available, then I'm thinking of writing an
> >> AGI and calling this AGI when a call is being answered. This AGI will
> >> then write to the MySQL-DB the state "busy" for this SIP peer.
> >> Off course when the call ends, I need another AGi in the h-exten which
> >> writes the state "free" for this SIP peer.
> >>
> >> You think this will work ? Or will it put too much load on my system ?
> >>
> >>
> >> Kind regards,
> >> Jonas.
> >>
> >>  
> > You could write a shell script to do what you suggested and pop it on a
> > cron. The info wouldn't be 100% realtime that way though but I think the
> > load would be very low.
> >
> > Also, as someone else has suggested, you could use hints but you have to
> > add some of the code for hints directly into the extensions.conf which
> > sort of goes against the point of RealTime unless you use scripts to
> > handle that part as I myself have done.
> >
> 
> Why should I use a cron ? I can just use an AGI in extensions.conf. 
> That's the closest to "realtime" I think.
> 
> How can I write information to a MySQL-DB using hints ? Please explain.
> 
> 
> Kind regards,
> Jonas.
> 
> --
TBH I hadn't though as far as how to write it to a DB, was just thinking
on ways of extension state reporting...
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Realtime SIP & peer status

[Jonas Kellens]

On 04/13/2011 11:28 AM, Andrew Thomas wrote:

Maybe I should have asked 'why do you want to put the status in to a
mySQL database'?

BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.


Well, I read out this information in a website which serves as a 
comprehensible GUI.


I know I can use mysql-functions in the dialplan, but when I need to 
write something on answering, then I need the AGI-option of the 
Dial()-command.



Kind regards,
Jonas.

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Re: [asterisk-users] Realtime SIP & peer status

[Ishfaq Malik]

On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
> BTW - extensions.conf has mySQL functions built in - so no external
> script is actually needed.
> 
>   
Could you point me in the right direction for that?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Realtime SIP & peer status

[Andrew Thomas]

Maybe I should have asked 'why do you want to put the status in to a
mySQL database'?

BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2011 10:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP & peer status


On 04/13/2011 11:20 AM, Ishfaq Malik wrote:
> On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
>
>> On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
>>  
>>> On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
>>>
>>>
 Hello,

 I'm using SIP realtime with MySQL DB.

 Is it possible to get the status of the SIP peer (free / calling) 
 from this realtime DB ?

 If not, is there another way to obtain the call state of a SIP peer

 ?

  
>>> You could use core show channels in the console/via AMI to determine

>>> if any extensions are on a call or even making a call.
>>>
>>>
>>>
>> If this information is not available, then I'm thinking of writing an

>> AGI and calling this AGI when a call is being answered. This AGI will

>> then write to the MySQL-DB the state "busy" for this SIP peer. Off 
>> course when the call ends, I need another AGi in the h-exten which 
>> writes the state "free" for this SIP peer.
>>
>> You think this will work ? Or will it put too much load on my system 
>> ?
>>
>>
>> Kind regards,
>> Jonas.
>>
>>  
> You could write a shell script to do what you suggested and pop it on 
> a cron. The info wouldn't be 100% realtime that way though but I think

> the load would be very low.
>
> Also, as someone else has suggested, you could use hints but you have 
> to add some of the code for hints directly into the extensions.conf 
> which sort of goes against the point of RealTime unless you use 
> scripts to handle that part as I myself have done.
>

Why should I use a cron ? I can just use an AGI in extensions.conf. 
That's the closest to "realtime" I think.

How can I write information to a MySQL-DB using hints ? Please explain.


Kind regards,
Jonas.

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 If you have received this communication in error we would appreciate
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and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
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any dissemination, distribution or copying of this communication and
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[asterisk-users] How to know extensions status ???

[virendra bhati]

Hi,

How to know the all SIP extensions status with AMI's  ExtensionState ?

What is the value should I pass in

Context: <> ??
which will be define at context here ? shell I use sip.conf's  context for
that extension or any other?

extension : <> ??
extension will be SIP/100 or just 100 ??


Please guide me ...

-
Thanks and regards

 Virendra Bhati
+91-9172341457
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Re: [asterisk-users] Realtime SIP & peer status

[Jonas Kellens]

On 04/13/2011 11:20 AM, Ishfaq Malik wrote:

On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
   

On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
 

On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:

   

Hello,

I'm using SIP realtime with MySQL DB.

Is it possible to get the status of the SIP peer (free / calling) from
this realtime DB ?

If not, is there another way to obtain the call state of a SIP peer ?

 

You could use core show channels in the console/via AMI to determine if
any extensions are on a call or even making a call.


   

If this information is not available, then I'm thinking of writing an
AGI and calling this AGI when a call is being answered. This AGI will
then write to the MySQL-DB the state "busy" for this SIP peer.
Off course when the call ends, I need another AGi in the h-exten which
writes the state "free" for this SIP peer.

You think this will work ? Or will it put too much load on my system ?


Kind regards,
Jonas.

 

You could write a shell script to do what you suggested and pop it on a
cron. The info wouldn't be 100% realtime that way though but I think the
load would be very low.

Also, as someone else has suggested, you could use hints but you have to
add some of the code for hints directly into the extensions.conf which
sort of goes against the point of RealTime unless you use scripts to
handle that part as I myself have done.
   


Why should I use a cron ? I can just use an AGI in extensions.conf. 
That's the closest to "realtime" I think.


How can I write information to a MySQL-DB using hints ? Please explain.


Kind regards,
Jonas.

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Re: [asterisk-users] Realtime SIP & peer status

[Ishfaq Malik]

On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
> On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
> > On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
> >
> >> Hello,
> >>
> >> I'm using SIP realtime with MySQL DB.
> >>
> >> Is it possible to get the status of the SIP peer (free / calling) from
> >> this realtime DB ?
> >>
> >> If not, is there another way to obtain the call state of a SIP peer ?
> >>  
> > You could use core show channels in the console/via AMI to determine if
> > any extensions are on a call or even making a call.
> >
> >
> If this information is not available, then I'm thinking of writing an 
> AGI and calling this AGI when a call is being answered. This AGI will 
> then write to the MySQL-DB the state "busy" for this SIP peer.
> Off course when the call ends, I need another AGi in the h-exten which 
> writes the state "free" for this SIP peer.
> 
> You think this will work ? Or will it put too much load on my system ?
> 
> 
> Kind regards,
> Jonas.
> 
You could write a shell script to do what you suggested and pop it on a
cron. The info wouldn't be 100% realtime that way though but I think the
load would be very low.

Also, as someone else has suggested, you could use hints but you have to
add some of the code for hints directly into the extensions.conf which
sort of goes against the point of RealTime unless you use scripts to
handle that part as I myself have done.

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Realtime SIP & peer status

[Andrew Thomas]

Why not use hints instead?


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2011 10:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP & peer status


On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
> On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
>
>> Hello,
>>
>> I'm using SIP realtime with MySQL DB.
>>
>> Is it possible to get the status of the SIP peer (free / calling) 
>> from this realtime DB ?
>>
>> If not, is there another way to obtain the call state of a SIP peer ?
>>  
> You could use core show channels in the console/via AMI to determine 
> if any extensions are on a call or even making a call.
>
>
If this information is not available, then I'm thinking of writing an 
AGI and calling this AGI when a call is being answered. This AGI will 
then write to the MySQL-DB the state "busy" for this SIP peer. Off
course when the call ends, I need another AGi in the h-exten which 
writes the state "free" for this SIP peer.

You think this will work ? Or will it put too much load on my system ?


Kind regards,
Jonas.

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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Realtime SIP & peer status

[Jonas Kellens]

On 04/13/2011 10:57 AM, Ishfaq Malik wrote:

On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
   

Hello,

I'm using SIP realtime with MySQL DB.

Is it possible to get the status of the SIP peer (free / calling) from
this realtime DB ?

If not, is there another way to obtain the call state of a SIP peer ?
 

You could use core show channels in the console/via AMI to determine if
any extensions are on a call or even making a call.

   
If this information is not available, then I'm thinking of writing an 
AGI and calling this AGI when a call is being answered. This AGI will 
then write to the MySQL-DB the state "busy" for this SIP peer.
Off course when the call ends, I need another AGi in the h-exten which 
writes the state "free" for this SIP peer.


You think this will work ? Or will it put too much load on my system ?


Kind regards,
Jonas.

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Re: [asterisk-users] [OT] Yealink Phones

[Russell Brown]

I've just started deploying these (well the T28P model) after years of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).

Anyway, for provisioning I use HTTP with a DHCP entry like:-

#
#   Yealink Phones
#
group {
#
# The phone should pickup the
# model config file (y0.cfg for the
# T28P) first and then the MAC.cfg file
#
# Yes tftp-server-name to set the DHCP option 
but
# the http:// tells the phone to get it's files 
via
# http.
option tftp-server-name 
"http://192.168.1.13/yealink";;
#
host yealinkT28P {
hardware ethernet 00:15:65:1b:d9:12;
fixed-address 192.168.1.33;
option host-name "yealinkT28P";
}
}

As the comments say, the phone's first pick up the model dependant
config file (y0.cfg for the T28P model) and then the
MAC.cfg file.

This is nice as you have one model.cfg file for the site-wide config and
then fine tune specific phones (setup different BLF keys and, obviously,
SIP logins for each device) in the MAC.cfg files.

In the y0.cfg file I have:-

#
#   Auto Provision
[ autoprovision ]
path = /config/Setting/autop.cfg
server_address = http://192.168.1.13/yealink
[ autop_mode ]
path = /config/Setting/autop.cfg
# Mode 7 = at Power On and Weekly
mode = 7
#   Sunday between 0100 and 0500
schedule_dayofweek=0
schedule_time = 01:00
schedule_time_end = 05:00
#


Re non-web based access.  Obviously the config files are on your
DHCP/Apache/Asterisk server so you can edit them however you like.

You can also enable telnet access to the phones with a 'hidden'
config option of:-

#
[ telnet ]
path=/config/Network/Network.cfg
telnet_enable=1
#

but the login/password are the admin defaults so a bit of a security
hole there.  Not really found much useful telnetting into the phone but
I've not played around with it much.

One other useful tip:  If you play around in the web interface, set the
phone up and then export the config, you end up with a config.bin file
which is just tar of the config files.  A quick diff and you can easily
find out what you need to tweak in your Autoprovision config files.

Hope that helps.

PS - anyone else with useful Yealink tips?

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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Re: [asterisk-users] Poor call quality - line drop, chopping sound, like robotic voice, Both party could not hear caller voice

[Andrew Thomas]

7. Take an Asterisk training course and become a dCAP.

As for "we have try to solve it but we're lack of asterisk knowledge" -
would you get a plumber to service your car?  Why not employ (as in 'pay
money') somebody to investigate this further.  As Satish pointed out -
QoS type issues take a lot of debugging and that usually has to be done
on-site.

BTW - I doubt any of this is caused by the PRI circuit.




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 12 April 2011 15:56
To: asterisk-users
Subject: Re: [asterisk-users] Poor call quality - line drop, chopping
sound, like robotic voice, Both party could not hear caller voice


It is Hard to say there is a problem but there are few points you should
verify. 

1. Codec (ulaw is good if you have enough bandwidth) 
2. System CPU load 
3. Bandwidth 
4. PRI card ( Hardware echo cancellation or software )
5. Kernel option CONFIG_HZ=1000  (Worth have this option) 
6. Finally Google your issue.   



Date: Tue, 12 Apr 2011 09:42:37 +0800
From: man.evolut...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Poor call quality - line drop, chopping sound,
like robotic voice, Both party could not hear caller voice

One of our client facing this issue, we have try to solve it but we're
lack of asterisk knowledge. Anybody can help us? Isn't any problem with
asterisk configuration or the problem come from PRI E1 itself?

[Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer
capability: 0x00 - SPEECH
[Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for
channel '0xb67f3d50'
[Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- Called g0/0X
[Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel DAHDI/2-1 to read
format ulaw
[Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel SIP/2130-06fb
to write format ulaw
[Apr 11 15:32:48] DEBUG[9231] channel.c: Set channel SIP/2130-06fb
to read format alaw
[Apr 11 15:32:48] DEBUG[2993] manager.c: Manager received command
'GetVar'
[Apr 11 15:32:48] NOTICE[9231] rtp.c: Unknown RTP codec 126 received
from '192.168.100.130'
[Apr 11 15:32:48] DEBUG[2993] manager.c: Manager received command
'GetVar'
[Apr 11 15:32:48] DEBUG[6914] chan_dahdi.c: Queuing frame from
PRI_EVENT_PROCEEDING on channel 0/2 span 1
[Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- DAHDI/2-1 is proceeding
passing it to SIP/2130-06fb
[Apr 11 15:32:48] DEBUG[9231] rtp.c: Ooh, format changed from unknown to
ulaw
[Apr 11 15:32:48] DEBUG[9232] audiohook.c: Failed to get 160 samples
from read factory 0xb7817dd0
[Apr 11 15:32:48] DEBUG[9231] rtp.c: Created smoother: format: 4 ms: 20
len: 160
[Apr 11 15:32:48] DEBUG[6915] chan_dahdi.c: Queuing frame from
PRI_EVENT_PROGRESS on channel 0/4 span 2
[Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making
progress passing it to SIP/2052-06fa
[Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making
progress passing it to SIP/2052-06fa
[Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for
656de8c01fcfde12371cfaa41a6cc357@127.0.1.1 - OPTIONS (No RTP)
[Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Initializing initreq for
method OPTIONS - callid 0ca5e0f16cc3027a450c5ce920189bc5@192.168.100.238
[Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Stopping retransmission on
'0ca5e0f16cc3027a450c5ce920189bc5@192.168.100.238' of Request 102: Match
Found
[Apr 11 15:32:49] DEBUG[30773] rtp.c: Got RTCP report of 76 bytes
[Apr 11 15:32:49] DEBUG[9169] rtp.c: Got RTCP report of 76 bytes
[Apr 11 15:32:49] DEBUG[9198] rtp.c: Got RTCP report of 64 bytes
[Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for
43bdc76362a70a0f138c364455fa976d@127.0.1.1 - OPTIONS (No RTP)
[Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Initializing initreq for
method OPTIONS - callid 645d2453470d5e3f1b3300d36c1f336b@192.168.100.238
[Apr 11 15:32:49] DEBUG[9189] rtp.c: Got RTCP report of 72 bytes
[Apr 11 15:32:49] DEBUG[6893] chan_sip.c: Stopping retransmission on
'645d2453470d5e3f1b3300d36c1f336b@192.168.100.238' of Request 102: Match
Found
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 160 samples
from read factory 0x87b8ee8
[Apr 11 15:32:49] DEBUG[9204] audiohook.c: Failed to get 1

Re: [asterisk-users] Realtime SIP & peer status

[Ishfaq Malik]

On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
> Hello,
> 
> I'm using SIP realtime with MySQL DB.
> 
> Is it possible to get the status of the SIP peer (free / calling) from
> this realtime DB ?
> 
> If not, is there another way to obtain the call state of a SIP peer ?
> 
> 
> Kind regards,
> Jonas.
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

You could use core show channels in the console/via AMI to determine if
any extensions are on a call or even making a call.


-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] send voicemail to multiple emails

[Andrew Thomas]

One of those points being to turn off a MWI if the total number of msgs
reaches 0.

Not everyone uses the same extension number for the user as for the
device they are on (think hot-desking).

I'm glad you got it to work your way :)


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 12 April 2011 22:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] send voicemail to multiple emails


On Tuesday 12 April 2011 15:28:50 vip killa wrote:
> Honestly, I don't understand why "externnotify" should run when 
> someone checks their voicemail... the change i made, makes sense so 
> maybe that should be contributed to the asterisk source.

The point of it is to run whenever there is a potential change in state.
It runs after someone checks voicemail, because checking voicemail can
clear new messages (or change the number of new messages).  You are
certainly welcome, in the external notification script to exit early, if
you determine that further action is not necessary.

The idea that we even need an extra flag for this is ridiculous.  If
there is that much penalty in starting up another script instance,
you're using the wrong scripting language or the wrong machine.

-- 
Tilghman

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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] Realtime SIP & peer status

[Michel Verbraak]

On 13-04-11 10:15, Jonas Kellens wrote:
> Hello,
>
> I'm using SIP realtime with MySQL DB.
>
> Is it possible to get the status of the SIP peer (free / calling) from
> this realtime DB ?
>
NO

> If not, is there another way to obtain the call state of a SIP peer ?
>
AMI (Asterisk Manager Interface)
>
> Kind regards,
> Jonas.
>
>
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[asterisk-users] 0018818: [patch] Crashing when using local channels and realtime on asterisk 1.8.3-rc2

[Ishfaq Malik]

Hi

Does anyone know if this fix is going to go into the next release of
1.8.x?

Also, is there even any rough timeframe of that release coming out?
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] Realtime SIP & peer status

[Jonas Kellens]

Hello,

I'm using SIP realtime with MySQL DB.

Is it possible to get the status of the SIP peer (free / calling) from 
this realtime DB ?


If not, is there another way to obtain the call state of a SIP peer ?


Kind regards,
Jonas.
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Re: [asterisk-users] Basic queue question

[federico cabiddu]

Hi,
in this case the Agents will be called in the order you defined them
in the queue.

Regards,

Federico

2011/4/12 satish patel :
>> strategy=linear
>
> In this case always call first land on A and then B right ? or random ?
>
> -Satish
>
>> Date: Tue, 12 Apr 2011 22:42:25 +0200
>> From: federico.cabi...@gmail.com
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Basic queue question
>>
>> Hi,
>> have a look here:
>> http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf
>> I think that in your case the functionality you want can be achieved
>> setting for your queue
>>
>> strategy=linear
>>
>> Regards,
>>
>> Federico
>>
>> 2011/4/12 satish patel :
>> > Hey Guys!
>> >
>> > We have very simple queue with basic options. We have two agent in queue
>> > A
>> > and B. Issue is if i dial in queue and A is unavailable then call not
>> > rollover to B just playing moh and then putting call in voicemail. I
>> > want
>> > call rollover thing like if A is not available or in case not able to
>> > pick
>> > call then call should ring B.
>> >
>> > what would be the best option for this kind of queue functionality.
>> >
>> > -Satish
>> >
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>> >
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