Re: [asterisk-users] asterisk with 1000 extensions

[Steve Edwards]

Please don't top-post.

On Thu, 7 Mar 2013, Bharat Lalcheta wrote:


You can use ATA box with pstn phone to reduce cost.


Are you wiring a building where multiple-line SIP gateways make sense?

How about a description of what you are trying to do?

Personally, I like Polycom SIP phones but I don't have to buy 1,000 of 
them :)


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Re: [asterisk-users] asterisk with 1000 extensions

[Kamlesh Kumar]

Server side installation with recent hardware is fine, we can build two 
parallel system for redundancy. We are more concern with the cost of SIP client 
(hardphone). What are the various ways to make this setup functional with low 
cost for SIP clients. Thanks,Kamlesh
 
> On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
> 
> > softphone is not going to be used in this setup. Hardphone is required. 
> > Around 60-70 simultaneous calls would be required.
> 
> OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big 
> deal.
> 
> I'd look for a reliable system and build 2 so you can swap between them as 
> needed. Going the full redundant, heartbeat kind of setup may be more 
> trouble than it is worth depending on how tolerant your users are to the 
> very occasional outage.
> 
> A couple of years ago, I bought a used Supermicro server with a 3.2 Ghz P4 
> off Ebay for $150 including shipping. Earlier this week, I updated the OS 
> and rebooted it. The uptime was 574 days.
> 
> -- 
> 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] asterisk with 1000 extensions

[Steve Edwards]

Please don't top-post.

On Thu, 7 Mar 2013, Kamlesh Kumar wrote:

softphone is not going to be used in this setup. Hardphone is required. 
Around 60-70 simultaneous calls would be required.


OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big 
deal.


I'd look for a reliable system and build 2 so you can swap between them as 
needed. Going the full redundant, heartbeat kind of setup may be more 
trouble than it is worth depending on how tolerant your users are to the 
very occasional outage.


A couple of years ago, I bought a used Supermicro server with a 3.2 Ghz P4 
off Ebay for $150 including shipping. Earlier this week, I updated the OS 
and rebooted it. The uptime was 574 days.


--
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] asterisk with 1000 extensions

[Kamlesh Kumar]

softphone is not going to be used in this setup. Hardphone is required. Around 
60-70 simultaneous calls would be required. Thanks,Kamlesh
 Date: Wed, 6 Mar 2013 21:15:51 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk with 1000 extensions

On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
 
> Technology is SIP and asterisk is not handling the media, what is 
> cheapest solution to be used for SIP client.
 
Client? How about a free SIP softphone?
 
Server? How many calls per second? How many simultaneous calls? Any 
half-way recent box should do. An Atom, i3, etc. Reliability and 
redundancy are going to be important unless you want 1,000 people calling 
you :)
 
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] asterisk with 1000 extensions

[Steve Edwards]

On Thu, 7 Mar 2013, Kamlesh Kumar wrote:

Technology is SIP and asterisk is not handling the media, what is 
cheapest solution to be used for SIP client.


Client? How about a free SIP softphone?

Server? How many calls per second? How many simultaneous calls? Any 
half-way recent box should do. An Atom, i3, etc. Reliability and 
redundancy are going to be important unless you want 1,000 people calling 
you :)


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Re: [asterisk-users] asterisk with 1000 extensions

[Kamlesh Kumar]

Technology is SIP and asterisk is not handling the media, what is cheapest 
solution to be used for SIP client. Thanks,Kamlesh
 Date: Wed, 6 Mar 2013 20:43:52 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk with 1000 extensions

On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
 
> We need to setup asterisk server for 1000 extensions and in this 
> setup only extension to extension dialling is required (without call 
> recording and voicemail), like intercom calling. Please let us know what 
> can be the best economic solution/setup for this.
 
The number of extensions is not the key factor. The number of simultaneous 
calls is.
 
What technology? SIP? Dahdi?
 
If all you are going to do is call from endpoint to endpoint, maybe 
something like Kamailio or OpenSIPS is appropriate.
 
If Asterisk is not handling the media, probably any old crappy computer 
can handle the call setup/call teardown load.
 
-- 
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] asterisk with 1000 extensions

[Steve Edwards]

On Thu, 7 Mar 2013, Kamlesh Kumar wrote:

We need to setup asterisk server for 1000 extensions and in this 
setup only extension to extension dialling is required (without call 
recording and voicemail), like intercom calling. Please let us know what 
can be the best economic solution/setup for this.


The number of extensions is not the key factor. The number of simultaneous 
calls is.


What technology? SIP? Dahdi?

If all you are going to do is call from endpoint to endpoint, maybe 
something like Kamailio or OpenSIPS is appropriate.


If Asterisk is not handling the media, probably any old crappy computer 
can handle the call setup/call teardown load.


--
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-
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Re: [asterisk-users] RPM updates

[joakimsen]

I am still facing this issue. Is AsteriskNOW and the CentOS repositories
depreciated?


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Steven Howes
> Sent: Wednesday, February 06, 2013 9:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RPM updates
> 
> On 28 Jan 2013, at 13:55, Steven Howes wrote:
> > Who do I need to poke to get the yum repository / RPM files updated? The
> dahdi RPMs are not up to date with the CentOS kernel versions any more,
it's
> making doing an installation a bit tricky due to dependancies, I'd rather
not
> roll back / remove new kernels if I don't have to..
> 
> 
> Cheers for the replies regarding alternative repos. I'm looking to keep
using
> the Digium ones, but they're still broken. Guess I'll just have to wait
until
> someone at Digium notices :S
> 
> Steve
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[asterisk-users] asterisk with 1000 extensions

[Kamlesh Kumar]

Hello, We need to setup asterisk server for 1000 extensions and in this setup 
only extension to extension dialling is required (without call recording and 
voicemail), like intercom calling. Please let us know what can be the best 
economic solution/setup for this. Thanks,Kamlesh
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Re: [asterisk-users] Asterisk crashed

[Bharat Lalcheta]

Can you provide OS details ? Its seems problem of abrt. Did u tested
asterisk without abrt

Regards,

Bharat Lalcheta

On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza
 wrote:
> Hi,
>
> I am running asterisk 1.8.14.0, It was running fine for last few days and
> suddenly crashed today
>
> In logs I can see that abrt tried to save the core dump but it couldn't
>
> Mar  6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip
> 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000]
> Mar  6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528
> (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528
> (450703360 bytes)
> Mar  6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12:11:09-26528'
> creation detected
> Mar  6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't
> belong to any package
> Mar  6 12:11:15 localhost abrtd: 'post-create' on
> '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1
>
> *Asterisk was running as root user
>
> Any suggestions?
>
> Regards,
> Zohair Raza
>
>
>
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Re: [asterisk-users] Redirect incoming call to SIP trunk.

[Emiliano Vazquez]

El 06/03/13 17:48, Administrator TOOTAI escribió:

Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :

Solved.


Great, happy for you.

What would be nice is to explain how you solve it for archives. Other 
people can run in the same problematic that yours and would be happy 
to see your way to get out of it 

+1


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Re: [asterisk-users] Redirect incoming call to SIP trunk.

[Steve Totaro]

On Wed, Mar 6, 2013 at 3:48 PM, Administrator TOOTAI  wrote:
> Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :
>>
>> Solved.
>
>
> Great, happy for you.
>
> What would be nice is to explain how you solve it for archives. Other people
> can run in the same problematic that yours and would be happy to see your
> way to get out of it
>

I would bet you that is exactly what he did.  This list has died off
so much because you can find almost every answer in the archives now.

Thanks,
Steve Totaro

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Re: [asterisk-users] Redirect incoming call to SIP trunk.

[Administrator TOOTAI]

Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :

Solved.


Great, happy for you.

What would be nice is to explain how you solve it for archives. Other 
people can run in the same problematic that yours and would be happy to 
see your way to get out of it




2013/3/5 Luis H. Forchesatto >


Greetings.

I got two asterisk servers, one is connected to another via sip
trunk. The incoming calls are routed to the time period an if
matches is transfered to the designed extension. If don't, is
redirected to a second time period. Then, if the call matches the
second time period it need to be transfered to the trunk that
directs to the second server.

How do I do to configure it this way?

The trunk it must be transfered has a outbound route configured too.

Any further details just ask.


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Re: [asterisk-users] AGI Script

[Steve Edwards]

Please don't top-post...

On 06/03/2013, at 13:24, Steve Edwards  
wrote:


Can you enable AGI debugging on the Asterisk console and see if that 
yields any clues?


On Wed, 6 Mar 2013, Gustavo Salvador wrote:

I have run asterisk in verbose mode, and also set on the debug, but the 
only clue I have is just this error.


Did that include 'agi set debug on?'

Can you 'cut-n-paste' the relevant 'sanitized' console output?

I looked through my AGIs and find I always set channel variables and let 
the dialplan do the actual dial().


1) Is your AGI exiting before the dial() completes?

2) If you execute the same dial() command from the 'AGI debug output' 
(which should show the expanded variables) in your dialplan, does that 
yield andy clues?


3) If you use another technology like SIP can you enable SIP debugging and 
observe the SIP dialog?


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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] AGI Script

[Gustavo Salvador]

Thanks Steve,

That was just a snippet, the complete script is as follow:

#!/usr/bin/perl

use strict;


$|=1;


# Setup some variables

my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;

my $key; my $value;


while() {

chomp;

last unless length($_);

if (/^agi_(\w+)\:\s+(.*)$/) {

$AGI{$1} = $2;

}

}


print STDERR "AGI Environment Dump:\n";

foreach my $i (sort keys %AGI) {

print STDERR " -- $i = $AGI{$i}\n";

}


sub checkresult {

my ($res) = @_;

my $retval;

$tests++;

chomp $res;

if ($res =~ /^200/) {

$res =~ /result=(-?\d+)/;

if (!length($1)) {

print STDERR "FAIL ($res)\n";

$fail++;

} else {

print STDERR "PASS ($1)\n";

$pass++;

}

} else {

print STDERR "FAIL (unexpected result '$res')\n";

$fail++;

}

}


my $count = keys(%AGI);

print "EXEC Dial('DAHDI/g2/$AGI{dnid},,W');

my $result = ;

&checkresult($result);

There I'm not using the AGI class because as test not works ( seems to me 
because it is not there within this asterisk version) . I have tested the 
Script communication with asterisk works making it to execute a voice prompt.

I have run asterisk in verbose mode, and also set on the debug, but the only 
clue I have is just this error.

Regards,

Gustavo

On 06/03/2013, at 13:24, Steve Edwards  wrote:

> On Wed, 6 Mar 2013, Gustavo Salvador wrote:
> 
>>> I'm writing an AGI Perl Script...
> 
>>> =
>>> #!/usr/bin/perl
>>> use strict;
>>> 
>>> my %AGI;
>>> :
>>> print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)";
>>> =
> 
> Is this your entire script or just a snippet? If this is all, this is not an 
> AGI.
> 
> An AGI is an executable that follows the AGI protocol. At a minimum, this 
> means:
> 
> 1) Read the AGI variables from STDIN.
> 
> 2) Write an AGI request to STDOUT.
> 
> 3) Read the AGI response from STDIN.
> 
> 4) Repeat steps 2 & 3 as needed.
> 
> Asterisk creates the process executing your AGI and sends a bunch of cruft to 
> it via the process's STDIN. If you don't read this, even if you don't need to 
> use the variables, your AGI will not execute correctly and reliably.
> 
> If you do not read the response after every request, your AGI will not 
> execute correctly and reliably.
> 
> Most people use an established AGI library since nobody understands the 
> implications of the protocol correctly the first time.
> 
> Can you enable AGI debugging on the Asterisk console and see if that yields 
> any clues?
> 
>>> When dialplan executes the AGI, asterisk throw the following error: 
>>> "Dropping incompatible voice fraile on SIP/INCONCERT-4796 of formar 
>>> ulaw since our native format has changed to 0x8 (alaw)."
> 
> (Actual 'cut-n-paste' is better than 'retyping' console output.)
> 
> I think if you take another look, this is a warning, not an error. No big 
> deal, it means just what it says, just ignore it.
> 
> -- 
> 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] Asterisk crashed

[Zohair Raza]

Hi,

I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed today

In logs I can see that abrt tried to save the core dump but it couldn't

Mar  6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac
ip 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000]
Mar  6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528
(/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528
(450703360 bytes)
Mar  6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12:11:09-26528'
creation detected
Mar  6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't
belong to any package
Mar  6 12:11:15 localhost abrtd: 'post-create' on
'/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1

*Asterisk was running as root user

Any suggestions?

Regards,
Zohair Raza
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Re: [asterisk-users] Error to install Asterisk

[Alex Villací­s Lasso]

El 06/03/13 11:52, Carlos Alvarez escribió:
I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact.  If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers.  You really need to know how to use the OS before you can learn to 
manage services running on it.  I strongly suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is simply no way to run a production server without having to do systems management regularly.



On Wed, Mar 6, 2013 at 3:01 AM, termo termosel mailto:fermit...@hotmail.com>> wrote:

Hi,

this is the outpu to df -h command:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi


Look carefully at the df -h output. It seems that the OP is trying to install 
and run asterisk from inside an Ubuntu livecd session. Whatever the result of 
the installation, it will be wiped out on the next restart.
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Re: [asterisk-users] AGI Script

[Steve Edwards]

On Wed, 6 Mar 2013, Gustavo Salvador wrote:


I'm writing an AGI Perl Script...



=
#!/usr/bin/perl
use strict;

my %AGI;
:
print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)";
=


Is this your entire script or just a snippet? If this is all, this is not 
an AGI.


An AGI is an executable that follows the AGI protocol. At a minimum, this 
means:


1) Read the AGI variables from STDIN.

2) Write an AGI request to STDOUT.

3) Read the AGI response from STDIN.

4) Repeat steps 2 & 3 as needed.

Asterisk creates the process executing your AGI and sends a bunch of cruft 
to it via the process's STDIN. If you don't read this, even if you don't 
need to use the variables, your AGI will not execute correctly and 
reliably.


If you do not read the response after every request, your AGI will not 
execute correctly and reliably.


Most people use an established AGI library since nobody understands the 
implications of the protocol correctly the first time.


Can you enable AGI debugging on the Asterisk console and see if that 
yields any clues?


When dialplan executes the AGI, asterisk throw the following error: 
"Dropping incompatible voice fraile on SIP/INCONCERT-4796 of formar 
ulaw since our native format has changed to 0x8 (alaw)."


(Actual 'cut-n-paste' is better than 'retyping' console output.)

I think if you take another look, this is a warning, not an error. No big 
deal, it means just what it says, just ignore it.


--
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 Script

[Gustavo Salvador]

Thanks,

But SIP uses the caller box to send the call to the second box where is running 
the AGI script, the second box uses DAHDI to routes the call to E1. I've tested 
the codec routing a call between a E1 extension and a local one with the 
originate extension command and works.
So that is because I'm loose with this

Regards,

Gustavo

On 06/03/2013, at 12:12, Gertjan Baarda  wrote:

> Might be a codec issue, try allow=all in your sip.conf
> 
> Sent from my iPhone
> 
> On 6 mrt. 2013, at 17:49, Gustavo Salvador
>  wrote:
> 
>>> 
>>> Hi every body,
>>> 
>>> Please if some one could help me with this:
>>> I'm writing an AGU Perl Script which basically makes a call using an 
>>> extension provided by other asterisk box to an E1. The asterisk version is 
>>> 1.6.0.28, so it hasn't the Wellington know AGI class. The code is as 
>>> follows:
>>> 
>>> =
>>> #!/usr/bin/perl
>>> use strict;
>>> 
>>> my %AGI;
>>> :
>>> print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)";
>>> =
>>> 
>>> When dialplan executes the AGI, asterisk throw the following error: 
>>> "Dropping incompatible voice fraile on SIP/INCONCERT-4796 of formar 
>>> ulaw since our native format has changed to 0x8 (alaw)."
>>> 
>>> And connection is never make.
>>> 
>>> It's the code OK?, I'm newbie on asterisk, and really don't know what is 
>>> going wrong.
>>> 
>>> Regards,
>>> 
>>> Gustavo
>> 
>> --
>> _
>> -- 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
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>>  http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [asterisk-users] Error to install Asterisk

[Carlos Alvarez]

On Wed, Mar 6, 2013 at 10:02 AM, Gertjan Baarda wrote:

> Couldn't agree more, Carlos. But then again, haven't we all started this
> way? ;-) The best way to understand Linux is learning the hard way. After
> all, it takes a genius to understand the simplicity of Linux.
>

If you're going to learn Linux, then learn it, not via some service running
on it.  It's clear in context that the original poster believes that he can
install and run Asterisk without knowing the OS.  This is obviously not
true.  If it's going to be someone's production server, that is scary.  It
also has led to many "ASTERISK SUCKS!" discussions I've had because there
were problems at the OS level that made the Asterisk server unreliable.

-- 
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TelEvolve
602-889-3003
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Re: [asterisk-users] AGI Script

[Gertjan Baarda]

Might be a codec issue, try allow=all in your sip.conf

Sent from my iPhone

On 6 mrt. 2013, at 17:49, Gustavo Salvador
 wrote:

>>
>> Hi every body,
>>
>> Please if some one could help me with this:
>> I'm writing an AGU Perl Script which basically makes a call using an 
>> extension provided by other asterisk box to an E1. The asterisk version is 
>> 1.6.0.28, so it hasn't the Wellington know AGI class. The code is as follows:
>>
>> =
>> #!/usr/bin/perl
>> use strict;
>>
>> my %AGI;
>> :
>> print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)";
>> =
>>
>> When dialplan executes the AGI, asterisk throw the following error: 
>> "Dropping incompatible voice fraile on SIP/INCONCERT-4796 of formar ulaw 
>> since our native format has changed to 0x8 (alaw)."
>>
>> And connection is never make.
>>
>> It's the code OK?, I'm newbie on asterisk, and really don't know what is 
>> going wrong.
>>
>> Regards,
>>
>> Gustavo
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Error to install Asterisk

[Gertjan Baarda]

Couldn't agree more, Carlos. But then again, haven't we all started this
way? ;-) The best way to understand Linux is learning the hard way. After
all, it takes a genius to understand the simplicity of Linux.

Sent from my iPhone

On 6 mrt. 2013, at 17:53, Carlos Alvarez  wrote:

I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact.  If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers.  You really need to know how to use the
OS before you can learn to manage services running on it.  I strongly
suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is
simply no way to run a production server without having to do systems
management regularly.


On Wed, Mar 6, 2013 at 3:01 AM, termo termosel wrote:

> Hi,
>
> this is the outpu to df -h command:
>
> root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
> S.ficherosTam.  Usado Disp. % Uso Montado en
> /cow   14G  4,5G  8,7G  34% /
> udev  999M  4,0K  999M   1% /dev
> tmpfs 403M  860K  402M   1% /run
> /dev/sdb1 799M  693M  106M  87% /cdrom
> /dev/loop0668M  668M 0 100% /rofs
> tmpfs1006M   44K 1006M   1% /tmp
> none  5,0M 0  5,0M   0% /run/lock
> none 1006M  100K 1006M   1% /run/shm
>
> Jordi
>
> --
> From: fermit...@hotmail.com
> To: asterisk-users@lists.digium.com
> Date: Tue, 5 Mar 2013 17:40:32 +
>
> Subject: Re: [asterisk-users] Error to install Asterisk
>
> Hi,
>
> Ok, tomorrow I will send the output when I will be in the office!
>
> Thanks!
>
> > From: asterisk_l...@earthshod.co.uk
> > To: asterisk-users@lists.digium.com
> > Date: Tue, 5 Mar 2013 16:11:01 +
> > Subject: Re: [asterisk-users] Error to install Asterisk
> >
> > On Tuesday 05 March 2013, termo termosel wrote:
> > > Hi,
> > > when I try to install Asterisk 11.2.1 the console return error which it
> > > tells: /usr/bin/ld: final link failed: No space left on device
> > > and the process exits installation.
> > > How can I solve this problem? Tmp folder is empty.
> > > Thanks,Jordi
> >
> > Try entering this command:
> > # df -h
> > and paste the complete output in a message.
> >
> > This will show the amount of space used and remaining on all
> filesystems, in
> > human-readable notation (i.e. it will automatically select the units:
> bytes,
> > kilo, mega, giga or terabytes, so as to get a sensible figure).
> >
> > You'll almost certainly have to move some files out of the way. Have you
> got,
> > or can you get, a USB external HDD; which either already has a Linux
> ext4 file
> > system on it, or contains only sacrificial data?
> >
> > --
> > AJS
> >
> > Answers come *after* questions.
> >
> > --
> > _
> > -- 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
>
> -- _
> -- 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
>
> --
> _
> -- 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
>



-- 
Carlos Alvarez
TelEvolve
602-889-3003

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Re: [asterisk-users] Redirect incoming call to SIP trunk.

[Luis H. Forchesatto]

Solved.

2013/3/5 Luis H. Forchesatto 

> Greetings.
>
> I got two asterisk servers, one is connected to another via sip trunk. The
> incoming calls are routed to the time period an if matches is transfered to
> the designed extension. If don't, is redirected to a second time period.
> Then, if the call matches the second time period it need to be transfered
> to the trunk that directs to the second server.
>
> How do I do to configure it this way?
>
> The trunk it must be transfered has a outbound route configured too.
>
> Any further details just ask.
>
> --
> Att.*
> ***
>
>


-- 
Att.*
***
Luis H. Forchesatto
Mail: luis_forchesa...@hotmail.com
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Re: [asterisk-users] Error to install Asterisk

[Carlos Alvarez]

I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact.  If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers.  You really need to know how to use the
OS before you can learn to manage services running on it.  I strongly
suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is
simply no way to run a production server without having to do systems
management regularly.


On Wed, Mar 6, 2013 at 3:01 AM, termo termosel wrote:

> Hi,
>
> this is the outpu to df -h command:
>
> root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
> S.ficherosTam.  Usado Disp. % Uso Montado en
> /cow   14G  4,5G  8,7G  34% /
> udev  999M  4,0K  999M   1% /dev
> tmpfs 403M  860K  402M   1% /run
> /dev/sdb1 799M  693M  106M  87% /cdrom
> /dev/loop0668M  668M 0 100% /rofs
> tmpfs1006M   44K 1006M   1% /tmp
> none  5,0M 0  5,0M   0% /run/lock
> none 1006M  100K 1006M   1% /run/shm
>
> Jordi
>
> --
> From: fermit...@hotmail.com
> To: asterisk-users@lists.digium.com
> Date: Tue, 5 Mar 2013 17:40:32 +
>
> Subject: Re: [asterisk-users] Error to install Asterisk
>
> Hi,
>
> Ok, tomorrow I will send the output when I will be in the office!
>
> Thanks!
>
> > From: asterisk_l...@earthshod.co.uk
> > To: asterisk-users@lists.digium.com
> > Date: Tue, 5 Mar 2013 16:11:01 +
> > Subject: Re: [asterisk-users] Error to install Asterisk
> >
> > On Tuesday 05 March 2013, termo termosel wrote:
> > > Hi,
> > > when I try to install Asterisk 11.2.1 the console return error which it
> > > tells: /usr/bin/ld: final link failed: No space left on device
> > > and the process exits installation.
> > > How can I solve this problem? Tmp folder is empty.
> > > Thanks,Jordi
> >
> > Try entering this command:
> > # df -h
> > and paste the complete output in a message.
> >
> > This will show the amount of space used and remaining on all
> filesystems, in
> > human-readable notation (i.e. it will automatically select the units:
> bytes,
> > kilo, mega, giga or terabytes, so as to get a sensible figure).
> >
> > You'll almost certainly have to move some files out of the way. Have you
> got,
> > or can you get, a USB external HDD; which either already has a Linux
> ext4 file
> > system on it, or contains only sacrificial data?
> >
> > --
> > AJS
> >
> > Answers come *after* questions.
> >
> > --
> > _
> > -- 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
>
> -- _
> -- 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
>
> --
> _
> -- 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
>



-- 
Carlos Alvarez
TelEvolve
602-889-3003
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[asterisk-users] AGI Script

[Gustavo Salvador]

> 
> Hi every body,
> 
> Please if some one could help me with this:
> I'm writing an AGU Perl Script which basically makes a call using an 
> extension provided by other asterisk box to an E1. The asterisk version is 
> 1.6.0.28, so it hasn't the Wellington know AGI class. The code is as follows:
> 
> =
> #!/usr/bin/perl
> use strict;
> 
> my %AGI;
> :
> print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)";
> =
> 
> When dialplan executes the AGI, asterisk throw the following error: "Dropping 
> incompatible voice fraile on SIP/INCONCERT-4796 of formar ulaw since our 
> native format has changed to 0x8 (alaw)."
> 
> And connection is never make.
> 
> It's the code OK?, I'm newbie on asterisk, and really don't know what is 
> going wrong.
> 
> Regards,
> 
> Gustavo

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Re: [asterisk-users] Change RX Signalling Bits in Dahdi drivers

[Optical Phoenix]

Thanks, will do

On Wed, Mar 6, 2013 at 11:24 AM, Justin Killen <
jkil...@allamericanasphalt.com> wrote:

>  You’d probably be better off sending this to the dev list (asterisk-dev)*
> ***
>
> ** **
>
> Justin Killen
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Optical Phoenix
> *Sent:* Tuesday, March 05, 2013 5:56 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Change RX Signalling Bits in Dahdi drivers
>
> ** **
>
> Greeting,
>
> I am trying to setup PLAR signalling in asterisk. I have modified the
> FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls.
>
> ** **
>
> .sig_type = DAHDI_SIG_FXSLS,
>
> .bits[DAHDI_TXSIG_ONHOOK]  =
> DAHDI_BITS_ABCD, /*changed by  for PLAR*/
>
> .bits[DAHDI_TXSIG_OFFHOOK] = (0),
> /*changed by  for PLAR*/
>
> .bits[DAHDI_TXSIG_START]   =
> DAHDI_BITS_ABCD, /*changed by for PLAR*/
>
> ** **
>
> When I got to change the rx, its a bit more complex. I have learned from
> this list that dahdi_rbsbits() handles the rx bits, but my changes seem to
> have no effect. Does anyone have a good understanding of this function? I
> would appreciate any help you can provide.
>
> ** **
>
> ** **
>
> case DAHDI_SIG_FXSLS:
>
> if (!(cursig & DAHDI_BBIT)) {  /*Dennis
> RINGING  */  /*<- I think this is checking if the state is different
> from a set value? needs clarification*/
>
> /* Check for ringing first */
>
> __dahdi_hooksig_pvt(chan,
> DAHDI_RXSIG_RING);
>
> break;
>
> }
>
> if ((chan->sig != DAHDI_SIG_FXSLS) && (cursig &
> DAHDI_ABIT)) {  /*<-- Why is it checking DAHDI_SIG_FXSLS? do I need to
> modify this to be  also?*/
>
>/* if went on hook */
>
> __dahdi_hooksig_pvt(chan,
> DAHDI_RXSIG_ONHOOK);  /*<--I think this is passing it to a function
> that reacts to the signal in this case the onhook signal?*/
>
> } else {
>
> __dahdi_hooksig_pvt(chan,
> DAHDI_RXSIG_OFFHOOK);  /* <-- same here but with the off hook? */
>
> }
>
> break;
>
>   case DAHDI_SIG_CAS:
>
> /* send event that something changed */
>
> __qevent(chan, DAHDI_EVENT_BITSCHANGED);
>
> break;
>
> ** **
>
> Thanks
>
> Dennis
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] Change RX Signalling Bits in Dahdi drivers

[Justin Killen]

You'd probably be better off sending this to the dev list (asterisk-dev)

Justin Killen

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Optical Phoenix
Sent: Tuesday, March 05, 2013 5:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Change RX Signalling Bits in Dahdi drivers

Greeting,
I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX 
bits in dahdi-base.c on line 2580, and I can make calls.

.sig_type = DAHDI_SIG_FXSLS,
.bits[DAHDI_TXSIG_ONHOOK]  = 
DAHDI_BITS_ABCD, /*changed by  for PLAR*/
.bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed 
by  for PLAR*/
.bits[DAHDI_TXSIG_START]   = 
DAHDI_BITS_ABCD, /*changed by for PLAR*/

When I got to change the rx, its a bit more complex. I have learned from this 
list that dahdi_rbsbits() handles the rx bits, but my changes seem to have no 
effect. Does anyone have a good understanding of this function? I would 
appreciate any help you can provide.


case DAHDI_SIG_FXSLS:
if (!(cursig & DAHDI_BBIT)) {  /*Dennis RINGING  */ 
 /*<- I think this is checking if the state is different from a set value? 
needs clarification*/
/* Check for ringing first */
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_RING);
break;
}
if ((chan->sig != DAHDI_SIG_FXSLS) && (cursig & 
DAHDI_ABIT)) {  /*<-- Why is it checking DAHDI_SIG_FXSLS? do I need to 
modify this to be  also?*/
   /* if went on hook */
__dahdi_hooksig_pvt(chan, 
DAHDI_RXSIG_ONHOOK);  /*<--I think this is passing it to a function 
that reacts to the signal in this case the onhook signal?*/
} else {
__dahdi_hooksig_pvt(chan, 
DAHDI_RXSIG_OFFHOOK);  /* <-- same here but with the off hook? */
}
break;
  case DAHDI_SIG_CAS:
/* send event that something changed */
__qevent(chan, DAHDI_EVENT_BITSCHANGED);
break;

Thanks
Dennis
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Re: [asterisk-users] Error to install Asterisk‏

[Thorsten Göllner]

Did you execute the "make command" in the same environment so that make 
really uses the TMPDIR directory? (no su or other shell)


Am 06.03.2013 13:37, schrieb termo termosel:

Hi,

the same error, I write your commands:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

but the same error happens

/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1»
make: *** [_cleantest_all] Error 2

Jordi


Date: Wed, 6 Mar 2013 13:29:24 +0100
From: t...@ovm-group.com
To: fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Try to set the tmp variable. In your case:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Am 06.03.2013 13:20, schrieb termo termosel:

Hi,

I read it but I don't find the solution. How Can I alocate more
free space in tmp?

Thanks,
Jordi


Date: Wed, 6 Mar 2013 13:12:34 +0100
From: t...@ovm-group.com 
To: asterisk-users@lists.digium.com

CC: fermit...@hotmail.com 
Subject: Re: [asterisk-users] Error to install Asterisk‏

Take a look here:

http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
 df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G 34% /
udev  999M  4,0K  999M 1% /dev
tmpfs 403M  860K  402M 1% /run
/dev/sdb1 799M  693M  106M 87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M 1% /tmp
none  5,0M 0  5,0M 0% /run/lock
none 1006M  100K 1006M 1% /run/shm

Jordi



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Re: [asterisk-users] Error to install Asterisk‏

[termo termosel]

Hi,

the same error, I write your commands:

mkdir /var/ext_tmp

export TMPDIR=/var/ext_tmp

make

but the same error happens

/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1»
make: *** [_cleantest_all] Error 2

Jordi

Date: Wed, 6 Mar 2013 13:29:24 +0100
From: t...@ovm-group.com
To: fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]  Error to install Asterisk‏


  

  
  
Try to set the tmp variable. In your case:



mkdir /var/ext_tmp

export TMPDIR=/var/ext_tmp

make



Am 06.03.2013 13:20, schrieb termo termosel:


  
  Hi,



I read it but I don't find the solution. How Can I alocate more
free space in tmp?



Thanks,

Jordi




  Date: Wed, 6 Mar 2013 13:12:34 +0100

  From: t...@ovm-group.com

  To: asterisk-users@lists.digium.com

  CC: fermit...@hotmail.com

  Subject: Re: [asterisk-users] Error to install Asterisk‏

  

  Take a look here:

  
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

  

  Am 06.03.2013 13:00, schrieb
termo termosel:

  
  

  
  Hi,



df -h output:




root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
df -h

S.ficherosTam.  Usado Disp. % Uso
Montado en

/cow   14G  4,5G  8,7G  34% /

udev  999M  4,0K  999M   1% /dev

tmpfs 403M  860K  402M   1% /run

/dev/sdb1 799M  693M  106M  87%
/cdrom

/dev/loop0668M  668M 0 100%
/rofs

tmpfs1006M   44K 1006M   1% /tmp

none  5,0M 0  5,0M   0%
/run/lock

none 1006M  100K 1006M   1%
/run/shm



Jordi

  
  

  

  
  

  
  


  

  
  


  
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Re: [asterisk-users] Error to install Asterisk‏

[Thorsten Göllner]

Try to set the tmp variable. In your case:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Am 06.03.2013 13:20, schrieb termo termosel:

Hi,

I read it but I don't find the solution. How Can I alocate more free 
space in tmp?


Thanks,
Jordi


Date: Wed, 6 Mar 2013 13:12:34 +0100
From: t...@ovm-group.com
To: asterisk-users@lists.digium.com
CC: fermit...@hotmail.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
 df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi



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Re: [asterisk-users] Error to install Asterisk‏

[Thorsten Göllner]

Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi




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Re: [asterisk-users] Error to install Asterisk

[termo termosel]

Hi,

this is the outpu to df -h command:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi

From: fermit...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Mar 2013 17:40:32 +
Subject: Re: [asterisk-users] Error to install Asterisk




Hi,
 
Ok, tomorrow I will send the output when I will be in the office!
 
Thanks!
 
> From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Tue, 5 Mar 2013 16:11:01 +
> Subject: Re: [asterisk-users] Error to install Asterisk
> 
> On Tuesday 05 March 2013, termo termosel wrote:
> > Hi,
> > when I try to install Asterisk 11.2.1 the console return error which it
> > tells: /usr/bin/ld: final link failed: No space left on device
> > and the process exits installation.
> > How can I solve this problem? Tmp folder is empty.
> > Thanks,Jordi
> 
> Try entering this command:
> # df -h
> and paste the complete output in a message.
> 
> This will show the amount of space used and remaining on all filesystems, in 
> human-readable notation  (i.e. it will automatically select the units: bytes, 
> kilo, mega, giga or terabytes, so as to get a sensible figure).
> 
> You'll almost certainly have to move some files out of the way.  Have you 
> got, 
> or can you get, a USB external HDD; which either already has a Linux ext4 
> file 
> system on it, or contains only sacrificial data?  
> 
> -- 
> AJS
> 
> Answers come *after* questions.
> 
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