Re: [asterisk-users] Configure Voicemail for Large Systems

2010-06-14 Thread Jonathan González
Would it be possible to see an example on extensions.conf and voicemail.conf
to see how to do that?

Thanks in advance,
Jonathan

On Mon, Jun 14, 2010 at 10:18 PM, Zeeshan Zakaria wrote:

> You can use realtime architecture. I have a similar setup, voicemails works
> just fine.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-06-14 6:08 PM, "Jonathan González"  wrote:
>
> Hi there,
>
> I have been taking a look on how to configure voicemail systems with
> asterisk and I would like to know if there's any way to define mailbox in
> a dynamic way.
>
> I have 100 users and I would like to know if there's any way to avoid the
> definition of the 100 mailboxes in voicemail.conf and use for example the
> variables that I am getting from extensions.conf.
>
> Thanks in advance,
> Jonathan
> --
> Personal webpage - www.jonbaraq.eu
>
> --
>
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[asterisk-users] Skype for SIP

2010-06-14 Thread Randy R
By the way, I am currently testing this product from Skype. I would
like to be able to receive calls ona Skype name on our pbx.

1) It works beautifully and you don't have to do anything in particular.

2) It's disproportionally expensive which is why I want Skype for
Asterisk to work.

SfS costs $5 per month per channel just to test the beta! I find that
insane, but I wanted to test it.
In October, they will begin charging for Skype Manager (required for
SfS) and a per seat charge for that.

My guess is that the one channel would cost me about $15/month. For
our use, I'd much rather use SfA.

You can test SfS is you like by just registering a SIP phone to their
servers. That's what I am doing now. I haven't tested outgoing calls
to SKype names, but they are possible. Probably, like OpenSkY they are
made using numbers in a table that equate to Skype names.

/r

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[asterisk-users] Skype for Asterisk - what processors/platforms does it run on?

2010-06-14 Thread Randy R
I understand that SfA is a binary module? There are processors it will
not work on, correct? Are there limits as to operating system or
distros?

tia,

/r

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Re: [asterisk-users] a2billing for residential voip usage

2010-06-14 Thread ram
you see lot of documentation on wiki

Google them many success case you see

Ram

On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy wrote:

> Hello List.
>
> I just installed a2billing with asterisk 1.6 and got it working. The only
> problem is that I'm trying to setup something to manage who's using the most
> minutes in the house. I noticed a2billing only works for callin cards
> setups, or maybe I didn't configure it correctly for what I want. Can I use
> a2billing for "•VoIP residential services"? if yes, how? if no, please guide
> me to another application I can use along side asterisk.
>
> Thanks in advanced for your time.
>
>
>
>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Steve Edwards

On Mon, 14 Jun 2010, bruce bruce wrote:

However, for this project, it seems that I can use php system() along 
with grep to see the status of a peer with one line of code:


asterisk -rx "sip show peer $sip_peer" | grep -c "X-Lite"'

Above ^^^ in Linux prompt returns 1 if $sip_peer is registered with 
X-Lite else it returns 0.


But using system() I think the "" confuse php and value is always 
returned as 0. $peer_count = system('asterisk -rx "sip show peer 
$sip_peer" | grep -c "X-Lite"', $retval);


Should $sip_peer be inside another set of parenthesis?


The single quotes are skipping the evaluation of $sip_peer.

Try something like this:

system('asterisk -rx "sip show peer ' . $sip_peer . '" | grep -c "X-Lite"')

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[asterisk-users] a2billing for residential voip usage

2010-06-14 Thread Landy Landy
Hello List.

I just installed a2billing with asterisk 1.6 and got it working. The only 
problem is that I'm trying to setup something to manage who's using the most 
minutes in the house. I noticed a2billing only works for callin cards setups, 
or maybe I didn't configure it correctly for what I want. Can I use a2billing 
for "•VoIP residential services"? if yes, how? if no, please guide me to 
another application I can use along side asterisk.

Thanks in advanced for your time.


  

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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
I do have a rather bigger project coming my way and I would really like to
know how to do the *"feeding a text file into the STDIN of your AGI so you
can debug completely external to Asterisk"*
*
*
*However, for this project, it seems that I can use php system() along with
grep to see the status of a peer with one line of code:*
*
*
*
 asterisk -rx "sip show peer $sip_peer" | grep -c "X-Lite"'
Above ^^^ in Linux prompt returns 1 if $sip_peer is registered with X-Lite
else it returns 0.

But using system() I think the "" confuse php and value is always returned
as 0.
$peer_count = system('asterisk -rx "sip show peer $sip_peer" | grep -c
"X-Lite"', $retval);

Should $sip_peer be inside another set of parenthesis?

Thanks,
Bruce

*
On Mon, Jun 14, 2010 at 6:44 PM, Steve Edwards wrote:

> On Mon, 14 Jun 2010, bruce bruce wrote:
>
>  Thanks for the input. I actually did use verbose() and that is when I
>> noticed that my path to phpagi was not right since nothing was coming
>> through. For return value prior to fixing phpagi path, I was getting:
>>
>> NoOp("SIP/64.111.222.111-0ca7", "")
>>
>> which actually is just right because if you notice the last three
>> charecters of that line is ""). So, when the phpagi path is correct, it
>> looks like:  "415444555").
>>
>
> Which is why enabling AGI debugging and posting actual console code can
> speed problem resolution. Sometimes it's just another set of eyes on the
> same output :)
>
> If you're going to be developing a significant bit of AGI code, there is a
> "not immediately obvious" technique of feeding a text file into the STDIN of
> your AGI so you can debug completely external to Asterisk -- even within an
> IDE like emacs+gdb. (Or whatever the PHP equivalence is.)
>
> --
> 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] Qwest PRIs

2010-06-14 Thread Steve Edwards
On Mon, 14 Jun 2010, C F wrote:

> One more thing, read the comments here:
> http://www.voip-info.org/wiki/index.php?page_id=573&tk=2ff846f8169b7694aed5&comments_page=1
> Don't forget to have a beer ready :P

Now that's really funny.

I read along with this and was thinking this was exactly my experience 
with some Qwest PRIs a couple of years ago.

Then I noticed -- it was me :)

I guess I had too many beers.

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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Qwest PRIs

2010-06-14 Thread C F
One more thing, read the comments here:
http://www.voip-info.org/wiki/index.php?page_id=573&tk=2ff846f8169b7694aed5&comments_page=1
Don't forget to have a beer ready :P


On Mon, Jun 14, 2010 at 7:18 PM, C F  wrote:
> Your configs looks good, the only thing left is to figure out:
> 1. You sure you have the right cable plugged in to the right port? The
> reason I'm asking this is because you started out with the dchannel
> being 48.
> 2. PRI Debug, what are the messages?
> 3. Contact your provider and try troubleshooting it with them.
>
> Please when you have it working report back what the issue was.
>
> On Mon, Jun 14, 2010 at 2:58 AM, Voip Asterisk  wrote:
>> Ya I'm passed that part now.  I have dahdi properly loading the card, and
>> both links are green.  Asterisk recognizes the channels, but still shows the
>> span as down.
>>
>> On Sun, Jun 13, 2010 at 6:35 PM, C F  wrote:
>>>
>>> Not sure what version you are running but I'm still running 1.2x in
>>> 1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
>>> assuming layer 2+) part loads with Asterisk.
>>>
>>> On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk 
>>> wrote:
>>> > Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi
>>> > to
>>> > bring up the PRIs without alarms.
>>> >
>>> > On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle  wrote:
>>> >>
>>> >> Voip Asterisk wrote:
>>> >> > Hi,
>>> >> >
>>> >> > I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
>>> >> >  I'm using an OpenVox D410E and the drivers are loaded.  My
>>> >> > system.conf looks like this:
>>> >> >
>>> >>
>>> >> Google turned up this:
>>> >>
>>> >> http://www.voip-info.org/wiki/view/NFAS
>>> >>
>>> >> Doug
>>> >>
>>> >> --
>>> >> Ben Franklin quote:
>>> >>
>>> >> "Those who would give up Essential Liberty to purchase a little
>>> >> Temporary
>>> >> Safety, deserve neither Liberty nor Safety."
>>> >>
>>> >>
>>> >> --
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>>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>> >>               http://www.asterisk.org/hello
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>>> >> To UNSUBSCRIBE or update options visit:
>>> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >
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>>
>>
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Re: [asterisk-users] Unable to pickup an extension, trying everything

2010-06-14 Thread Philipp von Klitzing
Quickly:

Do some reading on PICKUPMARK: You need to set this on the channel that 
you want to pick up, not the channel that is doing the pickup.

Philipp


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Re: [asterisk-users] Qwest PRIs

2010-06-14 Thread C F
Your configs looks good, the only thing left is to figure out:
1. You sure you have the right cable plugged in to the right port? The
reason I'm asking this is because you started out with the dchannel
being 48.
2. PRI Debug, what are the messages?
3. Contact your provider and try troubleshooting it with them.

Please when you have it working report back what the issue was.

On Mon, Jun 14, 2010 at 2:58 AM, Voip Asterisk  wrote:
> Ya I'm passed that part now.  I have dahdi properly loading the card, and
> both links are green.  Asterisk recognizes the channels, but still shows the
> span as down.
>
> On Sun, Jun 13, 2010 at 6:35 PM, C F  wrote:
>>
>> Not sure what version you are running but I'm still running 1.2x in
>> 1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
>> assuming layer 2+) part loads with Asterisk.
>>
>> On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk 
>> wrote:
>> > Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi
>> > to
>> > bring up the PRIs without alarms.
>> >
>> > On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle  wrote:
>> >>
>> >> Voip Asterisk wrote:
>> >> > Hi,
>> >> >
>> >> > I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
>> >> >  I'm using an OpenVox D410E and the drivers are loaded.  My
>> >> > system.conf looks like this:
>> >> >
>> >>
>> >> Google turned up this:
>> >>
>> >> http://www.voip-info.org/wiki/view/NFAS
>> >>
>> >> Doug
>> >>
>> >> --
>> >> Ben Franklin quote:
>> >>
>> >> "Those who would give up Essential Liberty to purchase a little
>> >> Temporary
>> >> Safety, deserve neither Liberty nor Safety."
>> >>
>> >>
>> >> --
>> >> _
>> >> -- 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
>> >
>> >
>> > --
>> > _
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>> >               http://www.asterisk.org/hello
>> >
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>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
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>
>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Steve Edwards

On Mon, 14 Jun 2010, bruce bruce wrote:

Thanks for the input. I actually did use verbose() and that is when I 
noticed that my path to phpagi was not right since nothing was coming 
through. For return value prior to fixing phpagi path, I was getting:


NoOp("SIP/64.111.222.111-0ca7", "")

which actually is just right because if you notice the last three 
charecters of that line is ""). So, when the phpagi path is correct, it 
looks like:  "415444555").


Which is why enabling AGI debugging and posting actual console code can 
speed problem resolution. Sometimes it's just another set of eyes on the 
same output :)


If you're going to be developing a significant bit of AGI code, there is a 
"not immediately obvious" technique of feeding a text file into the STDIN 
of your AGI so you can debug completely external to Asterisk -- even 
within an IDE like emacs+gdb. (Or whatever the PHP equivalence is.)


--
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] Configure Voicemail for Large Systems

2010-06-14 Thread Zeeshan Zakaria
You can use realtime architecture. I have a similar setup, voicemails works
just fine.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-14 6:08 PM, "Jonathan González"  wrote:

Hi there,

I have been taking a look on how to configure voicemail systems with
asterisk and I would like to know if there's any way to define mailbox in
a dynamic way.

I have 100 users and I would like to know if there's any way to avoid the
definition of the 100 mailboxes in voicemail.conf and use for example the
variables that I am getting from extensions.conf.

Thanks in advance,
Jonathan
-- 
Personal webpage - www.jonbaraq.eu

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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Thanks for the input. I actually did use verbose() and that is when I
noticed that my path to phpagi was not right since nothing was coming
through.

For return value prior to fixing phpagi path, I was getting:

*NoOp("SIP/64.111.222.111-0ca7", "")*

which actually is just right because if you notice the last three charecters
of that line is* "")*. So, when the phpagi path is correct, it looks like:
*"415444555")*.

-Bruce

On Mon, Jun 14, 2010 at 6:09 PM, Steve Edwards wrote:

> > On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
>
> >> Carlos, Thanks a lot for getting me started. That helps a great deal.
>
> >> exten => _x.,1,NoOp(${EXTEN})
>
> Since you're just getting started, there is an application specifically
> written to send output to the CLI -- verbose(). It's more "obvious" and it
> has additional functionality.
>
> On Mon, 14 Jun 2010, Carlos Chavez wrote:
>
> > As you can see agi_extension and agi_dnid should contain the number that
> > the user dialed.  Those are all the variables that are automatically
> > sent to the AGI script from Asterisk.
>
> agi_extension contains the extension (the bit after the equals sign) of
> the statement that executes the AGI application. For example,
>
>exten = s,n,agi(foo)
>
> When "foo" examines agi_extension, it will contain "s"
>
> > I do not know why you are getting the channel instead of the extension,
> > you could try giving the extension as a parameter to the AGI script if
> > you cannot get that from the included request variable.
>
> Better to fix the really simple stuff before he gets to the complex stuff.
>
> --
> 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] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Carlos Chavez
Go to http://phpagi.sourceforge.net/ and check the documentation.
Basically you can run any command you can run on the CLI or
extensions.conf.  To get the status of a SIP peer I would recommend
using AMI.  PHPAGI also has a library to connect to the AMI and send
commands.

On Mon, 2010-06-14 at 17:38 -0400, bruce bruce wrote:
> Thanks for pointing to the debug. I found that the path to phpagi was
> not right and since I fixed that everything seems to work fine.
> 
> 
> Now, I want to know if I can use a phpagi command to check the status
> of SIP Peer if it is online and registered or not. I know I can use
> grep with asterisk rx "sip show peers" and use that as a shell script
> but I think there are better methods in Asterisk dialplan or in phpagi
> that it can be check.
> 
> 
> Thanks,
> Bruce
> 
> On Mon, Jun 14, 2010 at 5:27 PM, Carlos Chavez
>  wrote:
>If you do an "agi set debug on" from the CLI and run
> your AGI you
> should see something like this:
> 
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_request: auth.agi
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_channel: SIP/206-0d9b
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_language: es
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_type: SIP
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_uniqueid: 1276192380.5115
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_version: 1.6.2.8
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_callerid: 206
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_calleridname: Carmen Saavedra
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_callingpres: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_callingani2: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_callington: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_callingtns: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_dnid: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_rdnis: unknown
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_context: oficina
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_extension: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_priority: 2
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_enhanced: 0.0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_accountcode:
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_threadid: -1234945136
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c:
> AGI Tx >>
> agi_arg_1: ldnac
> 
>As you can see agi_extension and agi_dnid should
> contain the number
> that the user dialed.  Those are all the variables that are
> automatically sent to the AGI script from Asterisk.  I do not
> know why
> you are getting the channel instead of the extension, you
> could try
> giving the extension as a parameter to the AGI script if you
> cannot get
> that from the included request variable.
> 
> 
> On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
> > Carlos, Thanks a lot for getting me started. That helps a
> great deal.
> >
> >
> > Currently, the $agi->request['agi_extension'];  returns the
> SIP
> > channel info with IP and I want that to be the incoming DID
> number.
> >
> >
> > My dialplan output is this for line one:
> >
> >
> > exten => _x.,1,NoOp(${EXTEN})
> >
> >
> > 415444555
> >
> >
> > But with the agi_extension it comes back as:
> > NoOp("SIP/64.111.222.111-0ca7", "")
> >
> >
> > Where can I find the list of command requests that can be
> sent to
> > Asterisk? Specially that for DID.
> >
> >
> > Thanks
> >
> >
> > On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez
> >  wrote:
> > On Mon, 2010-06-14 at 13:41 -0400, bruce bruce
> wrote:
> > > Hi Carlso,
> > >
> > >
> > > Thanks for the input. I have done this in php and
> am not
> > fami

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Steve Edwards
> On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:

>> Carlos, Thanks a lot for getting me started. That helps a great deal.

>> exten => _x.,1,NoOp(${EXTEN})

Since you're just getting started, there is an application specifically 
written to send output to the CLI -- verbose(). It's more "obvious" and it 
has additional functionality.

On Mon, 14 Jun 2010, Carlos Chavez wrote:

> As you can see agi_extension and agi_dnid should contain the number that 
> the user dialed.  Those are all the variables that are automatically 
> sent to the AGI script from Asterisk.

agi_extension contains the extension (the bit after the equals sign) of 
the statement that executes the AGI application. For example,

exten = s,n,agi(foo)

When "foo" examines agi_extension, it will contain "s"

> I do not know why you are getting the channel instead of the extension, 
> you could try giving the extension as a parameter to the AGI script if 
> you cannot get that from the included request variable.

Better to fix the really simple stuff before he gets to the complex stuff.

-- 
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] Configure Voicemail for Large Systems

2010-06-14 Thread Jonathan González
Hi there,

I have been taking a look on how to configure voicemail systems with
asterisk and I would like to know if there's any way to define mailbox in
a dynamic way.

I have 100 users and I would like to know if there's any way to avoid the
definition of the 100 mailboxes in voicemail.conf and use for example the
variables that I am getting from extensions.conf.

Thanks in advance,
Jonathan
-- 
Personal webpage - www.jonbaraq.eu
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[asterisk-users] Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards)

2010-06-14 Thread bruce bruce
Hi Guys,

Looking for a powerful box that is compact, can take two hard drives for
Raid-1 (no SSD, too expensive), have at least two Gig ports or two
10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card
which needs as much room as a PCIe and doesn't need the actual slot. That is
for Sangoma A200 + Daughter card.

I would be really glad if I can get some sort of a KVM over IP for this box
built in as this is going to be deployed overseas.

One of these servers is to support 30 extensions with transacoding g.729 and
the other is to support 10 extensions with transcoding. I don't think there
is anything in between in terms of hardware but any and all suggestions are
welcome.

Remember: Compact, quite, good airflow, powerful, 2 x empty slots one of
which is a PCIe and 2xNIC ports.

Thanks
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Thanks for pointing to the debug. I found that the path to phpagi was not
right and since I fixed that everything seems to work fine.

Now, I want to know if I can use a phpagi command to check the status of SIP
Peer if it is online and registered or not. I know I can use *grep with
asterisk rx "sip show peers" *and use that as a shell script but I think
there are better methods in Asterisk dialplan or in phpagi that it can be
check.

Thanks,
Bruce

On Mon, Jun 14, 2010 at 5:27 PM, Carlos Chavez wrote:

>If you do an "agi set debug on" from the CLI and run your AGI you
> should see something like this:
>
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_request: auth.agi
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_channel: SIP/206-0d9b
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_language: es
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_type: SIP
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_uniqueid: 1276192380.5115
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_version: 1.6.2.8
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callerid: 206
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_calleridname: Carmen Saavedra
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingpres: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingani2: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callington: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingtns: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_dnid: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_rdnis: unknown
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_context: oficina
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_extension: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_priority: 2
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_enhanced: 0.0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_accountcode:
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_threadid: -1234945136
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_arg_1: ldnac
>
>As you can see agi_extension and agi_dnid should contain the number
> that the user dialed.  Those are all the variables that are
> automatically sent to the AGI script from Asterisk.  I do not know why
> you are getting the channel instead of the extension, you could try
> giving the extension as a parameter to the AGI script if you cannot get
> that from the included request variable.
>
> On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
> > Carlos, Thanks a lot for getting me started. That helps a great deal.
> >
> >
> > Currently, the $agi->request['agi_extension'];  returns the SIP
> > channel info with IP and I want that to be the incoming DID number.
> >
> >
> > My dialplan output is this for line one:
> >
> >
> > exten => _x.,1,NoOp(${EXTEN})
> >
> >
> > 415444555
> >
> >
> > But with the agi_extension it comes back as:
> > NoOp("SIP/64.111.222.111-0ca7", "")
> >
> >
> > Where can I find the list of command requests that can be sent to
> > Asterisk? Specially that for DID.
> >
> >
> > Thanks
> >
> >
> > On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez
> >  wrote:
> > On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > > Hi Carlso,
> > >
> > >
> > > Thanks for the input. I have done this in php and am not
> > familiar with
> > > phpagi.
> > > So, there is absolutely no way to temporarily solve this
> > problem by
> > > getting the value back from php file?
> > >
> > >
> > > Wondering if it would require a lot of work to change the
> > php file to
> > > phpagi?
> > > Thanks,
> > > Bruce
> > >
> >
> >Here is an example:
> >
> > exten => _x.,1,AGI(sample.agi)
> > exten => _x.,n,NoOp(${var})
> >
> >
> > sample.agi:
> >
> > #!/usr/bin/php -q
> >  > set_time_limit(30);
> > require('phpagi/phpagi.php');
> > $agi = new AGI();
> > $exten = $agi->request['agi_extension']; //Dialed extension
> > // the result is stored in $exten
> > // do something with your data
> > $agi->set_variable("var", $result);
> > $agi->verbose("The result was: $result", 3);
> > ?>
> >
> >You can even send parameters to the AGI via the command
> > line like:
> >
> > exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for
> > 1.6 or |
> > for 1.4 or below
> >
> >And access them via $argv[1], $argv[2] is there is some
> > extra
> > information that you cannot get via the AGI variables.
> >
> > --
> >
> > Telecomunicaci

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Carlos Chavez
If you do an "agi set debug on" from the CLI and run your AGI you
should see something like this:

[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_request: auth.agi
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_channel: SIP/206-0d9b
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_language: es
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_type: SIP
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_uniqueid: 1276192380.5115
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_version: 1.6.2.8
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_callerid: 206
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_calleridname: Carmen Saavedra
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_callingpres: 0
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_callingani2: 0
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_callington: 0
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_callingtns: 0
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_dnid: *2208101282711
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_rdnis: unknown
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_context: oficina
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_extension: *2208101282711
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_priority: 2
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_enhanced: 0.0
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_accountcode: 
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_threadid: -1234945136
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_arg_1: ldnac

As you can see agi_extension and agi_dnid should contain the number
that the user dialed.  Those are all the variables that are
automatically sent to the AGI script from Asterisk.  I do not know why
you are getting the channel instead of the extension, you could try
giving the extension as a parameter to the AGI script if you cannot get
that from the included request variable.

On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
> Carlos, Thanks a lot for getting me started. That helps a great deal.
> 
> 
> Currently, the $agi->request['agi_extension'];  returns the SIP
> channel info with IP and I want that to be the incoming DID number. 
> 
> 
> My dialplan output is this for line one:
> 
> 
> exten => _x.,1,NoOp(${EXTEN})
> 
> 
> 415444555
> 
> 
> But with the agi_extension it comes back as:
> NoOp("SIP/64.111.222.111-0ca7", "")
> 
> 
> Where can I find the list of command requests that can be sent to
> Asterisk? Specially that for DID.
> 
> 
> Thanks
> 
> 
> On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez
>  wrote:
> On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > Hi Carlso,
> >
> >
> > Thanks for the input. I have done this in php and am not
> familiar with
> > phpagi.
> > So, there is absolutely no way to temporarily solve this
> problem by
> > getting the value back from php file?
> >
> >
> > Wondering if it would require a lot of work to change the
> php file to
> > phpagi?
> > Thanks,
> > Bruce
> >
> 
>Here is an example:
> 
> exten => _x.,1,AGI(sample.agi)
> exten => _x.,n,NoOp(${var})
> 
> 
> sample.agi:
> 
> #!/usr/bin/php -q
>  set_time_limit(30);
> require('phpagi/phpagi.php');
> $agi = new AGI();
> $exten = $agi->request['agi_extension']; //Dialed extension
> // the result is stored in $exten
> // do something with your data
> $agi->set_variable("var", $result);
> $agi->verbose("The result was: $result", 3);
> ?>
> 
>You can even send parameters to the AGI via the command
> line like:
> 
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for
> 1.6 or |
> for 1.4 or below
> 
>And access them via $argv[1], $argv[2] is there is some
> extra
> information that you cannot get via the AGI variables.
> 
> --
> 
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
> 
> 
> --
> _
> -- 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
> 
> 
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Hi again,

So, I have this but NoOp shows a random SIP info value rather then the one
passed to it. Just to test, I am sending $didin as argument to test.php and
then expect it back as $didgot back into dialplan. But it seems that either
send or receive has problem because no matter what I put as the NoOp in _x,4
it all comes back the same as SIP channel info.

*[custom-phpagi]*
*exten => _x.,1,Set(didin=${EXTEN})*
*exten => _x.,2,NoOp(${didin}*
*exten => _x.,3,deadAGI(test.agi,${didin})*
*exten => _x.,4,NoOp(${didgot})*
*exten => _x.,5,Hangup()*

*/var/lib/asterisk/agi-lib/test.agi:*

#!/usr/bin/php -q
get_variable("didgot", $didgot);  /// both of these don't seem to
send didgot back to dialplan
$agi->set_variable("didgot", $didgot);  /// both of these don't seem to
send didgot back to dialplan


exit(0);
?>

Thanks,
Bruce


On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > Hi Carlso,
> >
> >
> > Thanks for the input. I have done this in php and am not familiar with
> > phpagi.
> > So, there is absolutely no way to temporarily solve this problem by
> > getting the value back from php file?
> >
> >
> > Wondering if it would require a lot of work to change the php file to
> > phpagi?
> > Thanks,
> > Bruce
> >
> Here is an example:
>
> exten => _x.,1,AGI(sample.agi)
> exten => _x.,n,NoOp(${var})
>
> sample.agi:
>
> #!/usr/bin/php -q
>  set_time_limit(30);
> require('phpagi/phpagi.php');
> $agi = new AGI();
> $exten = $agi->request['agi_extension']; //Dialed extension
> // the result is stored in $exten
> // do something with your data
> $agi->set_variable("var", $result);
> $agi->verbose("The result was: $result", 3);
> ?>
>
>You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
> for 1.4 or below
>
>And access them via $argv[1], $argv[2] is there is some extra
> information that you cannot get via the AGI variables.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Danny Nicholas
Also cheaper to replace flash card than hard drive.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Monday, June 14, 2010 4:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Small PC to build and run Asterisk

Why no flash?

> * Small pre-built PC (not buying board, case, all parts separately)
> * Low power consumption
> * No fan or very small fan
> * Hard drive (not flash memory)

An ssd uses less power, so generates less warmth, hence less need for
fan in the drive area. Also less noise..

I like this one, or its smaller brother:
http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/

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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Hans Witvliet
Why no flash?

> * Small pre-built PC (not buying board, case, all parts separately)
> * Low power consumption
> * No fan or very small fan
> * Hard drive (not flash memory)

An ssd uses less power, so generates less warmth, hence less need for
fan in the drive area. Also less noise..

I like this one, or its smaller brother:
http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/

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Re: [asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun
Thank You Tarek!

That was the case, and i saw now i had a typo in the extension further down, 
but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t 
play, the settings are default.
Maybe i'll figure it out.

Thank you


Regards 

Aksel


Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tarek Sawah
Sendt: 14. juni 2010 15:00
Til: Asterisk Users
Emne: Re: [asterisk-users] Call queues - issues, can't make it work.

when you add an agent to a queue the agent should log in
try adding
member=SIP/301
member=SIP/302
instead of agent directives.
this will ring both phones.. from your output it doesn't seem to be ringing the 
agents at all.

-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 
562 2308



From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue “teknisk”
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings…and rings…until it hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no



The New Busy think 9 to 5 is a cute idea. Combine multiple calendars wit

[asterisk-users] Re : Re : Asterisk Call routing problem

2010-06-14 Thread Adil Zaaraoui


The hangup is after Dial.

anay suggestion


De : Vardan Harutyunyan 
À : asterisk-users@lists.digium.com
Envoyé le : Lun 14 juin 2010, 20h 31min 22s
Objet : Re: [asterisk-users] Re :  Asterisk Call routing problem

Hangup is comming after Dial or AGI?

-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

Adil Zaaraoui wrote:
> Dear Vardan,
> I had before the same problem, i reinstalled asterisk and it worked; now
> i get the same problem;
> i am using asterisk 1.4.22, it forwards if i call operator A (the one i
> was testing); but all other numbers are not forwarded, it just hangup,
> here is the output:
>
> Accepting AUTHENTICATED call from 192.168.1.4:
>
>  > requested format = unknown,
>
>  > requested prefs = (ulaw|slin|alaw),
>
>  > actual format = ulaw,
>
>  > host prefs = (gsm|ulaw|alaw),
>
>  > priority = mine
>
> -- Executing [00212675410...@outbound:1] Set("IAX2/#000105-8303",
> "calleeNumber=011212675410113") in new stack
>
> -- Executing [00212675410...@outbound:2] AGI("IAX2/#000105-8303")
>
> HANGUP 'IAX2/#000105-8303'
>
> thanks for the help
>
>
>
> 
> *De :* Vardan Harutyunyan 
> *À :* asterisk-users@lists.digium.com
> *Envoyé le :* Lun 14 juin 2010, 7h 36min 11s
> *Objet :* Re: [asterisk-users] Asterisk Call routing problem
>
> Whats about log?
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: i...@eif.am 
> www.eif-it.com
>
> Adil Zaaraoui wrote:
>  > Deal list,
>  > I have a problem with asterisk call routing.
>  > I configured my asterisk in a way that it forwards calls to a viop
>  > provider using IAX2 protocol.
>  > some cell phones numbers are routed and others are not!!! but when i try
>  > directly zoiper with my provider it works perfectly.
>  > Before, my asterisk was working perfectely. but i was testing just
>  > calling one number, i think that asterisk keeps in its cache the
>  > operator i was calling.
>  > It seems like asterisk record in its cache the dialed numbers. i tried
>  > to reload asterisk, i tried also to reboot the system, and the problem
>  > still persist.
>  > here is my config:
>  > [outbound]
>  > exten=>_00X.,1,set(calleNumber=_011${EXTEN:2})
>  > exten=>_00X.,n,AGI(http://localhost/myScript.agi)
>  >
>  > exten=>_00X.,n,Dial(IAX2/mylo...@outbound/${calleNumber
>  > })
>  >
>  > exten=>h,n,deadAGI(http://localhost/myScript2.agi)
>  >
>  > i tried disabling my scripts and still getting the same problem.
>  >
>  > Do you have any idea
>  >
>  >
>
>
> --
> _
> -- 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
>


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Re: [asterisk-users] WARNING message when play

2010-06-14 Thread equis software
If I use CONTROL STREAM FILE the messages disapearIt´s this ok?


On Mon, Jun 14, 2010 at 12:22 PM, Warren Selby wrote:

>
>
> On Mon, Jun 14, 2010 at 8:27 AM, equis software 
> wrote:
>
>> In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and aboveappear 
>> this messages
>>
>>
> This message was added around 1.4.23 to let you know that you're violating
> the AGI protocol.  Read up on the AGI protocol then check through your AGI
> file to see what you're missing.  Usually it's not reading something that
> asterisk is sending back to your script, or not sending a response that
> asterisk is expecting.  Everything else typically will still do what you
> want, but technically you're violating the protocol.
>
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Steve Edwards
On Mon, 14 Jun 2010, Carlos Chavez wrote:

>   You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or | 
> for 1.4 or below

Comma works fine in 1.2.

-- 
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 library for C/C++

2010-06-14 Thread David Backeberg
On Sun, Jun 13, 2010 at 2:59 PM, Vieri  wrote:
> I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6 
> compatible).
> I've taken a look at CAGI and QUIVR but their latest code releases date back 
> to 2006.
> I've also seen a more recent project (wildpbx) dated 2009:
> http://github.com/comradeb14ck/wildpbx/tree/master/libraries/agi/c/
>
> Any suggestions/recommendations for a C AGI library?
>
> Thanks,
>
> Vieri

I don't have a good answer to your real question. But my experience
with the AGI libraries with Perl is that the real core of AGI is
consistent. You can build on your own libraries to pass arguments
correctly, etc. But the mean idea of AGI should work pretty well from
1.2 to 1.4 to 1.6.

Why don't you try one of those 'old' ones, and see what problems you
run into. In my experience with my Perl AGI, the problems were at the
highest-level 'layer' that was just simple changes like feed it
arguments with commas rather than pipes. Easy changes.

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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Carlos, Thanks a lot for getting me started. That helps a great deal.

Currently, the *$agi->request['agi_extension'];  returns the SIP channel
info with IP and I want that to be the incoming DID number. *
*
*
*My dialplan output is this for line one:*

**exten => _x.,1,NoOp(${EXTEN})

*415444555*

But with the agi_extension it comes back as:
NoOp("SIP/64.111.222.111-0ca7", "")

Where can I find the list of command requests that can be sent to Asterisk?
Specially that for DID.

Thanks


On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > Hi Carlso,
> >
> >
> > Thanks for the input. I have done this in php and am not familiar with
> > phpagi.
> > So, there is absolutely no way to temporarily solve this problem by
> > getting the value back from php file?
> >
> >
> > Wondering if it would require a lot of work to change the php file to
> > phpagi?
> > Thanks,
> > Bruce
> >
> Here is an example:
>
> exten => _x.,1,AGI(sample.agi)
> exten => _x.,n,NoOp(${var})
>
> sample.agi:
>
> #!/usr/bin/php -q
>  set_time_limit(30);
> require('phpagi/phpagi.php');
> $agi = new AGI();
> $exten = $agi->request['agi_extension']; //Dialed extension
> // the result is stored in $exten
> // do something with your data
> $agi->set_variable("var", $result);
> $agi->verbose("The result was: $result", 3);
> ?>
>
>You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
> for 1.4 or below
>
>And access them via $argv[1], $argv[2] is there is some extra
> information that you cannot get via the AGI variables.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>
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Re: [asterisk-users] Unable to pickup an extension, trying everything

2010-06-14 Thread Jonas Kellens

Using asterisk 1.4.30

sip.conf is realtime sip_buddies in mysql database. What settings here 
affect the Pickup() ?? If you think about pickup/call-groups, I have 
none defined.


extensions.conf :

exten => _**XX,1,NoOp()
exten => _**XX,n,Macro(GetKlantIDfromCALLnum,${CALLERID(num)})
exten => _**XX,n,Macro(QueryAstDB,${klantID})
exten => _**XX,n,Pickup(${EXTEN:2...@${knummer})
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(${EXTEN:2})
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(${EXTEN:2...@from-testcorp)
exten => _**XX,n,Pickup(testco...@${knummer})
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(testcorp1)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(testco...@from-testcorp)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(testco...@testcorp)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(${EXTEN:2...@testcorp)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(${EXTEN:2...@sub-intern)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(testco...@sub-intern)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(${EXTEN:2...@sub-routing)
exten => _**XX,n,NoOp()
exten => _**XX,n,Pickup(testco...@sub-routing)
exten => _**XX,n,NoOp()
exten => _**XX,n,Set(PICKUPMARK=${EXTEN:2})
exten => _**XX,n,Pickup(${EXTEN:2...@pickupmark)


Do you have an idea about what setting I need for pickup() to work ??? I 
think I just need to know the "channel", as the error says something 
about the "channel".



Jonas.


On 06/14/2010 08:00 PM, Peder wrote:


sip.conf and extensions.conf would be helpful as well as knowing what 
version you are running.  Based on what you went, I would say you have 
a config error, but I can't tell where without seeing the config.




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Re: [asterisk-users] Re : Asterisk Call routing problem

2010-06-14 Thread Vardan Harutyunyan
Hangup is comming after Dial or AGI?

-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

Adil Zaaraoui wrote:
> Dear Vardan,
> I had before the same problem, i reinstalled asterisk and it worked; now
> i get the same problem;
> i am using asterisk 1.4.22, it forwards if i call operator A (the one i
> was testing); but all other numbers are not forwarded, it just hangup,
> here is the output:
>
> Accepting AUTHENTICATED call from 192.168.1.4:
>
>  > requested format = unknown,
>
>  > requested prefs = (ulaw|slin|alaw),
>
>  > actual format = ulaw,
>
>  > host prefs = (gsm|ulaw|alaw),
>
>  > priority = mine
>
> -- Executing [00212675410...@outbound:1] Set("IAX2/#000105-8303",
> "calleeNumber=011212675410113") in new stack
>
> -- Executing [00212675410...@outbound:2] AGI("IAX2/#000105-8303")
>
> HANGUP 'IAX2/#000105-8303'
>
> thanks for the help
>
>
>
> 
> *De :* Vardan Harutyunyan 
> *À :* asterisk-users@lists.digium.com
> *Envoyé le :* Lun 14 juin 2010, 7h 36min 11s
> *Objet :* Re: [asterisk-users] Asterisk Call routing problem
>
> Whats about log?
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: i...@eif.am 
> www.eif-it.com
>
> Adil Zaaraoui wrote:
>  > Deal list,
>  > I have a problem with asterisk call routing.
>  > I configured my asterisk in a way that it forwards calls to a viop
>  > provider using IAX2 protocol.
>  > some cell phones numbers are routed and others are not!!! but when i try
>  > directly zoiper with my provider it works perfectly.
>  > Before, my asterisk was working perfectely. but i was testing just
>  > calling one number, i think that asterisk keeps in its cache the
>  > operator i was calling.
>  > It seems like asterisk record in its cache the dialed numbers. i tried
>  > to reload asterisk, i tried also to reboot the system, and the problem
>  > still persist.
>  > here is my config:
>  > [outbound]
>  > exten=>_00X.,1,set(calleNumber=_011${EXTEN:2})
>  > exten=>_00X.,n,AGI(http://localhost/myScript.agi)
>  >
>  > exten=>_00X.,n,Dial(IAX2/mylo...@outbound/${calleNumber
>  > })
>  >
>  > exten=>h,n,deadAGI(http://localhost/myScript2.agi)
>  >
>  > i tried disabling my scripts and still getting the same problem.
>  >
>  > Do you have any idea
>  >
>  >
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>  --
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> http://www.asterisk.org/hello
>
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>


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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread SIP
Randy R wrote:
> Hi,
>
> I'm looking to build an Asterisk box that can run at a remote
> location. Here are most of the specs of what I'm looking for:
>
> Physical hardware
>
> * Small pre-built PC (not buying board, case, all parts separately)
> * Low power consumption
> * No fan or very small fan
> * Hard drive (not flash memory)
>
> Capabilities/capacity
>
> * No GUI, no X
> * Register to multiple SIP servers
> * There will be no PSTN
> * No analog phones
> * Small number of SIP devices will register - maybe 10 max
> * Three simultaneous channels active
> * Skype for Asterisk needs to run on this <- so this means x86, right?
>
> Recommendations wanted
>
> * What hardware
> * What distro
> * Which Asterisk version
>
> Comments and suggestions welcome. This is going to be discussed on VUC
> as well, so if you're comfortable with it, come on by: http://vuc.me
>
> Thanks in advance,
>
> /r
>
>   
We use the Acer AspireRevo AR1600-U910H in a lot of locations. It's
enough to handle a few dozen remote office employees on a full asterisk
install with transcoding.  160G hard drive, couple of gigs of ram (comes
with 1), gigabit networking, and it's $200.   We slap CentOS 5.X on
there, and Asterisk 1.4.X (we don't do 1.6.X).   It DOES have a fan, but
it's a very VERY quiet fan.  For pure fanless, you might try the Lenovo
Q110, but it will run you a bit more (should come standard with 320G hd
and 2G of ram, though).

We love the Acer AspireRevos.  I have one at home I use as a media
centre, as well.

N.

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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Carlos Chavez
On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> Hi Carlso,
> 
> 
> Thanks for the input. I have done this in php and am not familiar with
> phpagi.
> So, there is absolutely no way to temporarily solve this problem by
> getting the value back from php file?
> 
> 
> Wondering if it would require a lot of work to change the php file to
> phpagi?
> Thanks,
> Bruce
> 
Here is an example:

exten => _x.,1,AGI(sample.agi)
exten => _x.,n,NoOp(${var})

sample.agi:

#!/usr/bin/php -q
request['agi_extension']; //Dialed extension
// the result is stored in $exten
// do something with your data
$agi->set_variable("var", $result);
$agi->verbose("The result was: $result", 3);
?>

You can even send parameters to the AGI via the command line like:

exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
for 1.4 or below

And access them via $argv[1], $argv[2] is there is some extra
information that you cannot get via the AGI variables.

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


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Re: [asterisk-users] Unable to pickup an extension, trying everything

2010-06-14 Thread Peder
sip.conf and extensions.conf would be helpful as well as knowing what
version you are running.  Based on what you went, I would say you have a
config error, but I can't tell where without seeing the config.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, June 14, 2010 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Unable to pickup an extension, trying everything

 

Hello list,

I try to pick up a ringing extension but nothing works.

To be clear, I'm trying to pick up extension 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:4]
Pickup("SIP/testcorp3-0041", "1...@123456") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:5]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:6]
Pickup("SIP/testcorp3-0041", "10") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:7]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:8]
Pickup("SIP/testcorp3-0041", "1...@from-testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:9]
Pickup("SIP/testcorp3-0041", "testco...@123456") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:10]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:11]
Pickup("SIP/testcorp3-0041", "testcorp1") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:12]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:13]
Pickup("SIP/testcorp3-0041", "testco...@from-testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:14]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:15]
Pickup("SIP/testcorp3-0041", "testco...@testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:16]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:17]
Pickup("SIP/testcorp3-0041", "1...@testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:18]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:19]
Pickup("SIP/testcorp3-0041", "1...@sub-intern") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:20]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:21]
Pickup("SIP/testcorp3-0041", "testco...@sub-intern") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:22]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:23]
Pickup("SIP/testcorp3-0041", "1...@sub-routing") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:24]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:25]
Pickup("SIP/testcorp3-0041", "testco...@sub-routing") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for testcorp1.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:26]
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:27]
Set("SIP/testcorp3-0041", "PICKUPMARK=10") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:28]
Pickup("SIP/testcorp3-0041", "1...@pickupmark") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No
target channel found for 10.


These are the channels :

asterisk*CLI> core show channels

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Hi Carlso,

Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by getting
the value back from php file?

Wondering if it would require a lot of work to change the php file to
phpagi?
Thanks,
Bruce

On Mon, Jun 14, 2010 at 12:12 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> >
> > I have a php file that if an argument is passed to it, it will echo a
> > number back. I am looking to use system() in dial-plan to send
> > ${EXTEN} to it and then to get that processed value back from the php
> > file and put it in $var back into asterisk dial-plan. While trying
> > this method doesn't work:
> >
> >
> > exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})
> > exten => _x.,n,NoOp(${var})
> >
> >
>
> This is exactly what AGI was made for.  Look into the PHPAGI
> library.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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[asterisk-users] Re : Asterisk Call routing problem

2010-06-14 Thread Adil Zaaraoui
Dear Vardan,

I had before the same problem, i reinstalled asterisk and it worked; now i get 
the same problem;

i am using asterisk 1.4.22, it forwards if i call operator A (the one i was 
testing); but all other numbers are not forwarded, it just hangup, here is the 
output:

Accepting AUTHENTICATED call from 192.168.1.4:
> requested format = unknown,
> requested prefs = (ulaw|slin|alaw),
> actual format = ulaw,
> host prefs = (gsm|ulaw|alaw),
> priority = mine
-- Executing [00212675410...@outbound:1] Set("IAX2/#000105-8303", 
"calleeNumber=011212675410113") in new stack
-- Executing [00212675410...@outbound:2] AGI("IAX2/#000105-8303")
HANGUP 'IAX2/#000105-8303'
 
 
thanks for the help





De : Vardan Harutyunyan 
À : asterisk-users@lists.digium.com
Envoyé le : Lun 14 juin 2010, 7h 36min 11s
Objet : Re: [asterisk-users] Asterisk Call routing problem

Whats about log?

-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com

Adil Zaaraoui wrote:
> Deal list,
> I have a problem with asterisk call routing.
> I configured my asterisk in a way that it forwards calls to a viop
> provider using IAX2 protocol.
> some cell phones numbers are routed and others are not!!! but when i try
> directly zoiper with my provider it works perfectly.
> Before, my asterisk was working perfectely. but i was testing just
> calling one number, i think that asterisk keeps in its cache the
> operator i was calling.
> It seems like asterisk record in its cache the dialed numbers. i tried
> to reload asterisk, i tried also to reboot the system, and the problem
> still persist.
> here is my config:
> [outbound]
> exten=>_00X.,1,set(calleNumber=_011${EXTEN:2})
> exten=>_00X.,n,AGI(http://localhost/myScript.agi)
>
> exten=>_00X.,n,Dial(IAX2/mylo...@outbound/${calleNumber
> })
>
> exten=>h,n,deadAGI(http://localhost/myScript2.agi)
>
> i tried disabling my scripts and still getting the same problem.
>
> Do you have any idea
>
>


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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Randy R


Thanks everyone for your suggestions.

Is it feasable to run Skype for Asterisk on the Atom processors? It's
a feature I'd really like to have. As for conferencing, we rarely use
it but never would need more than 3 seats.



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[asterisk-users] calling peer from server

2010-06-14 Thread nikhil singhania
Thanks for the reply.
Actually my problem is not related to sip.conf and extensions.conf. I have
used only standard files from martin pdf which are given as example.
I am able to call some system connected over LAN, when each has a softphone
and are registered on a asterisk server. But now what i want is instead of
using the softphone I write a function in my file which will be executed
when the call is placed.
In that file i wrote
   $agi=new AGI();
   $agi->exec_dial(SIP,2002,NULL,NULL,NULL);
I have used the exec_dial function found in phpagi.php . It is built above
basic dial function. Here 'agi' is an instance of class AGI, which has a
method exec_dial.

When i execute the php file, over command line on my unix machine, I am
expecting a call on the softphone which I have registered on the asterisk
server.

For ip clarification:
all have static ip:
server ip:172.26.48.208:5060
i have configured twinkle as a softphone client on 172.26.48.113:5061, since
on the asterisk server cli when i use 'sip debug set peer 2002' it shows the
registered ip, as i said earlier, but again on 'sip show registry' no value
is displayed. I don't know what's going on.
  In the softphone I give domin information as the ip of the asterisk server
i.e. 172.26.48.208. In the softphone it shows registration successful.

Thanks in advance
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Gordon Henderson
On Mon, 14 Jun 2010, Chris Bagnall wrote:

> Actually, the Atom seems to be surprisingly powerful. We have a couple of
> Atom boxes with transcoding and conferences enabled without issue. I
> wouldn't pretend it'll cope with hundreds of conference participants, but
> with ~10 or so it seems to be fine.

I'll second the Atoms - I have several in the data centre handling VoIP, 
virtual PBXs, etc. And you can now get fanless motherboards. Bliss.

Even using a few as general purpose LAMP servers too - the data centre I 
use doesn't charge per amp, but it's coming, and already there in most big 
places - in the UK, anyway - it seems Amps cost more than Gb)

> Likewise with transcoding - we've only really tested up to ~30 channels with
> G.711 to GSM, not any of the "heavier CPU workload" translations (e.g. iLBC
> or G.729).
>
> For a small to medium office (e.g. 30 extensions, 10 concurrent calls) it
> works fine, even with a little conferencing and transcoding.

I do that with a 500MHz AMD Geode ... (no transcoding though - benchmarked 
it to 85 concurrent calls, handling the media streams - limit these boxes 
to 60 extensions though)

Gordon

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Re: [asterisk-users] How to disable day light saving on Snom 360 phones?

2010-06-14 Thread Zeeshan Zakaria
I knew somebody will send me these links, so I mentioned in my question that
wiki was of no help. These were the first pages I went through, and they
don't tell how to *disable* DST. To have DST picked up automatically by
these phones based on timezone is so nice of Snom but that is exactly what I
*don't* want in this case. Setting it to '0' seconds doesn't help either.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-14 12:44 PM, "Philipp von Klitzing" <
klitz...@pool.informatik.rwth-aachen.de> wrote:

Hi!


> Snom wiki was not helpful. My client wants to keep his phones pointed
> to UTC time, no DST, no c...
How about this?

XML Syntax: Settings/dst/xml
http://wiki.snom.com/Settings/dst

Settings/utc offset:
Signed UTC offset in seconds. This value is retrieved automatically from
the timezone configuration. Usually there will be no need to change this
setting.
http://wiki.snom.com/Settings/utc_offset

NOTE: The timezone setting sets utc_offset and dst automatically !

Philipp


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Re: [asterisk-users] How to disable day light saving on Snom 360 phones?

2010-06-14 Thread Philipp von Klitzing
Hi!

> Snom wiki was not helpful. My client wants to keep his phones pointed
> to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours
> difference. 

How about this?

XML Syntax: Settings/dst/xml
http://wiki.snom.com/Settings/dst

Settings/utc offset:
Signed UTC offset in seconds. This value is retrieved automatically from 
the timezone configuration. Usually there will be no need to change this 
setting. 
http://wiki.snom.com/Settings/utc_offset

NOTE: The timezone setting sets utc_offset and dst automatically ! 

Philipp


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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread William Stillwell (Lists)
I would think AGI would be better. ?

 

I don't think system() returns anything, except maybe a success/fail ?

 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, June 14, 2010 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to pass variable back and forth from dialplan
to php file?

 

Hi Everyone,

 

I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method doesn't work:

 

exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})

exten => _x.,n,NoOp(${var})

 

What is right syntax for line 1?

 

Thanks,

Bruce

 

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Re: [asterisk-users] How to stop intruder from registering sip?

2010-06-14 Thread Dave Platt
> As I mentioned, I'm not inclined to mess with the secrets, too much 
> hassle for users. 

I'm afraid that I have to consider that attitude to be a bit like
saying "It's too much hassle for us to insist that our employees
lock their desk drawers and the front door... or wash their
hands after going to the bathroom... or cover their mouths when
they sneeze.  Oh, yeah, we keep the combination to the corporate
safe on a yellow sticky-note on the bulletin board, so that anyone
who forgets it can figure it out quickly."

There are ways to make stronger secrets easier to work with.
One method creates secret phrases by concatenating a bunch
of randomly-chosen dictionary words.  If you have enough
such words in the dictionary you can create phrases which have
enough randomness to survive brute-force attacks but which
aren't too difficult to type in correctly.  For example, such
a gibberish-generator might output

fizzy.basal.nerfy.dogma.colma.flinx

It's your choice... but these basic security principles about
setting secrets/passwords have the fruits of many peoples'
expen$ive experience at the high cost of *not* doing things
properly.

If the cost of doing things securely is that you have to spend
a few minutes of IT-guru time setting up each user's phone
or softphone, or need to write a document-generator which
prints out step-by-step instructions for each user with the
necessary user-name and secret included... it could be a
*very* good investment.


> That's why I'm considering deny/permit.

> Does that solve my problem?

*Only* if you have complete physical control over *every*
network on which those phones will be used, *and* all of
your employees are completely trustworthy.

It's really no solution at all if you need to have "road warriors"
using soft-phones on networks across the world, since you won't
be able to deny IP addresses meaningfully in that case.  All it
would take would be one such employee using a softphone via an
insecure network (e.g. open WiFi access point), somebody sniffs
the protocol and sees the registration and records the extension
number and then does a brute-force secret-guessing attack.  Boom.
You're out hundreds or thousands of dollars of calling costs before
you can react.  Scammers can use your SIP system to make calls to
"premium" phone numbers that cost several dollars per minute... and
the scammer may well get a portion of this revenue.

Big companies have ended up losing tens of thousands of dollars
to this sort of attack against their PBX systems.

Or, worse... your SIP secrets end up in the hands of a cybergang
which starts using your system for criminal activities (e.g.
drug-trafficing, making scam calls to homeowners, etc.), and
you find your company facing investigation by law enforcement,
or your SIP provider cuts you off due to abuse complaints.  The
secondary cost of either of these to your business could be
severe.

As Dirty Harry said, "How lucky do you feel?".  You've already
been hit once.

> But I'm struck with your notion of having sip user ids different from 
> extensions. That would not require any user effort, or messing with each 
> phone. But...
> 
> We use a combo of aastra 9133i and 57i's. Don't the user id and the 
> extension HAVE to be the same? I had thought the aastra's used the 
> extension as the SIP id to register.

By no means - at least, not in the 9133i, and I'd be surprised if
the 57i had that requirement.

Look in the Administration manual for the 9133i, Appendix A,
"SIP Basic, Global Settings", "SIP Global Authentication".
This is where you can set the "authentication name" and
"sip password", which are what the phone uses to register with
the server (e.g. the SIP user name and secret).  Make this name
*different* from the extension name, and provide a good secret.

You can also set the "SIP display name", which is what
shows up on the screen, and is sent as the "From" field
in the SIP protocol.  You can set this to the user's primary
extension number.

A bit further down, there are per-line registration fields
which do the same thing for individual line-presence
buttons... screen name (also used for From:), user name
(for SIP registration), password (SIP registration secret).




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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread Carlos Chavez
On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
> Hi Everyone,
> 
> 
> I have a php file that if an argument is passed to it, it will echo a
> number back. I am looking to use system() in dial-plan to send
> ${EXTEN} to it and then to get that processed value back from the php
> file and put it in $var back into asterisk dial-plan. While trying
> this method doesn't work:
> 
> 
> exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})
> exten => _x.,n,NoOp(${var})
> 
> 

This is exactly what AGI was made for.  Look into the PHPAGI library.

-- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] How to pass variable back and forth from dialplan to php file?

2010-06-14 Thread bruce bruce
Hi Everyone,

I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method doesn't work:

exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})
exten => _x.,n,NoOp(${var})

What is right syntax for line 1?

Thanks,
Bruce
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[asterisk-users] How to disable day light saving on Snom 360 phones?

2010-06-14 Thread Zeeshan Zakaria
Greetings,

Sounds like a simple thing to do, but I was not able to do it on these
particular phones. Snom wiki was not helpful. My client wants to keep his
phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0
hours difference.

The phones are provisioned from a tftp server.

If I remove 'dst' value from the provisioning file, on bootup phones force
users to pickup a time zone. The only suitable timezone is 'GBR-0', but it's
Greenwich time, which does take in account DST. If I do 'utc_offset: -3600',
it'll be good only until next DST.

Does anybody khow can I achieve this?

Thanks,

Zeeshan A Zakaria

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[asterisk-users] Unable to pickup an extension, trying everything

2010-06-14 Thread Jonas Kellens

Hello list,

I try to pick up a ringing extension but nothing works.

To be clear, I'm trying to pick up extension 10.

[Jun 14 17:37:34] -- Executing [*...@from-testcorp:4] 
Pickup("SIP/testcorp3-0041", "1...@123456") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:5] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:6] 
Pickup("SIP/testcorp3-0041", "10") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:7] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:8] 
Pickup("SIP/testcorp3-0041", "1...@from-testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:9] 
Pickup("SIP/testcorp3-0041", "testco...@123456") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:10] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:11] 
Pickup("SIP/testcorp3-0041", "testcorp1") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:12] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:13] 
Pickup("SIP/testcorp3-0041", "testco...@from-testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:14] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:15] 
Pickup("SIP/testcorp3-0041", "testco...@testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:16] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:17] 
Pickup("SIP/testcorp3-0041", "1...@testcorp") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:18] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:19] 
Pickup("SIP/testcorp3-0041", "1...@sub-intern") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:20] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:21] 
Pickup("SIP/testcorp3-0041", "testco...@sub-intern") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:22] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:23] 
Pickup("SIP/testcorp3-0041", "1...@sub-routing") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:24] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:25] 
Pickup("SIP/testcorp3-0041", "testco...@sub-routing") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for testcorp1.


[Jun 14 17:37:34] -- Executing [*...@from-testcorp:26] 
NoOp("SIP/testcorp3-0041", "") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:27] 
Set("SIP/testcorp3-0041", "PICKUPMARK=10") in new stack
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:28] 
Pickup("SIP/testcorp3-0041", "1...@pickupmark") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: 
No target channel found for 10.



These are the channels :

asterisk*CLI> core show channels
Channel  Location State   Application(Data)
SIP/testcorp1-00 s...@from-testcorp:1Ringing AppDial((Outgoing Line))
IAX2/testlocal-1612  s...@sub-routing:13 RingDial(SIP/testcorp1|30)
2 active channels
1 active call


I don't know what the "target channel" is and I try every context that I 
have defined in my dialplan..


I try to pickup extension 10 or also the SIPaccount testcorp1 (which is 
reachable through ext

Re: [asterisk-users] WARNING message when play

2010-06-14 Thread Warren Selby
On Mon, Jun 14, 2010 at 8:27 AM, equis software wrote:

> In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and aboveappear 
> this messages
>
>
This message was added around 1.4.23 to let you know that you're violating
the AGI protocol.  Read up on the AGI protocol then check through your AGI
file to see what you're missing.  Usually it's not reading something that
asterisk is sending back to your script, or not sending a response that
asterisk is expecting.  Everything else typically will still do what you
want, but technically you're violating the protocol.


-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] Hint priority in RealTime

2010-06-14 Thread Ishfaq Malik

Hi

I've just had a request from a customer who wants to use Busy Lamp Feed. 
I've had a look around and it would appear that you have top use the 
'hint' priority. We are using asterisk 1.4.17 with realtime and the 
priority column in the extensions table is a tinyint so obviously I 
can't put hint in there.


Has anyone any experience of working round this problem?

Thanks

Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062
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Re: [asterisk-users] How to stop intruder from registering sip?

2010-06-14 Thread Warren Selby
On Sun, Jun 13, 2010 at 3:06 PM, sean darcy  wrote:

> But I'm struck with your notion of having sip user ids different from
> extensions. That would not require any user effort, or messing with each
> phone. But...
>

It'd be just as much effort as changing the passwords for each phone.
You'll have to modify the SIP USERNAME setting on each phone you want to
change the username for, the same as modifying the SIP PASSWORD setting for
each phone.

I'd recommend changing all of the passwords, modifying them on the phones
themselves, and then setting up a fail2ban solution that will ban anyone who
has more than 5 failed password attempts in less than a few minutes.  You
can even leave iptables setup to allow all, and just block the IPs that
fail2ban triggers on.

In your situation, using a password like , you may not end up with 5
failed password attempts, as that's usually one of the first things the
scripts out there will try, so fail2ban will only help you if you up your
password security.

I've had trouble getting the permit/deny trick to work as an IP filter in
the past, so instead I went with an iptables / fail2ban solution, along with
difficult to guess passwords.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] Multiple parking lots - 1.6

2010-06-14 Thread Mike
Hi,

 

I am trying to find documents on multiple parking lots in Asterisk 1.6,
which was announced as a new feature.  Although 1.6 has been out a while, I
see no info on how to set this up.

 

The wiki is stuck in the past (apr 2008).  Is there anything available on
how to create multiple parking lots on a single Asterisk install?

 

Thank you,

 

Mike

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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Chris Bagnall
Actually, the Atom seems to be surprisingly powerful. We have a couple of 
Atom boxes with transcoding and conferences enabled without issue. I 
wouldn't pretend it'll cope with hundreds of conference participants, but 
with ~10 or so it seems to be fine.

Likewise with transcoding - we've only really tested up to ~30 channels with 
G.711 to GSM, not any of the "heavier CPU workload" translations (e.g. iLBC 
or G.729).

For a small to medium office (e.g. 30 extensions, 10 concurrent calls) it 
works fine, even with a little conferencing and transcoding.

Regards,

Chris


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Re: [asterisk-users] WARNING message when play

2010-06-14 Thread equis software
In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and above appear
this messages


On Mon, Jun 14, 2010 at 9:08 AM, equis software wrote:

> This are the console messages with AGI debugging
>
> AGI Rx << STREAM FILE msgBienvenida112 1234567890*#
> -- Playing 'msgBienvenida112' (escape_digits=1234567890*#)
> (sample_offset 0)
> [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
> failed: Broken pipe
> [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
> failed: Broken pipe
> [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
> failed: Broken pipe
> [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
> failed: Broken pipe
> [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
> failed: Broken pipe
>
>
>
> On Fri, Jun 11, 2010 at 4:23 PM, Steve Edwards 
> wrote:
>
>> On Fri, 11 Jun 2010, equis software wrote:
>>
>>  When I use an eagi script when play a message appear a lot of warning
>>> messages, but it play very well
>>> I´m using
>>> Asterisk 1.4.32
>>> dahdi-linux-2.3.0.1
>>> chan_ss7-1.4.1
>>>
>>> Any ideas??
>>>
>>> -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
>>> [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write()
>>> failed: Broken pipe
>>>
>>
>> I don't use EAGI, and coming from file.c may indicate otherwise, but
>> "broken pipe" messages and AGI may mean you are violating the AGI protocol.
>> Enabling AGI debugging and watching the console output may give you a clue.
>>
>> EAGI implies you are doing something with the incoming audio on FD 3. Any
>> chance you are closing FD3?
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>> Newline  Fax: +1-760-731-3000
>> --
>> _
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] calling peer from server

2010-06-14 Thread Tarek Sawah

does that phon has a static IP? does it register with the server? posting your 
SIP.con and extensions.conf related to this issue could help us to understand 
what you are doing.

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






From: niksingha...@gmail.com
Date: Mon, 14 Jun 2010 17:49:37 +0530
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] calling peer from server

Hi everybody,
  This is the console output of the asterisk server.
debian-te410*CLI> sip set debug peer 2002
SIP Debugging Enabled for IP: 172.26.48.113:5061


I have a sofphone with user 2002 registered on the server on the ip 113. 

 I am trying to place a call to the sofphone on this ip. I have written a 
simple php script which utilises the exec_dial function inbuilt in phpagi.php 
file.


I have tried diff ways but can't seem to get it work.
  Can please some one suggest me anything in this regard.
-- 
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem


IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
  niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/



  
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Re: [asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Tarek Sawah

when you add an agent to a queue the agent should log in try adding 
member=SIP/301member=SIP/302instead of agent directives.this will ring both 
phones.. from your output it doesn't seem to be ringing the agents at all.

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
















Hello there

 

 

I have been struggling with queues, because
i think this is the right module for our business.

My main goal, is when we receive external
calls, the receptionist should be able to transfer the call to us 

Technicians, and I am trying to add 2
extensions to a queue name [teknisk]

Extension 301 and 302.

 

I have a test setup now which I thought
should look like this:

When a external call come to my external
number (67209611) this will ring for 5 seconds, and then transferred to queue 
“teknisk”

And I thought that internal
phonex/extensions 301 and 302 would ring.

 

But, when I ring the external number, it
just rings…and rings…until it hang-ups.

 

CLI output shows that the commands are
running, but maybe the wrong way, are the queue command routed to my sip
provider?

 

Info: 67209611 is my public phone number.

90015103 is my cell phone number

301 and 302 are internal extensions in
technician department, which I am trying to route the queue to with the ringall
argument.

This happens:

Reloading MGCP

  == Using SIP RTP TOS bits 184

  == Using SIP RTP CoS mark 5

-- Executing
[4767209...@internal:1]
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new
stack

-- Executing
[4767209...@internal:2]
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num
90015103") in new stack

Callerid num 90015103

-- Executing
[4767209...@internal:3]
Dial("SIP/odin.service.ipallover.net-00d1",
"SIP/301,5") in new stack

  == Using SIP RTP TOS bits 184

 == Using SIP RTP CoS mark 5

-- Called 301

-- SIP/301-00d2 is
ringing

-- Nobody picked up in
5000 ms

-- Executing
[4767209...@internal:4]
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk")
in new stack

-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'

-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1

--
 Playing 'queue-youarenext.gsm'
(language 'en')

-- Told
SIP/odin.service.ipallover.net-00d1 in teknisk their queue position (which
was 1)

--
 Playing 'queue-thankyou.gsm'
(language 'en')

-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'

-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1

  == Spawn extension (internal,
4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-00d1'

 

asterisk*CLI>

 

---

Agents.conf is default and  i have two
extensions/agents

agent => 301,301

agent => 302,302

 

 

--

[r...@asterisk asterisk]# more queues.conf

 

[teknisk]

music = default

announce = queue-callswaiting.gsm

strategy = ringall

timeout = 15

retry = 0

maxlen = 0

announce-frequency = 120

announce-holdtime = yes

 

member => Agent/301

member => Agent/302

 

-

Sip.conf

[301]

type=friend

secret=xx

host=dynamic

context=phones

mailbox=...@default


qualify=yes

callgroup=teknisk

-

extensions.conf snipped

 

;exten 301

exten => 4767209611,1,NoOp();

exten => 4767209611,n,Verbose(Callerid
num ${CALLERID(num)});

exten => 4767209611,n,Dial(SIP/301,5);

exten => 4767209600,n,Queue(teknisk);

exten =>
4767209611,n,Voicemail(301);  
;Added 06.Mai.10-Aksel

 

 

 

 

Could someone please help me in the right
direction here?

 

 

Med vennlig hilsen

Abacus IT AS

- din Visma Software Partner

 

Tor Aksel Celasun

Mobilnummer 900 15 103

Sentralbord/Support 4000 1850

ak...@abacus-it.no

 

  
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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread John Ervin
How limited are you in the no-no category when using a small machine 
like this.  Do you set you system to prevent transcoding?  Does it 
prevent conferencing?  Just curious.  I kind of like those features but 
love the idea of the small machines.


On 06/14/2010 07:37 AM, Chris Bagnall wrote:

I'm looking to build an Asterisk box that can run at a remote
location.
 

We've used the Asus eeeBox (desktop version of their little netbooks) quite
successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.

Generally we run Gentoo Linux with Asterisk 1.4., but no reason why
you couldn't run another version you're more comfortable with.

For 3 concurrent calls, even a machine of this spec might be overkill, but
it's a good general-purpose server to have on-site at a remote location and
might be useful for other things (general fileserver, rsync backup server,
etc.).

The eeeBox also has the advantage of being cheap (quite probably cheaper
than smaller/lower power units), which means keeping a spare around in case
of hardware failure isn't an unrealistic option.

Regards,

Chris

   



--
John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
jer...@jervin.com




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[asterisk-users] calling peer from server

2010-06-14 Thread nikhil singhania
Hi everybody,
  This is the console output of the asterisk server.
debian-te410*CLI> sip set debug peer 2002
SIP Debugging Enabled for IP: 172.26.48.113:5061
I have a sofphone with user 2002 registered on the server on the ip 113.

 I am trying to place a call to the sofphone on this ip. I have written a
simple php script which utilises the exec_dial function inbuilt in
phpagi.php file.
I have tried diff ways but can't seem to get it work.
  Can please some one suggest me anything in this regard.
-- 
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
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Re: [asterisk-users] WARNING message when play

2010-06-14 Thread equis software
This are the console messages with AGI debugging

AGI Rx << STREAM FILE msgBienvenida112 1234567890*#
-- Playing 'msgBienvenida112' (escape_digits=1234567890*#)
(sample_offset 0)
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe



On Fri, Jun 11, 2010 at 4:23 PM, Steve Edwards wrote:

> On Fri, 11 Jun 2010, equis software wrote:
>
>  When I use an eagi script when play a message appear a lot of warning
>> messages, but it play very well
>> I´m using
>> Asterisk 1.4.32
>> dahdi-linux-2.3.0.1
>> chan_ss7-1.4.1
>>
>> Any ideas??
>>
>> -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
>> [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write()
>> failed: Broken pipe
>>
>
> I don't use EAGI, and coming from file.c may indicate otherwise, but
> "broken pipe" messages and AGI may mean you are violating the AGI protocol.
> Enabling AGI debugging and watching the console output may give you a clue.
>
> EAGI implies you are doing something with the incoming audio on FD 3. Any
> chance you are closing FD3?
>
> --
> 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] Small PC to build and run Asterisk

2010-06-14 Thread Randy R
On Mon, Jun 14, 2010 at 1:37 PM, Chris Bagnall
 wrote:
> We've used the Asus eeeBox (desktop version of their little netbooks) quite
> successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.

Wow, we used to benefit from the space program that handed down
technologies madre cheaper, now it's the Netbook revolution that gives
us inexpensive "Nettops"!

No lack of choice in the Atom series from several manufacturers.

Thanks, Chris!

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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Chris Bagnall
> I'm looking to build an Asterisk box that can run at a remote
> location.

We've used the Asus eeeBox (desktop version of their little netbooks) quite 
successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.

Generally we run Gentoo Linux with Asterisk 1.4., but no reason why 
you couldn't run another version you're more comfortable with.

For 3 concurrent calls, even a machine of this spec might be overkill, but 
it's a good general-purpose server to have on-site at a remote location and 
might be useful for other things (general fileserver, rsync backup server, 
etc.).

The eeeBox also has the advantage of being cheap (quite probably cheaper 
than smaller/lower power units), which means keeping a spare around in case 
of hardware failure isn't an unrealistic option.

Regards,

Chris

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[asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun
Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue "teknisk"
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings...and rings...until it 
hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no

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Re: [asterisk-users] logging stopped suddenly

2010-06-14 Thread Jonas Kellens

I don't think it's a disk space issue :

bash-3.2# df -h
FilesystemSize  Used Avail Use% Mounted on
/dev/sda1  25G  5.0G   19G  21% /
tmpfs 256M 0  256M   0% /dev/shm

bash-3.2# df -h /var/log/
FilesystemSize  Used Avail Use% Mounted on
/dev/sda1  25G  5.0G   19G  21% /


Jonas


On 06/14/2010 12:22 PM, Ishfaq Malik wrote:


Hi

Could it be a disk space issue?

Ish

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Re: [asterisk-users] logging stopped suddenly

2010-06-14 Thread Ishfaq Malik

On 14/06/10 11:04, Jonas Kellens wrote:

Hello list,

I noticed today that the last logfiles dates 3 days ago !

The logfiles are rotated every night. The logfiles of 2 days ago, 1 
day ago and today are empty !


vps*CLI> module show like logger
Module 
Description  Use Count

0 modules loaded
vps*CLI> logger reload
[Jun 14 11:57:19]   == Parsing '/etc/asterisk/logger.conf': [Jun 14 
11:57:19] Found

[Jun 14 11:57:19] Asterisk Queue Logger restarted
vps*CLI> module show like logger
Module 
Description  Use Count

0 modules loaded

ls -l /var/log/asterisk :

-rw-rw-r-- 1 root root 0 Jun 14 02:02 messages.vps.hosting.net
-rw-rw-r-- 1 root root 0 Jun 13 02:02 messages.vps.hosting.net.1
-rw-rw-r-- 1 root root592918 Jun  5 02:30 messages.vps.hosting.net.10
-rw-rw-r-- 1 root root 145037679 Jun  4 02:30 messages.vps.hosting.net.11
-rw-rw-r-- 1 root root866083 Jun  3 02:30 messages.vps.hosting.net.12
-rw-rw-r-- 1 root root   1395911 Jun  2 02:30 messages.vps.hosting.net.13
-rw-rw-r-- 1 root root   2151387 Jun  1 02:30 messages.vps.hosting.net.14
-rw-rw-r-- 1 root root 0 Jun 12 02:02 messages.vps.hosting.net.2


How come my logging suddenly stopped and how do I reactivate the 
logging ??



Jonas.


Hi

Could it be a disk space issue?

Ish

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[asterisk-users] logging stopped suddenly

2010-06-14 Thread Jonas Kellens

Hello list,

I noticed today that the last logfiles dates 3 days ago !

The logfiles are rotated every night. The logfiles of 2 days ago, 1 day 
ago and today are empty !


vps*CLI> module show like logger
Module Description  
Use Count

0 modules loaded
vps*CLI> logger reload
[Jun 14 11:57:19]   == Parsing '/etc/asterisk/logger.conf': [Jun 14 
11:57:19] Found

[Jun 14 11:57:19] Asterisk Queue Logger restarted
vps*CLI> module show like logger
Module Description  
Use Count

0 modules loaded

ls -l /var/log/asterisk :

-rw-rw-r-- 1 root root 0 Jun 14 02:02 messages.vps.hosting.net
-rw-rw-r-- 1 root root 0 Jun 13 02:02 messages.vps.hosting.net.1
-rw-rw-r-- 1 root root592918 Jun  5 02:30 messages.vps.hosting.net.10
-rw-rw-r-- 1 root root 145037679 Jun  4 02:30 messages.vps.hosting.net.11
-rw-rw-r-- 1 root root866083 Jun  3 02:30 messages.vps.hosting.net.12
-rw-rw-r-- 1 root root   1395911 Jun  2 02:30 messages.vps.hosting.net.13
-rw-rw-r-- 1 root root   2151387 Jun  1 02:30 messages.vps.hosting.net.14
-rw-rw-r-- 1 root root 0 Jun 12 02:02 messages.vps.hosting.net.2


How come my logging suddenly stopped and how do I reactivate the logging ??


Jonas.
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Re: [asterisk-users] How to stop intruder from registering sip?

2010-06-14 Thread Tarek Sawah

along with all the previous suggestions.. i found out that fail2ban is a good 
safe tool to be used along with hard passwords and not using numeric 
usernames.. for me using A2Billing along with Asterisk was a pain because it 
needs to create usernames numeric.. so i had to create strong SIP users and 
passwords then assign a2billing accounts to them to make it safer.. plus the 
fail2ban .. give it a try.

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308






> Date: Sun, 13 Jun 2010 22:28:38 -0700
> To: asterisk-users@lists.digium.com
> From: i...@extrasensory.com
> Subject: Re: [asterisk-users] How to stop intruder from registering sip?
> 
> At 01:06 PM 6/13/2010, you wrote:
> >We use a combo of aastra 9133i and 57i's. Don't the user id and the
> >extension HAVE to be the same? I had thought the aastra's used the
> >extension as the SIP id to register.
> 
> So in your extensions.conf you need lines like:
> 
> exten => 123,1,dial(SIP/123_thisisAfunnyextension)
> 
> Well, that should give you the idea. Don't know if it's the best way, 
> but it's worked for me.
> 
> Ira 
> 
> 
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Re: [asterisk-users] Qwest PRIs

2010-06-14 Thread Doug Lytle
Voip Asterisk wrote:
> Ya I'm passed that part now.  I have dahdi properly loading the card, 
> and both links are green.  Asterisk recognizes the channels, but still 
> shows the span as down.

Some telcos turn down a span when too many errors have occurred.   You 
may still want to contact your provider, at least they can tell you if 
you're throwing errors.

Doug

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Re: [asterisk-users] Issues running Asterisk + Iaxmodem + Hylafax on same machine

2010-06-14 Thread Doug Lytle
Andrew Joakimsen wrote:
> I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on
> the same machine. After rebooting the iaxmodems don't register to
> asterisk. Stoping and starting the relevant services gets it working,
>
>

I'm using init to spawn a script that kicks off iaxmodem.  I don't know 
if this is the difference, but it's working fine that way for me.

cat start.iaxmodem
#!/bin/sh

/usr/local/bin/iaxmodem

sleep 3

/usr/local/sbin/faxgetty ttyIAX01 &
/usr/local/sbin/faxgetty ttyIAX02 &
/usr/local/sbin/faxgetty ttyIAX03 &
/usr/local/sbin/faxgetty ttyIAX04 &


Doug


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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread --[ UxBoD ]--
> * Skype for Asterisk needs to run on this <- so this means x86, right?

or x86_64 is fine

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[asterisk-users] debug message: Internal timing is disabled

2010-06-14 Thread Daniel Knoll
Hi all,
i got a lot of this messages if only one caller is in a meetme
conference and it playing a MusicOnHold Sound. If a second Caller
entry the Conference the messages ended.

DEBUG[11794] channel.c: Internal timing is disabled
(option_internal_timing=0 chan->timingfd=61

What does this message mean?

Thanx for answers
Daniel

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Re: [asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Saflis ST
AsteriskNow is better

On Mon, Jun 14, 2010 at 2:26 PM, Randy R  wrote:

> Hi,
>
> I'm looking to build an Asterisk box that can run at a remote
> location. Here are most of the specs of what I'm looking for:
>
> Physical hardware
>
>* Small pre-built PC (not buying board, case, all parts separately)
>* Low power consumption
>* No fan or very small fan
>* Hard drive (not flash memory)
>
> Capabilities/capacity
>
>* No GUI, no X
>* Register to multiple SIP servers
>* There will be no PSTN
>* No analog phones
>* Small number of SIP devices will register - maybe 10 max
>* Three simultaneous channels active
>* Skype for Asterisk needs to run on this <- so this means x86, right?
>
> Recommendations wanted
>
>* What hardware
>* What distro
>* Which Asterisk version
>
> Comments and suggestions welcome. This is going to be discussed on VUC
> as well, so if you're comfortable with it, come on by: http://vuc.me
>
> Thanks in advance,
>
> /r
>
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[asterisk-users] Small PC to build and run Asterisk

2010-06-14 Thread Randy R
Hi,

I'm looking to build an Asterisk box that can run at a remote
location. Here are most of the specs of what I'm looking for:

Physical hardware

* Small pre-built PC (not buying board, case, all parts separately)
* Low power consumption
* No fan or very small fan
* Hard drive (not flash memory)

Capabilities/capacity

* No GUI, no X
* Register to multiple SIP servers
* There will be no PSTN
* No analog phones
* Small number of SIP devices will register - maybe 10 max
* Three simultaneous channels active
* Skype for Asterisk needs to run on this <- so this means x86, right?

Recommendations wanted

* What hardware
* What distro
* Which Asterisk version

Comments and suggestions welcome. This is going to be discussed on VUC
as well, so if you're comfortable with it, come on by: http://vuc.me

Thanks in advance,

/r

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[asterisk-users] Issues running Asterisk + Iaxmodem + Hylafax on same machine

2010-06-14 Thread Andrew Joakimsen
I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on
the same machine. After rebooting the iaxmodems don't register to
asterisk. Stoping and starting the relevant services gets it working,
but what is the point of using init scripts if it does not work right?
I already tried to adjust the init scripts in /etc/rc3.d so I have:

S50asterisk
s90iaxmodem
S95hylafax

So it should be starting in the correct order. I've previously done
this so Asterisk is on one server and IAXmodem and Hylafax on another,
I am stumped.


[r...@pbxserver ~]# rasterisk
Asterisk 1.6.1.20, 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.
=
Connected to Asterisk 1.6.1.20 currently running on pbxserver (pid = 2218)
Verbosity is at least 3
pbxserver*CLI> iax2 show pee
peer   peers
pbxserver*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status
ttyIAX0  (Unspecified)   (D)  255.255.255.255  0 (E) UNKNOWN
ttyIAX1  (Unspecified)   (D)  255.255.255.255  0 (E) UNKNOWN
ttyIAX2  (Unspecified)   (D)  255.255.255.255  0 (E) Unmonitored
ttyIAX3  (Unspecified)   (D)  255.255.255.255  0 (E) Unmonitored
4 iax2 peers [0 online, 2 offline, 2 unmonitored]
pbxserver*CLI> exit
[r...@pbxserver ~]# /etc/init.d/hylafax stop
Shutting down HylaFAX queue manager (faxq):[  OK  ]
Shutting down HylaFAX server (hfaxd):  [  OK  ]
[r...@pbxserver ~]# /etc/init.d/asterisk stop
Stopping safe_asterisk:[  OK  ]
Shutting down asterisk:[  OK  ]
[r...@pbxserver ~]# /etc/init.d/iaxmodem stop
[r...@pbxserver ~]# /etc/init.d/asterisk start
Starting asterisk: [  OK  ]
[r...@pbxserver ~]# /etc/init.d/iaxmodem start
Starting IAXmodem: [  OK  ]
[r...@pbxserver ~]# /etc/init.d/hylafax start
Starting HylaFAX queue manager (faxq): [  OK  ]
Starting HylaFAX server (hfaxd):   [  OK  ]
Restarting HylaFAX modem manager (faxgetty):   [  OK  ]
[r...@pbxserver ~]# rasterisk
Asterisk 1.6.1.20, 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.
=
Connected to Asterisk 1.6.1.20 currently running on pbxserver (pid = 4175)
Verbosity is at least 3
pbxserver*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status
ttyIAX0  192.168.3.28(D)  255.255.255.255  4570  (E) OK (1 ms)
ttyIAX1  192.168.3.28(D)  255.255.255.255  56869 (E) OK (1 ms)
ttyIAX2  192.168.3.28(D)  255.255.255.255  38332 (E) Unmonitored
ttyIAX3  192.168.3.28(D)  255.255.255.255  56565 (E) Unmonitored
4 iax2 peers [2 online, 0 offline, 2 unmonitored]
pbxserver*CLI>

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