Re: [asterisk-users] Remote Unix Connection

[Zeeshan Zakaria]

Do you use FreePBX by any chance?

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-10-16 6:38 PM, "Dan Journo"  wrote:

> Serious answer:
> Looks like a process running asterisk -r. Do you have any sort of
> AGI, cron j...
Thanks for lightning my day!

Is there any way to debug this because as far as i'm aware, there's nothing
running that command, (except for me)


-- 
_
-- Bandwidth and Colocati...
-- 
_
-- 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

Re: [asterisk-users] DMTF Mode

[Dan Journo]

Hi All,

Regarding the DTMF issue I reported where the tones werent being sent through 
the provider to the pstn phones, 
I ended up being told to switch to "inband".

However, now, asterisk is not recognising my features (*1, etc).

Any ideas?

I've checked using tcpdump, and asterisk is still part of the media path.

Thanks
Dan

-- 
_
-- 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

Re: [asterisk-users] (no subject)

[Sherwood McGowan]

On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo 
wrote:

>  Hi,
>
>
>
> Does anyone know where this is suddenly coming from?
>
>
>
> --
> _
> -- 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
>

Where what is suddenly coming from?
Cheers - The Mick
-- 
_
-- 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

Re: [asterisk-users] Remote Unix Connection

[Dan Journo]

> Serious answer:
> Looks like a process running asterisk -r.  Do you have any sort of
> AGI, cron job or perhaps a nagios check which does this?

> Not so serious answer:
> IT IS COMING FROM INSIDE OF THE HOUSE

Thanks for lightning my day!

Is there any way to debug this because as far as i'm aware, there's nothing 
running that command, (except for me)

-- 
_
-- 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

Re: [asterisk-users] Remote Unix Connection

[Mark Deneen]

On Sat, Oct 16, 2010 at 5:36 PM, Dan Journo
 wrote:
> Hi,
>
>
>
> Does anyone know where this is suddenly coming from?
>
>
>
>     -- Remote UNIX connection
>
>     -- Remote UNIX connection disconnected
>
>     -- Remote UNIX connection
>
>     -- Remote UNIX connection disconnected
>
>     -- Remote UNIX connection
>
>     -- Remote UNIX connection disconnected
>

Serious answer:
Looks like a process running asterisk -r.  Do you have any sort of
AGI, cron job or perhaps a nagios check which does this?

Not so serious answer:
IT IS COMING FROM INSIDE OF THE HOUSE

-M

-- 
_
-- 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

Re: [asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?

[Paul Belanger]

On Sat, Oct 16, 2010 at 4:59 PM, Frank Tarczynski  wrote:
> Any pointers to share?
>
chan_dahdi.conf
faxdetect=incoming

extensions.conf
exten => fax,1,Dial(DAHDI/4)

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- 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

[asterisk-users] Remote Unix Connection

[Dan Journo]

Hi,

Does anyone know where this is suddenly coming from?

-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected


Thanks
Dan

p.s. sorry about the last post. hit the mouse by mistake and it sent the email.
-- 
_
-- 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

[asterisk-users] (no subject)

[Dan Journo]

Hi,

Does anyone know where this is suddenly coming from?

-- 
_
-- 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

[asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?

[Frank Tarczynski]

 I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine.  Both are connected to a DAHDI board.  I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.

What's the best way to go about doing this?  I've looked into Fax for
Asterisk, bit I'm not sure that I want it or NVFax detection.

Any pointers to share?

-- 
_
-- 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

Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

[Shaun Ruffell]

On 10/16/10 12:47 PM, Karsten Wemheuer wrote:
> Hi,
>
> Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell:
>> On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
>>
>>> I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
>>> dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
>>> older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
>>> card. The system starts without any errors.
>>>
>>> I discovered a severe issue. The kernel panics on a very small load. The
>>> first call normally gets through. If I start the second or third call
>>> and sometimes when I terminate the first call, the system panics (Oops
>>> text on console).
>>>
>>> After solving some difficulties (the relevant part of the Oops text
>>> scrolls out of the monitor, no serial interface), I get the text via
>>> netconsole. It seems to me, that the panic occurred in oslec (function
>>> "oslec_update"). But maybe I am wrong with this. In the oslec code there
>>> is a patch to enable MMX. After switching this off, the problem
>>> disappeared. AFAIK the cpu supports mmx.
>>>
>>> Where should I address this issue to? Is it a known issue?
>>>
>>
>> Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h?
>
> No, I don't think so. (This is from memory, I currently have no access
> to the test system). But there was a patch (I think from debian
> packages, original from bug tracker (dahdi_mmx_auto.diff from
> http://bugs.digium.com/view.php?id=13500)) which enables mmx at least
> for the echo canceler oslec (I think). Disabling this patch let the
> kernel panic disappear.
>

Hmmm...I can't be certain since there are many parts coming from out of 
the tree here (besides just olsec itself), but looking at 
https://issues.asterisk.org/file_download.php?file_id=22366&type=bug 
doesn't *seem* right.  It appears that DAHDI_USE_MMX is exported and 
therefore the olsec in git://gitorious.org/dahdi-extra/dahdi-extra.git 
uses the MMX instructions, but -DCONFIG_DAHDI_MMX is only added onto 
CFLAGS_zaptel_base.o and not CFLAGS_dahdi_base.o.  Therefore, oslec most 
likely is killing the FPU registers since it believes that dahdi-base.c 
is taking care of saving and restoring them by hand.

I would recommend changing CFLAGS_zaptel_base.o to CFLAGS_dahdi_base.o, 
or hand edit include/dahdi/dahdi_config.h to make sure CONFIG_DAHDI_MMX 
is defined and see if you still get the crash.

Cheers,
Shaun

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

-- 
_
-- 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

Re: [asterisk-users] How to connect asterisk PBX to PSTN

[Sebastian]

On 10/14/2010 03:09 PM, Gopalakrishnan A.N wrote:
> Hi Joshi,
>
> To connect with PSTN line you need FXO / FXS card. FXO is used to
> connect CO line and FXS is used to connect internal station line. With
> help of FXO you can connect the outside world and with help of FXS you
> can connect normal analog phones. Inspite of normal analog phones you
> can connect SIP phones (soft phones) also.

Actually to connect PSTN lines (regular telephone lines coming from your 
telecom provider) to Asterisk you only need FXO cards. Or ATA's 
(analogue telephone adapters) - specially if your Asterisk box doesn't 
have PCI or PCI-e slots. The FXO cards can be PCI or PCI-express (or 
other flavours for more complex setups) with 1-8 (or even more) 
connectors per card for the same number of telephone lines to be 
connected to Asterisk. Digium, Sangoma, OpenVox and others make them.

You only need FXS cards if you have analogue phones which you want to 
re-use. Otherwise there are many manufacturers of SIP hardware phones - 
looking perfectly similar to regular business desk phones - so there 
isn't much point IMO to buy FSX cards unless you already have the 
analogue phones and need to re-use them.

Sebastian


Some vendors are there for
> these PSTN cards like Digium, Sangoma, Openvox.
>
> Good luck:)
>
>
>
> On Thu, Oct 14, 2010 at 7:05 PM, Jigar Joshi  > wrote:
>
> Hello community,
>
> I have successfully set up asterisk free PBX server and I am also
> able to connect to it by softphone.
>
> Now as next step I want to extend this to PSTN ,
>
> My Required scenario:
>
> I need a number which will connect outside PSTN world to my PBX and
> by applying extension particular softphone or connected normal phone
> should get connected.
>
> Which hardware I need for it.
> Also please explain a bit of dial plans.
>
> Thanks
>
>
>
> --
> _
> -- 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
>
>
>
>
> --
> Thank you  with regards,
> Gopalakrishnan A.N,
>
>

-- 
_
-- 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

Re: [asterisk-users] How to find ".gsm" audio file length or duration

[Barry Miller]

On Sat, Oct 16, 2010 at 06:46:20PM +0100, Tiago Geada wrote:
> r you would have to convert that gsm to another format first like ogg

Why on earth would you have to do that?  Did you even try doing what
I suggested?

> 
> On 16 October 2010 18:23, Barry Miller  wrote:
> 
> > On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
> > > Hi Friends,
> > >
> > > I need to find ".gsm" file length or duration.
> > >
> > > *E.g.*
> > > demo-congrats.gsm
> > >
> > > sox demo-congrats.gsm -e stat
> > >
> > > Above command is display file length in seconds. like as
> > > Length (seconds): 27.96
> > >
> > > I want to ".gsm" file length or duration in dialplan.
> >
> >Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} /
> > 1650])
> >   Verbose(Length (seconds): ${DUR})
> >
> > for asterisk >= 1.6

-- 
Barry

-- 
_
-- 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

Re: [asterisk-users] How to find ".gsm" audio file length or duration

[Tiago Geada]

r you would have to convert that gsm to another format first like ogg

On 16 October 2010 18:23, Barry Miller  wrote:

> On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
> > Hi Friends,
> >
> > I need to find ".gsm" file length or duration.
> >
> > *E.g.*
> > demo-congrats.gsm
> >
> > sox demo-congrats.gsm -e stat
> >
> > Above command is display file length in seconds. like as
> > Length (seconds): 27.96
> >
> > I want to ".gsm" file length or duration in dialplan.
>
>Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} /
> 1650])
>   Verbose(Length (seconds): ${DUR})
>
> for asterisk >= 1.6
>
> --
> Barry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

[Karsten Wemheuer]

Hi,

Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell:
> On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
> 
> > I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
> > dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
> > older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
> > card. The system starts without any errors.
> > 
> > I discovered a severe issue. The kernel panics on a very small load. The
> > first call normally gets through. If I start the second or third call
> > and sometimes when I terminate the first call, the system panics (Oops
> > text on console).
> > 
> > After solving some difficulties (the relevant part of the Oops text
> > scrolls out of the monitor, no serial interface), I get the text via
> > netconsole. It seems to me, that the panic occurred in oslec (function
> > "oslec_update"). But maybe I am wrong with this. In the oslec code there
> > is a patch to enable MMX. After switching this off, the problem
> > disappeared. AFAIK the cpu supports mmx.
> > 
> > Where should I address this issue to? Is it a known issue?
> > 
> 
> Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h?

No, I don't think so. (This is from memory, I currently have no access
to the test system). But there was a patch (I think from debian
packages, original from bug tracker (dahdi_mmx_auto.diff from
http://bugs.digium.com/view.php?id=13500)) which enables mmx at least
for the echo canceler oslec (I think). Disabling this patch let the
kernel panic disappear.

Maybe Alex suggestion points in an interesting direction. The kernel is
indeed compiled with CONFIG_PREEMPT=y.

If I have enough time, I'll try compiling the kernel with another
preemption model ("voluntary" or "no") and patch applied.

Thanks,

Karsten



-- 
_
-- 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

Re: [asterisk-users] How to find ".gsm" audio file length or duration

[Barry Miller]

On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
> Hi Friends,
> 
> I need to find ".gsm" file length or duration.
> 
> *E.g.*
> demo-congrats.gsm
> 
> sox demo-congrats.gsm -e stat
> 
> Above command is display file length in seconds. like as
> Length (seconds): 27.96
> 
> I want to ".gsm" file length or duration in dialplan.

   Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} / 1650])
   Verbose(Length (seconds): ${DUR})

for asterisk >= 1.6

-- 
Barry

-- 
_
-- 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

Re: [asterisk-users] fraud advice (Also advice on using ipbanning)

[Bryant Zimmerman]

When we designed our systems on asterisk we designed it to me multi-tenant. 
Se we use customer prefixes on all extensions. This allows us to have 
multiple customers using the same extension pools. It also reduces the hack 
foot print as hackers must know the prefix for a customer to try and brute 
force things. All passwords use 8+ characters with alfa/numeric and special 
characters. 

As I see it Asterisk does very good keeping out the hackers if you use a 
solid design in your peer and dialplans. At the least put an alpha 
character post or pre other wise you are just asking for it.  Use your head 
you can be smarter then they are.

We are looking into ipban as well. If any one has an example of ipban I 
would love to see how best to implement it.  In a 4 year period we have not 
had a breach but we do get about 10 to 15 hack attempts a week. We have 
blocking scripts that block ip's at the primary firewall but I would like 
to trigger the ipban at each switch level. Could I also use the ipban 
method to trigger the audo updates to our primary firewalls? Any advice is 
appreciated. 

 Bryant


 From: "Steve Totaro" 
Sent: Friday, October 15, 2010 11:22 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] fraud advice

On Fri, Oct 15, 2010 at 10:29 AM, Steve Edwards
 wrote:
> On Thu, 14 Oct 2010, bruce bruce wrote:
>
>> But it also sickens me at how badly Asterisk is made to not cope with
>> situations like this and worse than that is FreePBX.
>
> Kind of like blaming the gun manufacturer instead of the criminal with
> their finger on the trigger?
>
> Is there some gaping hole in Asterisk security or are you just asleep at
> the wheel?
>
> --
> Thanks in advance,
> 
-
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 
PST
> Newline  Fax: 
+1-760-731-3000
>

This is nothing new. Trunk to trunk transfers and other exploits
could be used on old school phone systems to do the same thing.

I would start with getting the current balance, if over $10k call the
FBI, call them anyways, it couldn't hurt. You want the Feds to check
things out before local police if possible.

Gather as much info as possible, along with police and FBI case
numbers and then call the carrier and see what can be done.

A friend of mine took what was supposed to be my one month rotation to
Iraq. I had too much going on to be in Iraq for a month and a half
and had taken the last rotation so it wasn't even my turn.

The phone bill came for his cell (company provided on Asia Cell) for
$4k in just a couple weeks. It turns out that he was not using the
cell and one of the cleaning people stole his SIM.

After contacting Asia Cell a few times about the matter, they credited
the whole amount back. So you never know.

As for security, I assume you need to allow these extensions to
register from outside the LAN? If not, then only allow them to
register via a LAN IP, I would do it with iptables, only allow the
provider IP through.

I am curious what your user:pass was? something like 1000:1000, I see
many systems setup like this and am surprised they haven't been hit
yet.

In the future, you could use a scheme that makes it much more secure
and also pretty easy to maintain.

The username could be the MAC and the pass could be the serial number
or asset tags if you use them.

I know there must be dozens of people reading this that have had the
same issue but are embarrassed to speak up.

(BTW Sierra Leone is in West Africa, not the Middle East.)

Thanks,
Steve T

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DAHDI, PRI and callerid

[Steve Edwards]

On Sat, 16 Oct 2010, Kent Varmedal wrote:

> I have just set up Asterisk to use an E1 line with a Digium card. And I
> can call both in and out, but my outgoing line is all ways identifying
> itself as the same number, and i can't even change it to another number
> in the same number series.
>
> Do anyone have some clue on how to fix this.

You didn't show the console log of an outgoing call, but I suspect the 
problem is with your carrier.

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

-- 
_
-- 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

[asterisk-users] DAHDI, PRI and callerid

[Kent Varmedal]

Hi,

I have just set up Asterisk to use an E1 line with a Digium card. And I
can call both in and out, but my outgoing line is all ways identifying
itself as the same number, and i can't even change it to another number
in the same number series. 

Do anyone have some clue on how to fix this. 

I'm using Asterisk 1.6.2.13, libpri 1.4.11.4 and DAHDI 2.4.0.

/etc/dahdi/system.conf:

span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-12
unused=13-15,17-31
dchan=16
echocanceller=mg2,1-12

# Global data

loadzone= no
defaultzone = no



/etc/asterisk/chan_dahdi.conf
[trunkgroups]

trunkgroup => 1,16,31
spanmap => 1,1,1

[channels]

context=incoming

usecallerid=yes
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

usecallerid=yes
callerid=asreceived

usecallingpres=yes

callgroup=1

switchtype = euroisdn
signalling = pri_cpe
group = 1

channel => 1-12




Best regards,
Kent Varmedal




-- 
_
-- 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

Re: [asterisk-users] Audiocodes firmware

[Mark Deneen]

On Thu, Oct 14, 2010 at 11:46 PM, Mark Murawski
 wrote:
> Crazy.  What do you plan on using for an ATA now?
>
> The problems I'm having are getting 500 "Server Internal Error" on just
> about every other call placed out of this mp-118.  The box has been
> installed and in use for quite some time and recently started having
> problems.  Reboots, etc don't make a difference.  I noticed it had newer
> firmware than what I had on some other boxes that had no issues whatsoever.
> I do have a 5.80 firmware I had downloaded a while back and put that on.
> Now the internal server errors are happening on 70-80% of sip->pstn calls.
> pstn->sip calls seem to be coming in just fine.
>
> Ever since I did the firmware downgrade, now my ssh sessions to the box get
> disconnected after about 30 seconds with invalid packet errors.
>
> I've had problems with earlier firmware as well... once the 5.x firmware
> started shipping on audiocodes it seemed they were just about DOA.  The web
> interface worked but nothing else worked right.  Perfectly working
> configurations on other boxes that were copied to the new boxes with new
> firmware would just fail in various ways... disconnect supervision not
> working, internal routing not working.  Finally I managed to get a hold of
> the 5.80 firmware which got rid of all those problems.
>
> Now I'm stuck again.  I have a box in service that's having problems and I
> can't get new firmware.

This doesn't sound like something which firmware would fix, especially
if it had been running fine before.  It sounds like either the mp-118
is failing or you have a damaged cable.

-M

-- 
_
-- 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

Re: [asterisk-users] MySQL and Channel Event Logging

[Sherwood McGowan]

Hey, Murph! Good to see you're still active 'round these parts!

I actually have not delved too far into CEL, I had just started looking into
it when I noticed that there wasn't anything for MySQL and so I began
searching and finally making this inquiry :)

I will take a peek at the source code for the other backends and see if I
can't hack something together to submit to the community.

Past that, once I've learned a bit more about CEL in general, I'll let you
know how I plan on using "that table" (sorry, I currently have no idea
what's in the table yet, but I'll learn).

Thanks, by the way, for your work on CEL (I remember drooling over a blog
post or two of yours that mentioned the beginnings of work on CEL), on AEL
v2, and last but not least, listening to my suggestions and whatnot
regarding AEL :)

Cheers,
Sherwood McGowan

On Wed, Oct 13, 2010 at 11:17 PM, Steve Murphy  wrote:

>
>
> On Wed, Oct 13, 2010 at 9:52 PM, Sherwood McGowan <
> sherwood.mcgo...@gmail.com> wrote:
>
>>
>>
>> On Wed, Oct 13, 2010 at 9:53 PM, Paul Belanger <
>> paul.belan...@polybeacon.com> wrote:
>>
>>> On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan
>>>  wrote:
>>> > Hey all, sorry if this has been covered, but I've not found anything
>>> after a
>>> > couple hours' worth of googling. I can see (and I'm familiar with) all
>>> the
>>> > usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find
>>> any
>>> > reference to MySQL and the new CEL logging tool other than ODBC. Is
>>> this the
>>> > only method available to use MySQL with CEL at this time?
>>> >
>>> Looking at the CEL config files, I don't see one specifically for
>>> MySQL.  I do have it up and running via ODBC, for what it's worth.
>>>
>>> --
>>> Paul Belanger | dCAP
>>>
>> Thanks mate, I appreciate the reply. That's what I've seen from looking at
>> the configs right now.
>>
>>
> When I wrote the CEL backends, I probably skipped the MySQL stuff, because
> it would be in the "addons" stuff
> for Asterisk.
>
> But, if you look at the similarities between the CEL backends, and the CDR
> backends, you'll probably
> notice that you could pump out a myseql backend with the same mods in a
> short amount of time.
>
> I would be curious to see how you plan to use that table! Have you mapped
> out your sql statements yet?
>
> murf
>
>
>
> --
> Steve Murphy
> ParseTree Corp
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to find ".gsm" audio file length or duration

[RAJNIKANT VANZA]

Hi Friends,

I need to find ".gsm" file length or duration.

*E.g.*
demo-congrats.gsm

sox demo-congrats.gsm -e stat

Above command is display file length in seconds. like as
Length (seconds): 27.96

I want to ".gsm" file length or duration in dialplan.


-- 
Best Regards,
Rajnikant Vanza
-- 
_
-- 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

Re: [asterisk-users] Audio problems on cable modem link

[Kevin Keane]

I've had to rip out VoIP in two cable-modem situations because the call quality 
was too poor.

Bandwidth isn't the main characteristic you are looking for; most Internet 
connections have plenty of that. Latency and jitter matter far more. Latency 
describes how long each packet travels from your Asterisk system to the other 
end of the IAX connection. The easiest way to measure it is with ping. Jitter 
describes how consistent the data transfer speed is. Cable modems are to some 
extent designed for burst data. This will obviously kill call quality.

You can go to http://myvoipspeed.visualware.com/ to get an idea. Run the same 
test multiple times, both with no traffic on the cable modem, and with a VoIP 
call going on. Then compare the jitter numbers. What I found was that on my 
connection (also cable modem) the average jitter was supposedly acceptable, but 
it was highly variable - in three tests with minimal other traffic all in the 
middle of the night, the jitter was 1.6, 3.5 and 4.6 milliseconds. 4.6 is in 
the borderline quality area. And if there is more traffic, quality may well go 
down further, into the poor-quality zone.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, October 15, 2010 7:53 AM
To: Asterisk Users List
Subject: [asterisk-users] Audio problems on cable modem link

We have a small office installation running over a cable modem.  (8M down, 500k 
up confirmed with numerous speed test sites)

When a single call is up, call quality is fine.  When a second call is up, 
outbound audio is immediately choppy.  We're using ulaw, and confirmed that 
traffic with 2 calls is <175kbps in/out.  (IAX connection out)

Asterisk doesn't report any dropped frames, the internet connection looks fine, 
etc.   We have a linux router in place running wondershaper that seems to be 
running fine (same as our other installations).

Can someone suggest where to look?  Could this be the ITSP?  
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to 
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users