date:20100713

Re: [asterisk-users] Recording from g729 to wav means transcoding ?

2010-07-13 Thread Gordon Henderson

On Tue, 13 Jul 2010, Paul Belanger wrote:

> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens  
> wrote:
>> I have no licenses and I want to avoid transcoding all together.
>>
> For terminating a call into Asterisk, you need g729 licenses.  It is
> that simple.

The sounds package is avalable in g729 formats and voicemail can write in 
g729 format too without transcoding from a g729 source. So what else is 
there if you're not interfacing with PSTN hardware or using meetme?

Gordon

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[asterisk-users] Asterisk core dumping on SendFax with FFA

2010-07-13 Thread Ilmars Knipšis


 Hi !

I have the same problem with the latest asterisk and FFA.
Does anybody know how to eliminate the situations when this problem occurs?

I looked in the Digium download section and found just old 
res_fax_digium version 1.2.0.

Is there any chance to get driver update?


Thanks!
--

*Ilmars*

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[asterisk-users] Silence RTP

2010-07-13 Thread kawanobe tomohito



hello
 
I found silence RTP packet from Asterisk in early dialog.
I want to know reason and how to solve.
 
RTP packet
   80 00 40 22 00 0c 74 58 06 98 eb 44 ff ff ff ff  ..@"..tX...D
0010   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0020   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0030   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0040   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0050   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0060   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0070   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0080   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
0090   ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff  
00a0   ff ff ff ff ff ff ff ff ff ff ff ff  
  
_


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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

2010-07-13 Thread bruce bruce

Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job
for a 35, square feet warehouse with environmental noise level of
slightly higher than standard but not those of industrial.

Only two speakers and done deal. Though I know that three speaker would have
been the perfect solution but 4 would cover every single little corner and
be an overkill.

-Bruce

On Tue, Jul 13, 2010 at 8:47 PM, C F  wrote:

> I agree with horns you'll usually get better coverage. I have done
> this in the past with 5 speakers for a 30k sq ft warehouse very good
> coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
> Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
> side I installed a horn every 60ft alternating facing one north and
> the other south, which ended up 3 facing one way and 2 the other. You
> can get double horn speakers which will face 2 sides. I wouldn't mount
> them on the wall specifically not so low as fork lifts and what not
> will damage them.
>
>
> On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce  wrote:
> > Well, these are horn speakers with 30 Watt which will receive 10 Watt
> only
> > from Amplifer. I am not connecting them to ceiling so maybe 10 feet off
> the
> > ground. I guess my coverage would be better???
> > Based on your calculations for for 40k sqfeet that would be 33 speakers.
> I
> > think that's way too much of an overkill.
> > thanks,
> > Bruce
> >
> > On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:
> >>
> >> In my experience using height for radius works, for example if you
> >> have a 20 ft high ceiling then the coverage for one speaker would be
> >> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
> >> has never killed anyone, but this really depends on the power of the
> >> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
> >> depending on how many speakers I put on one amplifier and the output
> >> wattage of that amplifier.
> >>
> >>
> >>
> >> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce 
> wrote:
> >> > Hi Guys,
> >> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to
> use
> >> > 2
> >> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
> feet
> >> > height. Is that enough? Is there calculator online I can use to
> >> > determine
> >> > the number of speakers needed? I guess these speakers go in chain so I
> >> > am
> >> > not sure if the full capacity of the speaker (30 watt) will be used.
> >> > I appreciate your advice.
> >> > Thanks,
> >> > Bruce
> >> > --
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> >> >
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> >
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> >
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[asterisk-users] Can't compile DAHDI - wrong kernel source

2010-07-13 Thread Thermal Wetland

I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.

The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest
version
Nothing to do

uname -a returns:
Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed
Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux

When I try to compile DAHDI it fails with:
make[2]: Leaving directory
`/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-028stab064.7 kernel
installed.

Is there a way to trick DAHDI to use the installed kernel?

Thanks for the help!

-- 
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

2010-07-13 Thread khalid touati

>'Fax for Asterisk' is a commercial application sold by Digium. This is not
their official support channel. Since you paid for the >product, why not
contact them directly about your problem?

i did get this version for free after buying a (actually several) digium
telephony card, but i realized that they're not supporting the free version
after talking and emailing them, actually i was calling Digium support for
all the past year and i can say that (for me) it was good:4/5 satisfaction,
but this time with fax, i didn't get much help, i was redirected to the
community and that why i posted. by the way is there a reliable alternative?
is for 1.6 rfax is doing good (if anyone worked with it)?
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[asterisk-users] How to pass through supported 100rel

2010-07-13 Thread kawanobe tomohito



hello
 
I want to know how to pass through 100rel header.
and I hope that asterisk PRACK to UAS.(RFC3262 behavior)
 
  
_


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Re: [asterisk-users] Unable to open pseudo device

2010-07-13 Thread Malvin Rito

Thanks for the reply. There is no folder dahdi under /dev folder. I cannot
also find /udev.d on /etc folder.

Under /dev folder I only see /dev/zap/pseudo.

Regards,
Malvin

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, July 13, 2010 9:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to open pseudo device

On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote:
> Hi List,
>
> I'm new to asterisk and currently running the newest of version. I'm
> encountering the error below when I dial my meetme conference #:
> "WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo
> device"
>
> I already tried googling this issue and found some procedure but still no
> luck on fixing it. My server does not have any digium hardware and I'm
> trying this via ztdummy.

It's likely an issue of permissions.  Check the permissions of
/dev/dahdi/pseudo versus the user your Asterisk daemon runs at.  If
necessary,
change your permission script in /etc/udev.d/ to match the ownership of the
pseudo device to the user running Asterisk.

-- 
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Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] power outage

2010-07-13 Thread C F

It has nothing to do with the D-channel, however you will never know
if the B-channels work if the D-channel is down. D-channel is what
allows the B-channels to work, and is the first place to troubleshoot.
If something is screwed up with the power the symptom you'll get is a
non working PRI, the way to check it is by means of seeing if the
D-channel synced up or not.


On Mon, Jul 12, 2010 at 2:17 AM, Justin Case
 wrote:
> What would the power have to do with the D Channel ? Isn't which channel
> used a logical setting (as opposed to physical). I am not saying your wrong
> I am just trying to understand why it happens.
>
> On Mon, Jul 12, 2010 at 7:56 AM, C F  wrote:
>>
>> I have found that sometimes shutting down the machine waiting a full
>> minute while the power cable is unplugged then restarting can fix such
>> problems if it's power related.
>>
>> On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis  wrote:
>> > I have a TE205P that has been working fine for 2 years.
>> > power outage yesterday took out my everything for over an hour.
>> >
>> > Everything has come back up except the PRI. My provider has checked it
>> > to the box
>> > and says everything looks good on their end.
>> >
>> > I get this message:
>> > [Jul  9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels available!
>> > Using Primary channel 24 as D-channel anyway!
>> >
>> > ztcfg -vvv
>> >
>> > Zaptel Version: 1.4.12.1
>> > Echo Canceller: MG2
>> > Configuration
>> > ==
>> >
>> > SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
>> >
>> > Channel map:
>> >
>> > Channel 18: Clear channel (Default) (Slaves: 18)
>> > Channel 19: Clear channel (Default) (Slaves: 19)
>> > Channel 20: Clear channel (Default) (Slaves: 20)
>> > Channel 21: Clear channel (Default) (Slaves: 21)
>> > Channel 22: Clear channel (Default) (Slaves: 22)
>> > Channel 23: Clear channel (Default) (Slaves: 23)
>> > Channel 24: D-channel (Default) (Slaves: 24)
>> >
>> > 7 channels to configure.
>> >
>> > and show status gives me condition RED of course.
>> >
>> > How do I find out whats wrong here?
>> >
>> > Jerry
>> >
>> > --
>> > _
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>> >
>>
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>
>
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

2010-07-13 Thread C F

I agree with horns you'll usually get better coverage. I have done
this in the past with 5 speakers for a 30k sq ft warehouse very good
coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
side I installed a horn every 60ft alternating facing one north and
the other south, which ended up 3 facing one way and 2 the other. You
can get double horn speakers which will face 2 sides. I wouldn't mount
them on the wall specifically not so low as fork lifts and what not
will damage them.


On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce  wrote:
> Well, these are horn speakers with 30 Watt which will receive 10 Watt only
> from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the
> ground. I guess my coverage would be better???
> Based on your calculations for for 40k sqfeet that would be 33 speakers. I
> think that's way too much of an overkill.
> thanks,
> Bruce
>
> On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:
>>
>> In my experience using height for radius works, for example if you
>> have a 20 ft high ceiling then the coverage for one speaker would be
>> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
>> has never killed anyone, but this really depends on the power of the
>> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
>> depending on how many speakers I put on one amplifier and the output
>> wattage of that amplifier.
>>
>>
>>
>> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce  wrote:
>> > Hi Guys,
>> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use
>> > 2
>> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet
>> > height. Is that enough? Is there calculator online I can use to
>> > determine
>> > the number of speakers needed? I guess these speakers go in chain so I
>> > am
>> > not sure if the full capacity of the speaker (30 watt) will be used.
>> > I appreciate your advice.
>> > Thanks,
>> > Bruce
>> > --
>> > _
>> > -- 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 --
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>>               http://www.asterisk.org/hello
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>
>
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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

2010-07-13 Thread Paul Belanger

On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens  wrote:
> I have no licenses and I want to avoid transcoding all together.
>
For terminating a call into Asterisk, you need g729 licenses.  It is
that simple.

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

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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

2010-07-13 Thread Steve Totaro

On Tue, Jul 13, 2010 at 4:43 PM, Philipp von Klitzing <
klitz...@pool.informatik.rwth-aachen.de> wrote:

> Hi!
>
> > 'Fax for Asterisk' is a commercial application sold by Digium. This is
> > not their official support channel. Since you paid for the product, why
> > not contact them directly about your problem?
>
> Maybe because having to deal with Digium support is an ... uncomfortable
> experience that I've made twice. It really feels like a "support
> avoidance" system, and you are an unwelcome guest that should please
> leave as soon as possible.
>
> I really hope that Digium takes steps to a) make their behaviour less
> bureaucratic when establishing a support call (try to do that on behalf
> of a clue-less customer of yours and you know what I mean). Also my
> feeling is that b) the level of competence (or familiarity with the
> product in question when it comes to software prodcuts) could use some
> improvement.
>
> If you take a look at the Digium web forum for Skype for Asterisk, for
> example, you will see that any kind of "half-official" answer or helpful
> reaction from Digium's side is by now non-existent - most probably due to
> internal policies. This just doesn't feel right, and many other companies
> haven proven that I can be done differently.
>
> Philipp
>
>
I couldn't have said it better myself.   http://tinyurl.com/3x4yt9k
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

2010-07-13 Thread Philipp von Klitzing

Hi!

> 'Fax for Asterisk' is a commercial application sold by Digium. This is
> not their official support channel. Since you paid for the product, why
> not contact them directly about your problem? 

Maybe because having to deal with Digium support is an ... uncomfortable 
experience that I've made twice. It really feels like a "support 
avoidance" system, and you are an unwelcome guest that should please 
leave as soon as possible.

I really hope that Digium takes steps to a) make their behaviour less 
bureaucratic when establishing a support call (try to do that on behalf 
of a clue-less customer of yours and you know what I mean). Also my 
feeling is that b) the level of competence (or familiarity with the 
product in question when it comes to software prodcuts) could use some 
improvement.

If you take a look at the Digium web forum for Skype for Asterisk, for 
example, you will see that any kind of "half-official" answer or helpful 
reaction from Digium's side is by now non-existent - most probably due to 
internal policies. This just doesn't feel right, and many other companies 
haven proven that I can be done differently.

Philipp


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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Warren Selby

On Tue, Jul 13, 2010 at 2:36 PM, Steve Edwards wrote:

> Good catch. Looks like a bug to me.
>
>
I'll create an issue on the tracker later today.

> Just out of curiosity, why is the time the dialplan is reloaded of
> interest?
>  
>

I hadn't thought about it, but I suppose if it's evaluating at reload and
not runtime, this would indeed be the case.  I'm hoping to get the various
date-related values at runtime, not the reload time.  Hmmm...it looks like
I'll have to re-evaluate what I'm trying to do...

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

2010-07-13 Thread Tim Nelson

- "khalid touati"  wrote: 
> so nobody seems to like dealing with fax!! 
> 
> 

'Fax for Asterisk' is a commercial application sold by Digium. This is not 
their official support channel. Since you paid for the product, why not contact 
them directly about your problem? 

And yes, to answer your question, not many people like dealing with fax. Some 
of us however are so "lucky" that we get to deal with fax every day. :-) 

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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!

2010-07-13 Thread khalid touati

so nobody seems to like dealing with fax!!

2010/7/12 khalid touati 

> Hi Guys,
> i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
> and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
> i'm having is that i'm able to receive faxes from a website (that offer this
> service) but not able to receive from a regular fax machine (that is working
> perfect).
>
> [fax-rx]
>
> exten => receive,1,NoOp( FAX RECEIVE ) exten =>
> receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ]) exten =>
> receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
>
> exten =>
> receive,n,Set(FAXFILE=${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})}.tif)
>
> exten =>
> receive,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})})
>
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
>
> exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
>
> exten => receive,n,NoOp( SETTING FAXOPT ) exten =>
> receive,n,Set(FAXOPT(ecm)=yes) exten => receive,n,Set(FAXOPT(headerinfo)=MY
> FAXBACK RX) exten => receive,n,Set(FAXOPT(localstationid)=15184893772)
>
> exten => receive,n,Set(FAXOPT(maxrate)=14400)
>
> exten => receive,n,Set(FAXOPT(minrate)=2400)
>
> exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten =>
> receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten =>
> receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten =>
> receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten =>
> receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten =>
> receive,n,NoOp( RECEIVING FAX : ${FAXFILE} ) exten =>
> receive,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE})
>
> exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}"
> --cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name
> "Sir/Madam"')
>
>
>
> a previous debugging showed:
>
>
> *- for a fax from myfax.com that was received successfully:*
>
> pbx1*CLI>
>
>> Channel 'DAHDI/1-1' fax session '53', [ 034.021683 ], channel
> sent 59 frames (1180 ms) of energy.
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53', [ 040.489601 ],
> STAT_EVT_HW_CLOSE  st: WT_HW_CLSrt: WCLSNCLS
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53', [ 040.489798 ],
> STAT_SES_COMPLETE
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '53' is complete, result: 'SUCCESS'
> (FAX_SUCCESS), error: 'NO_ERROR', pages: 2, resolution: '204x196', transfer
> rate: '14400', remoteSID: 'FAX'
>
> pbx1*CLI>
>
> -- Executing [rece...@fax-rx:21] System("DAHDI/1-1",
> "/usr/local/bin/fax2mail -p -f
> "/var/spool/asterisk/fax/2010-05-18_03:59:42_" --cid-number  --cid-name ""
> --dest-name "Sir/Madam"") in new stack
>
>
>
> pbx1*CLI>
>
>   == Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
>
>
>
> pbx1*CLI>
>
> -- Hungup 'DAHDI/1-1'
>
>
>
> -* for a fax from regular machine that failed:*
>
> *
> *
>
> pbx1*CLI>
>
>> Channel 'DAHDI/1-1' fax session '54', [ 032.782251 ], channel
> sent 3 frames (60 ms) of energy.
>
>
>
> pbx1*CLI>
>
> -- Channel 0/1, span 1 got hangup request, cause 16
>
>
>
> pbx1*CLI>
>
> [May 17 19:02:41] NOTICE[1316]: res_fax.c:993 generic_fax_exec: Channel
> 'DAHDI/1-1' did not return a frame; probably hung up.
>
>
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54', [ 038.131701 ],
> STAT_EVT_HW_CLOSE  st: WT_HW_CLSrt: WCLSNCLS
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54', [ 038.131879 ],
> STAT_SES_COMPLETE
>
> pbx1*CLI>
>
> -- Channel 'DAHDI/1-1' fax session '54' is complete, result: 'SUCCESS'
> (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x196', transfer
> rate: '14400', remoteSID: '518 489 3772'
>
>
>
> pbx1*CLI>
>
>   == Spawn extension (fax-rx, receive, 20) exited non-zero on 'DAHDI/1-1'
>
>
>
> pbx1*CLI>
>
> -- Hungup 'DAHDI/1-1'
>
>
> I would really appreciate any help! thanks!
>
> --
> Abdullah
>



-- 
Abdullah
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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Steve Edwards

On Tue, 13 Jul 2010, Warren Selby wrote:

> YEAR = ${STRFTIME(${EPOCH},,%Y)}

On Tue, 13 Jul 2010, Danny Nicholas wrote:

> YEAR = ${STRFTIME(${EPOCH}||%Y)}

Good catch. Looks like a bug to me.

Not that anybody cares, but the 2 statements exhibit the same bug in 1.2.

Just out of curiosity, why is the time the dialplan is reloaded of 
interest?

-- 
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] OT: fail2ban, spam and mail servers

2010-07-13 Thread Hans Witvliet

On Tue, 2010-07-13 at 06:53 -0400, cov...@ccs.covici.com wrote:
> What you can do -- I don't know about nomad, but can you make them use
> authentication?
> 
> Randy R  wrote:
> 
> > On Tue, Jul 13, 2010 at 12:29 PM,   wrote:
> > > What I do, is only open port 25 to the list of ips of the spam filtering
> > > service -- I use an iptables script called rc.firewall which I found
> > > several years ago which works well and has a nice syntax for this and I
> > > get no direct spam, I get some which gets by the filters.
> > 
> > Hi John,
> > 
> > I'd like to do that, but there are nomad users who might be anywhere
> > in the world. True maybe I could ask them to use port 587 and then
> > allow ONLY the service IPs access to port 25.
> > 
> > /r
> > 

Just wondering,
Most spammers or cases of joe-jobs originate from an other URL then they
claim to come from. 

Can this not be dealt with using certificates?
Something like for nomads, only accepting signed messages...

hw

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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Warren Selby

On Tue, Jul 13, 2010 at 1:49 PM, Danny Nicholas  wrote:

>  --
>
> Since you never know when you’ll need this, I slapped the code into my
> 1.4.30.
>
> Here is the “corrected” code that works
>
> YEAR = ${STRFTIME(${EPOCH}||%Y)}
>
> MONTH = ${STRFTIME(${EPOCH}||%m)}
>
> DAY = ${STRFTIME(${EPOCH}||%d)}
>
> TIMESTAMP = ${STRFTIME(${EPOCH}||%Y%m%d-%H%M%S)}
>
>
Wow, I'm surprised that worked, but it did!  Thanks very much!

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Danny Nicholas

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, July 13, 2010 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] STRFTIME function declared in globals context

 

On Tue, Jul 13, 2010 at 1:15 PM, Tilghman Lesher  wrote:

When you load the dialplan, do you see the global variables getting set?
That would at least tell you whether the problem lies at the point where the
values are loaded into memory, or later, at evaluation time.




  == Setting global variable 'YEAR' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'MONTH' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'DAY' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'TIMESTAMP' to 'Tue Jul 13 13:36:55 2010'

So apparently they're loaded into memory when the dialplan is reloaded?

-- 
Thanks,
--Warren Selby
http://www.selbytech.com

 

--

Since you never know when you'll need this, I slapped the code into my
1.4.30.  

Here is the "corrected" code that works

YEAR = ${STRFTIME(${EPOCH}||%Y)}

MONTH = ${STRFTIME(${EPOCH}||%m)}

DAY = ${STRFTIME(${EPOCH}||%d)}

TIMESTAMP = ${STRFTIME(${EPOCH}||%Y%m%d-%H%M%S)}

 

 

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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Warren Selby

On Tue, Jul 13, 2010 at 1:15 PM, Tilghman Lesher  wrote:

> When you load the dialplan, do you see the global variables getting set?
> That would at least tell you whether the problem lies at the point where
> the
> values are loaded into memory, or later, at evaluation time.
>
>
  == Setting global variable 'YEAR' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'MONTH' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'DAY' to 'Tue Jul 13 13:36:55 2010'
  == Setting global variable 'TIMESTAMP' to 'Tue Jul 13 13:36:55 2010'

So apparently they're loaded into memory when the dialplan is reloaded?

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Tilghman Lesher

On Tuesday 13 July 2010 11:30:44 Warren Selby wrote:
> I'm trying to declare a few date-related global variables to ease my
> dialplan.  When I declare the following in the [globals] context of
> extensions.conf, I get unexpected results:
>
> YEAR = ${STRFTIME(${EPOCH},,%Y)}
> MONTH = ${STRFTIME(${EPOCH},,%m)}
> DAY = ${STRFTIME(${EPOCH},,%d)}
> TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}

When you load the dialplan, do you see the global variables getting set?
That would at least tell you whether the problem lies at the point where the
values are loaded into memory, or later, at evaluation time.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] asterisk un-registering from provider

2010-07-13 Thread Eddie Mikell

All:

Starting switching over my phone lines.

I got phone line 1 switched.  Everyone working.

I switched the second phone line, and it worked about an hour, then I 
started getting errors from the cli saying the server could not register 
with the providing.  I restarted the system, and it worked ok for about 
30 minutes, and then started giving he same errors.

The error is
[Jul 13 11:21:14] NOTICE[27331]: chan_sip.c:10169 sip_reg_timeout:-- 
Registration for '4342201...@ia.ntelos.net' timed out, trying again 
(Attempt #19)
 > doing dnsmgr_lookup for 'ia.ntelos.net'

It keeps doing this until I restart asterisk.

No, the password hasn't changed - the system works fine for anywhere 
from 5 minutes to 30 minutes, but again, I suspect it depends on the 
call load.

I suspect it has something to do with the call load, but there really 
wasn't that many calls in progress - maybe 5 at the time of the 
failure.  I have since switched the last phone back to the old system 
(it handles the volume of our client calls).

Anyone else experience this?  Where should I start looking - at my 
server, or at the provider?

Thanks,

Eddie

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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Barry Miller

On Tue, Jul 13, 2010 at 01:07:34PM -0400, Barry Miller wrote:
> 
> Try adding "preload => func_strings.so" to modules.conf

Ah, sorry.  I just saw your earlier response that said you're on 1.4 -
I was remembering that after I migrated from 1.4 -> 1.6, I had to preload
func_db.so so that I could use the DB function in [globals].

-- 
Barry

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Re: [asterisk-users] Delay between answer and pickup ?

2010-07-13 Thread Olivier

2010/7/11 Julian Lyndon-Smith 

> Anyone got a clue  ? (he asks in desperation!)
>
> Julian
>
> On 9 July 2010 17:48, Julian Lyndon-Smith  wrote:
> > We are having a situation on our dialler here where our agents are
> > claiming that when they receive a call because it has been answered,
> > it seems as if the call had been answered several seconds earlier -
> > IOW, they are hearing "hello ? Hello ?" and often hear the phone being
> > put down as an initial part of the call.
> >
> > We have verified this by checking the voice recordings.
> >
> > Yet, the logs of asterisk don't show this discrepancy.
> >
> > We are using a local channel to dial a landline through a sip
> > provider.
>

Could it be possible that the firewall is waiting for outbound RTP media,
before letting inbound RTP come in ?


> When the call is answered, the agent's phone is then
> > dialled.
> >
> > the logs go something like this
> >
> >
> > [Jul  9 13:29:26] VERBOSE[23396] logger.c: [Jul  9 13:29:26] --
> > SIP/provider-0001ed6e is making progress passing it to
> > Local/somenum...@dialleroutbound-4c93,2
> > [Jul  9 13:29:44] VERBOSE[23396] logger.c: [Jul  9 13:29:44] --
> > SIP/provider-0001ed6e answered
> > Local/01577864...@dialleroutbound-4c93,2
>

What would rtp debug show ?


> > ..
> >
> > [Jul  9 13:29:45] VERBOSE[23416] logger.c: [Jul  9 13:29:45] --
> > Executing [*00...@diallerconnected:2]
> > Dial("Local/somenum...@dialleroutbound-4c93,1",
> > "SIP/*0086*|5|iA(autoanswer)") in new stack
> > [Jul  9 13:29:45] VERBOSE[23416] logger.c: [Jul  9 13:29:45] --
> > Local/somenum...@dialleroutbound-4c93,1 requested special control 20,
> > passing it to SIP/*0086*-0001ed73
> > [Jul  9 13:29:46] VERBOSE[23416] logger.c: [Jul  9 13:29:46] --
> > Local/somenum...@dialleroutbound-4c93,1 requested special control 20,
> > passing it to SIP/*0086*-0001ed73
> > [Jul  9 13:29:46] VERBOSE[23416] logger.c: [Jul  9 13:29:46] --
> > Local/somenum...@dialleroutbound-4c93,1 requested special control 20,
> > passing it to SIP/*0086*-0001ed73
> > [Jul  9 13:29:46] VERBOSE[23416] logger.c: [Jul  9 13:29:46] --
> > SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1
> >
> > ..
> >
> > as you can see, the call is answered at 13:29:44 and the agent gets
> > called (auto-answer phones) at 13:29:46, yes if you listen to the call
> > recording, there is a 6 second gap between the person saying "hello"
> > and the agent being connected.
> >
> > Is it possible that the call was answered 5 seconds *before* I get
> > notification of the answer ? i.e. is the provider taking too long
> > notifying me of the answer ?
> >
> > Julian
> >
>
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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Barry Miller

On Tue, Jul 13, 2010 at 11:30:44AM -0500, Warren Selby wrote:
> I'm trying to declare a few date-related global variables to ease my
> dialplan.  When I declare the following in the [globals] context of
> extensions.conf, I get unexpected results:
> 
> YEAR = ${STRFTIME(${EPOCH},,%Y)}
> MONTH = ${STRFTIME(${EPOCH},,%m)}
> DAY = ${STRFTIME(${EPOCH},,%d)}
> TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
> 
> If I evaluate these variables in the dialplan later, using
> 
> exten => ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})
> 
> My output is as follows:
> 
> -- Executing [7...@phones:3] Verbose("SIP/2625-d5f0", "Tue Jul 13
> 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
> Jul 13 11:08:42 2010") in new stack
> Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
> 2010 - Tue Jul 13 11:08:42 2010
> 
> However, the following line:
> 
> exten => ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
> ${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
> ${STRFTIME(${EPOCH},,%d)})
> 
> evaluates with what I expect:
> 
>     -- Executing [7...@phones:4] Verbose("SIP/2625-d5f0",
> "20100713-110853 - 2010 - 07 - 13") in new stack
> 20100713-110853 - 2010 - 07 - 13
> 
> Is what I'm trying to do possible?  It seems like it's at least recognizing
> that I'm trying to grab a date, but it's not taking the date format
> parameters that I want.

Try adding "preload => func_strings.so" to modules.conf

-- 
Barry

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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Warren Selby

On Tue, Jul 13, 2010 at 11:47 AM, Danny Nicholas  wrote:

>
>  You don’t state which version you are on (These things change from 1.2 to
> 1.4 to 1.6/8), but that being said, you would probably more likely to
> succeed doing Set(GLOBAL) in an isolated context instead of using the
> [global] context for this bit of voodoo.  Looking forward to a better answer
> (folks like to correct my shots across the bow).
>
>
Sorry about that.  I'm on version 1.4.33.1.  I'm wanting to use these
variables in multiple contexts throughout the dialplan, and also as
arguments to some AGI scripts, etc.  I'd rather they be set in the [globals]
context if at all possible, that way I'm not limited to where I can use
them.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Danny Nicholas

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, July 13, 2010 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] STRFTIME function declared in globals context

I'm trying to declare a few date-related global variables to ease my
dialplan.  When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:

YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}

If I evaluate these variables in the dialplan later, using 

exten => ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})

My output is as follows:

-- Executing [7...@phones:3] Verbose("SIP/2625-d5f0", "Tue Jul 13
11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
Jul 13 11:08:42 2010") in new stack
Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
2010 - Tue Jul 13 11:08:42 2010

However, the following line:

exten => ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})

evaluates with what I expect:

-- Executing [7...@phones:4] Verbose("SIP/2625-d5f0",
"20100713-110853 - 2010 - 07 - 13") in new stack
20100713-110853 - 2010 - 07 - 13

Is what I'm trying to do possible?  It seems like it's at least recognizing
that I'm trying to grab a date, but it's not taking the date format
parameters that I want.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com

--

You don't state which version you are on (These things change from 1.2 to
1.4 to 1.6/8), but that being said, you would probably more likely to
succeed doing Set(GLOBAL) in an isolated context instead of using the
[global] context for this bit of voodoo.  Looking forward to a better answer
(folks like to correct my shots across the bow).

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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-13 Thread covici

cov...@ccs.covici.com wrote:

> Dan Austin  wrote:
> 
> > Manmohan wrote:
> > 
> > > My Web-MeetMe_v4.0.1, i followed the instructions in the 
> > > README File in the same package.
> > Good.  There are other instruction packages, but since I wrote
> > the README it is the one I am most familiar with.
> > 
> > >> Are you using RealTime enabled app_meetme or app_cbmysql 
> > >> from the WMM package? > 
> > > i didnt get this actually what do i need to check here? Please 
> > > dont mind but m not so good in opensource world. I try to read and
> > > understand and on trial n error basis try  to implement things. 
> > > Though had very much interest in learning things.
> > Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
> > was in a separate Asterisk application (app_cbmysql).  With version 4 of
> > WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
> > application.
> > 
> > The README in 4.0.1 lists the steps to setup RealTime (database) support
> > for Asterisk and MeetMe.  This narrows down the possible problems, since
> > we do not need to consider app_cbmysql.
> > 
> > Did you install Asterisk from a package with yum, or did you compile it
> > yourself?  
> > 
> > Dan
> 
> I am getting this error without webmeetme at all, after upgrading to
> svn-275706 from an earlier version 262801.  Its a certain argument of
> meetme which I have not trafcked down yet which is causing this.

OK, if the argument to meetme is conference number,TcMsrm it does not
crash, but if it is , cMs then it dies -- asterisk
dies.  Is this enough for someone to figure out?

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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[asterisk-users] STRFTIME function declared in globals context

2010-07-13 Thread Warren Selby

I'm trying to declare a few date-related global variables to ease my
dialplan.  When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:

YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}

If I evaluate these variables in the dialplan later, using

exten => ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})

My output is as follows:

-- Executing [7...@phones:3] Verbose("SIP/2625-d5f0", "Tue Jul 13
11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
Jul 13 11:08:42 2010") in new stack
Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
2010 - Tue Jul 13 11:08:42 2010

However, the following line:

exten => ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})

evaluates with what I expect:

-- Executing [7...@phones:4] Verbose("SIP/2625-d5f0",
"20100713-110853 - 2010 - 07 - 13") in new stack
20100713-110853 - 2010 - 07 - 13

Is what I'm trying to do possible?  It seems like it's at least recognizing
that I'm trying to grab a date, but it's not taking the date format
parameters that I want.


-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-07-13 Thread covici

Dan Austin  wrote:

> Manmohan wrote:
> 
> > My Web-MeetMe_v4.0.1, i followed the instructions in the 
> > README File in the same package.
> Good.  There are other instruction packages, but since I wrote
> the README it is the one I am most familiar with.
> 
> >> Are you using RealTime enabled app_meetme or app_cbmysql 
> >> from the WMM package? > 
> > i didnt get this actually what do i need to check here? Please 
> > dont mind but m not so good in opensource world. I try to read and
> > understand and on trial n error basis try  to implement things. 
> > Though had very much interest in learning things.
> Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk
> was in a separate Asterisk application (app_cbmysql).  With version 4 of
> WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe
> application.
> 
> The README in 4.0.1 lists the steps to setup RealTime (database) support
> for Asterisk and MeetMe.  This narrows down the possible problems, since
> we do not need to consider app_cbmysql.
> 
> Did you install Asterisk from a package with yum, or did you compile it
> yourself?  
> 
> Dan

I am getting this error without webmeetme at all, after upgrading to
svn-275706 from an earlier version 262801.  Its a certain argument of
meetme which I have not trafcked down yet which is causing this.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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[asterisk-users] How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?

2010-07-13 Thread bruce bruce

Hi Everyone,

I have done "yum install speex libspeex-devel speex-devel" and it was
succesful on CentOS. I then tried "yum install asterisk16 asterisk16-addons
asterisk16-configs" but "core show translation" doesn't show speex loaded.
Is there a way to or an option that I can append to the asterisk install to
make sure it compiles with speex in mind?

Thanks,
Bruce
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Re: [asterisk-users] MyFuel Express FO - Shortcomings PLEASE DELETE THREAD

2010-07-13 Thread Kevin P. Fleming

On 07/13/2010 08:27 AM, Alphonse Ogulla wrote:
> My sincere apologies for inadvertently sending this mail note to this
> list which happens to be the first entry in my address book.
> 
> If you are a list administrator, kindly delete this thread from the list.
> 
> My apologies once again and please do not reply.

Threads cannot be deleted from the list; once messages are posted, they
appear in the archives (of which there are many) and are delivered to
thousands of subscribers. Sorry.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

On Tue, Jul 13, 2010 at 2:45 PM, Gordon Henderson
 wrote:
> Good luck!

A few have written me off list (thanks) so I thought I'd close out my
own "thoughts" on this. It's been about two hours and it does look
like things are working great. I removed the huge number of
CONNECT...REJECT statements in sendmail (not needed since the port
isn't there any more). I put the authorized IP list in a pf table and
all that is working just fine.

Yes, many people over the past few years have complained they weren't
able to send mail and were told to change the port to 587, which we
opened last year. Our situation is unusual as the people out on the
road vary a lot, and some of them use a VPN to access the customer
EXCHANGE directly. That can connect directly to our box, so it causes
no problems for them.

So unless we hear from stragglers on port 25, this is looking very
good. I don't know how many IP can be put in a pf table but it was
obvious that there would be tens of thousands in a very small number
of days.

/r

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Re: [asterisk-users] MyFuel Express FO - Shortcomings PLEASE DELETE THREAD

2010-07-13 Thread Alphonse Ogulla

My sincere apologies for inadvertently sending this mail note to this list
which happens to be the first entry in my address book.

If you are a list administrator, kindly delete this thread from the list.

My apologies once again and please do not reply.

Rgds,
Alphonse

On Tue, Jul 13, 2010 at 4:04 PM, Steve Howes wrote:

> Did you mean to send this to a mailing list?..
>
>
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Re: [asterisk-users] Unable to open pseudo device

2010-07-13 Thread Tilghman Lesher

On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote:
> Hi List,
>
> I'm new to asterisk and currently running the newest of version. I'm
> encountering the error below when I dial my meetme conference #:
> "WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo
> device"
>
> I already tried googling this issue and found some procedure but still no
> luck on fixing it. My server does not have any digium hardware and I'm
> trying this via ztdummy.

It's likely an issue of permissions.  Check the permissions of
/dev/dahdi/pseudo versus the user your Asterisk daemon runs at.  If necessary,
change your permission script in /etc/udev.d/ to match the ownership of the
pseudo device to the user running Asterisk.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] MyFuel Express FO - Shortcomings

2010-07-13 Thread Steve Howes

Did you mean to send this to a mailing list?..

S

On 13 Jul 2010, at 13:33, Alphonse Ogulla wrote:

> Re-sent copying UNON and Expand Technologies. Apologies for the omission.
> 
> Rgds,
> Alphonse
> 
> On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla  wrote:
> Dear Esther,
> 
> The foregoing mail notes sent to Expand Technology refer, in view that you 
> were not copied in the initial correspondence.
> 
> We were hoping that Expand Technology would make a comeback today with a 
> course of action to resolve certain shortcomings flagged in MyFuel Express 
> but unfortunately this has not been the case. 
> 
> Kindly contact Expand Technology and make it clear that we need the five 
> critical elements resolved in the next 7 days to enable us progress with the 
> system upgrade as planned.
> 
> An improved version of MyFuel Express should be released speedily without 
> unnecessary mention to the ToR, more so because we are requesting minute 
> revision to code in beta stage and not significant modification or new 
> functionality in a final product.
> 
> Please follow up on our behalf and revert with firmed up dates when we can 
> get a new version of MyFuel Express without the listed drawbacks.
> 
> Best Regards,
> Alphonse Ogulla
> 
> 
> -- Forwarded message --
> From: Jacques de Gersigny 
> Date: Mon, Jul 12, 2010 at 8:34 PM
> Subject: Re: MyFuel Express FO - Shortcomings
> To: Alphonse Ogulla 
> Cc: Lovena Modelly , James Gathoga 
> , Simon Beamish , 
> Sanjita Sehmi , Sheila Cardovillis 
> 
> 
> 
> Hi Alphonse, Simon,
> I'm in a Business trip and I will get back on monday next.
> Rgds,
> JDG
> 
> 
> On 8 July 2010 16:05, Alphonse Ogulla  wrote:
> Dear Jacques,
> We tried getting you on phone in the office at noon (Kenya time) but 
> unfortunately you had stepped out for lunch. We however managed to get hold 
> of Lovena and briefly deliberated the critical items in the ensuing email. In 
> principle, we agreed to address these drawbacks in the following manner:
> 
> 1) Expand Technologies to resolve items 1a (card printing) and 1c (bank card 
> & cheque payment currency) without further deliberations..
> 2) Refer to the final signed TOR for items 1b (card transfer) and 1d (FO 
> direct topup). I shall get the final TOR from Easther Wanjoga of Kenya Shell 
> Ltd.
> 3) Lastly, Expand Technologies to check if the chip card has "sufficient" 
> space to store the expiry date in order to implement item 1e (card validity).
> 
> I'm also made to understand that you called Simon Beamish and discussed 
> further the items listed above. Kindly look into these issues keenly and 
> revert with a proposal on headway latest by Monday 12th July AM. Please 
> remember to copy Shell in your rejoinder.
> 
> Looking forward to hearing from you soon.
> 
> Best Regards,
> Alphonse Ogulla
> Tel: +254 20 7621510
> Mobile: +254 723 465172
> 
> 
> On Mon, Jul 5, 2010 at 12:15 PM, Alphonse Ogulla  wrote:
> Dear Jacques et alia,
> 
> We have identified certain shortcomings in MyFuel Express Front Office (FO) 
> software that we need rectified as soon as possible and in-time for the go 
> live scheduled for next month. The critical elements should be given 
> uttermost priority as it is impossible to commence printing and issuing of 
> cards with the listed drawbacks still in place.
> 
> 1) CRITICAL ELEMENTS
> 
> a) Card Printing and Personalisation
> Increase padding on the left margin so that the card holder name, description 
> and vehicle registration do not print on the UN logo.
> 
> b) Card Transfer
> The required functionality should be transfer of card value on the e-purse 
> and not transfer of the card-holder particulars to another card as is 
> currently the case.
> 
> c) Local Epurse Remote Top-up (Bank card & cheque payment)
> Currently only the cash top-up function has the option of selecting the 
> paying currency. A similar option is required for the bank card and cheque 
> top-up since many clients run US$ transactions on their credit/debit bank 
> cards. Similarly, US$ account holders have cheque books for US$ transactions 
> only.
> 
> d) Front Office Direct Top-up
> The FO lacks direct top-up capability whereas there is a card reader/writer 
> directly connected to the FO. A work-around has been implemented by 
> connecting a hand held POS to the Ethernet network to download and effect the 
> actual top-up on the card. This two step procedure is time consuming and 
> shall drastically slow down the top-up process at the station.
> 
> e) Client Management - Card Validity
> Provide entry for expiry date i.e. dd/mm/ instead of number of years 
> since client contracts expire on specific dates and not at the end of the 
> year.
> 
> 2) IMPROVEMENTS
> 
> a) User Management -> User Rights
> Group the list of functions into various roles to ease creation of a new 
> user. We acknowledge that new users shall not be created frequently but 
> currently one has to chose from over 200 different functio

Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Gordon Henderson

On Tue, 13 Jul 2010, Randy R wrote:

> Hi Gordon,
>
> On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson
>  wrote:
>
>> Technically/pedantically, users ought to be connecting to port 587 to submit
>> their email anyway, with port 25 being reserved for MTA to MTA
>> communications, so block 25 for everyone but the MX relaying host and insist
>> your users connect on port 587 to relay their outgoing email (with
>> smtp-auth)
>
> Yes. The only thing that is delicate here is the "insist" part, but
> they'll get over it. "Users" are customers.

Indeed - and with another 'hat' on, I run an ISP business, providing email 
and web hosting facilities to clients - and facing exactly the same 
issues. It's been (being) a struggle to get people to change their 
settings, but we're slowly getting there.

Because I have multiple servers, I can run both in parallel and are giving 
groups of customers cut-off dates for final migration based on the servers 
their using to relay outbound email...

>> I'd assume that most MTAs listen on 587 these days (as well as 25) - it's
>> been in the standards for quite a number of years now. (Introduced in 1998
>> in RFC2476)
>
> Yes, if you have that port open. (we do)
>
>> And I don't know about where you are, but where I am (UK) some ISPs are now
>> blocking outbound SMTP connections on port 25, or force-proxying them via
>> their own email servers, making the use of port 587 almost mandatory -
>> BTretail and Orange, and I think AOL do, but there's probably others.
>> However it's only a matter of time before they catch up and as soon as the
>> spammers start to use that port, the ISPs will block them too.
>
> Yes, more and more providers do this.
>
> So (even before I read your message) I decided to limit port 25 access
> to the restricted IP set we know about. This will be an interesting 48
> hours or so while we see if the users are still using port 25 :-)

Good luck!

Gordon

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Re: [asterisk-users] MyFuel Express FO - Shortcomings

2010-07-13 Thread Alphonse Ogulla

Re-sent copying UNON and Expand Technologies. Apologies for the omission.

Rgds,
Alphonse

On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla  wrote:

> Dear Esther,
>
> The foregoing mail notes sent to Expand Technology refer, in view that you
> were not copied in the initial correspondence.
>
> We were hoping that Expand Technology would make a comeback today with a
> course of action to resolve certain shortcomings flagged in MyFuel Express
> but unfortunately this has not been the case.
>
> Kindly contact Expand Technology and make it clear that we need the five
> critical elements resolved in the next 7 days to enable us progress with the
> system upgrade as planned.
>
> An improved version of MyFuel Express should be released speedily without
> unnecessary mention to the ToR, more so because we are requesting minute
> revision to code in beta stage and not significant modification or new
> functionality in a final product.
>
> Please follow up on our behalf and revert with firmed up dates when we can
> get a new version of MyFuel Express without the listed drawbacks.
>
> Best Regards,
> Alphonse Ogulla
>
>
> -- Forwarded message --
> From: Jacques de Gersigny 
> Date: Mon, Jul 12, 2010 at 8:34 PM
> Subject: Re: MyFuel Express FO - Shortcomings
> To: Alphonse Ogulla 
> Cc: Lovena Modelly , James Gathoga <
> j.gath...@expand-technology.com>, Simon Beamish ,
> Sanjita Sehmi , Sheila Cardovillis <
> sheila.cardovil...@unon.org>
>
>
> Hi Alphonse, Simon,
> I'm in a Business trip and I will get back on monday next.
> Rgds,
> JDG
>
>
> On 8 July 2010 16:05, Alphonse Ogulla  wrote:
>
>> Dear Jacques,
>> We tried getting you on phone in the office at noon (Kenya time) but
>> unfortunately you had stepped out for lunch. We however managed to get hold
>> of Lovena and briefly deliberated the critical items in the ensuing email.
>> In principle, we agreed to address these drawbacks in the following manner:
>>
>> 1) Expand Technologies to resolve items 1a (card printing) and 1c (bank
>> card & cheque payment currency) without further deliberations..
>> 2) Refer to the final signed TOR for items 1b (card transfer) and 1d (FO
>> direct topup). I shall get the final TOR from Easther Wanjoga of Kenya Shell
>> Ltd.
>> 3) Lastly, Expand Technologies to check if the chip card has "sufficient"
>> space to store the expiry date in order to implement item 1e (card
>> validity).
>>
>> I'm also made to understand that you called Simon Beamish and discussed
>> further the items listed above. Kindly look into these issues keenly and
>> revert with a proposal on headway latest by Monday 12th July AM. Please
>> remember to copy Shell in your rejoinder.
>>
>> Looking forward to hearing from you soon.
>>
>> Best Regards,
>> Alphonse Ogulla
>> Tel: +254 20 7621510
>> Mobile: +254 723 465172
>>
>>
>> On Mon, Jul 5, 2010 at 12:15 PM, Alphonse Ogulla wrote:
>>
>>> Dear Jacques et alia,
>>>
>>> We have identified certain shortcomings in MyFuel Express Front Office
>>> (FO) software that we need rectified as soon as possible and in-time for the
>>> go live scheduled for next month. The critical elements should be given
>>> uttermost priority as it is impossible to commence printing and issuing of
>>> cards with the listed drawbacks still in place.
>>>
>>> 1) CRITICAL ELEMENTS
>>>
>>> a) Card Printing and Personalisation
>>> Increase padding on the left margin so that the card holder name,
>>> description and vehicle registration do not print on the UN logo.
>>>
>>> b) Card Transfer
>>> The required functionality should be transfer of card value on the
>>> e-purse and not transfer of the card-holder particulars to another card as
>>> is currently the case.
>>>
>>> c) Local Epurse Remote Top-up (Bank card & cheque payment)
>>> Currently only the cash top-up function has the option of selecting the
>>> paying currency. A similar option is required for the bank card and cheque
>>> top-up since many clients run US$ transactions on their credit/debit bank
>>> cards. Similarly, US$ account holders have cheque books for US$ transactions
>>> only.
>>>
>>> d) Front Office Direct Top-up
>>> The FO lacks direct top-up capability whereas there is a card
>>> reader/writer directly connected to the FO. A work-around has been
>>> implemented by connecting a hand held POS to the Ethernet network to
>>> download and effect the actual top-up on the card. This two step procedure
>>> is time consuming and shall drastically slow down the top-up process at the
>>> station.
>>>
>>> e) Client Management - Card Validity
>>> Provide entry for expiry date i.e. dd/mm/ instead of number of years
>>> since client contracts expire on specific dates and not at the end of the
>>> year.
>>>
>>> 2) IMPROVEMENTS
>>>
>>> a) User Management -> User Rights
>>> Group the list of functions into various roles to ease creation of a new
>>> user. We acknowledge that new users shall not be created frequently but
>>> currently one has to chose from over 200 differe

Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

Hi Gordon,

On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson
 wrote:

> Technically/pedantically, users ought to be connecting to port 587 to submit
> their email anyway, with port 25 being reserved for MTA to MTA
> communications, so block 25 for everyone but the MX relaying host and insist
> your users connect on port 587 to relay their outgoing email (with
> smtp-auth)

Yes. The only thing that is delicate here is the "insist" part, but
they'll get over it. "Users" are customers.

> I'd assume that most MTAs listen on 587 these days (as well as 25) - it's
> been in the standards for quite a number of years now. (Introduced in 1998
> in RFC2476)

Yes, if you have that port open. (we do)

> And I don't know about where you are, but where I am (UK) some ISPs are now
> blocking outbound SMTP connections on port 25, or force-proxying them via
> their own email servers, making the use of port 587 almost mandatory -
> BTretail and Orange, and I think AOL do, but there's probably others.
> However it's only a matter of time before they catch up and as soon as the
> spammers start to use that port, the ISPs will block them too.

Yes, more and more providers do this.

So (even before I read your message) I decided to limit port 25 access
to the restricted IP set we know about. This will be an interesting 48
hours or so while we see if the users are still using port 25 :-)

/r

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Re: [asterisk-users] Asterisk + Hylafax + Iiaxmodem - Outbound number.

2010-07-13 Thread Doug Lytle

Marta Silva wrote:
>  Hi there,
> Thank you for your response. So I can use the ModemSetOriginCmd 
> command to assign the outbound number on the iaxmodem, but how do I 
> choose which modem to use for my specific sip client (GXW-4004), as I 
> have 2 faxes connected to my GXW box?
>

Why would you need to choose a specific modem?  I believe everything can 
be set from the command line, when generating your fax.  Granted, I 
haven't read the complete thread.

If you have specific HylaFAX+ or iaxmodem questions, it'd probably be 
better to move this thread to either of those lists.

Doug

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Gordon Henderson


On Tue, 13 Jul 2010, Randy R wrote:


On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles
 wrote:

On Tuesday 13 Jul 2010, Randy R wrote:

I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change their configs to 587 and most
spammers will be trying 25 which is closed.


Can't you just insist on SMTP AUTH?  Or  (crude but still just about usable)
require a POP3 connection before allowing an SMTP connection?


The problem is that mail to legitimate users is being sent here
although "here" is NOT the MX. On the other hand, when the users on
the road try to connect to use the server to send on port 25, it needs
to be open. I'm pretty sure closing 25 would kill the spam. But the
users would need to connect to a port for SMTP.


Technically/pedantically, users ought to be connecting to port 587 to 
submit their email anyway, with port 25 being reserved for MTA to MTA 
communications, so block 25 for everyone but the MX relaying host and 
insist your users connect on port 587 to relay their outgoing email (with 
smtp-auth)


I'd assume that most MTAs listen on 587 these days (as well as 25) - it's 
been in the standards for quite a number of years now. (Introduced in 1998 
in RFC2476)


And I don't know about where you are, but where I am (UK) some ISPs are 
now blocking outbound SMTP connections on port 25, or force-proxying them 
via their own email servers, making the use of port 587 almost mandatory - 
BTretail and Orange, and I think AOL do, but there's probably others. 
However it's only a matter of time before they catch up and as soon as the 
spammers start to use that port, the ISPs will block them too.


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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

2010-07-13 Thread Jonas Kellens


I have no licenses and I want to avoid transcoding all together.

When the phone supports G729 and the SIP provider support G729, then the 
audio can just pass through...


However, in some cases the audio is recorded. Any change that we can 
record in G729 format then ??


And how about voicemail, is this then also translated ??


Jonas.


On 07/13/2010 01:15 PM, Paul Belanger wrote:

On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellens  wrote:
   

when the conversation is using the G729-codec and the conversation is
recorded with the Monitor()-application in wav-format, will there be
transcoding (and thus a need for licenses ?)

 

I believe so, Yes.  You can check your license use by using:

*CLI>  g729 show licenses
   
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[asterisk-users] Unable to open pseudo device

2010-07-13 Thread Malvin Rito

Hi List,

I'm new to asterisk and currently running the newest of version. I'm
encountering the error below when I dial my meetme conference #:
"WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device"

I already tried googling this issue and found some procedure but still no
luck on fixing it. My server does not have any digium hardware and I'm
trying this via ztdummy.

Please advise,

Malvin


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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

2010-07-13 Thread Paul Belanger

On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellens  wrote:
> when the conversation is using the G729-codec and the conversation is
> recorded with the Monitor()-application in wav-format, will there be
> transcoding (and thus a need for licenses ?)
>
I believe so, Yes.  You can check your license use by using:

*CLI> g729 show licenses

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles
 wrote:
> On Tuesday 13 Jul 2010, Randy R wrote:
>> I was thinking of closing port 25 and using an alternate port (587?)
>> setup if the spam service is able to connect to an alternate port.
>> That way, the users can also change their configs to 587 and most
>> spammers will be trying 25 which is closed.
>
> Can't you just insist on SMTP AUTH?  Or  (crude but still just about usable)
> require a POP3 connection before allowing an SMTP connection?

The problem is that mail to legitimate users is being sent here
although "here" is NOT the MX. On the other hand, when the users on
the road try to connect to use the server to send on port 25, it needs
to be open. I'm pretty sure closing 25 would kill the spam. But the
users would need to connect to a port for SMTP.

/r

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

On Tue, Jul 13, 2010 at 12:53 PM,   wrote:
> What you can do -- I don't know about nomad, but can you make them use
> authentication?

They do identify, but they have to connect first :)

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread A J Stiles

On Tuesday 13 Jul 2010, Randy R wrote:
> I was thinking of closing port 25 and using an alternate port (587?)
> setup if the spam service is able to connect to an alternate port.
> That way, the users can also change their configs to 587 and most
> spammers will be trying 25 which is closed.

Can't you just insist on SMTP AUTH?  Or  (crude but still just about usable)  
require a POP3 connection before allowing an SMTP connection?

-- 
AJS

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread covici

What you can do -- I don't know about nomad, but can you make them use
authentication?

Randy R  wrote:

> On Tue, Jul 13, 2010 at 12:29 PM,   wrote:
> > What I do, is only open port 25 to the list of ips of the spam filtering
> > service -- I use an iptables script called rc.firewall which I found
> > several years ago which works well and has a nice syntax for this and I
> > get no direct spam, I get some which gets by the filters.
> 
> Hi John,
> 
> I'd like to do that, but there are nomad users who might be anywhere
> in the world. True maybe I could ask them to use port 587 and then
> allow ONLY the service IPs access to port 25.
> 
> /r
> 
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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

On Tue, Jul 13, 2010 at 12:29 PM,   wrote:
> What I do, is only open port 25 to the list of ips of the spam filtering
> service -- I use an iptables script called rc.firewall which I found
> several years ago which works well and has a nice syntax for this and I
> get no direct spam, I get some which gets by the filters.

Hi John,

I'd like to do that, but there are nomad users who might be anywhere
in the world. True maybe I could ask them to use port 587 and then
allow ONLY the service IPs access to port 25.

/r

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread covici

What I do, is only open port 25 to the list of ips of the spam filtering
service -- I use an iptables script called rc.firewall which I found
several years ago which works well and has a nice syntax for this and I
get no direct spam, I get some which gets by the filters.

Randy R  wrote:

> Many of you are interested in and have used or recommended fail2ban
> for your linux boxes. I finally installed it on our FreeBSD server (no
> asterisk, hence the OT) with the help of a friend from the VoIP Users
> Conference and Asterisk community.
> 
> After a lot of new learning about regex, I extended the actions and
> filters to look at our mail server, plagued by spammers - who isn't?
> Our server has a unique setup now. The customer found a spam filtering
> service that works VERY well as the MX for the domain. Their server
> then connects to ours to deliver. Obviously, the IPs of that service
> are entered as RELAY in the sendmail config. Here is my question:
> 
> We are still getting a lot of direct spam. Being that only account
> holders and the spam filtering servers should be connecting, I started
> blocking various connections bith in /etc/mail/access and in pf.
> However, I soon saw that I'll need to block the en tire Internet IP
> space. Blocking by IP is a problem for a small number of nomad users
> whose IP may just be in China, Russia or Argentina at some point.
> 
> I was thinking of closing port 25 and using an alternate port (587?)
> setup if the spam service is able to connect to an alternate port.
> That way, the users can also change their configs to 587 and most
> spammers will be trying 25 which is closed.
> 
> Is this a tenable idea? What are your experiences and opinions?
> 
> tia
> 
> /r
> 
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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

On Tue, Jul 13, 2010 at 11:04 AM, dotnetdub  wrote:
> Hi Randy,
> How many users are on this 'domain'? Google Apps Free is a great solution
> for upto 50 users with 7.6GB per user. Their spam filtering usually does the
> job for our customers.

Hi Brian,

Thanks for the reply. I'm familiar with Google's pro services, we have
other customers on them. However, this doesn't address the question I
asked, which is regarding a FreeBSD mail server we run.

/r

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Re: [asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread dotnetdub

On 13 July 2010 09:52, Randy R  wrote:

> Many of you are interested in and have used or recommended fail2ban
> for your linux boxes. I finally installed it on our FreeBSD server (no
> asterisk, hence the OT) with the help of a friend from the VoIP Users
> Conference and Asterisk community.
>
> After a lot of new learning about regex, I extended the actions and
> filters to look at our mail server, plagued by spammers - who isn't?
> Our server has a unique setup now. The customer found a spam filtering
> service that works VERY well as the MX for the domain. Their server
> then connects to ours to deliver. Obviously, the IPs of that service
> are entered as RELAY in the sendmail config. Here is my question:
>
> We are still getting a lot of direct spam. Being that only account
> holders and the spam filtering servers should be connecting, I started
> blocking various connections bith in /etc/mail/access and in pf.
> However, I soon saw that I'll need to block the en tire Internet IP
> space. Blocking by IP is a problem for a small number of nomad users
> whose IP may just be in China, Russia or Argentina at some point.
>
> I was thinking of closing port 25 and using an alternate port (587?)
> setup if the spam service is able to connect to an alternate port.
> That way, the users can also change their configs to 587 and most
> spammers will be trying 25 which is closed.
>
> Is this a tenable idea? What are your experiences and opinions?
>
> tia
>
> /r
>
>
Hi Randy,

How many users are on this 'domain'? Google Apps Free is a great solution
for upto 50 users with 7.6GB per user. Their spam filtering usually does the
job for our customers.

Regards,
Brian
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[asterisk-users] OT: fail2ban, spam and mail servers

2010-07-13 Thread Randy R

Many of you are interested in and have used or recommended fail2ban
for your linux boxes. I finally installed it on our FreeBSD server (no
asterisk, hence the OT) with the help of a friend from the VoIP Users
Conference and Asterisk community.

After a lot of new learning about regex, I extended the actions and
filters to look at our mail server, plagued by spammers - who isn't?
Our server has a unique setup now. The customer found a spam filtering
service that works VERY well as the MX for the domain. Their server
then connects to ours to deliver. Obviously, the IPs of that service
are entered as RELAY in the sendmail config. Here is my question:

We are still getting a lot of direct spam. Being that only account
holders and the spam filtering servers should be connecting, I started
blocking various connections bith in /etc/mail/access and in pf.
However, I soon saw that I'll need to block the en tire Internet IP
space. Blocking by IP is a problem for a small number of nomad users
whose IP may just be in China, Russia or Argentina at some point.

I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change their configs to 587 and most
spammers will be trying 25 which is closed.

Is this a tenable idea? What are your experiences and opinions?

tia

/r

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[asterisk-users] Recording from g729 to wav means transcoding ?

2010-07-13 Thread Jonas Kellens


Hello list,

when the conversation is using the G729-codec and the conversation is 
recorded with the Monitor()-application in wav-format, will there be 
transcoding (and thus a need for licenses ?)



Kind regards,

Jonas.
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[asterisk-users] How to trace incoming AMI requests ?

2010-07-13 Thread Olivier

Hello,

Is there a way to trace into a log file, incoming AMI requests ?

For instance, I've got several apps accross the LAN, sending AMI requests
such as :
Action: originate
Channel: Local/7...@internal
Exten: 00123456789
Priority: 1
...
Some of them might be sometimes producing some erroneous requests.

I would like to check from server side, that each received request is
formatted and compliant.
Ideally, I would to read in a log file, a (reliable) copy of AMI originate
requests.
Any suggestion ?

If this helps, I can use astmanproxy.

I also saw a debug=on option inside /etc/asterisk/manager.conf but I
couldn't find a change in produced output.
And
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.confis
not helpful on this specific option.

Regards
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