Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Andrew Furey]

Ditto; a Gmail issue?

Andrew

On 12 June 2017 at 16:00, Marcelo Terres  wrote:

> It is happening the same with me.
>
> Regards,
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 12 June 2017 at 08:07, Olivier  wrote:
> > Hello,
> >
> > I'm a faithful reader of this mailing list, for several years now.
> >
> > Lately, I'm receiving emails asking me to re-enable my list subscription
> due
> > to "excessive bouncing".
> >
> > What does this exactly mean and why am I receiving this ?
> > Beside re-enabling my subscription, what can I do to improve things ?
> >
> > Regards
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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Re: [asterisk-users] What linux distro most popular for Asterisk

[Andrew Furey]

[Apologies, top-posting, Gmail, yadda yadda]

As with a lot of software, I suspect the best answer is whichever distro
YOU are most comfortable with. You're the one who has to support it, after
all... Just my 2c.

Andrew

On Thursday, 17 October 2013, Rusty Newton wrote:

 On Tue, Oct 15, 2013 at 11:58 PM, Michelle Dupuis 
 mdup...@ocg.cajavascript:;
 wrote:
  Is there a recent survey of that Linux distro and version people are
 using
  for the Asterisk installations?  I recall seeing a pie chart over a year
 ago
  (I think on a wiki but I can't find it again)also hoping for
 something
  more current.
 
  I suspect RH5 and RH6 are most popular...but I'm looking for facts

 I don't have any numbers, but I watch the issue tracker a lot and I
 see pretty much CentOS, Debian and Ubuntu. Which seems to match what
 everyone else is saying on this thread.

 --
 Rusty Newton
 Digium, Inc. | Community Support Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct: +1 256 428 6200

 Check us out at: http://digium.com  http://asterisk.org

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-- 
Linux supports the notion of a command line or a shell for the same
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Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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Re: [asterisk-users] how to set iaxmodem receiving speed

[Andrew Furey]

On 17 May 2012 22:40, gincantalupo gincantal...@fgasoftware.com wrote:
 That could seem counter-intuitive but it is not. Not to mention the fact
 that information technology is not science, the solution to broken faxes is
 to lower down speed. This works even with normal telco lines even if you DO
 NOT have a pbx (telco technicians even say not to make faxes pass thru your
 PBX). I could ask my customer's telco to lower the speed down but it depends
 on the guy working at the call-center...sometime you talk to dummy people
 who ARE sure it is impossible. But it is not. So, I do not want to spend
 days to convince people working at that telco call-center that what I'm
 asking is feasible and I do not want to tell my customer to tell their
 customer to lower their faxes speed (before installing our PBX they were
 able to send perfect faxes so, why should they?).

 My idea was to tell iaxmodem not to accept fast speed rates so the fax
 machine on the other side should be forced to negotiate a slower speed as if
 my customer fax weren't virtual as iaxmodem is but a real one.

 I suspect that the problem is about the primary lines because I tested
 iaxmodem many times on my LAN and it is (surprisingly :) ) working fine (10
 good received faxes out of 10 sent!!!) but, as you may know, talking to
 telco technician is a nightmarethey always say problems are always on
 the PBX side... :(

 Moreover, after sending a fax, the fax machine beeps correctly as the fax
 was correctly sent without corruption. :o

And as a matching data point, we use ActiveFax for sending (interfaced
from an ERP package) and often get Comm Error 283 and incomplete
faxes. If it's just making a bad situation worse, how is it that our
solution of turning off ECM mode fixes it 98% of the time? I'm
curious. (We know it's fixed as often we're the receiver, and we can
see the correct fax come through.)

Sometimes we lower the speed too (to 9600) but often simply disabling
ECM is the solution.

Regards,
Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
                          -- Bill Garrett

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Re: [asterisk-users] asterisk distributions

[Andrew Furey]

On 2 March 2012 01:43, A J Stiles asterisk_l...@earthshod.co.uk wrote:
 Look, the command line is a fact of life.  Microsoft have spent a fortune
 telling you that you're not smart enough to use it.  You do not have to fall
 for that.  Are you going to sit back and let them call you stupid?

 Think of trying to make yourself understood in a foreign country by pointing
 and gesturing.  There comes a point where you will actually have expended
 *more* effort than if you had just bitten the bullet and learned the language
 in the first place.

My signature is probably applicable here (out of the fortune database)...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
                          -- Bill Garrett

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Re: [asterisk-users] Interesting attack tonight fail2ban them

[Andrew Furey]

On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
 I thought that it might be worth adding a line to my fail2ban filter, but am
 looking for a hand with the regex.  I have come up with:
     NOTICE.* .*: Call from '' to extension '.*' rejected because
 extension not found

 but I realize that anyone misdialling a valid extension a few times gets cut
 off. Can someone suggest an improvement?  (How could I limit this to 4 or
 more digits dialled for example?)

[ Caveat - I have never used fail2ban ]

If it supports Perl-style regexps, you could do:

NOTICE.* .*: Call from '' to extension '[0-9]{4,}' rejected because
extension not found

That will do at least 4 digits.

Or the long way (Bash-style etc):

NOTICE.* .*: Call from '' to extension '[0-9][0-9][0-9][0-9][0-9]*'
rejected because extension not found

HTH,
Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
                          -- Bill Garrett

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Re: [asterisk-users] How to have Asterisk respond from the IP address used for registration

[Andrew Furey]

On 28/05/2010, Mike l...@virtutel.ca wrote:
 That was a simplified example. I actually have two links from different
  ISPs, totally different networks.  Those on provider A should talk to
  provider`s A IP address and have their answers come back from provider's A
  IP, and those on provider B should talk to my provider B NIC and get the
  response back from that IP.

I think this is more a router issue - we do this with three links,
going into a single Linux-based Linksys which acts as the single
gateway for the LAN (so it has 4 interfaces). You need to look into
the ip command, and packet mangling to mark connections as coming
from each provider (so that all related packets go back the same way).

HTH
Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Are there AMI commands to manipulate a voice mailbox?

[Andrew Furey]

On 14/05/2010, Motiejus Jakštys desired@gmail.com wrote:
 Talking about file permissions, on Linux everything is possible using
  POSIX ACLs. You can set specific rights to files/directories for
  certain users.
  Note 1: if setting group permissions is enough, use that.
  Note 2: Asterisk and web server should be on separate machines (at
  least virtual machines) for many reasons... I would mount my voicemail
  over NFS... which itself has enough access control.
  Note 3: if you decide to experiment with ACLs (IMHO, most flexible) -
  do not forget to remout your file system:
  mount -o remount,acl /usr

Not quite everything - you're still limited to read/write/execute
granularity (unless something has changed in the 5 years since I
experimented with it). If you're expecting Full Control / Modify /
Delete etc as per Windows 2000 and its ilk, you might have to look at
something else...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Voicemail() app not available?

[Andrew Furey]

On 13/05/2010, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 Specifically, builds 3 different variants of app_voicemail.so as
  different modules (app_voicemail.so, app_voicemail_imap.so,
  app_voicemail_odbc.so).

Correct; the other two were noload(ed) by default so I left them.


 What happens if you run:

   module unload app_voicemail.so
   module   load app_voicemail.so

Aha; I get the following:

asteriskdemo*CLI module unload app_voicemail.so
Unable to unload resource app_voicemail.so
[May 14 11:06:04] WARNING[15747]: loader.c:501 ast_unload_resource:
Firm unload failed for app_voicemail.so
asteriskdemo*CLI module load app_voicemail.so
[May 14 11:06:22] ERROR[15747]: app_voicemail.c:7941 load_module:
app_voicemail.so depends upon res_adsi.so

... which I had noload(ed) as well - didn't realise it was required.
(Further looking shows it was in the messages log the whole time;
didn't think to check, been connecting to -r console...)

Loading res_adsi.so fixes the problem. I'll have to go over the
Asterisk Slimming wiki entry...

Thanks Tzafrir!

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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[asterisk-users] Voicemail() app not available?

[Andrew Furey]

Hi all,

I have a demo machine I'm running up on Lenny - it has the packaged
Asterisk version installed (1.4.21.2+stuff).

I'm trying to add an extension to leave a voicemail message, just with
Voicemail(1234), which I've done before (on 1.2 at least), but it's
saying no application 'Voicemail' .

module show like voi shows app_voicemail.so and
app_hasnewvoicemail.so loaded (I have autoload=yes in modules.conf
and have noload= a bunch, but I even explicitly set
load=app_voicemail.so just in case.

However, core show applications like voi only lists HasVoicemail
and HasNewVoicemail, with no sign of Voicemail. The wiki seems to
show that it should all be included, with no sign of deprecation...

Any ideas where I can look?

TIA,
Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

-- 
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Re: [asterisk-users] test

[Andrew Furey]

On 25/10/2009, Matt mhop...@gmail.com wrote:
 This is a test... I am being told I am subscribed, but I am not getting
 messages.

Gmail always seems to hide receipt of your own messages to mailing lists...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] PHP AGI Not Working and Odd Behavior

[Andrew Furey]

On 25/06/2009, David Quinton gna...@bizorg.co.uk wrote:
 May be a  total red herring (I'm using an old version of Trixbox)
  but if I edit my PHPs on a Windows machine and upload using FTP, they
  will only run if I fire up Nano and save the file on the Asterisk box.

I haven't used TrixBox, but that sounds a lot like CRLF-LF issues to me...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Voicemail config help - require password

[Andrew Furey]

On 19/03/2009, Jonathan Thurman jthurma...@gmail.com wrote:
  Also, is there a way to retain deleted messages for a length of time
  before they are purged?  We currently have that feature on our
  production VM server that I am trying to replicate.  Thanks!

Could this be done with a simple nightly cron job? Something like

find /var/lib/asterisk/messages [I forget the path] -name *.wav -mtime
+90 -exec rm {} \;

That'll delete any WAV files older than 90 days. Mind you, there might
be an index file that needs editing as well (haven't used the VM
system for a while) so a script may be needed...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] chan_ss7 0.10

[Andrew Furey]

On 27/02/2008, Joel Solanki [EMAIL PROTECTED] wrote:
 I tried 3 times to send this message. It goes out but i dont recieve mail
 sent on asterisk-users@lists.digium.com but when someone replies to that
 email i recieve the email like you did.
 I thought mails were not going to mailling list to tried 3 times.  But it is
 strange. As far as i know if i sent mail to [EMAIL PROTECTED]
 then even i should also recieve my email right ?

I don't know whether the list server is refraining from sending it to
yourself, or if (more likely) Gmail is deciding not to show it since
you already know what it says... but yes, it happens for me too (as
well as on another mailing list I'm on). Standard Gmail behaviour, at
least.

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Wanted: tutorial on troubleshooting SIP issues

[Andrew Furey]

On 12/11/2007, randulo [EMAIL PROTECTED] wrote:
 I thought Wireshark was the cute Mac OS X name.

The author changed the name of the codebase last year due to employment changes:

http://en.wikipedia.org/wiki/Wireshark#History

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Net2Phone Multiple SIP Trunk Not Working

[Andrew Furey]

On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote:

I'm sorry that's because I didn't get a visibility of ny post, I though that
was a network problem (as I cannot see my post on the mailing list)


You never do with mailing lists on Gmail, I presume it hides it based
on the message ID (since you already have a copy).

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
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Re: [asterisk-users] Re: MeetMe and ChannelRedirect

[Andrew Furey]

On 17/05/07, Tony Mountifield [EMAIL PROTECTED] wrote:

In article [EMAIL PROTECTED],
Rafael Vidal Aroca [EMAIL PROTECTED] wrote:
 i'm trying to implement the following scenario:

 - A user calls number 700
 - Asterisk then dials to extensions 100, 200, 300, 400 and 500
 - And then bridges all calls to a conference room

 I tried to use MeetMe and ChannelRedirect, but seems that after
 channel redirect nothing more is executed. So, this seem to work for the
 caller and first called, but the others stay outside.

 Could anyone help or give me a hint?

The way I did this kind of thing was like this:

1. Extension 700 calls an AGI script which generates a .call file in
/var/spool/asterisk/outgoing for each of the calls to the other extensions.
Extension 700 then drops into the Meetme room to wait for the others.

2. Each call file specifies a Local channel to make the call to the
extension,
and uses the Context, Extension and Priority fields to direct the answered
call into the Meetme room.

If any of the calls to the other extensions fails (e.g. busy), you don't get
any notification of that. If you want such notification, you will need to
get a lot more complex, probably involving a controlling process using the
Manager API.

Hope this helps.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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 -- Bill Garrett
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Re: [asterisk-users] Re: MeetMe and ChannelRedirect

[Andrew Furey]

On 17/05/07, Andrew Furey [EMAIL PROTECTED] wrote:


[nothing]


Ugh, what happened there? must have clicked the wrong button. Sorry
for the noise folks.

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
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Re: [asterisk-users] Softphone that supports central provisioning?

[Andrew Furey]

On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote:

 Tzafrir Cohen wrote:
 Dear Senad,

 The setup program for your soft phone can be downloaded from here:
 a href=http://malwareserver.com/malware.exe;http://LINK/a

 During the setup you will be asked for configuration file. Please use
 attached file.

Tzafrir is referring to possible link that user can receive from
someone...

Since I was referring to SYSTEM email message generated from within PBXware,
above is not possible without some serious hacking of the network, the box,
the chroot etc... If one is at that level it then becomes a criminal issue.


Not denying the criminal aspect, but who says the email has to really
come from that box? If there's one thing SMTP is good at, it's
allowing forged emails... it wouldn't take a decent phisher 10 minutes
to craft an email that has all the same content including From
addresses.

Sure, the full headers would give up the game - but how many of your
users would (a) check them, and (b) understand what they're seeing?
I'd be surprised if it's more than 5% - and in many cases it only
takes one person to fall for it...

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [asterisk-users] SLA more than 100% ?

[Andrew Furey]

On 24/02/07, Tim Connolly [EMAIL PROTECTED] wrote:

How does one answer more than 100% of the calls in less than 60 seconds?


techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s
holdtime),
W:0, C:3, A:2, SL:166.7% within 60s


Probably talking out of my hat (I've never particularly looked at
those figures), but might there have been some calls in progress at
the start of the 60s which have finished? If it works on subtract for
call started, add for call-stopped...?

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context

[Andrew Furey]

On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote:

What Lee suggested is to have the AGI script to actually parse, insert a new
context in extensions.conf, or deleting from it, then reload
extensions.conf.  This would at least achieve what you wanted to do.


Or alternatively, to avoid complete disaster, why not have
extensions.conf include another file (#include somefile.conf) and
edit that one with your script? I've done that before (although I was
actually recreating the entire file each time by populating from an
external database).

Andrew

--
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Re: [asterisk-users] Why dont my messages get through

[Andrew Furey]

On 11/8/06, Nick Hoffman [EMAIL PROTECTED] wrote:

 They do get through. Messages you send to the list won't get sent back
 to you, because you sent them.

Hi Alex. Are you sure about that? I receive a copy of every email I send to
the list.


I think it's just Gmail that hides them, especially since you have the
one you sent already there. It does the same for another mailing list
I'm on. This doesn't apply to Christian, of course.

(Interesting experiment would be to delete your sent copy as soon as
you send it, before the list server sends it back, and see if it
reappears...)

Andrew

--
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Re: [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)

[Andrew Furey]

On 6/15/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:

On Wed, 2006-06-14 at 19:58 -0400, Daniel Salama wrote:
 Mainly GXP-2000 (with silence suppression off) and Eyebeam (with
 Enable microphone noise reduction off)

its safe to ignore that too, it just means that asterisk doesnt support
a sip feature that your phone does and its telling you hey I know that
the feature exists but I really dont support it.  If you disable CNG in
the phone you wont see the messages anymore, alternatively if you ignore
the messages they wont bother you anymore :)


Was going to post this separately but since there's a thread here... :-P

Has anyone has problems with this with Engin? We asked them about it
and they said they don't have any settings at their end, we should
check our client. But since Asterisk is the client in this case, how
do you change it?

Thanks,
Andrew

--
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Re: [Asterisk-Users] Re: Forcing Marker bit

[Andrew Furey]

On 6/1/06, Tony Mountifield [EMAIL PROTECTED] wrote:

# script /tmp/output.txt
Script started, file is /tmp/output.txt
# exec asterisk -rv
... do asterisky stuff ...
host*CLI exit
Script done, file is /tmp/output.txt
#


Actually you need another exit in there:

# script /tmp/output.txt
Script started, file is /tmp/output.txt
# exec asterisk -rv
... do asterisky stuff ...
host*CLI exit
Executing last minute cleanups
# exit
Script done, file is /tmp/output.txt
#

Andrew

--
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Re: [Asterisk-Users] Quad BRI card

[Andrew Furey]

On 5/18/06, Craig Guy [EMAIL PROTECTED] wrote:

Any device to legally connect to the PSTN in Australia must be approved by
the regulatory body.  A process that usually costs at least $20,000 and only
allows the permit holder to sell the product for conneciton to the pstn.  It
is a very high barrier to entry for the Australian market.  There is a guy
in Victoria who certified the Fritz! card and charges $400 each for them.
Paralell imports are not allowed to be connected.


Ah, so that's why they're so expensive :(

Sorry, what do you mean by that last sentence?

Andrew

--
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Re: [Asterisk-Users] Re: problems with emailing voicemail

[Andrew Furey]

On 3/23/06, Avi Miller [EMAIL PROTECTED] wrote:
 Not only does 'host.domain.com' need to resolve to an IP
 address, but that IP address must resolve to 'host.domain.com' in the
 reverse lookup table.

Not technically true, AFAIK... the reverse doesn't have to be the same
(how would multiple domain hosting work?) but it should resolve to
_something_ (NXDOMAIN is not an option).

Andrew

--
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Re: [Asterisk-Users] chan_capi setting ${DNIS}

[Andrew Furey]

On 2/20/06, Nathan Alberti [EMAIL PROTECTED] wrote:
 Is there a reason the variable ${DNIS} does not get set with incoming
 calls via chan_capi ?

 Is it related to the MSN=X in capi.conf ?

Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki variables page, but ${DNID} works for
me with a BRI...

Andrew

--
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Re: [Asterisk-Users] Any way to grep through fast moving console messages?

[Andrew Furey]

On 2/10/06, Eric Bishop [EMAIL PROTECTED] wrote:
 Or perhaps slow them down or pipe to a file?

I usually run Asterisk in a screen session, and use Ctrl-A, [ to
scroll through screen's buffer... I'm sure there's other ways too :)

Andrew

--
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Re: [Asterisk-Users] Re: Polycom FW

[Andrew Furey]

On 1/23/06, Doug Lytle [EMAIL PROTECTED] wrote:
  Further, Polycom SIP phones have the longest boot time of any phone
  I've ever seen (something like 5 min, compared to a Sipure, less than

 Give a SIP based Cisco 79XX phone a try, just about as long in boot time.

Huh? My 7905 takes well under 10 seconds, including Asterisk
registration and NTP update. Granted, if it were DHCP it might take
marginally longer, but 5 _minutes_?

Andrew

--
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reason that only children read books with only pictures in them.
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Re: [Asterisk-Users] WG: App_rxfax problem

[Andrew Furey]

  Ouch ... error while writing audio data: : Broken pipe

 If you are talking about the Ouch message, yes lots of people have seen
 the error and its usually the result of some misconfiguration in one of
 your files (likely zapata.conf).

Correct me if I'm wrong, but isn't that message from mpg123 itself? It
appears in the binary (via strings), and I've seen it at non-asterisk
times too. AFAIK it comes up whenever the parent application (asterisk
in this case) quits without closing it properly (hence, broken
pipe).

As such, this means that the above error simply shows that asterisk
crashed (which they presumably already knew), and has nothing to do
with the problem itself...

Andrew

--
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reason that only children read books with only pictures in them.
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[Asterisk-Users] Dial plan logic documentation?

[Andrew Furey]

Hi all,

What methods (software or even on paper) would you folks use /
recommend for the purposes of documenting how a dial plan is
constructed? ie. what extensions jump to other extensions, etc? This
is as a means of getting the big picture rather than having pages
and pages of printed extensions.conf output...

If you consider priorities as line numbers, extensions as
functions/subroutines and contexts as source files, you could compare
the dialplan to a regular programming language source... I've thought
of various things like flowcharts but I don't know of any really good
flowcharting programs. Besides, the analogy breaks down in that
programming languages don't generally jump to specific line numbers in
a function (whereas using priorities other than 1 is quite common).

Any thoughts?

Andrew

--
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reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
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Re: [Asterisk-Users] Motherboard and processor recommendations

[Andrew Furey]

On 9/10/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 IIRC Linux's raid support will support hot-swapping disks, but I'm not
 sure which disks are are supported.

The software RAID is no problem (raidsetfaulty, raidhotremove,
raidhotadd)... the only question is whether the BIOS and/or disks
themselves will hot-swap.

Andrew

-- 
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Re: [Asterisk-Users] Cannot native bridge on licensed G729

[Andrew Furey]

On 7/25/05, Jay Milk [EMAIL PROTECTED] wrote:
 I'd say your hardware is out of codecs.  Sipura SPA-2000's, for example,
 only allow one G729-call at a time because of licensing issues.  Allow
 GSM as a secondary codec and you should be fine.  Yep, it's more
 bandwidth, but...

Thanks Jay, that makes sense (although it's a bit inconvenient...). I
was still having trouble with the phones picking the free codec first,
but that seems to be solved with the RxCodec and TxCodec settings in
the web interface.

Thanks again,
Andrew

-- 
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[Asterisk-Users] Cannot native bridge on licensed G729

[Andrew Furey]

Hi folks,

In an effort to save bandwidth (our 7905s run over a WAN) we've
switched from ulaw to g729a. We purchased 4 licenses from Digium (4
SIP clients, low call volume), and they seem to have been accepted:

 [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator)
  == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e
  == Found license 'G729-253D0C86' providing 4 channels
  == Found total of 4 G.729 licenses
  == Registered translator 'g729tolin' from format G729A to SLINR, cost 5
  == Registered translator 'lintog729' from format SLINR to G729A, cost 24

*CLI show translation
 Translation times between formats (in milliseconds)
  Source Format (Rows) Destination Format(Columns)

 G723   GSM  ULAW  ALAW  G726 ADPCM SLINR LPC10 G729A SPEEX  ILBC
   G723 - - - - - - - - - - -
GSM - - 4 5 - - 3 -27 - -
   ULAW -11 - 1 - - 1 -25 - -
   ALAW -12 1 - - - 2 -26 - -
   G726 - - - - - - - - - - -
  ADPCM - - - - - - - - - - -
  SLINR -10 1 2 - - - -24 - -
  LPC10 - - - - - - - - - - -
  G729A -15 6 7 - - 5 - - - -
  SPEEX - - - - - - - - - - -
   ILBC - - - - - - - - - - -

*CLI sip show peer andrew
[snip]
  Codecs   : G.729A

But when we try to use more than one (such as transferring an incoming
BRI call to a second phone), when the phone answers, the transfer
fails and we get the following:

*CLI Jul 25 16:49:25 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
-- Executing Dial(SIP/andrew-89e3, SIP/jeremy|20) in new stack
-- Called jeremy
-- SIP/jeremy-b7a9 is ringing
-- SIP/jeremy-b7a9 answered SIP/andrew-89e3
-- Attempting native bridge of SIP/andrew-89e3 and SIP/jeremy-b7a9
Jul 25 16:49:36 WARNING[851980]: rtp.c:1392 ast_rtp_bridge: codec0 =
12 is not codec1 = 256, cannot native bridge.
  == Spawn extension (default, 801, 1) exited non-zero on 'SIP/andrew-89e3'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back
from 192.168.200.226
Jul 25 16:49:42 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!


Any ideas?

Andrew

-- 
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reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
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Re: [Asterisk-Users] MOH - request to schdule in the past

[Andrew Furey]

 From your system command line (not asterisk), type 'mpg123' and tell
 us what version of mpg123 you're running.
 
 If its not v0.59r or v0.59q, then get one of those installed.
 (Lots of notes say v0.59r only, however I've been using v0.59q
 on RHv9 and Fedora 3 boxes with no problems.)

FWIW, I have 0.59r (on Sarge) and I still get this from time to time
(usually when the system is temporarily busy). I don't have a timing
source, but nor do I have any particular problems... I presume the
music jitters at the time but there's usually no one using it at that
moment.

Andrew

-- 
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enough to accomplish a sufficiently broad range of tasks.
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Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Andrew Furey]

 Only if you have your clothes on and they don't... ;-)

I should _hope_ they don't have your clothes on :)

Andrew

-- 
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Re: [Asterisk-Users] Maximum retries exceeded on call

[Andrew Furey]

 I see this message in my asterisk log sometimes.  Can someone explain to me
 what this means and how to correct the problem?
 
 May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
 exceeded on call 795fcf0c6
 [EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
 May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
 exceeded on call 0ec0538d9
 [EMAIL PROTECTED] for seqno 40146 (Non-critical Response)
 May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
 exceeded on call 0ec0538d9
 [EMAIL PROTECTED] for seqno 40147 (Non-critical Response)

We tend to get this when asterisk tries to call a SIP extension which
has lost its connection for some reason (network troubles, power
outage, whatever). See if there are any calls being attempted at that
time...

Andrew

-- 
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Re: [Asterisk-Users] Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors

[Andrew Furey]

 I'm having major problems getting a Fritz card to dial out in the UK (or 
 indeed
 answer, but I've been concentrating on dialing out). I'm getting the 0x3301 or
 0x3302 error.

Not a direct answer, but are you _absolutely sure_ the card works? I
had the exact same thing late last year here in Australia, there was
nothing wrong with the config but once we got around to testing it at
another known installation it was found to be faulty. Replaced it
under warranty, and everything worked as we had it.

Might save you some head-banging, at least...

Andrew

-- 
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reason that only children read books with only pictures in them.
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enough to accomplish a sufficiently broad range of tasks.
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[Asterisk-Users] Outgoing faxes with chan_capi?

[Andrew Furey]

Hi all,

Does anyone know of a way to send faxes over CAPI?

I'm using asterisk on Debian sarge on 2.4.27, with 2 Fritz! PCI cards
and the appCapiFax patch. Incoming faxes work perfectly with
capiAnswerFax (I have dedicated numbers so I don't have to detect fax
calls). But I can't find any method of _sending_ with CAPI, other than
this comment on the list a few weeks back:

[snip]
In this little corner of the world [Australia] we pay $400 for a
single port Fritz! card
that costs $30 anywhere else and have to use chan_capi.  Chan_capi is ok for
basic service but is not supported by spandsp or AMP, and only partially
supported by FOP.
[/snip]

And since most fax systems use spandsp (including app_txfax, which
would be my first choice) it doesn't seem to leave many options.

The wiki comments mention CapiSuite, but that appears to be a
completely separate package to Asterisk, so I'd (I think) need to
fully shut down Asterisk, send the fax, then start it up again? Not
really acceptable, even for a lightly-loaded system like ours
(incoming calls wait for no man).

Thanks in advance,
Andrew

-- 
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reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
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Re: [Asterisk-Users] GotoIfTime Discrete weekdays (Mon,Wed,Fri)

[Andrew Furey]

 Does anyone know if there is a way to get GotoIfTime to accept
 individual weekdays instead of a range?
 
 Example Dr. Office is closed on Thursday and Sunday.

Think the easiest option would be to use two statements:

exten = s,1,GotoIfTime(*|thu|*|*?closed,s,1)
exten = s,2,GotoIfTime(*|sun|*|*?closed,s,1)
exten = s,3,[restofdialplan]

Andrew

-- 
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Re: [Asterisk-Users] TDM400 Problem

[Andrew Furey]

  Has anyone seen this message trying to install an TDM400.. spurious
  8259A interrupt: IRQ7
 
 I used to get this message a lot on my computer (before I even heard
 of Asterisk, mind you). When I looked into it and asked people
 questions, I was told that it is a harmless message.

I even get this on production servers (ISP level) that have nothing to
do with Asterisk. Given that IRQ 7 is the parallel port, I would guess
it comes up when the parallel drivers are loaded but there is nothing
plugged in to the port. Completely harmless AFAIK...

Andrew

-- 
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reason that only children read books with only pictures in them.
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enough to accomplish a sufficiently broad range of tasks.
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[Asterisk-Users] Indication of transfer on display

[Andrew Furey]

Hi all,

I'm using asterisk 1.0.2 (the Debian Sarge package) with Cisco 7905G
phones (SIP firmware). I've defined a macro to do some custom CallerID
stuff for our 100-number ISDN range (so we can see what line they've
called).

What I'd like to do is have the phone display update (to the original
called number?) when an incoming call is transferred from one phone to
another. At present there's no way to tell, short of looking at the
asterisk console for when the channel hangs up - the phone still
displays the internal caller ID of the first phone.

I see that the wiki shows this features.conf option for CVS HEAD,
which would almost do the job:

xfersound = beep   ; to indicate an attended transfer is complete

However, am I correct in assuming that this only applies to transfers
done with the # key, rather than the phone's own transfer function?

Any suggestions?

Thanks,
Andrew

-- 
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enough to accomplish a sufficiently broad range of tasks.
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Re: [Asterisk-Users] Cisco 7940G

[Andrew Furey]

 AFAIK, Cisco 79xx phones don't have web-based configuration. They have a
 telnet interface, though it's enabled/disabled based on the config files the 
 phone
 gets from the TFTP server.

FWIW, my 7905A with SIP firmware 1.01.00(030807A) has web but not
telnet. The web interface has a number of settings not found in the
keypad+LCD interface (including silence suppression).

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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[Asterisk-Users] Re: Busy message on ISDN cards? (SOLVED)

[Andrew Furey]

 The secondary problem I reported earlier (Outgoing MSN andrew not
 allowed) seems to have fixed itself. But when I try to call, I get:
 
 *CLI -- Executing Dial(SIP/andrew-4e2d,
 Capi/91234567:0412345678) in new stack
 -- data = 91234567:0412345678
 -- capi request omsn = 91234567
   == found capi with omsn = 91234567
[snip]

   == received CONNECT_CONF PLCI = 0x101 INFO = 0
 -- DISCONNECT_IND ID=001 #0x000f LEN=0014
   Controller/PLCI/NCCI= 0x101
   Reason  = 0x3301


Well, after much head-banging, we finally worked out that the Fritz
card itself was faulty. Returned it under warranty for a replacement,
and all works perfectly (line synchronisation, outgoing calls,
iincoming calls). Argh...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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[Asterisk-Users] Re: Busy message on ISDN cards?

[Andrew Furey]

Something of an update...

At the recommendation of a consultant I called in last week, I've now
switched to an AVM Fritz! PCI card, using CAPI. At this stage I'm only
using the one card so I don't need the patches to run multiple cards
yet.

Upon loading the modules (capi/capifs and fcpci), I get:

CAPI-driver Rev 1.1.4.1: loaded
capifs: Rev 1.1.4.1
capi20: started up with major 68
kcapi: capi20 attached
capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs)
fcpci: AVM FRITZ!Card PCI driver, revision 0.5.2
fcpci: (fcpci built on Dec 24 2004 at 14:17:43)
fcpci: Loading...
fcpci: Driver 'fcpci' attached to stack
kcapi: driver fcpci attached
fcpci: Auto-attaching...
PCI: Found IRQ 5 for device 00:12.0
fcpci: Stack version 3.11-02
kcapi: Controller 1: fritz-pci attached
kcapi: card 1 fritz-pci ready.
fcpci: Loaded.
kcapi: notify up contr 1
capi: controller 1 up

Also this looks correct:

netmagic:~# cat /proc/capi/controllers/1
name fritz-pci
io   0xD800
irq  5
type A1
class14
ver_driver   3.11-02
ver_cardtype fritz-pci
protocol DSS1
linetype point to multipoint

And in the asterisk output:

 [chan_capi.so] = (Common ISDN API for Asterisk)
  == Parsing '/etc/asterisk/capi.conf': Found
-- This box has 1 capi controller(s).
-- CAPI[contr1] supports DTMF
-- CAPI[contr1] supports supplementary services
HOLD/RETRIEVE
TERMINAL PORTABILITY
ECT
3PTY
CF
CD
MCID
CCBS
MWI
CCNR
  == ast_capi_pvt(92130800,*,demo,0,2) (1,2,64)
  == ast_capi_pvt(92130800,*,demo,0,2) (1,2,64)
-- listening on contr1 CIPmask = 0x1fff03ff
  == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5)
aLaw CVS HEAD)


Of course I'm not using modem.conf any more, but capi.conf contains:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=91234567
incomingmsn=*
outgoingmsn=91234567
controller=1
softdtmf=1
accountcode=
context=demo
devices=2


The secondary problem I reported earlier (Outgoing MSN andrew not
allowed) seems to have fixed itself. But when I try to call, I get:

*CLI -- Executing Dial(SIP/andrew-4e2d,
Capi/91234567:0412345678) in new stack
-- data = 91234567:0412345678
-- capi request omsn = 91234567
  == found capi with omsn = 91234567
  == CAPI Call CAPI[contr1/91234567]/0 -- Called 91234567:0412345678
-- CONNECT_CONF ID=001 #0x0004 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- CONNECT_CONF ID=001 #0x0004 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
-- DISCONNECT_IND ID=001 #0x000f LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3301

  == DISCONNECT_IND PLCI=0x101 REASON=0x3301
-- CAPI Hangingup
  == No one is available to answer at this time
-- Timeout on SIP/andrew-4e2d


Sometimes the reason is given as 0x3302 instead. I should also mention
that unlike the Eicon cards, plugging an Onramp into the Fritz card
does not cause the Onramp to synchronise... the LED just keeps
flashing.

I noted that the output above lists the card as point-to-multipoint.
Finding a couple of pages which suggested that both the card and the
line need to be in this mode, I rang Telstra on Friday, who said that
the line was currently in point-to-point mode, but they would change
it to point-to-multipoint immediately. Of course there's no one there
until Wednesday to check if it's actually been done, but there's no
apparent change from before (the existing PBX can synchronise but the
Fritz card can't).

Any further ideas?

Thanks,
Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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[Asterisk-Users] Busy message on ISDN cards?

[Andrew Furey]

Hi all,

I'm new to asterisk and not too knowledgeable on ISDN, so please be gentle :)

I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with
kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway
ISDN line. I'm pretty certain the card and line both work since
they've been used in this machine for PPP before this (but with an
older kernel with DoV patches, which are no longer to be used).


If I do

# modprobe hisax type=11,11 protocol=2,2 id=HiSax

it responds (in the syslog) with:

kernel: ISDN subsystem Rev:
1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded
kernel: HiSax: Linux Driver for passive ISDN cards
kernel: HiSax: Version 3.5 (module)
kernel: HiSax: Layer1 Revision 1.1.4.1
kernel: HiSax: Layer2 Revision 1.1.4.1
kernel: HiSax: TeiMgr Revision 1.1.4.1
kernel: HiSax: Layer3 Revision 1.1.4.1
kernel: HiSax: LinkLayer Revision 1.1.4.1
kernel: HiSax: Total 2 cards defined
kernel: HiSax: Card 1 Protocol EDSS1 Id=HiSax (0)
kernel: HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2
kernel: PCI: Found IRQ 9 for device 00:09.0
kernel: PCI: Sharing IRQ 9 with 00:04.2
kernel: Diva: IPAC PCI card configured at 0xd0862000 IRQ 9
kernel: Diva: IPAC PCI space at 0xd086
kernel: Diva: IPAC version 1
kernel: Eicon.Diehl Diva: IRQ 9 count 1697
kernel: Eicon.Diehl Diva: IRQ 9 count 1705
kernel: HiSax: DSS1 Rev. 1.1.4.1
kernel: HiSax: 2 channels added
kernel: HiSax: MAX_WAITING_CALLS added

so it appears to be detected. I'm using the following modem.conf:

[interfaces]
context=remote
driver=i4l
language=en
type=autodetect
dialtype=tone
mode=immediate

group=1
msn=91234567
incomingmsn=*
device = /dev/ttyI0


Starting asterisk with -c returns:

 == Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] = (Generic Voice Modem Driver)
 == Parsing '/etc/asterisk/modem.conf': Found
 == Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated Modem Driver)


But if I define a test extension such as:

TRUNK=Modem/g1
exten = 2468,1,Dial(${TRUNK}/91234567:0412345678)

and try to dial it, the console says:

Dec 14 13:29:17 WARNING[15375]: chan_modem_i4l.c:608 i4l_dial:
Outgoing MSN andrew not allowed (see outgoingmsn=,, in modem.conf)
-- Called g1/91234567:0412345678
-- Modem[i4l]/ttyI0 is busy
-- Hungup 'Modem[i4l]/ttyI0'

I gather than busy is used for pretty much everything except for no
connection, but are there any suggestions of where to look?

Thanks in advance,

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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