Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Doug Crompton]

Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys functions in
that way on many of those.

I still like my "wired" phones to work like they use to. You dial a number
and it executes the call immediately.

Ok I came up with one that I think would work, maybe needs some
refinement

[out-international]
exten => _011,1,goto(process-international,s,1)

[process-international]

exten => s,1,read(number)
exten => s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten => s,3,Macro(failann,${DIALSTATUS})

This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.

The read command has many options including saying a file. You could for
instance hear "Country Code" after dialing 011. This would clue you into
the fact that you  were dialing and international call. There are also
digit limits and timeouts that can be set.

So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.

Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.

Doug

On Tue, 19 Dec 2006, Gordon Henderson wrote:

> On Tue, 19 Dec 2006, Anthony Kepler wrote:
>
> > Do you, Gordon or Doug, happen to place international calls with early-dial
> > enabled?  What kind of extensions.conf magic do you work to allow this?
> > I have been trying for some time to get this to work.  (My message from
> > 2006.11.03 regarding this is quoted just below)
>
> Not me (& I'm in the UK FWIW).
>
> I'm trying to get my users into thinking of the phones in the same terms
> as they'd treat their mobiles - so get them to dial the full area code
> starting with a zero (no 9 for outside line here, although I do support it
> in addition to zero), and then pushing the send key after they have
> entered the number... My reasoning for this is that it then mimics the way
> they use their mobiles, (and who doesn't have a mobile these days?) and
> you can dial the full number in the UK anyway without incuring any cost or
> call routing issues (just time to dial the 4 or 5 digit prefix)
>
> Gordon


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[asterisk-users] MATH function

[Rilawich Ango]

I try to use function MATH or GoToIf for checking the negative value
but CLI shows an error as following.

exten => s,11,Set(bt=${MATH(-1>0)})
func_math.c:164 builtin_function_math: '' is not a valid number

exten => s,11,GoToIf($[-1 < 0]?20)
WARNING[12926]: ast_expr2.y:729 op_negate: non-numeric argument

It seems both functions can't accept negative number for comparison.
What function can I use to compare negative number?
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Re: [asterisk-users] Billing solution

[Andrew Joakimsen]

Astpp runs two cron jobs, it writes the rate to the CDR, does it by the
accountcode.

On 12/18/06, C F <[EMAIL PROTECTED]> wrote:


Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you
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Re: [asterisk-users] Billing solution

[Giedrius Augys]

2006/12/20, C F <[EMAIL PROTECTED]>:


Well I did:
astpp
http://www.astpp.org/


On 12/20/06, Vicky <[EMAIL PROTECTED]> wrote:
> I am looking for exactly same kind of billing stuff but i dont think you
> will get it without letting ur billing program make some changes in
asterisk
> .
>
>
> On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> > a2billing
> >
> > Is very good
> >
> >
> >
> > On 12/19/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:
> > >
> > >
> > >
> > > 2006/12/19, C F <[EMAIL PROTECTED]>:
> > >
> > > > Can anyone recommend a call accounting solution with rating for
post
> > > > paid billing that works well with asterisk using the account code
or
> > > > any other info from the CDR?
> > > >
> > > > I don't want the billing software to any phone calls for me,
therefore
> > > > any solution that modifies my extensions.conf is out, nor does it
have
> > > > to allow for customers the ability to log in to check their
> > > > usage/balances.
> > > > I have looked at astbill but it looks to be way overcomplicated
for
> > > > what I want, as well as it requires realtime.
> > > > Thank you
> > > > ___
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> > >
> > >
> > > Mor and Mcc
> > >
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As I said , MCC would the best solution for you (
http://www.kolmisoft.com/). You will compile app mcc2 , and you use
this app as Dial command in
extensions.conf .
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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Lee]

Maxim Veksler wrote:


I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally unfriendly to 3rd party
changes.


That is a very good point.  I'm writing a (windows gui) tool myself and 
the way that it behaves now is to completely replace existing conf files 
on the server.  That seems a bit short sighted now in light of your 
argument.



Thanks for posting that.

--

Warm Regards,

Lee

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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Lee]

Colin Anderson wrote:

The only other thing that comes to mind is that .call files are very
sensitive to whitespace; you may have unintentially padded the .call file
with whitespace or tabs that it does not like. 


The attached .call file works on my 1.0.9 server. Maybe it can give you some
insight. 


Looking at your file, there are tabs between the pairname/separator and 
the actual value.  The example on the wiki didn't see to use tabs and I 
guess that was it because it's working now...kinda.


Good eye and thanks to everyone else of the help.


--

Warm Regards,

Lee

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Re: [asterisk-users] Echo problem

[Scott Gifford]

pixiesfr <[EMAIL PROTECTED]> writes:

> Hi,
>
> Did you try to increase echotraining ??
> echo training = 800 ..

Yes, I tried 800, 1200, and 2000; none seemed to make any difference.

Thanks!

---Scott.
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Re: [asterisk-users] Match a Numer - then continue with dialplan

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 05:19:57PM -0700, Douglas Garstang wrote:
> > -Original Message-
> > From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, December 19, 2006 4:16 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Match a Numer - then continue with
> > dialplan
> > 
> > 
> > Please correct me if I'm misunderstanding your requirements, but see
> > below (inline) for what I would do: 
> > 
> > > -Original Message-
> [snip]
> > > 
> > > [coo1_CallStart]
> > > include => coo1_OnNet
> > > include => syst_OnNet
> > > include => syst_OffNet
> > 
> > Instead of including your system-wide logic for offnet calling,
> > introduce a per-company offnet and include that instead:
> > 
> > [coo1_CallStart]
> >  include => coo1_OnNet
> >  include => syst_OnNet
> >  include => coo1_OffNet 
> > 
> > [coo1_OffNet]
> > 
> > exten => _X.,1,Set(CALLERID(NUM)=3254000)
> > exten => _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
> > exten => _X.,3,Goto(syst_OffNet,${EXTEN},1)
> 
> Bradley, If I do this, then I can no longer continue with further 
> extensions in my dialplan as Asterisk has already matched a number. 

An explicit WaitExten?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Need a 630 DID

[Shady]

Need a 630 (Naperville, IL) DID over SIP.

Please contact offlist.

Thanks,
Shady
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Re: [asterisk-users] No music on hold?

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote:
> Heya,
> 
> > I've got Asterisk 1.2.10 up and running on Debian using the back ports.
> > I noticed that it didn't come with mpg123 or depend on it and I believe
> > I read somewhere that asterisk now handles it's own mp3 playback?  Is
> > this true?  If so I must have a problem, because I hear no music when
> > putting someone on hold.  When looking at the console when putting
> > someone on hold, I see the following:
> >
> > -- Started music on hold, class 'default', on channel
> > 'IAX2/voicepulse01-3'
> > -- Stopped music on hold on IAX2/voicepulse01-3
> >
> > It says music starts and then it instantly stops.  Any ideas?
> 
> Do you have asterisk-addons installed? That could be the issue.

Why?

-- 
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] AGI Help Please

[William Piper]

Ok, further testing shows that the AGI is apparently running however the
information is still not displaying on the CLI.

Below are a few errors in the script and on a google search, although I
found people with the same error, I didn't find a resolution.

Any thoughts on what is causing this error?
Any thoughts as to why the output is not showing on the CLI without doing a
debug?

Thanks

   -- Executing AGI("SIP/216-e866", "test.php") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
AGI Tx >> agi_request: test.php
AGI Tx >> agi_channel: SIP/216-e866
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1166590405.17780
AGI Tx >> agi_callerid: 216
AGI Tx >> agi_calleridname: Billy Test
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 255
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: admin
AGI Tx >> agi_extension: 255
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: test216
AGI Tx >> >
AGI Rx << >
AGI Tx >> 510 Invalid or unknown command
AGI Rx << >
AGI Tx >> 510 Invalid or unknown command
AGI Rx << VERBOSE"There have been"
AGI Tx >> 510 Invalid or unknown command
AGI Rx << VERBOSE"125 calls made"
AGI Tx >> 510 Invalid or unknown command
   -- AGI Script test.php completed, returning 0
   -- Executing Hangup("SIP/216-e866", "") in new stack



On 12/19/06, William Piper <[EMAIL PROTECTED]> wrote:


Jay,

I just tried the suggested changes... same response.
I tested the script via command-line & it works fine.

[EMAIL PROTECTED] agi-bin]# php test.php
Content-type: text/html
X-Powered-By: PHP/4.3.9

VERBOSE"There have been"
VERBOSE"1 calls made"
[EMAIL PROTECTED] agi-bin]#

The permissions are correct:
-rwxr-xr-x  1 root root  1004 Dec 19 23:42 test.php

Any other thoughts?
Thanks,

bp

On 12/19/06, Jay Milk <[EMAIL PROTECTED] > wrote:

> Does the script run from command-line?  Without taking a close look at
> this, the include statements in the function body of connect_db look
> potentially messy.
>
> Also, any output to stdout is interpreted by asterisk as a command, so
> those fputs statements would be a problem -- do
> fputs($stdout,"VERBOSE \"There have been\"\n");
> fputs($stdout,"VERBOSE \"$row_count calls made\"\n");
>
> instead.
>
> William Piper wrote:
> > List,
> >
> > I finally decided to break down & start playing with AGI scripts, but
> > for the life of me, I can't figure out what I am doing wrong.
> >
> > Below is a super simple script to run a query in mysql to see how many
> > call records there are for the extension calling in, then print the
> > total in the CLI.
> >
> > This is all I get on the CLI:
> > -- Executing AGI("SIP/216-0baa", "test.php") in new stack
> > -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
> > -- AGI Script test.php completed, returning 0
> > -- Executing Hangup("SIP/216-0baa", "") in new stack
> >
> >
> > Here is the script:
> > #!/usr/bin/php -q
> >  > ob_implicit_flush(false);
> > set_time_limit(6);
> > $stdin  = fopen("php://stdin","r");
> > $stdout = fopen('php://stdout', 'w');
> >
> > function read() {
> >   global $stdin, $debug;
> >   $input = str_replace("\n", "", fgets($stdin, 4096));
> >   return $input;
> > }
> > function connect_db() {
> > $database="asteriskcdrdb";
> >  include("./common.php");
> >  include("./dbconnect.php");
> >  }
> >
> > // parse agi headers into array
> > while ($env=read()) {
> >   $env = str_replace("\"","",$env);
> >   $s = split(": ",$env);
> >   $agi[str_replace("agi_","",$s[0])] = trim($s[1]);
> >  if (($env == "") || ($env == "\n")) {
> >  break;
> >   }
> > }
> >
> > // main program
> > $clid = $agi[callerid];
> > connect_db();
> >
> > $query1 = "SELECT * FROM cdr WHERE dst = '$clid' ";
> > $query_result1 = @mysql_query($query1);
> > $row_count = mysql_num_rows($query_result1);
> > $row1 = @mysql_fetch_array ($query_result1);
> >
> > fputs($stdout,"There have been\n");
> > fputs($stdout,"$row_count calls made\n");
> >
> > fflush($stdout);
> > fclose($stdin);
> > fclose($stdout);
> > exit;
> > ?>
> >
> > There are no debug errors and the query is going through just fine...
> > and yes, I chmod 755.
> > Does anyone have a clue what I am doing wrong?
> >
> > Thanks,
> >
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Re: [asterisk-users] Billing solution

[C F]

Well I did:
astpp
http://www.astpp.org/


On 12/20/06, Vicky <[EMAIL PROTECTED]> wrote:

I am looking for exactly same kind of billing stuff but i dont think you
will get it without letting ur billing program make some changes in asterisk
.


On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> a2billing
>
> Is very good
>
>
>
> On 12/19/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:
> >
> >
> >
> > 2006/12/19, C F <[EMAIL PROTECTED]>:
> >
> > > Can anyone recommend a call accounting solution with rating for post
> > > paid billing that works well with asterisk using the account code or
> > > any other info from the CDR?
> > >
> > > I don't want the billing software to any phone calls for me, therefore
> > > any solution that modifies my extensions.conf is out, nor does it have
> > > to allow for customers the ability to log in to check their
> > > usage/balances.
> > > I have looked at astbill but it looks to be way overcomplicated for
> > > what I want, as well as it requires realtime.
> > > Thank you
> > > ___
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > Mor and Mcc
> >
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> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
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>
>


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Re: [asterisk-users] Billing solution

[Vicky]

I am looking for exactly same kind of billing stuff but i dont think you
will get it without letting ur billing program make some changes in asterisk
.

On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:


a2billing

Is very good

On 12/19/06, Giedrius Augys <[EMAIL PROTECTED]> wrote:
>
>
>
> 2006/12/19, C F <[EMAIL PROTECTED]>:
> >
> > Can anyone recommend a call accounting solution with rating for post
> > paid billing that works well with asterisk using the account code or
> > any other info from the CDR?
> >
> > I don't want the billing software to any phone calls for me, therefore
> >
> > any solution that modifies my extensions.conf is out, nor does it have
> > to allow for customers the ability to log in to check their
> > usage/balances.
> > I have looked at astbill but it looks to be way overcomplicated for
> > what I want, as well as it requires realtime.
> > Thank you
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> Mor and Mcc
>
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>
>

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[asterisk-users] Need Wholesale Termination

[Shady]

Looking for a good termination provider for US/Canada

Please contact offlist.

Shady
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Re: [asterisk-users] AGI Help Please

[William Piper]

Jay,

I just tried the suggested changes... same response.
I tested the script via command-line & it works fine.

[EMAIL PROTECTED] agi-bin]# php test.php
Content-type: text/html
X-Powered-By: PHP/4.3.9

VERBOSE"There have been"
VERBOSE"1 calls made"
[EMAIL PROTECTED] agi-bin]#

The permissions are correct:
-rwxr-xr-x  1 root root  1004 Dec 19 23:42 test.php

Any other thoughts?
Thanks,

bp

On 12/19/06, Jay Milk <[EMAIL PROTECTED]> wrote:


Does the script run from command-line?  Without taking a close look at
this, the include statements in the function body of connect_db look
potentially messy.

Also, any output to stdout is interpreted by asterisk as a command, so
those fputs statements would be a problem -- do
fputs($stdout,"VERBOSE \"There have been\"\n");
fputs($stdout,"VERBOSE \"$row_count calls made\"\n");

instead.

William Piper wrote:
> List,
>
> I finally decided to break down & start playing with AGI scripts, but
> for the life of me, I can't figure out what I am doing wrong.
>
> Below is a super simple script to run a query in mysql to see how many
> call records there are for the extension calling in, then print the
> total in the CLI.
>
> This is all I get on the CLI:
> -- Executing AGI("SIP/216-0baa", "test.php") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
> -- AGI Script test.php completed, returning 0
> -- Executing Hangup("SIP/216-0baa", "") in new stack
>
>
> Here is the script:
> #!/usr/bin/php -q
>  ob_implicit_flush(false);
> set_time_limit(6);
> $stdin  = fopen("php://stdin","r");
> $stdout = fopen('php://stdout', 'w');
>
> function read() {
>   global $stdin, $debug;
>   $input = str_replace("\n", "", fgets($stdin, 4096));
>   return $input;
> }
> function connect_db() {
> $database="asteriskcdrdb";
>  include("./common.php");
>  include("./dbconnect.php");
>  }
>
> // parse agi headers into array
> while ($env=read()) {
>   $env = str_replace("\"","",$env);
>   $s = split(": ",$env);
>   $agi[str_replace("agi_","",$s[0])] = trim($s[1]);
>  if (($env == "") || ($env == "\n")) {
>  break;
>   }
> }
>
> // main program
> $clid = $agi[callerid];
> connect_db();
>
> $query1 = "SELECT * FROM cdr WHERE dst = '$clid' ";
> $query_result1 = @mysql_query($query1);
> $row_count = mysql_num_rows($query_result1);
> $row1 = @mysql_fetch_array ($query_result1);
>
> fputs($stdout,"There have been\n");
> fputs($stdout,"$row_count calls made\n");
>
> fflush($stdout);
> fclose($stdin);
> fclose($stdout);
> exit;
> ?>
>
> There are no debug errors and the query is going through just fine...
> and yes, I chmod 755.
> Does anyone have a clue what I am doing wrong?
>
> Thanks,
>
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Re: [asterisk-users] Echo problem

[Jason Bachman]

As I understand it, the echo cancelers in Asterisk only work with the 
Analog cards (FXS/FXO).  If you are getting echo on a digital line, 
there is a problem with either a DAC, the T1 clocking, or you are 
getting bit errors.  You have a Switch in the middle - perhaps the 
switch is doing doing digital-analog conversions instead of sending the 
digital data straight through. The cause of the echo could very well be 
there, and the echo cancelers (even if they worked on a digital line) 
would not help because the cause of the echo is somewhere else, not at 
the Digium card.  Check your Tadiran switch for any echo cancel 
options.  I'm not familiar with that switch so I am no help to you on 
that, but I am pretty sure that its not the Digium card or Asterisk.


Regards,
--Jason Bachman

Scott Gifford wrote:

Hello,

We're in the process of setting up an Asterisk server, and are having
echo problems.  We have a Digium TE110P, and have tried the MG and
MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and
training times, and with both trunk and 1.2 branch versions of
Zaptel, Libpre, and Asterisk.  In all cases, callers from the PSTN
hear their own voice echoed back after 1.5-2 seconds; none of these
adjustments made a difference, except adjusting gain made the echo
quieter.

We followed these instructions in trying to eliminate echo:


http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html

Our lines come in from the telco in a PRI, then connect to a Tadiran
switch which hands the lines off to Asterisk over a T1 card.

We have done loopback tests with the Digium card with a loop plug in
it.

We're a bit stumped as to what to try next.  Any suggestions or
advice?

Thanks

---Scott.
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[asterisk-users] SIP and ZAP

[Maxi Belino]

Hi all,

I'm doing some coding, i'll be thankfull if anybody can help me here.

does anybody knows if ast_request_and_dial() returns differents "reasons"
when dialing to SIP and ZAP devices?
For example, if the phone of the callee is BUSY, i think
ast_request_and_dial() should return the same reason (int) no matter if the
phone is a SIP device or ZAP device
but apparently is not like that, why? is that the correct behaviour ?

may be i'm wrong,..

Thanks!!!

Maxi
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Re: [asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME

[Ex Vitorino]

 James,


 Thanks a lot for sharing the result of those debugging hours ! :-)

 I'm now left with two choices to begin with:

 1. Replacing the "." with a "-" within the dialplan

 2. Replacing the "Ubuntu Server" packaged sox version (12.17.9)
 with the most recent (12.18.2) which no longer seems to
 suffer from that sillyness...

 (yes, I did a quick new sox download/compile/test in a separate system
  and "soxmix file1.this.ext file2.that.ext mix.good.ext" started working
  with the new version !)


 Kind Regards,
--
 Ex Vito

On 12/19/06, James Fromm <[EMAIL PROTECTED]> wrote:

I spent hours debugging this a few weeks ago.

The ${UNIQUEID} contains a period (".").  Mine are something like
.xx.  When soxmix is executed to mix the in and out files, the
file types are not specified.  This causes soxmix to attempt to
determine the file type by the filename's extension.  The routine in sox
that looks for the filename's extension doesn't expect multiple periods
in the filename.  So it finds the file type to be xx.wav (or xx.gsm) and
that's not a format sox can handle.

You can add an AGI call to your dialplan immediately after the Queue
application to join the files.


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Re: [asterisk-users] No music on hold?

[Jerry]

Heya,

> I've got Asterisk 1.2.10 up and running on Debian using the back ports.
> I noticed that it didn't come with mpg123 or depend on it and I believe
> I read somewhere that asterisk now handles it's own mp3 playback?  Is
> this true?  If so I must have a problem, because I hear no music when
> putting someone on hold.  When looking at the console when putting
> someone on hold, I see the following:
>
> -- Started music on hold, class 'default', on channel
> 'IAX2/voicepulse01-3'
> -- Stopped music on hold on IAX2/voicepulse01-3
>
> It says music starts and then it instantly stops.  Any ideas?

Do you have asterisk-addons installed? That could be the issue.

J.
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Re: [asterisk-users] Is logic right?

[Mayo Jordanov]

Shouldn't the second line (the one with HANGUP) have priority 2  
rather than 1? So it would read:

exten => s/9185415897,2,Hangup
?

regards,
mayo


On Dec 19, 06, at 14:54 , Michael Sullivan wrote:


OK.  My basic asterisk install seems to be working.  I can get caller
ID.  My dialplan says:

[incoming]
; incoming calls from the FXO port are directed to this context from
zapata.conf
exten => s/9185415897,1,Set(CALLERID(name)=Michael Sullivan)
exten => s/9185415897,1,HANGUP(1)
exten => s,1,Set(CALLERID(name)=Someone Else)

This is for testing.  It's supposed to check the caller ID to see  
who is

calling, and if it's my cell phone, hang up, but let any other number
through to ring on our handsets.  Instead it rewrites the caller ID  
name

to Michael Sullivan and allows the call to pass through.  Here is the
output of the call to the CLI:

-- Starting simple switch on 'Zap/1-1'
-- Executing Set("Zap/1-1", "CALLERID(name)=Michael Sullivan") in
new stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Dec 19 16:51:37 NOTICE[7413]: chan_zap.c:6073 ss_thread: Got event 18
(Ring Begin)...
-- Executing Set("Zap/1-1", "CALLERID(name)=Someone Else") in new
stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'

In fact, the "hang up on my number" clause isn't even shown in the
dialplan in the CLI:

camille*CLI> show dialplan incoming
[ Context 'incoming' created by 'pbx_config' ]
  's' =>1. Set(CALLERID(name)=Michael Sullivan)
[pbx_config]
  's' =>1. Set(CALLERID(name)=Someone Else)
[pbx_config]
camille*CLI>
-= 2 extensions (2 priorities) in 1 context. =-
camille*CLI>


What am I doing wrong?





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Re: [asterisk-users] Follow-me challenge

[Chris Johnson]

On 12/18/06, Eric Jacksch <[EMAIL PROTECTED]> wrote:


 Is the problem just when you don't answer the cell phone?  Many cell
phones go to a voice announcement when they're turned off or not answered,
and Asterisk thinks the call has been answered. The other issue could be
that your gateway (asterisk1) is answering the call before the outbound leg
is answered.  One workaround would be to use a macro that requires you to
press a key to accept the call on your cell.  (See the M option to the dial
command and 
http://www.voip-info.org/wiki/view/Asterisk+tips+findme)

Also, I see that you're using the r option — you might want to drop that.

I'm also not convinced that it will ever find 102,107 in your dialplan.
 You might want to look at using ${DIALSTATUS} and making it a bit more
explicit.

Cheers,
Eric



Dropped the r option and line 107.
The M option had the same result.
Based on another comment :

Is your other server patching through to a Zap channel (analog)?
If so, as soon as the dial goes out, an analog Zap channel is considered
"answered", which could be your issue.

Doesn't sound like follow-me will work properly with an analog trunk
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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

Leo, sorry I completely don't follow you. I don't see how the registry 
(astdb) can help me here.


-Original Message-
From:   Leo Ann Boon [mailto:[EMAIL PROTECTED]
Sent:   Tue 12/19/2006 6:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 
Subject:Re: [asterisk-users] Match a Numer - then continue with dialplan

Douglas Garstang wrote:
> I just know someone is going to ask 'why would you ever want to do that?'. 
> Here's my answer.
>
> We have two companies, each with a dialplan similar to what's below. In the 
> event that the number being dialled does not match any number within our OWN 
> company, we want to set the caller id to be a generic one for the company, 
> NOT one for the user. This is a pretty normal requirement that most companies 
> want. So, in the event that the logic flows beyond coo1_OnNet, we want to 
> reset the caller id of say, 3254001 , to 3254000 . If 
> there was a way to match against a number in the dialplan, and then continue 
> execution after that point, we could put this statement at the end of the 
> coo1_OnNet context and it would all be sweet. Without that, I don't have a 
> clue how to do this... unless we stick with out current 3,000 line python 
> script.
>   
If you're not using realtime to store your SIP registry, you should be 
able to look up the number in the family SIP/Registry (case sensitive) 
using the DB functions. If you're using realtime, then you'll have to do 
an SQL query.

Leo

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Re: [asterisk-users] BLF on GXP2000

[Chris Johnson]

Rebooted the phone...No luck

On 12/19/06, Ken Williams <[EMAIL PROTECTED]> wrote:


 One thing I've noticed, is any time I make changes to Asterisk I have to
reboot the phones to keep BLF up to date.  Have you tried that?

 --
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Chris Johnson
*Sent:* Monday, December 18, 2006 6:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] BLF on GXP2000

Well, I am making some progress. I have made some changes as defined below
and now have a green line on the BLF, but it still does not indicate when
the extension receives a call or goes off hook.

Here are the changes:
the [ext-local-custom] context no longer exists
the subscribecontext in sip.con no longer exists

[internal]
exten => 101,1,Macro(voicemail,${polycom430})
exten => 101,hint,${polycom430}

Asterisk 1.4.0b3
*CLI> show hints

-= Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED]: SIP/101
State:Idle   Watchers  1

- 1 hints registered


On 12/18/06, Ken Williams <[EMAIL PROTECTED]> wrote:
>
>  Here's what I have, it's to early for me to think so hopefully looking
> at mine helps :D
>
> extensions.conf:
>
> [ext-local]
> exten => 701,1,Macro(exten-vm,701,701)
> exten => 701,n,Hangup
> exten => 701,hint,SIP/701
> sip.conf:
>
> [701]
> type=friend
> secret=1234
> record_out=Adhoc
> record_in=Adhoc
> qualify=yes
> port=5060
> nat=yes
> host=dynamic
> dtmfmode=rfc2833
> dial=SIP/701
> context=from-internal
> canreinvite=no
> callerid=device <701>
> mailbox=701
> If this doesn't help in some fashion let me know and I'll think it
> through a little later...off to get some coffee.
>
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Chris Johnson
> *Sent:* Sunday, December 17, 2006 4:50 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] BLF on GXP2000
>
> I am trying to set up the BLF on a GXP2000.
> Currently what I have is
> extensions.conf:
> [globals]
> polycom430=SIP/101
>
> [internal]
> exten => 101,1,Macro(voicemail,${polycom430})
>
> [macro-voicemail]
> exten => s,1,Dial(${ARG1},10,tT)
> exten => s,2,VoiceMail([EMAIL PROTECTED] )
> exten => s,102,VoiceMail([EMAIL PROTECTED])
>
> [ext-local-custom]
> exten => 101,hint,${polycom430}
>
>
> sip.conf:
> [general]
> subscribecontext=ext-local-custom
>
> And have set up the key to Asterisk BLF with UserID101
>
> When I reload the phone, I get the following error:
> [Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064
> handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 
192.168.1.248
> , but there is no hint for that extension
>
>
> Any help is greatly appreciated.
>
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>
>

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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13

[Lee Howard]

Jean-Yves Avenard wrote:


Hi

On 12/20/06, Lee Howard <[EMAIL PROTECTED]> wrote:


This thread seems like an awfully crazy amount of work to get fax
working when using IAXmodem and HylaFAX would do it without the
headache, most likely.



Does IAXmodem allows you to receive faxes with any extensions
(auto-detecting incoming faxes). 



Sure, I guess.  The fax detection part comes from Asterisk or OpenPBX or 
whatever.  Same as with rxfax/txfax, etc.


Lee.
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13

[pixiesfr]

Hi,

No IaxModem is only a "modem simulator".

Let asterisk do the difference, and send it to you iax extension...


@++.

Jean-Yves Avenard a écrit :

Hi

On 12/20/06, Lee Howard <[EMAIL PROTECTED]> wrote:

This thread seems like an awfully crazy amount of work to get fax
working when using IAXmodem and HylaFAX would do it without the
headache, most likely.


Does IAXmodem allows you to receive faxes with any extensions
(auto-detecting incoming faxes).

JY
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Re: [asterisk-users] Match a Numer - then continue with dialplan

[Leo Ann Boon]

Douglas Garstang wrote:

I just know someone is going to ask 'why would you ever want to do that?'. 
Here's my answer.

We have two companies, each with a dialplan similar to what's below. In the event that the 
number being dialled does not match any number within our OWN company, we want to set the 
caller id to be a generic one for the company, NOT one for the user. This is a pretty 
normal requirement that most companies want. So, in the event that the logic flows beyond 
coo1_OnNet, we want to reset the caller id of say, 3254001 , to 3254000 
. If there was a way to match against a number in the dialplan, and then 
continue execution after that point, we could put this statement at the end of the 
coo1_OnNet context and it would all be sweet. Without that, I don't have a clue how to do 
this... unless we stick with out current 3,000 line python script.
  
If you're not using realtime to store your SIP registry, you should be 
able to look up the number in the family SIP/Registry (case sensitive) 
using the DB functions. If you're using realtime, then you'll have to do 
an SQL query.


Leo

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Re: [asterisk-users] Echo problem

[pixiesfr]

Hi,

Did you try to increase echotraining ??
echo training = 800 ..

@++

Hello,

We're in the process of setting up an Asterisk server, and are having
echo problems.  We have a Digium TE110P, and have tried the MG and
MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and
training times, and with both trunk and 1.2 branch versions of
Zaptel, Libpre, and Asterisk.  In all cases, callers from the PSTN
hear their own voice echoed back after 1.5-2 seconds; none of these
adjustments made a difference, except adjusting gain made the echo
quieter.

We followed these instructions in trying to eliminate echo:


http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html

Our lines come in from the telco in a PRI, then connect to a Tadiran
switch which hands the lines off to Asterisk over a T1 card.

We have done loopback tests with the Digium card with a loop plug in
it.

We're a bit stumped as to what to try next.  Any suggestions or
advice?

Thanks

---Scott.
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13

[Jean-Yves Avenard]

Hi

On 12/20/06, Lee Howard <[EMAIL PROTECTED]> wrote:

This thread seems like an awfully crazy amount of work to get fax
working when using IAXmodem and HylaFAX would do it without the
headache, most likely.


Does IAXmodem allows you to receive faxes with any extensions
(auto-detecting incoming faxes).

JY
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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Al Bochter]

Thanks Bob I will have to download the updated ver. then
Don't mind me I had a brain fart.. :-[

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Bob Chiodini wrote:


The free version 1.31 has all 16 "keys" on the keypad.

Bob...

Al Bochter wrote:


Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 
DTMF tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but 
first I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
"A B C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM





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Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 4:05:10 PM





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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

> -Original Message-
> From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 19, 2006 4:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Match a Numer - then continue with
> dialplan
> 
> 
> Please correct me if I'm misunderstanding your requirements, but see
> below (inline) for what I would do: 
> 
> > -Original Message-
[snip]
> > 
> > [coo1_CallStart]
> > include => coo1_OnNet
> > include => syst_OnNet
> > include => syst_OffNet
> 
> Instead of including your system-wide logic for offnet calling,
> introduce a per-company offnet and include that instead:
> 
> [coo1_CallStart]
>  include => coo1_OnNet
>  include => syst_OnNet
>  include => coo1_OffNet 
> 
> [coo1_OffNet]
> 
> exten => _X.,1,Set(CALLERID(NUM)=3254000)
> exten => _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
> exten => _X.,3,Goto(syst_OffNet,${EXTEN},1)

Bradley, If I do this, then I can no longer continue with further extensions in 
my dialplan as Asterisk has already matched a number. I still need to check 
black/white lists, set pic codes and rate centers, 4 digit extensions etc 
within the company context. I just need to set the caller id and then move on. 
If I goto over to ${EXTEN} within syst_OffNet, I'd have to put ALL this logic 
within that extension, which would mean potentiall several hundred priorities. 
Asterisk really does need a way to match a number, execute some code, and then 
go back to looking for extensions.
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13

[Lee Howard]

This thread seems like an awfully crazy amount of work to get fax 
working when using IAXmodem and HylaFAX would do it without the 
headache, most likely.


Lee.
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[asterisk-users] Echo problem

[Scott Gifford]

Hello,

We're in the process of setting up an Asterisk server, and are having
echo problems.  We have a Digium TE110P, and have tried the MG and
MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and
training times, and with both trunk and 1.2 branch versions of
Zaptel, Libpre, and Asterisk.  In all cases, callers from the PSTN
hear their own voice echoed back after 1.5-2 seconds; none of these
adjustments made a difference, except adjusting gain made the echo
quieter.

We followed these instructions in trying to eliminate echo:


http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html

Our lines come in from the telco in a PRI, then connect to a Tadiran
switch which hands the lines off to Asterisk over a T1 card.

We have done loopback tests with the Digium card with a loop plug in
it.

We're a bit stumped as to what to try next.  Any suggestions or
advice?

Thanks

---Scott.
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[asterisk-users] Using Asterisk/Digium card with Tadiran switch

[Scott Gifford]

Hello,

We've got an Asterisk server with a Digium TE110P card, connected to a
Tadiran Coral Flexicom IPX 500 switch using a T1 card.  We are having
echo problems on the lines coming in from the Digium card.

I was wondering if anybody is successfully using a Digium card and
Asterisk with a Tadiran switch, and if so whether they could share
some configuration information?

Thanks!

Scott.

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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Michiel van Baak]

On 00:57, Wed 20 Dec 06, Tzafrir Cohen wrote:
> I'm quite pessimistic regarding tools that are supposed to tolerate any
> sort of manual override with Asterisk's configuration. At least tools
> that are aimed to give a simplified/more powerful user interface rather
> than a glorified text editor with macros.

I agree here, if it's not for customers.
I'm not going to give them ssh/scp/ftp access so we need
some kind of tool to do the basic and not-so-basic configs.

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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[asterisk-users] AstManProxy - Manager

[Daniel Gradecak]

Hello,

I cannot find documnetation explaining how to access AstManProxy. I am 
working with Asterisk Java and accessing Asterisk Manager. I wonder if
AStManProxy is using the same API as Manager? Can I access it with 
Asterisk java too ?


Regards,
Daniel
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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Watkins, Bradley]

Please correct me if I'm misunderstanding your requirements, but see
below (inline) for what I would do: 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Douglas Garstang
> Sent: Tuesday, December 19, 2006 5:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Match a Numer - then continue 
> with dialplan
> 
> I just know someone is going to ask 'why would you ever want 
> to do that?'. Here's my answer.
> 
> We have two companies, each with a dialplan similar to what's 
> below. In the event that the number being dialled does not 
> match any number within our OWN company, we want to set the 
> caller id to be a generic one for the company, NOT one for 
> the user. This is a pretty normal requirement that most 
> companies want. So, in the event that the logic flows beyond 
> coo1_OnNet, we want to reset the caller id of say, 3254001 
> , to 3254000 . If there was a way to match 
> against a number in the dialplan, and then continue execution 
> after that point, we could put this statement at the end of 
> the coo1_OnNet context and it would all be sweet. Without 
> that, I don't have a clue how to do this... unless we stick 
> with out current 3,000 line python script.
> 
> [coo1_CallStart]
> include => coo1_OnNet
> include => syst_OnNet
> include => syst_OffNet

Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:

[coo1_CallStart]
 include => coo1_OnNet
 include => syst_OnNet
 include => coo1_OffNet 

[coo1_OffNet]

exten => _X.,1,Set(CALLERID(NUM)=3254000)
exten => _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten => _X.,3,Goto(syst_OffNet,${EXTEN},1)


The rest of this can stay untouched.


> [coo1_OnNet]
> 
> exten => 3254101,1,Dial(SIP/3254101,20,tr)
> exten => 3254102,1,Dial(SIP/3254102,20,tr)
> exten => 3254103,1,Dial(SIP/3254103,20,tr)
> 
> exten => 1000,1,Answer
> exten => 1000,2,Wait(1)
> exten => 1000,3,NoOp(${FOO})
> 
> [syst_OnNet]
> include => coo1_OnNet
> include => coo2_OnNet
> 
> [syst_OffNet]
> exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)

Regards,
- Brad
The contents of this e-mail are intended for the named addressee only. It 
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Re: [asterisk-users] bridging isdn calls to free up channels

[Paul Hales]

Option 7a - use a VOIP provider, and tie up no lines.

PaulH


On Mon, 2006-12-18 at 11:35 +1100, James Harper wrote:
> I was incorrect in a previous email... The situation in question is
> this:
> 
> Asterisk <---BRI---> PBX <---BRI---> PSTN
> 
> There are Samsung extensions on the PBX and SIP extensions on Asterisk.
> I want to be able to use TAPI to initiate dialling, and the PBX has no
> such feature so Asterisk must initiate it.
> 
> For an Asterisk initiated call from a PBX extension to a PSTN number,
> this works as follows:
> 1. TAPI (eg Outlook) sends the instruction to Asterisk
> 2. Asterisk calls the extension.
> 3. The extension answers
> 4. Asterisk dials the PSTN number
> 5. Asterisk joins the ends together
> 
> This works great except it ties up two BRI channels between asterisk and
> the PBX for the duration of the call. Is there a trick I can do to tell
> the PBX to join the channels together internally? A transfer or
> something?
> 
> I'm using mISDN at the moment, but I guess I could use CAPI if the
> required features were missing from mISDN...
> 
> Thanks
> 
> James
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[asterisk-users] Cisco devices (without STUN) and dynamic NAT

[Brad Templeton]

Cisco devices (7912, ata-168, 7960 etc.) don't support STUN. 

However, they do let you define a static external NAT IP
address, and parameters to send a keep-alive out through the
NAT on a regular basis.

However, I want to make these devices work in an environment
where they are behind a NAT which has a dynamic IP which
might change (though in fact it changes only rarely.)

They're talking to an external Asterisk server which is
of course not behind NAT.

The docs say that nat=yes will cause Asterisk to ignore the
IPs in the SIP headers and SDP, and replace them with the
actual address the packets are coming from.

I thought this meant that if I put any address in the Cisco's
NATIP field, this would work because Asterisk would rewrite
the SDP to the real address, which might have changed since
the NATIP field was set.

However, Asterisk is not doing this (though it's doing
some other interesting things, now noticing that the NAT
address in the Via header is local to it and talking
directly to that) and it's trying to do native bridged
channels to other devices, which isn't working with
the wrong address in the SDP.

I want native bridging of course, in fact it's a must
if you have a phone on the east coast, an asterisk
server on the west coast and a SIP terminator in
the middle.  No way you want to hairpin the audio, and
with STUN supporting SIP phones this works fine.


canreinvite is not involved here because there is no
need for reinvite on a simple call.

I'm using svn trunk, as of a few days ago.
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Gordon Henderson]

On Tue, 19 Dec 2006, Anthony Kepler wrote:

Do you, Gordon or Doug, happen to place international calls with early-dial 
enabled?  What kind of extensions.conf magic do you work to allow this?
I have been trying for some time to get this to work.  (My message from 
2006.11.03 regarding this is quoted just below)


Not me (& I'm in the UK FWIW).

I'm trying to get my users into thinking of the phones in the same terms 
as they'd treat their mobiles - so get them to dial the full area code 
starting with a zero (no 9 for outside line here, although I do support it 
in addition to zero), and then pushing the send key after they have 
entered the number... My reasoning for this is that it then mimics the way 
they use their mobiles, (and who doesn't have a mobile these days?) and 
you can dial the full number in the UK anyway without incuring any cost or 
call routing issues (just time to dial the 4 or 5 digit prefix)


Gordon


 >
On 11/3/06, *Anthony Kepler* <[EMAIL PROTECTED] 
> wrote:I am trying to allow users to 
place outgoing international calls from a
GXP-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1 


I have the following extension line:
exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial "011254"...etc.
and I get this on the asterisk console:
Executing Dial("SIP/1001-081fb718", "Zap/g1/0112") in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the "."
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set("SIP/1001-081fb718", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED]  | SIP/Email



Gordon Henderson wrote:

On Sun, 5 Nov 2006, Doug Crompton wrote:



On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 "Line"
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon



Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:



Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:



I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that 
when

dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?


Set the "Early Dial" option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the "send" button...

Gordon


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Re: [asterisk-users] NAT and Dial to two channels at once

[Brad Templeton]

On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote:
> You need to understand how NAT works, if you can chan2 and chan2 is behind a
> NAT and suddenly someone else is invited to chan2's IP address port 5060
> chan2's router willl say "WTF I dont have an estabished connection on port
> 5060" (to the client being reinvited to chan2) and it wont work. You need to
> have the media path go through asterisk in that case.

Actually, it's more complex than that.

If the NAT box has had a hole poked (in its config) for the RTP port (SIP
port is only used by Asterisk) then any machine can send it RTP on that
port.

In addition, if the NAT is of the "full cone" type, any host can send to
your port once you have sent a packet out that port.

With Restricted cone and Port restricted cones, it also works as long as
the Natted IP phone is sending packets out to the other host already.
Which it should be if we have symmetric RTP.

Symmetric NATs, which are rare, will change the port number when they
start talking to a different host for RTP.  This will screw up all but
the cleverest implementations.  (Though there are endpoints that notice
if the RTP is coming from a port other than they were told, and start
sending to that instead of the one in the SDP)

What doesn't work is assymetric RTP with NAT.   In this case we have
the audio going through asterisk in one direction, and directly in
the other direction.  That will fail if the direct direction tries
to go into a nat (it should work if it's only leaving a nat)

Asterisk currently does assymetric RTP if it thinks it only has to
listen to one end of the audio path.  That's a good idea in
general -- but not one that works through anything but a
manually opened NAT.


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Re: [asterisk-users] AGI Help Please

[William Piper]

I am running console. I'm a newbie for AGI's but not that new.

Thanks,

bp


On 12/19/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:


On Tue, Dec 19, 2006 at 09:35:56AM +0200, yusuf wrote:

> to see debug output for AGI's, you *must* be connected to the first Ast
> terminal.  So start Asterisk like 'asterisk -cvv', then you
> will see output from your AGI.

Actually, if you have not started Asterisk in a console, it will be sent
to the first remote client. Thus you won't exit Asterisk if you
accidentally press Ctrl-C, as you're used to ;-)

--
  Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] 2 devices using same sip account

[Gordon Henderson]

On Tue, 19 Dec 2006, Carla Schroder wrote:


Your phones only register once, when they first start up. Seems to me that
having multiple phones on the same account is asking for trouble- why not set
up multiple accounts in the usual way, and create a ring group for all the
phones you want to use? Like this example that rings two phones at the same
time:

exten => 100,1,Dial(SIP/101&SIP/102,30,t)
exten => 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the
same: one user with many phones, one extension, one voicemail box.


I've been setting up a few systems recently with a SIP account and an IAX 
account (same passwords, CLI, etc.) and having the users use a SIP 
hardphone for the office desk, and an IAX (idefisk) softphone for 
out-of-office calls. (My Dial() calls both accounts, so both phones ring)


It saves hassles with NAT, etc. for remote SIP phones too.

No good if you only have SIP phones though!

Gordon
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Re: [asterisk-users] Billing solution

[Carlos Rojas]

a2billing

Is very good

On 12/19/06, Giedrius Augys <[EMAIL PROTECTED]> wrote:




2006/12/19, C F <[EMAIL PROTECTED]>:
>
> Can anyone recommend a call accounting solution with rating for post
> paid billing that works well with asterisk using the account code or
> any other info from the CDR?
>
> I don't want the billing software to any phone calls for me, therefore
> any solution that modifies my extensions.conf is out, nor does it have
> to allow for customers the ability to log in to check their
> usage/balances.
> I have looked at astbill but it looks to be way overcomplicated for
> what I want, as well as it requires realtime.
> Thank you
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Mor and Mcc

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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 11:33:55PM +0200, Maxim Veksler wrote:
> On 12/17/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
> >> Hi list,
> >>
> >> It's been a while since I've done asterisk stuff, and I'm wondering if
> >> there any news in the field.
> >>
> >> What do you people use today for http management of debian based Asterisk
> >> setup?
> >> Preferably something with the proven ".deb" extension.
> >
> >"destar" has a "tar" extension, but a prefix quite similar to "Debian".
> >Availble in Etch.
> >
> >FreePBX is not yet availble, though a Sarge-based CD that includes it
> >and generally works could be downloaded from
> >http://updates.xorcom.com/iso/rapid-current.iso and the package is
> >availble from
> >
> >  deb http://updates.xorcom.com/rapid future main
> >
> 
> Hi Tzafrir,
> Thank you!
> 
> I am aware of both of these tools, I don't like them!
> They make absolute changes in your /etc/asterisk/* files, they assume
> that they are the only thing you will be using for managing your
> asterisk pbx and they are both totally unfriendly to 3rd party
> changes.
> 
> Are there any other tools then, perhaps some that has not been debianized 
> yet?
> I'd like something that could be more cooperative with user hand made 
> changes.

You refer to direct addition of content to the cnfig files. But this is
not the only way to affect the output.

If you consider those interfaces to be "code generators", you can
manipulate their inputs (the mysql database in the case of FreePBX, the
set of pthon ojects for DeStar) or the generator itself (patch the
code).

Sometimes patching the code is trivial (e.g: disabling a line from being
generaed). Sometimes it's more complicated.


I'm quite pessimistic regarding tools that are supposed to tolerate any
sort of manual override with Asterisk's configuration. At least tools
that are aimed to give a simplified/more powerful user interface rather
than a glorified text editor with macros.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Is logic right?

[Michael Sullivan]

OK.  My basic asterisk install seems to be working.  I can get caller
ID.  My dialplan says:

[incoming]
; incoming calls from the FXO port are directed to this context from
zapata.conf
exten => s/9185415897,1,Set(CALLERID(name)=Michael Sullivan)
exten => s/9185415897,1,HANGUP(1)
exten => s,1,Set(CALLERID(name)=Someone Else)

This is for testing.  It's supposed to check the caller ID to see who is
calling, and if it's my cell phone, hang up, but let any other number
through to ring on our handsets.  Instead it rewrites the caller ID name
to Michael Sullivan and allows the call to pass through.  Here is the
output of the call to the CLI:

-- Starting simple switch on 'Zap/1-1'
-- Executing Set("Zap/1-1", "CALLERID(name)=Michael Sullivan") in
new stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Dec 19 16:51:37 NOTICE[7413]: chan_zap.c:6073 ss_thread: Got event 18
(Ring Begin)...
-- Executing Set("Zap/1-1", "CALLERID(name)=Someone Else") in new
stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'

In fact, the "hang up on my number" clause isn't even shown in the
dialplan in the CLI:

camille*CLI> show dialplan incoming
[ Context 'incoming' created by 'pbx_config' ]
  's' =>1. Set(CALLERID(name)=Michael Sullivan)
[pbx_config]
  's' =>1. Set(CALLERID(name)=Someone Else)
[pbx_config]
camille*CLI> 
-= 2 extensions (2 priorities) in 1 context. =-
camille*CLI> 


What am I doing wrong?





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Re: [asterisk-users] No music on hold?

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 04:49:36PM -0500, Phil Finkler wrote:
> Hi all,
> 
>  
> 
> I've got Asterisk 1.2.10 up and running on Debian using the back ports.

Debian does not include the default MoH files that come with Debian for
legal reasons. Get some sound files in the moh directory, basically, and
use the naitve moh.

Grab
http://updates.xorcom.com/rapid/pool/main/f/freepbx/asterisk-sounds-moh-freepbx_2.1.3.dfsg-1.2902_all.deb
 
or an equivalent.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Colin Anderson]

The only other thing that comes to mind is that .call files are very
sensitive to whitespace; you may have unintentially padded the .call file
with whitespace or tabs that it does not like. 

The attached .call file works on my 1.0.9 server. Maybe it can give you some
insight. 

-Original Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
wo rk


Colin Anderson wrote:
> If you are using Windows to generate the .call files, make sure they are
in
> Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files.
Use
> Crimson Editor www.crimsoneditor.com to make the file, and click Document
>
> File Format > Unix Format. 
> 
> I ran into this same problem, and it turns out my Asterisk install would
not
> use Windows-formatted text files, it would just ignore them and delete
them.
> 
> 

Hi Colin,

Thanks for responding.  I've run into the problem elsewhere myself. 
Alas, I wrote the call file using nano on the linux box through ssh/putty.

-- 

Warm Regards,

Lee

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614.call
Description: Binary data
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Re: [asterisk-users] Need help getting started with asterisk

[Time Bandit]

Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all.  Is there some explicit thing I
need to put in to get the caller ID?

callerid=asreceived
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RE: [asterisk-users] T1 Pri Question

[Colin Anderson]

Short answer: a single group should be fine. Long answer: It depends. 

Your Dial() command determines the order in which Asterisk plucks channels
from your PRI. Most north american system call inbound channel 1 first, then
2, etc. It makes sense, then for you to take channels from the topmost
first, for outbound calls. This is dictacted by how you format your Dial()
command: 

g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group). 
G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group). 
r: use a round-robin search, starting at the next highest channel than last
time (aka. ascending rotary hunt group). 
R: use a round-robin search, starting at the next lowest channel than last
time (aka. descending rotary hunt group). 

(from voip-info.org)

So, a dial command like:

exten => 1,Dial(Zap/g0/5551212)

would take the channel from the bottom of your group. 

This may have bearing on your situation. However, from what you are
descibing, this seems to be a symptom of a larger problem, that of Asterisk
not correctly hanging up the zap channel and that has nothing to do with
channel selection preferences. Anything weird on your CLI during this
period? 

-Original Message-
From: Rob Schall [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] T1 Pri Question


When setting up the zaptel, etc Is it necessary to have a seperate
group for incoming and one for outgoing calls? Either way, does asterisk
always know which channels are open, and does it always clear up a
channel for use once a call completes?

Reason for asking... After dialing into our system over a few DID
numbers, I noticed you can only call 2-3 times before getting a busy
signal. However we have a full 24 channel PRI. During this time, you are
more than able to make outgoing calls over that same PRI. After hanging
up on the incoming call, (outside into the PRI), it can take upto a few
minutes to clear up for the next person to call in.

Any thoughts,
Rob

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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

> -Original Message-
> From: David Thomas [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 19, 2006 3:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Match a Numer - then continue with
> dialplan
> 
> 
> On 12/19/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > Anyone know if there's a way to match a dialplan extension, 
> execute some code, say set a variable, and then continue with 
> the dialplan?
> >
> > I want to set a variable when the dialplan flows beyond a 
> certain context. This would be a great feature.
> >
> > Doug.
> 
> Have you tried using the SetVar cmd? I haven't tried it but it sounds
> like it might work for this.
> 
> http://www.voip-info.org/wiki/view/Asterisk+variables
> 
> Regards,
> David

David,

If I call setvar, my variable will be set, but dialplan processing will stop...
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RE: [asterisk-users] No music on hold?

[Kevin Trumbull]

I had this same problem.  I also read that mpg123 was not required, but it 
actually is if you wish to use mp3 files.  I just decided to go with RAW files 
because I had problems converting some mp3's to the appropriate bit rate.

http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it

--
Kevin Trumbull

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Finkler
Sent: Tuesday, December 19, 2006 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No music on hold?


Hi all,

I've got Asterisk 1.2.10 up and running on Debian using the back ports.  I 
noticed that it didn't come with mpg123 or depend on it and I believe I read 
somewhere that asterisk now handles it's own mp3 playback?  Is this true?  If 
so I must have a problem, because I hear no music when putting someone on hold. 
 When looking at the console when putting someone on hold, I see the following:

-- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3'
-- Stopped music on hold on IAX2/voicepulse01-3

It says music starts and then it instantly stops.  Any ideas?

Thanks,
Phil
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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

I just know someone is going to ask 'why would you ever want to do that?'. 
Here's my answer.

We have two companies, each with a dialplan similar to what's below. In the 
event that the number being dialled does not match any number within our OWN 
company, we want to set the caller id to be a generic one for the company, NOT 
one for the user. This is a pretty normal requirement that most companies want. 
So, in the event that the logic flows beyond coo1_OnNet, we want to reset the 
caller id of say, 3254001 , to 3254000 . If there was a way 
to match against a number in the dialplan, and then continue execution after 
that point, we could put this statement at the end of the coo1_OnNet context 
and it would all be sweet. Without that, I don't have a clue how to do this... 
unless we stick with out current 3,000 line python script.

[coo1_CallStart]
include => coo1_OnNet
include => syst_OnNet
include => syst_OffNet

[coo1_OnNet]

exten => 3254101,1,Dial(SIP/3254101,20,tr)
exten => 3254102,1,Dial(SIP/3254102,20,tr)
exten => 3254103,1,Dial(SIP/3254103,20,tr)

exten => 1000,1,Answer
exten => 1000,2,Wait(1)
exten => 1000,3,NoOp(${FOO})

[syst_OnNet]
include => coo1_OnNet
include => coo2_OnNet

[syst_OffNet]
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)



~  


> -Original Message-
> From: Douglas Garstang 
> Sent: Tuesday, December 19, 2006 2:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Match a Numer - then continue with dialplan
> 
> 
> Anyone know if there's a way to match a dialplan extension, 
> execute some code, say set a variable, and then continue with 
> the dialplan?
> 
> I want to set a variable when the dialplan flows beyond a 
> certain context. This would be a great feature.
> 
> Doug.
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[asterisk-users] T1 Pri Question

[Rob Schall]

When setting up the zaptel, etc Is it necessary to have a seperate
group for incoming and one for outgoing calls? Either way, does asterisk
always know which channels are open, and does it always clear up a
channel for use once a call completes?

Reason for asking... After dialing into our system over a few DID
numbers, I noticed you can only call 2-3 times before getting a busy
signal. However we have a full 24 channel PRI. During this time, you are
more than able to make outgoing calls over that same PRI. After hanging
up on the incoming call, (outside into the PRI), it can take upto a few
minutes to clear up for the next person to call in.

Any thoughts,
Rob

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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Steve Edwards]

I downloaded the free 1.37 version. The "slide out" keypad displays "ABCD" 
buttons but they do not respond to clicks. You can enter "ABCD" using your 
keyboard and they will be sent to Asterisk.


There appears to be some "funkyness" with case though.

If you enter "" and click dial, "" is sent to Asterisk.

If you then enter "A", "Aaaa" is displayed, matching the previous entry 
ignoring case. If you continue typing "AAA" and click dial, "Aaaa" is sent 
to Asterisk.


On Tue, 19 Dec 2006, Zoa wrote:



Hmm, if the latest free version does not have all 16 keys, email 
[EMAIL PROTECTED], there should not be a difference in the amount of 
DTMF keys between biz and free version.


Zoa

Bob Chiodini wrote:

The free version 1.31 has all 16 "keys" on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF 
tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but first I 
must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will accept 
it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys "A B 
C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Match a Numer - then continue with dialplan

[David Thomas]

On 12/19/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

Anyone know if there's a way to match a dialplan extension, execute some code, 
say set a variable, and then continue with the dialplan?

I want to set a variable when the dialplan flows beyond a certain context. This 
would be a great feature.

Doug.


Have you tried using the SetVar cmd? I haven't tried it but it sounds
like it might work for this.

http://www.voip-info.org/wiki/view/Asterisk+variables

Regards,
David
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[asterisk-users] No music on hold?

[Phil Finkler]

Hi all,

 

I've got Asterisk 1.2.10 up and running on Debian using the back ports.
I noticed that it didn't come with mpg123 or depend on it and I believe
I read somewhere that asterisk now handles it's own mp3 playback?  Is
this true?  If so I must have a problem, because I hear no music when
putting someone on hold.  When looking at the console when putting
someone on hold, I see the following:

 

-- Started music on hold, class 'default', on channel
'IAX2/voicepulse01-3'

-- Stopped music on hold on IAX2/voicepulse01-3

 

It says music starts and then it instantly stops.  Any ideas?

 

Thanks,

Phil 

 

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[asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

Anyone know if there's a way to match a dialplan extension, execute some code, 
say set a variable, and then continue with the dialplan?

I want to set a variable when the dialplan flows beyond a certain context. This 
would be a great feature.

Doug.
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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Michiel van Baak]

On 23:33, Tue 19 Dec 06, Maxim Veksler wrote:
> Are there any other tools then, perhaps some that has not been debianized 
> yet?
> I'd like something that could be more cooperative with user hand made 
> changes.

Thanks to the folks at SpeakUp (http://www.speakup.nl) I
have a nice webtool that puts everything in seperate files
you can #include in your normal asterisk configs.
We use it on one of our PBX boxen next to several custom
made stuff.

Speakup granted me the rights to release the version in GPL.
I've been too busy to prepare a release but with holidays
coming etc it will be available somewhere Q1 2007 (I hope)

As soon as I have something I'll let the list know...

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Lee]

Colin Anderson wrote:

If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document >
File Format > Unix Format. 


I ran into this same problem, and it turns out my Asterisk install would not
use Windows-formatted text files, it would just ignore them and delete them.




Hi Colin,

Thanks for responding.  I've run into the problem elsewhere myself. 
Alas, I wrote the call file using nano on the linux box through ssh/putty.


--

Warm Regards,

Lee

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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Maxim Veksler]

On 12/17/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
> Hi list,
>
> It's been a while since I've done asterisk stuff, and I'm wondering if
> there any news in the field.
>
> What do you people use today for http management of debian based Asterisk
> setup?
> Preferably something with the proven ".deb" extension.

"destar" has a "tar" extension, but a prefix quite similar to "Debian".
Availble in Etch.

FreePBX is not yet availble, though a Sarge-based CD that includes it
and generally works could be downloaded from
http://updates.xorcom.com/iso/rapid-current.iso and the package is
availble from

  deb http://updates.xorcom.com/rapid future main



Hi Tzafrir,
Thank you!

I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally unfriendly to 3rd party
changes.

Are there any other tools then, perhaps some that has not been debianized yet?
I'd like something that could be more cooperative with user hand made changes.

Maybe not web based GUI then?

Thanks,
Maxim.


--
   Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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--
Cheers,
Maxim Veksler

"Free as in Freedom" - Do u GNU ?
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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Zoa]

Hmm, if the latest free version does not have all 16 keys, email 
[EMAIL PROTECTED], there should not be a difference in the amount 
of DTMF keys between biz and free version.


Zoa

Bob Chiodini wrote:

The free version 1.31 has all 16 "keys" on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 
DTMF tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but 
first I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
"A B C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Bob Chiodini]

The free version 1.31 has all 16 "keys" on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF 
tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but 
first I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
"A B C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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Re: [asterisk-users] AGI Help Please

[Jay Milk]

Does the script run from command-line?  Without taking a close look at 
this, the include statements in the function body of connect_db look 
potentially messy.


Also, any output to stdout is interpreted by asterisk as a command, so 
those fputs statements would be a problem -- do

fputs($stdout,"VERBOSE \"There have been\"\n");
fputs($stdout,"VERBOSE \"$row_count calls made\"\n");

instead.

William Piper wrote:

List,
 
I finally decided to break down & start playing with AGI scripts, but 
for the life of me, I can't figure out what I am doing wrong.
 
Below is a super simple script to run a query in mysql to see how many 
call records there are for the extension calling in, then print the 
total in the CLI.
 
This is all I get on the CLI:

-- Executing AGI("SIP/216-0baa", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
-- AGI Script test.php completed, returning 0
-- Executing Hangup("SIP/216-0baa", "") in new stack
 
 
Here is the script:

#!/usr/bin/php -q


There are no debug errors and the query is going through just fine... 
and yes, I chmod 755.

Does anyone have a clue what I am doing wrong?

Thanks,


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Re: [asterisk-users] Polycom ring backs and CID

[J. Oquendo]

Noah Miller wrote:

Hi -
We'll still need to see more of your dialplan.  By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate... 
Call comes in receptionist answers. For some ungodly reason this client 
does not want voicemail, so when a call is xferred, the call goes 
through fine, if no one answers it rings back to the receptionist 
*SUCCESSFULLY*. However, what the client is complaining about is, it 
sounds idiotic to repeat the company mantra "Thank you for calling 
Foobar Co. how can I xfer your call" to a caller they just answered but 
failed to be xferred successfully. Before someone asks "why identify the 
caller ID" this customer also (for some ungodly reason) only wants his 
CID showing up in and out. (Don't ask)


So again:

Call comes in --> Receptionist (How can I direct your call)
Receptionist --> Transfers to extension
Extension --> No answer --> Back to receptionist
Receptionist (same call) --> Thank you for calling Foobar

Easier to comprehend?

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RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Colin Anderson]

If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document >
File Format > Unix Format. 

I ran into this same problem, and it turns out my Asterisk install would not
use Windows-formatted text files, it would just ignore them and delete them.


-Original Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
work


Tzafrir Cohen wrote:
> On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
>>> In the CLI:
>>>
>>> sip show peer axVoice
>>> show dialplan main_menu
>>> set verbose 3
>>>
>>>
>>> Then drop the call file
>>>
>>> What is the CLI trace of the above?
>>>
>> Hi, thanks for responding.  Please see the output below.
>>
>> Please note that moving a call file into /var/spool/asterisk/outgoing 
>> did not produce any CLI output.  The file was copied correctly, I 
>> believe and not present in the /outgoing directory when I checked with a 
>> simple ls command.
>>
>> # cp lee.call test.call
>> # mv test.call /var/spool/asterisk/outgoing
> 
> Are both the current directory and /var/spool/asterisk/outgoing on the
> same filesystem? If not, a 'mv' is implemented through a copy.
> 
> Anyway, you left out the CLI output of dropping trhe file.
> 
> Can Asterisk read that file? Write to it?
> 

Hi,

As I mentioned above, the action of dropping a .call into the /outgoing 
directory did not produce any CLI output.  I did this through 2 putty 
sessions.  The first, we setup to watch the CLI output and the second 
was to use the commandline to move the .call into the /outgoing directory.

Asterisk must be doing *something* with the file (if just deleting it) 
because if I check the /outgoing directory after I move the file, there 
is no file there.  It's deleted.

-- 

Warm Regards,

Lee

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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Al Bochter]

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF tones

Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but first 
I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

--> For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
--> Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
--> Need Voice Mail?
http://www.bochterservices.com/?t=VMS&t=email
-->For new and used security items
http://www.bochterservices.com/?j=store&t=email
-->BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
"A B C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM





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Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

yes thats we i use 100,a,b,c etc.
do you can mail me your extensions how you do the dials and the vm?

regards rene

On Tue, 19 Dec 2006 13:15:53 -0600
 Aaron Daniel <[EMAIL PROTECTED]> wrote:

We use this function regularly (you should see my phone's
dialstring...).  If one phone responds that it's unavailable, the rest
of the phones will still ring through.  In the event that none of the
other phones are answered, the extension is considered unanswered, so
depending on how you program your dialplan, the call will go to the
unavailable voicemail.  If you watch the CLI in this situation, you'll
see Asterisk try all the devices in the group at the same time, and
it'll just bypass any devices that are unavailable.

Also, the problem with multiple phones registering with Asterisk at the
same name is that Asterisk only stores the information about the device
once, and is overwritten with each subsequent register.  If you have a
softphone and a hardphone both registered, whichever one has a faster
re-register rate will win out over the slower one.  The only way around
this is through the call groups, as several people have stated.

Aaron

On Tue, 2006-12-19 at 10:55 -0800, Carla Schroder wrote:
Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:


http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
> how isit possible to get the VM there when one line is busy?
>
> regards rene
>
> On Tue, 19 Dec 2006 09:48:01 -0800
>
>   Carla Schroder <[EMAIL PROTECTED]> wrote:
> > Your phones only register once, when they first start up. Seems to me
> > that having multiple phones on the same account is asking for trouble-
> > why not set
> > up multiple accounts in the usual way, and create a ring group for all
> > the phones you want to use? Like this example that rings two phones at
> > the same time:
> >
> > exten => 100,1,Dial(SIP/101&SIP/102,30,t)
> > exten => 100,2,VoiceMail([EMAIL PROTECTED])
> >
> > There are all kinds of fancy variations on this theme, but the idea is
> > the same: one user with many phones, one extension, one voicemail box.


--
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Doug Lytle]

Lee wrote:


As I mentioned above, the action of dropping a .call into the 
/outgoing directory did not produce any CLI output.  I did this 
through 2 putty sessions.  The first, we setup to watch the CLI output 
and the second was to use the commandline to move the .call into the 
/outgoing directory.


set verbose 50 and try again.

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] 2 devices using same sip account

[Aaron Daniel]

We use this function regularly (you should see my phone's
dialstring...).  If one phone responds that it's unavailable, the rest
of the phones will still ring through.  In the event that none of the
other phones are answered, the extension is considered unanswered, so
depending on how you program your dialplan, the call will go to the
unavailable voicemail.  If you watch the CLI in this situation, you'll
see Asterisk try all the devices in the group at the same time, and
it'll just bypass any devices that are unavailable.

Also, the problem with multiple phones registering with Asterisk at the
same name is that Asterisk only stores the information about the device
once, and is overwritten with each subsequent register.  If you have a
softphone and a hardphone both registered, whichever one has a faster
re-register rate will win out over the slower one.  The only way around
this is through the call groups, as several people have stated.

Aaron

On Tue, 2006-12-19 at 10:55 -0800, Carla Schroder wrote:
> Hmm, I don't know what happens when one of the lines is busy and none of the 
> lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
> then perhaps this is what you want:
> 
> http://www.voip-info.org/wiki/view/Asterisk+tips+findme
> 
> On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
> > how isit possible to get the VM there when one line is busy?
> >
> > regards rene
> >
> > On Tue, 19 Dec 2006 09:48:01 -0800
> >
> >   Carla Schroder <[EMAIL PROTECTED]> wrote:
> > > Your phones only register once, when they first start up. Seems to me
> > > that having multiple phones on the same account is asking for trouble-
> > > why not set
> > > up multiple accounts in the usual way, and create a ring group for all
> > > the phones you want to use? Like this example that rings two phones at
> > > the same time:
> > >
> > > exten => 100,1,Dial(SIP/101&SIP/102,30,t)
> > > exten => 100,2,VoiceMail([EMAIL PROTECTED])
> > >
> > > There are all kinds of fancy variations on this theme, but the idea is
> > > the same: one user with many phones, one extension, one voicemail box.

-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] Polycom ring backs and CID

[Noah Miller]

Hi -


>> (INTERNAL)
>> 1 Call comes in to receptionist and gets transferred to someone
>> 2 No one picks up that transfer
>> 3 Call goes back to receptionist
>>
>> Now when the call goes back to the receptionist, how can I change either
>> the ringer, the callerID or both?
>
> If you're looking for the technical aspects of how to do custom
> ringtones, see here:
> http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
> (this page is for setting the phone to auto-answer, but changing
> ringtones is the same procedure)
>
> For setting the caller ID, see here:
> http://www.voip-info.org/wiki/view/Setting+Callerid
>
Not what I needed but thanks... I'm using the standard Asterisk
transferring. I know there is a method to do so for parked calls:


We'll still need to see more of your dialplan.  By your description,
it looks like the call is failing because the Dial() times out.

blindxfer and atxfer won't automatically return a caller to the
receptionist.  You have to have something in the dialplan to do that.
When we know what it is that is redirecting your failed transfers back
to the receptionist (probably the 't' extension), we can just insert a
"Set(CALLERID=)" or "Set(_ALERT_INFO=)".  You may also have
transfers fail because they get sent to an invalid extension.  The
calls go to the 'i' extension.  You can modify it accordingly, too.

- Noah
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Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

nothing happen it only let ring all lines which are not in use but i want that 
the busy vm message is coming when one line is busy.


On Tue, 19 Dec 2006 10:55:34 -0800
 Carla Schroder <[EMAIL PROTECTED]> wrote:
Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:


http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:

how isit possible to get the VM there when one line is busy?

regards rene

On Tue, 19 Dec 2006 09:48:01 -0800

  Carla Schroder <[EMAIL PROTECTED]> wrote:
> Your phones only register once, when they first start up. Seems to me
> that having multiple phones on the same account is asking for trouble-
> why not set
> up multiple accounts in the usual way, and create a ring group for all
> the phones you want to use? Like this example that rings two phones at
> the same time:
>
> exten => 100,1,Dial(SIP/101&SIP/102,30,t)
> exten => 100,2,VoiceMail([EMAIL PROTECTED])
>
> There are all kinds of fancy variations on this theme, but the idea is
> the same: one user with many phones, one extension, one voicemail box.


--
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/

best book for sysadmins and power users
~
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

I am located on the west coast of the united states.
In order to dial an international number from within the US, we must 
first dial the special international access code that tells the PSTN 
"the following call is an international one" - in the US that is "011", 
followed by the country code, and then the actual number for our 
destination within that country.  (which would include whatever their 
concept of area code, prefix, and destination number are - which varies 
widely from country to country)


If you're generally interested in this, then you might find the 
following reading interesting as well:

http://en.wikipedia.org/wiki/North_American_Numbering_Plan
and
http://en.wikipedia.org/wiki/Area_code

  - Anthony Kepler
  [EMAIL PROTECTED] | SIP/Email

Doug Crompton wrote:

Sorry, I did not read the original message completely. The answer is no I
do not make international calls. I do not know anyone in any other 
country

to call! I do not have a rule for that but it should be easy to implement
as 01x would not match anything I currently have for early dial. Would 
you

always dial a 0 first for all international mumbers? Give me an example?

Are you outside the US? If so give me your number and I will try it!

Doug

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Re: [asterisk-users] Re: AEL2 on Asterisk 1.2.4

[Lee]

Steve Murphy wrote:


Lee, everyone--

Sorry about that. I've created a patches subdir in the AEL2-1.2
repository, and put that patch down in there. So, now, you do:

svn co http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/patches

and then


Excellent!  Thank you.  I will try this.


Again, you 1.2 users: It is not absolutely necessary to update your 1.2
installation to use AEL with 1.2; you can build a 1.4 somewhere, and use
the AEL in 1.4 to compile your extensions.ael into an extensions.conf
file via

aelparse -d -w

This will generate the file 'extensions.conf.aeldump', which you can
inspect and then copy into your appropriate /etc/asterisk directory on
the machine where your 1.2 installation resides.

The main thing to watch out for in this scenario, is that AEL uses a
slightly enhanced version of the $[] parser to do its thing. If you
don't use the new features, you should be quite OK.


Great idea.

Thanks again,

--

Warm Regards,

Lee

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Re: [asterisk-users] STUN with one public and one private IP?

[David Thomas]

Are you kidding? Lighten up people!
Al made a friendly recommendation based on the comments regarding TrixBox.

Go have a beer... take a load off... enjoy the holidays.

Regards,
David
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Lee]

Tzafrir Cohen wrote:

On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:

In the CLI:

sip show peer axVoice
show dialplan main_menu
set verbose 3


Then drop the call file

What is the CLI trace of the above?


Hi, thanks for responding.  Please see the output below.

Please note that moving a call file into /var/spool/asterisk/outgoing 
did not produce any CLI output.  The file was copied correctly, I 
believe and not present in the /outgoing directory when I checked with a 
simple ls command.


# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing


Are both the current directory and /var/spool/asterisk/outgoing on the
same filesystem? If not, a 'mv' is implemented through a copy.

Anyway, you left out the CLI output of dropping trhe file.

Can Asterisk read that file? Write to it?



Hi,

As I mentioned above, the action of dropping a .call into the /outgoing 
directory did not produce any CLI output.  I did this through 2 putty 
sessions.  The first, we setup to watch the CLI output and the second 
was to use the commandline to move the .call into the /outgoing directory.


Asterisk must be doing *something* with the file (if just deleting it) 
because if I check the /outgoing directory after I move the file, there 
is no file there.  It's deleted.


--

Warm Regards,

Lee

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Re: [asterisk-users] 2 devices using same sip account

[Carla Schroder]

Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:

http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
> how isit possible to get the VM there when one line is busy?
>
> regards rene
>
> On Tue, 19 Dec 2006 09:48:01 -0800
>
>   Carla Schroder <[EMAIL PROTECTED]> wrote:
> > Your phones only register once, when they first start up. Seems to me
> > that having multiple phones on the same account is asking for trouble-
> > why not set
> > up multiple accounts in the usual way, and create a ring group for all
> > the phones you want to use? Like this example that rings two phones at
> > the same time:
> >
> > exten => 100,1,Dial(SIP/101&SIP/102,30,t)
> > exten => 100,2,VoiceMail([EMAIL PROTECTED])
> >
> > There are all kinds of fancy variations on this theme, but the idea is
> > the same: one user with many phones, one extension, one voicemail box.

-- 
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/
best book for sysadmins and power users
~
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RE: [asterisk-users] Parsing Area Code from CallerID

[Jonathan k. Creasy]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Bruce Ferrell
> Sent: Tuesday, December 19, 2006 12:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Parsing Area Code from CallerID
> 
> 
> John French wrote:
> > How would I parse the area code from this variable?
Number=2515551212
> > Sorry for the dense question, I don't seem to be able to find an
> > appropriate function for parsing left to right.
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> NPA=${NUMBER:0:3}
> 
> --
> One day at a time, one second if that's what it takes


That works if the number is always NPA-NXX-. If you end up with
+1NPANXX or 1NPANXX then you don't have the right data. 

-Jonathan

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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Doug Crompton]

Sorry, I did not read the original message completely. The answer is no I
do not make international calls. I do not know anyone in any other country
to call! I do not have a rule for that but it should be easy to implement
as 01x would not match anything I currently have for early dial. Would you
always dial a 0 first for all international mumbers? Give me an example?

Are you outside the US? If so give me your number and I will try it!

Doug

On Tue, 19 Dec 2006, Anthony Kepler wrote:

> I understand how early dial works (484 response and all that jazz), I
> also understand the NANP and how to keep my extensions from
> overlapping... but thank you for the tips.
>
> My question was:  Do you place international calls from phones with
> early-dial enabled?
> If so, might you be willing to share the relevant portions of your dial
> plan that are concerned with placing said international calls?
>
> Thanks again,
> - Anthony Kepler
> [EMAIL PROTECTED] | SIP/Email
>

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Re: [asterisk-users] 2 devices using same sip account

[Noah Miller]

Hi Rene -


how isit possible to get the VM there when one line is busy?


If I understand your question correctly, the answer is you need two
incoming phone lines.


- Noah
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Re: [asterisk-users] Re: IBM Server / USB Ports

[Matt]

Ok.. so then for some reason the PCI slot that the Digium card is in
is following the IRQ of the Ethernet controller.  We will move the
Digium card and see what happens.

On 12/19/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
> Could the unknown device be a management card?
>
> The newer dells have a management card built into the fist ethernet 
controller.

D161:2400 is a Digium TDM2400P card.

Source: Debian has a nice 'update-pciids' command, which updates the
local pciids file from http://pciids.sourceforge.net/ .
http://pci-ids.ucw.cz/iii/?p=d
http://pci-ids.ucw.cz/iii/?i=d161
http://pci-ids.ucw.cz/iii/?i=d1612400

--
   Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
> >In the CLI:
> >
> >sip show peer axVoice
> >show dialplan main_menu
> >set verbose 3
> >
> >
> >Then drop the call file
> >
> >What is the CLI trace of the above?
> >
> 
> Hi, thanks for responding.  Please see the output below.
> 
> Please note that moving a call file into /var/spool/asterisk/outgoing 
> did not produce any CLI output.  The file was copied correctly, I 
> believe and not present in the /outgoing directory when I checked with a 
> simple ls command.
> 
> # cp lee.call test.call
> # mv test.call /var/spool/asterisk/outgoing

Are both the current directory and /var/spool/asterisk/outgoing on the
same filesystem? If not, a 'mv' is implemented through a copy.

Anyway, you left out the CLI output of dropping trhe file.

Can Asterisk read that file? Write to it?

> 
> 
> 
> === sip show peer axVoice ===
> =
> CLI>
> 
>   * Name   : axVoice
>   Secret   : 
>   MD5Secret: 
>   Context  : incoming
>   Subscr.Cont. : 
>   Language :
>   AMA flags: Unknown
>   CallingPres  : Presentation Allowed, Not Screened
>   FromUser : datatrak
>   FromDomain   : 216.143.130.36
>   Callgroup:
>   Pickupgroup  :
>   Mailbox  :
>   VM Extension : 555
>   LastMsgsSent : -1
>   Call limit   : 0
>   Dynamic  : No
>   Callerid : "" <>
>   Expire   : -1
>   Insecure : port,invite
>   Nat  : Always
>   ACL  : No
>   CanReinvite  : Yes
>   PromiscRedir : No
>   User=Phone   : No
>   Trust RPID   : No
>   Send RPID: No
>   DTMFmode : rfc2833
>   LastMsg  : 0
>   ToHost   : 216.143.130.36
>   Addr->IP : 216.143.130.36 Port 5060
>   Defaddr->IP  : 216.143.130.36 Port 0
>   Def. Username: 
>   SIP Options  : (none)
>   Codecs   : 0x4 (ulaw)
>   Codec Order  : (ulaw)
>   Status   : Unmonitored
>   Useragent:
>   Reg. Contact :
> 
> === show dialplan main_after_hours ===
> (I mistyped the name of the context in original post)
> 
> CLI> show dialplan main_after_hours
> [ Context 'main_after_hours' created by 'pbx_config' ]
>   '1' =>1. Playback(transfer) 
> [pbx_config]
> 2. Macro(DialExtenVM|111|30|tm) 
> [pbx_config]
> 3. Set(EXTEN=955) 
> [pbx_config]
> 4. GoTo(Management|955|1) 
> [pbx_config]
> 5. Playback(transfer) 
> [pbx_config]
> 6. Macro(DialExtenVM|111|30|tr) 
> [pbx_config]
> 7. Set(EXTEN=955) 
> [pbx_config]
> 8. GoTo(Management|955|1) 
> [pbx_config]
> 9. Playback(custom/no_tech_available) 
> [pbx_config]
> 10. Voicemail(111) 
> [pbx_config]
>   '2' =>1. 
> Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
> 2. Goto(support_non_emergency|s|1) 
> [pbx_config]
>   '444' =>  1. Set(LIMIT_PLAYAUDIO_CALLEE=yes) 
> [pbx_config]
> 2. Dial(SIP/111|30|mgL(1:1:5000)) 
> [pbx_config]
> 3. Wait(3) 
> [pbx_config]
> 4. Goto(main_after_hours|s|1) 
> [pbx_config]
>   '9' =>1. 
> Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
> 2. Goto(main_branch|s|1) 
> [pbx_config]
>   'i' =>1. GotoIf($[ ${FAIL_MENU} != ""]|?2:3) 
> [pbx_config]
> 2. Goto(${FAIL_MENU}|s|1) 
> [pbx_config]
> 3. Goto(main_branch|s|1) 
> [pbx_config]
>   's' =>1. Answer() 
> [pbx_config]
> 2. Wait(1) 
> [pbx_config]
> 3. Background(custom/after_hours) 
> [pbx_config]
>   't' =>1. GotoIf($[ ${TIMEOUT_MENU} != "" ]|?2:3) 
> [pbx_config]
> 2. Goto(${TIMEOUT_MENU}|s|1) 
> [pbx_config]
> 3. Goto(main_branch|s|1) 
> [pbx_config]
>   '_ZZZ' => 1. 
> Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m)
>  
> [pbx_config]
> 
> 
> 
> -- 
> 
> Warm Regards,
> 
> Lee
> 
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> 

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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

I understand how early dial works (484 response and all that jazz), I 
also understand the NANP and how to keep my extensions from 
overlapping... but thank you for the tips.


My question was:  Do you place international calls from phones with 
early-dial enabled?
If so, might you be willing to share the relevant portions of your dial 
plan that are concerned with placing said international calls?


Thanks again,
   - Anthony Kepler
   [EMAIL PROTECTED] | SIP/Email

Doug Crompton wrote:

Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.

As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.

However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.

Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.

Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.

Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.

You just have to use it and fix things as they come up. I think I have
most all cases trapped now!

Doug



On Tue, 19 Dec 2006, Anthony Kepler wrote:

  

Do you, Gordon or Doug, happen to place international calls with
early-dial enabled?  What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work.  (My message from
2006.11.03 regarding this is quoted just below)



On 11/3/06, *Anthony Kepler* <[EMAIL PROTECTED]
> wrote:I am trying to allow users to
place outgoing international calls from a
GXP-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1

I have the following extension line:
exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial "011254"...etc.
and I get this on the asterisk console:
Executing Dial("SIP/1001-081fb718", "Zap/g1/0112") in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the "."
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set("SIP/1001-081fb718", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED]  | SIP/Email
  

Gordon Henderson wrote:


On Sun, 5 Nov 2006, Doug Crompton wrote:


  

On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 "Line"
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon


  

Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:




Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:


  

I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?



Set the "Early Dial" option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the "send" button...

Gordon

  

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Re: [asterisk-users] Polycom ring backs and CID

[J. Oquendo]

Noah Miller wrote:

Hi -


(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist

Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?



If you're looking for the technical aspects of how to do custom
ringtones, see here:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
(this page is for setting the phone to auto-answer, but changing
ringtones is the same procedure)

For setting the caller ID, see here:
http://www.voip-info.org/wiki/view/Setting+Callerid

Not what I needed but thanks... I'm using the standard Asterisk 
transferring. I know there is a method to do so for parked calls:


exten => 7XX,1,SetVar(_ALERT_INFO=http://somewhere/alt.wav)
exten => 7XX,2,Set(CALLERID(name)=Parked Call)
exten => 7XX,n,ChanIsAvail(SIP/${EXTEN:1}|sj)
exten => 7XX,n,Dial(SIP/${EXTEN:1}|30)
exten => 7XX,n,Goto(default,${EXTEN},102)
exten => 7XX,102,Goto(main-aa,s,1)

I'm wondering if anyone has set it up differently


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Re: [asterisk-users] Changing CALLERIDNUM on the fly

[Doug Crompton]

Thanks Anselm, That did it!

Doug

On Tue, 19 Dec 2006, Anselm Martin Hoffmeister wrote:

> Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
> > Is what I am trying to do in this context possible. That is changing the
> > incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
> > preceeded by a "1" I want to add a "1". Often calls come in without the
> > preceeding "1" and this plays havoc with my redial if the 3 digit area
> > code matches a local 3 digit extension. All my outside calls are 10 digits
> > or 1+10 digits.
> >
> > Doug
> >
> >
> > [from-pstn]
> > exten => s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
> > exten => s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
> > exten => s,3,set(__CALLERIDNUM=1${CALLERIDNUM})   if not add 1
> > exten => s,4,noop(${CALLERIDNUM})  > and this still displays without
>
> Replace CALLERIDNUM with CALLERID(num) on all occasions, and you will
> not need underscores because this is a special variable anyway.
> CALLERIDNUM is obsolete.
>
> You could get along with one line less:
> exten => s,1,GotoIf($[A${CALLERID(num):0:1} = A1]?3:2)
> exten => s,2,Set(CALLERID(num)=1${CALLERID(num)})
> exten => s,3,NOOP(Continue in Dialplan)
>
> Note that my GotoIf contains the two additional "A" letters which is
> important to avoid syntax errors if the CALLERID(num) is empty for
> whatever reason. I do not know what ends up in your CALLERID(num) if the
> number of the caller is not available (like "anonymous" or "withheld") -
> anyway, with this statement it will end up being prepended by "1". You
> migth want to have a special case for that.
>
> If your phones happen to also display CALLERID(name) you can use this to
> lookup the phone number in a phone book (here in Germany there is an
> online service for number reverse lookup which works for about 50% of my
> callers) and set the variable.
>
> BR
> Anselm
>
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Doug Crompton]

Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.

As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.

However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.

Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.

Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.

Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.

You just have to use it and fix things as they come up. I think I have
most all cases trapped now!

Doug



On Tue, 19 Dec 2006, Anthony Kepler wrote:

> Do you, Gordon or Doug, happen to place international calls with
> early-dial enabled?  What kind of extensions.conf magic do you work to
> allow this?
> I have been trying for some time to get this to work.  (My message from
> 2006.11.03 regarding this is quoted just below)
>
> > On 11/3/06, *Anthony Kepler* <[EMAIL PROTECTED]
> > > wrote:I am trying to allow users to
> > place outgoing international calls from a
> > GXP-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1
> > 
> > I have the following extension line:
> > exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> >
> > When I attempt to place a call to a number in, for instance, Kenya, I
> > dial "011254"...etc.
> > and I get this on the asterisk console:
> > Executing Dial("SIP/1001-081fb718", "Zap/g1/0112") in new stack
> >-- Called g1/0112
> >
> > It is attempting to dial out as soon as it receives a single digit to
> > represent the "."
> > What I need is for it to wait a reasonable amount of time for additional
> > digits.
> > I have tried using set(TIMEOUT(digit)=5), and I see the following in the
> > asterisk console:
> >-- Executing Set("SIP/1001-081fb718", "TIMEOUT(digit)=5") in new stack
> >-- Digit timeout set to 5
> > However, this is printed far less than 5 seconds before the dial out
> > attempt.
> >
> > I assume there must be something relatively obvious I'm missing here...
> > if anyone can shed some light on this, it would be greatly appreciated.
> >
> >
> > Thank you,
> >- Anthony Kepler
> > [EMAIL PROTECTED]  | SIP/Email
>
>
> Gordon Henderson wrote:
> > On Sun, 5 Nov 2006, Doug Crompton wrote:
> >
> >
> >> On the Budgetone 200 it is in the account tab settings of the web setup
> >> and it does work here with asterisk and my dialplans..
> >>
> >
> > On the GPX2000's it's via the web interface under each of the 4 "Line"
> > configuration tabs. (so you'd have to set it on each account you were
> > using on the phone)
> >
> > Gordon
> >
> >
> >> Doug
> >>
> >> On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
> >>
> >>
> >>> Hi,
> >>>
> >>> Where can I find that option?
> >>>
> >>> Thanks
> >>> Jesus
> >>>
> >>> -Mensaje original-
> >>> De: [EMAIL PROTECTED]
> >>> [mailto:[EMAIL PROTECTED] En nombre de Gordon
> >>> Henderson
> >>> Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
> >>> Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> >>> Non-Commercial Discussion
> >>> Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
> >>>
> >>> On Wed, 1 Nov 2006, Henry.L.Coleman wrote:
> >>>
> >>>
>  I came to the same conclusion.
>  There is one thing however that the GXP2000 needs in my opinion.
>  There is no dial plan avaiable in the configuration, this means that when
>  dialing a number there is a slight delay before it actually dials.
>  With a dial plan the dialed number is sent immeadiately the pattern is
>  match ed so it saves a second or two. Maybe they will fix this?
> 
> >>> Set the "Early Dial" option - it's on a per-line basis, then as soon
> >>> as Asterisk gets a number it can dial, it will. No need to wait the 4
> >>> seconds or press the "send" button...
> >>>
> >>> Gordon
> >>>
> >> ___
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRI

Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

how isit possible to get the VM there when one line is busy?

regards rene

On Tue, 19 Dec 2006 09:48:01 -0800
 Carla Schroder <[EMAIL PROTECTED]> wrote:
Your phones only register once, when they first start up. Seems to me that 
having multiple phones on the same account is asking for trouble- why not 
set 
up multiple accounts in the usual way, and create a ring group for all the 
phones you want to use? Like this example that rings two phones at the same 
time:


exten => 100,1,Dial(SIP/101&SIP/102,30,t)
exten => 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the 
same: one user with many phones, one extension, one voicemail box.


On Tuesday 19 December 2006 8:18 am, rilawich ango wrote:

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue "database showkey
SIP/Registry/sip account" in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:
> 19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
> > Hi,
> >
> > It seems that they both can make calls, but only one can receive
> > call: the
> > last registered...
> >
> > Greg
> >
> >> Hi all,
> >>   What will happen if 2 devices using the same set of sip account to
> >> connect to the same asterisk?  Do they both can make call?  Can they
> >> receive call as normal?
> >> ___
> >> --Bandwidth and Colocation provided by Easynews.com --
>
> In Asterisk, you should only have one phone per account. We do not
> support
> multiple devices per account. The PBX core needs to know how many
> devices
> that we are calling each time we access it.
>
> /O


--
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/

best book for sysadmins and power users
~
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Re: [asterisk-users] 2 devices using same sip account

[Carla Schroder]

Your phones only register once, when they first start up. Seems to me that 
having multiple phones on the same account is asking for trouble- why not set 
up multiple accounts in the usual way, and create a ring group for all the 
phones you want to use? Like this example that rings two phones at the same 
time:

exten => 100,1,Dial(SIP/101&SIP/102,30,t)
exten => 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the 
same: one user with many phones, one extension, one voicemail box.

On Tuesday 19 December 2006 8:18 am, rilawich ango wrote:
> It seems that Greg is truth for the case.  Asterisk doesn't care how
> many devices register to the same account as it is a feature of sip
> protocol (please let me know if there is a method to restrict it).
>
> In my case, I use a soft phone an hard phone using the same sip
> account information to register to the same asterisk.  Soft phone
> register first and then hard phone register later.  I dial the number
> and hard phone ring.  Then I disconnect hard phone and expect soft
> phone will be ring after a couple of time.  However, soft phone didn't
> ring as the call is failed.  I issue "database showkey
> SIP/Registry/sip account" in CLI.  It displays the information which
> belongs to hard phone.  That's mean asterisk will keep the information
> of hard phone even it is disconnected with ignoring the soft phone
> registration.  Does asterisk can be set to refresh its registry in a
> couple of time to remove the old registry record?
>
> On 12/19/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:
> > 19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
> > > Hi,
> > >
> > > It seems that they both can make calls, but only one can receive
> > > call: the
> > > last registered...
> > >
> > > Greg
> > >
> > >> Hi all,
> > >>   What will happen if 2 devices using the same set of sip account to
> > >> connect to the same asterisk?  Do they both can make call?  Can they
> > >> receive call as normal?
> > >> ___
> > >> --Bandwidth and Colocation provided by Easynews.com --
> >
> > In Asterisk, you should only have one phone per account. We do not
> > support
> > multiple devices per account. The PBX core needs to know how many
> > devices
> > that we are calling each time we access it.
> >
> > /O

-- 
~
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Lee]

In the CLI:

sip show peer axVoice
show dialplan main_menu
set verbose 3


Then drop the call file

What is the CLI trace of the above?



Hi, thanks for responding.  Please see the output below.

Please note that moving a call file into /var/spool/asterisk/outgoing 
did not produce any CLI output.  The file was copied correctly, I 
believe and not present in the /outgoing directory when I checked with a 
simple ls command.


# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing



=== sip show peer axVoice ===
=
CLI>

  * Name   : axVoice
  Secret   : 
  MD5Secret: 
  Context  : incoming
  Subscr.Cont. : 
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  FromUser : datatrak
  FromDomain   : 216.143.130.36
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : 555
  LastMsgsSent : -1
  Call limit   : 0
  Dynamic  : No
  Callerid : "" <>
  Expire   : -1
  Insecure : port,invite
  Nat  : Always
  ACL  : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 216.143.130.36
  Addr->IP : 216.143.130.36 Port 5060
  Defaddr->IP  : 216.143.130.36 Port 0
  Def. Username: 
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : Unmonitored
  Useragent:
  Reg. Contact :

=== show dialplan main_after_hours ===
(I mistyped the name of the context in original post)

CLI> show dialplan main_after_hours
[ Context 'main_after_hours' created by 'pbx_config' ]
  '1' =>1. Playback(transfer) 
[pbx_config]
2. Macro(DialExtenVM|111|30|tm) 
[pbx_config]
3. Set(EXTEN=955) 
[pbx_config]
4. GoTo(Management|955|1) 
[pbx_config]
5. Playback(transfer) 
[pbx_config]
6. Macro(DialExtenVM|111|30|tr) 
[pbx_config]
7. Set(EXTEN=955) 
[pbx_config]
8. GoTo(Management|955|1) 
[pbx_config]
9. Playback(custom/no_tech_available) 
[pbx_config]
10. Voicemail(111) 
[pbx_config]
  '2' =>1. 
Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
2. Goto(support_non_emergency|s|1) 
[pbx_config]
  '444' =>  1. Set(LIMIT_PLAYAUDIO_CALLEE=yes) 
[pbx_config]
2. Dial(SIP/111|30|mgL(1:1:5000)) 
[pbx_config]
3. Wait(3) 
[pbx_config]
4. Goto(main_after_hours|s|1) 
[pbx_config]
  '9' =>1. 
Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
2. Goto(main_branch|s|1) 
[pbx_config]
  'i' =>1. GotoIf($[ ${FAIL_MENU} != ""]|?2:3) 
[pbx_config]
2. Goto(${FAIL_MENU}|s|1) 
[pbx_config]
3. Goto(main_branch|s|1) 
[pbx_config]
  's' =>1. Answer() 
[pbx_config]
2. Wait(1) 
[pbx_config]
3. Background(custom/after_hours) 
[pbx_config]
  't' =>1. GotoIf($[ ${TIMEOUT_MENU} != "" ]|?2:3) 
[pbx_config]
2. Goto(${TIMEOUT_MENU}|s|1) 
[pbx_config]
3. Goto(main_branch|s|1) 
[pbx_config]
  '_ZZZ' => 1. 
Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m) 
[pbx_config]




--

Warm Regards,

Lee

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Re: [asterisk-users] Polycom ring backs and CID

[Noah Miller]

Hi -


(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist

Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?


We'll need to see a little more info to help you out:

1) What mechanism are you using to transfer, built-in asterisk or the
Polycom transfer key(s)?
2) What does your dial plan look like - how is it that calls are
ringing back to your receptionist?

If you're looking for the technical aspects of how to do custom
ringtones, see here:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
(this page is for setting the phone to auto-answer, but changing
ringtones is the same procedure)

For setting the caller ID, see here:
http://www.voip-info.org/wiki/view/Setting+Callerid


- Noah
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[asterisk-users] Is MOH Still Broken in Asterisk 1.4 (beta3)?

[Douglas Garstang]

I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, 
when a callee put a caller on hold, the musiconhold class that was played was 
not the one the callee wanted the caller to hear, but something else. Even 
after using mohsuggest in Asterisk 1.4, it still appears that this is not 
working correctly.

Here's the results of a simple test:

CASE CALLER CALLEE HOLDER HOLDER HEARS MOH
--
1325410132541023254101moh1
2325410132541023254102default

3325410232541013254102moh2
4325410232541013254101default

For each extension, I have mohsuggest set. Test cases 1 and 3, where the caller 
puts the callee on hold, yield the expected behaviour. However, test cases 2 
and 4 where the callee puts the caller on hold, do not yield the correct 
results.

Here's what the results SHOULD be.

CASE CALLER CALLEE HOLDER HOLDER HEARS MOH
--
1325410132541023254101moh1
2325410132541023254102moh2

3325410232541013254102moh2
4325410232541013254101moh1

Am I possibly doing something wrong with mohsuggest?

sip.conf:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[3254101]
type = friend 
context = CallStart
username = 3254101
accountcode = 3254101
qualify = yes
canreinvite = no
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang <3254101>
secret = password
mohsuggest = moh1

[3254102]
type = friend 
context = CallStart
username = 3254102
accountcode = 3254102
qualify = yes
canreinvite = no
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang <3254101>
secret = password
mohsuggest = moh2
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

Do you, Gordon or Doug, happen to place international calls with 
early-dial enabled?  What kind of extensions.conf magic do you work to 
allow this?
I have been trying for some time to get this to work.  (My message from 
2006.11.03 regarding this is quoted just below)


On 11/3/06, *Anthony Kepler* <[EMAIL PROTECTED] 
> wrote:I am trying to allow users to 
place outgoing international calls from a
GXP-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1 


I have the following extension line:
exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial "011254"...etc.
and I get this on the asterisk console:
Executing Dial("SIP/1001-081fb718", "Zap/g1/0112") in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the "."
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set("SIP/1001-081fb718", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED]  | SIP/Email



Gordon Henderson wrote:

On Sun, 5 Nov 2006, Doug Crompton wrote:

  

On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 "Line"
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon

  

Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:



Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:

  

I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?


Set the "Early Dial" option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the "send" button...

Gordon
  

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Re: [asterisk-users] Parsing Area Code from CallerID

[Bruce Ferrell]

John French wrote:
How would I parse the area code from this variable? Number=2515551212  
Sorry for the dense question, I don't seem to be able to find an 
appropriate function for parsing left to right.

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NPA=${NUMBER:0:3}

--
One day at a time, one second if that's what it takes

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Re: [asterisk-users] 2 devices using same sip account

[Peder @ NetworkOblivion]

It doesn't have anything to do with hardphone versus softphone.  The 
issue is that it can only keep track of one registration per account. 
When the hardphone gets unplugged, it will not know about the softphone 
until it registers with asterisk.  It's initial registration was lost 
when the hardphone registered with the same info.


rilawich ango wrote:

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue "database showkey
SIP/Registry/sip account" in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:


19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:

> Hi,
>
> It seems that they both can make calls, but only one can receive
> call: the
> last registered...
>
> Greg
>
>> Hi all,
>>   What will happen if 2 devices using the same set of sip account to
>> connect to the same asterisk?  Do they both can make call?  Can they
>> receive call as normal?
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
In Asterisk, you should only have one phone per account. We do not
support
multiple devices per account. The PBX core needs to know how many
devices
that we are calling each time we access it.

/O
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--

Network stuff you didn't know
http://www.networkoblivion.com

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[asterisk-users] Re: AEL2 on Asterisk 1.2.4

[Steve Murphy]

On Tue, 2006-12-19 at 04:20:07 -0500, [EMAIL PROTECTED] wrote:
> 
> Hey all,
> 
> I am very interested in using AEL2 (don't want to upgrade to
> 1.4 to get 
> it though), but am having some problems upgrading/patching my
> asterisk 
> system.  I am following the instructions on the wiki:
> 
> http://www.voip-info.org/wiki/view/Asterisk
> +AEL2#AEL2AnnouncementsandNews
> 
> But get the following error:
> 
> 
> "'http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/diffs.AEL2.patch' 
> refers to a file, not a directory"
> 
> This refers to the process of including the patch as described
> in the 
> first portion of the wiki page.
> 
> Am am still new to linux so the problem could be just me, but
> I believe 
> I followed the instructions.  They are pretty simple after
> all.
> 
> BTW, I tried both ways described and I could not get either to
> work.
> 
> Thanks for any help,
> 
> -- 
> 
> Warm Regards,
> 
> Lee

Lee, everyone--

Sorry about that. I've created a patches subdir in the AEL2-1.2
repository, and put that patch down in there. So, now, you do:

svn co http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/patches

and then

patch -p0 < patches/diffs.AEL2.patch

Assuming that you are in a 1.2 source directory...

Hopefully this sequence will work better. I've updated the voip-info
wiki.

murf

Again, you 1.2 users: It is not absolutely necessary to update your 1.2
installation to use AEL with 1.2; you can build a 1.4 somewhere, and use
the AEL in 1.4 to compile your extensions.ael into an extensions.conf
file via

aelparse -d -w

This will generate the file 'extensions.conf.aeldump', which you can
inspect and then copy into your appropriate /etc/asterisk directory on
the machine where your 1.2 installation resides.

The main thing to watch out for in this scenario, is that AEL uses a
slightly enhanced version of the $[] parser to do its thing. If you
don't use the new features, you should be quite OK.

murf


-- 
Steve Murphyaka 'codefreeze' or 'wyoming' on FreeNode IRC
Software Developer
Digium


smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Re: IBM Server / USB Ports

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
> Could the unknown device be a management card?
> 
> The newer dells have a management card built into the fist ethernet 
> controller.

D161:2400 is a Digium TDM2400P card.

Source: Debian has a nice 'update-pciids' command, which updates the
local pciids file from http://pciids.sourceforge.net/ . 
http://pci-ids.ucw.cz/iii/?p=d
http://pci-ids.ucw.cz/iii/?i=d161
http://pci-ids.ucw.cz/iii/?i=d1612400

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Need help getting started with asterisk

[Michael Sullivan]

Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all.  Is there some explicit thing I
need to put in to get the caller ID?

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Re: [asterisk-users] 2 devices using same sip account

[rilawich ango]

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue "database showkey
SIP/Registry/sip account" in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:


19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:

> Hi,
>
> It seems that they both can make calls, but only one can receive
> call: the
> last registered...
>
> Greg
>
>> Hi all,
>>   What will happen if 2 devices using the same set of sip account to
>> connect to the same asterisk?  Do they both can make call?  Can they
>> receive call as normal?
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
In Asterisk, you should only have one phone per account. We do not
support
multiple devices per account. The PBX core needs to know how many
devices
that we are calling each time we access it.

/O
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[asterisk-users] db.c: Unable to open Asterisk database

[DODO BUBU]

Dear asterisk users,
I am using Asterisk and I a m a new user.
Before it was working properly.
Since two days, users can not get registered : users registered timeout.
Those are the results of commands 
1. /var/log/asterisk#asterisk-rvv  
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk 1.2.1 currently running on ns1 (pid = 4244)
Verbosity is at least 46

2. var/log/asterisk# tail  -200  messages
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Database unavailable
Dec 19 09:45:59 WARNING[4245] chan_iax2.c: Unable to open IAX timing interface: 
No such file or directory
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Database unavailable
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:54:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 09:54:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:02:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:02:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:02:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:10:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:10:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:10:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:19:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:19:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:19:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:27:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:27:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:27:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:35:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:35:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:35:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:44:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:44:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:44:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:52:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:52:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:52:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:00:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:00:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:00:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:09:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:09:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:09:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:17:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:17:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:17:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:25:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:25:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/s

RE: [asterisk-users] Parsing Area Code from CallerID

[Jonathan k. Creasy]

${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end. 

-Jonathan

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John French
> Sent: Tuesday, December 19, 2006 10:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Parsing Area Code from CallerID
> 
> How would I parse the area code from this variable? Number=2515551212
> Sorry for the dense question, I don't seem to be able to find an
> appropriate function for parsing left to right.
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Re: [asterisk-users] DTMF Tones "A-B-C-D"

[Zoa]

Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it).

Zoa


Al Bochter wrote:
Ok does anyone know of any softphones that will dial DTMF tone keys "A 
B C D"

And do you know if Asterisk will take the DTMF Tones for "A B C D"



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