Re: [asterisk-users] Asterisk as an IVR solution

[Douglas Garstang]

Well I can tell you that it makes a difficult programming environment, just a 
little more difficult. It means I can't implement a menu as a single reusable 
piece of code inside a macro. 


- Original Message 
From: Tilghman Lesher <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, July 10, 2008 6:07:36 PM
Subject: Re: [asterisk-users] Asterisk as an IVR solution

On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
> It's a known problem.
>
> If you call Background() in a macro, then Asterisk will look for the
> extensions to jump to in the CALLING Macro/context and NOT the Macro that
> the Background() app was called in.

I wouldn't call it a known problem.  It works precisely as it was designed to
work.  It may not work the way that you want it to, but it works like a Macro:
an independent set of instructions, with substitution, that acts as if it were
invoked inline with the calling location.  That is why Background will match
in the context of the calling location: it acts like it never left that
original context (and, in a way, it really didn't).

Subroutines are a different beast, and they are available with the Gosub/
Return set of routines in app_stack.so.

-- 
Tilghman

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Re: [asterisk-users] Asterisk as an IVR solution

[Al Baker]

SO does that mean that if he used BACKGROUND is a SubRoutine  he would
get the "correct" or "desired" action , from his point of view? It would 
jump to the "1" Extension in the SUBROUTINE ?

Tilghman Lesher wrote:
> On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
>   
>> It's a known problem.
>>
>> If you call Background() in a macro, then Asterisk will look for the
>> extensions to jump to in the CALLING Macro/context and NOT the Macro that
>> the Background() app was called in.
>> 
>
> I wouldn't call it a known problem.  It works precisely as it was designed to
> work.  It may not work the way that you want it to, but it works like a Macro:
> an independent set of instructions, with substitution, that acts as if it were
> invoked inline with the calling location.  That is why Background will match
> in the context of the calling location: it acts like it never left that
> original context (and, in a way, it really didn't).
>
> Subroutines are a different beast, and they are available with the Gosub/
> Return set of routines in app_stack.so.
>
>   

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[asterisk-users] Asterisk cant play sounds from AGI

[Edwin Quijada]

Hi! I am a newbie using Asterisk. I am developing an IVR using perl from AGI 
and Cepstral as voices
The AGI is this

use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
 
 
 $AGI = new Asterisk::AGI;
 %input = $AGI->ReadParse();
 #
$AGI->say_number('9865');
$AGI->say_digits('873746');
 
speak("Hello World");
 
 
 
sub speak
  {
$text = $_[0];
 
my $hash = md5_hex($text);
 
my $ttsdir = "/var/lib/asterisk/sounds/tts";
my $cepoptions = "-p audio/sampling-rate=8000,audio/channels=1";
 
my $wavefile = "$ttsdir/tts-$hash.wav";
 
unless (-f $wavefile)
  {
open(fileOUT, ">/var/lib/asterisk/sounds/tts/say-text-$hash.txt");
print fileOUT "$text";
close(fileOUT);
 
my $execf="/opt/swift/bin/swift -f $ttsdir/say-text-$hash.txt -o 
$wavefile $cepoptions";
system($execf);
 
unlink("$ttsdir/say-text-$hash.txt");
  }
$filename = 'tts/'.basename('tts/'.basename($wavefile,".wav"));
$AGI->stream_file($filename);
#  unlink("$wavefile");

This function I took from internet where i found it


My problem is that i cant hear anything when play the file sound using  
$AGI->stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound but I cant 
hear anything.

I thought maybe was the codec but asterisk can play .wav
But this works
$AGI->say_number('9865');


Any help or cluees will be so appreciate~!
Thks!


*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun"
*---*

_
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Re: [asterisk-users] Diagnosing dropped calls...

[John Faubion]

>Try dropping the IAX2 and only use SIP.  Don't ask "why?"  

Well in our case we were NOT using IAX at all. Strictly SIP.

>You could be hitting an overloaded router or whatever along 
>the way, 10mbs fiber does not mean low latency or lost packets.

So true, hence the reason I suggested using mtr to check it. Many times in
our case we saw gateways between networks that were dropping packets
presumably  due to overload conditions. RTP traffic over UDP would add far
more load than the ICMP packets used for mtr.

>Seriously though, if your business lives and dies by the 
>phone system, get T1 with SIP from your provider directly 
>(point to point) with G729 or just get a real ISDN or POTS lines.

With this particular customer we initially had unrealistic time and budget
constraints. We had 11 days between when they contacted us and when the
system had to be in place. With an average install time around here of 30-35
days that ruled out the T1. Their initial monthly telephone budget was $150
per month. Like I said, unrealistic.

>Usually, once they perceive a problem, then even if the other 
>side of the call is on a cell and the cell drops the call, 

Again so true. This customer got to the point that if a call dropped when
they we on their cell phone, regardless of the fact that the call was not
going through our system, we got a trouble call. Thankfully, a week after
cutting over to the T1 we called them to make sure they hadn't found someone
else. 8)

John


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Re: [asterisk-users] Mail Server

[Felipe Trevisan]

I´ve managed to put it to work, very simply.
Just created an A DNS entry pointing to my system. This procedure validates
the reverse lookup, gmail and others do, before accepting the mail in their
inboxes.

All my sendmail emails gets delivered with no need of smarthosts, therefore,
no need to SSl or TLS auth.

Sorry, for posting on the wrong list. When I started using asterisk I had no
clue, whatsoever on Linux, and tought that as the problem arose when trying
to build an asterisk server, that these should be the right forums.

But thanks all for the help.


Felipe


On Thu, Mar 20, 2008 at 12:16 AM, Al lists <[EMAIL PROTECTED]> wrote:

> Or maybe you can show him some links ;)
> Try this for send mail:
> http://docs.snake.de/smtp-auth.html
>
> this is very common these days and to make it more fun each mailserver
> (provider) has their own criteria to decide if your email is spam or not.
> to give you and example:
> make sure you are using static public IP address for outgoing mails, have a
> PTR record for that IP and also A record for the fqn that those mails are
> coming from.
> For smtp auth you need to have saslauth in place and most recent sendmails
> are compile with saslauth these days.
> I did not have 100% success with smtp and sasl and i believe that was
> caused due to have different TLS versions.
> anyway that link should put you in the right direction and if anyone else
> has a better/easier mta that handles smarthost and auth flawlessly, please
> comment.
>
>
>
> On Sun, Mar 16, 2008 at 3:48 PM, linuxian iandsd <[EMAIL PROTECTED]>
> wrote:
>
>> well, maybe ou're on the wrong list (talkin sendmail in an asterisk list
>> !!!) you're better in sendmail's list.
>>
>> anyway, you need to modify sendmail.cf file, just a few tweaks & it will
>> be ok.  you will need a smarthost, what is a smarthost ? thats an smtp
>> server that is allowed to send mail to the world, without it you can't send
>> mail, & this smarthost will be your isp's smtp server & noone else's unless
>> you know a lot of ppl around. otherwise your mails will get nowhere.
>>
>> if you need an sendmail.cf file example i can paste it for you here.
>> also dovecot.conf will be valuable for you.
>>
>>
>> hope this helps.
>>
>>
>> On Fri, Mar 14, 2008 at 1:52 PM, Felipe Trevisan <[EMAIL PROTECTED]>
>> wrote:
>>
>>> How would you relay on Google Apps, as Google requires SSL or TLS
>>> authentication?
>>>
>>> How can I configure sendmail to do this?
>>>
>>>
>>> Actually, sendmail is trying to send email directly, and I get the
>>> response below. I´ll now try Mike Hammett´s solution.
>>>
>>> Thanks,
>>>
>>> Felipe Trevisan
>>>
>>>
>>>
>>> *Message contents*
>>>
>>> The original message was received at Thu, 13 Mar 2008 23:49:31 -0300
>>> from trixbox1.localdomain [127.0.0.1]
>>>
>>>- The following addresses had permanent fatal errors -
>>>
>>>
>>>
>>> <[EMAIL PROTECTED]>
>>> (reason: 550-5.7.1 [201.6.192.115] The IP you're using to send email is 
>>> not authorized
>>>
>>>
>>> )
>>>
>>>- Transcript of session follows -
>>>
>>> ... while talking to gmail-smtp-in.l.google.com.:
>>> >>> DATA
>>> <<< 550-5.7.1 [201.6.192.115] The IP you're using to send email is not 
>>> authorized
>>>
>>>
>>>
>>> <<< 550-5.7.1 to send email directly to our servers. Please use
>>> <<< 550 5.7.1 the SMTP relay at your service provider instead. 
>>> a44si4966479rne.2
>>> 554 5.0.0 Service unavailable
>>>
>>>  *Failed delivery status*   *Final recipient* [EMAIL PROTECTED]  *Reason
>>> for failure* 550-5.7.1 [201.6.192.115] The IP you're using to send email
>>> is not authorized  *Remote mail server* gmail-smtp-in.l.google.com  
>>> *Reporting
>>> mail server* trixbox1.localdomain
>>>
>>>
>>>
>>> On Thu, Mar 13, 2008 at 7:13 PM, Mike Hammett <[EMAIL PROTECTED]>
>>> wrote:
>>>
  Through help from people on the lists and then further investigation
 based on those results, here is what I did.

 1)  I set the office to a statically assigned IP instead of from the
 pool.
 2)  I made an A entry on one of my domains aiur.ics-il.net (where aiur
 is the machine name).
 3)  I added aiur.ics-il.net directly after 127.0.0.1 in the /etc/hosts
 file (copied below).
 4)  I set the from email address (serveremail) in
 /etc/asterisk/voicemail.conf to something at the domain I created (
 [EMAIL PROTECTED]).
 5)  Presto!

 [EMAIL PROTECTED] ~]# cat /etc/hosts
 # Do not remove the following line, or various programs
 # that require network functionality will fail.
 127.0.0.1   aiur.ics-il.net Aiurlocalhost.localdomain
 localhost
 ::1 localhost6.localdomain6 localhost6


 --
 Mike Hammett
 Intelligent Computing Solutions
 http://www.ics-il.com



 - Original Message -
  *From:* Mike Hammett <[EMAIL PROTECTED]>
 *To:* Asterisk Users Mailing List - Non-Commercial 
 Discussion
  *Sent:* Thursday,

Re: [asterisk-users] US T1 Hangup Detection

[Daniel Hazelbaker]

Another update on the latest hookup attempt.  I can make it work  
reasonably well with callprogress=yes, it detects the hangup but only  
after about 7-9 seconds.  My config files are the same as the last  
time I posted (apparently last time I wasn't waiting long enough for  
callprogress to kick in).  If I turn callprogress=off then it never  
hangs up, if I turn it on, again after 7-9 seconds, it will hangup.   
My current Adtran unit hangs up almost instantly, so I think it is  
getting and detecting the "disconnect supervision" correctly.

I am going to try again in a few days, but I want to make some  
modifications to the zaptel driver (add in some debug code) and revert  
to Asterisk 1.4, not that I expect that to make a difference but I'll  
try anything at this point.  Before I do a question:

I am using esf,b8zs signalling on a true(/inband) T1 line.  Does the  
Zaptel driver use the rxwink timing to detect a hangup by the  
disconnect supervision?  If not, what does it use as it must be able  
to tell the difference (and I am using analog terms here) between a  
hangup and a "flash" for 3-way calling.

Thanks again.  If I can't figure this out I will have to call them  
out, but I don't think they will do anything other than say "yep it  
works on our side, fix your own equipment."

Daniel

On Jul 8, 2008, at 1:11 PM, Daniel Hazelbaker wrote:

> Just an update for the information I got from Verizon:
>
> It is a "true" T1, not a PRI for sure.  b8zs and esf signalling.  It
> is loop start with disconnect supervision (kewlstart as I understand
> it).  I know I had already tried kewlstart before, but I suppose it is
> possible that some other configuration option was making it not work.
> Since the only T1 line I have coming in is our live phone lines I will
> have to test this again late some night this week, so if somebody has
> an idea of some things to check while I am doing that I will certainly
> give it a shot, otherwise I will report back afterwords if I had
> success.


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Re: [asterisk-users] Asterisk as an IVR solution

[Steve Totaro]

On Thu, Jul 10, 2008 at 9:07 PM, Tilghman Lesher <
[EMAIL PROTECTED]> wrote:

> On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
> > It's a known problem.
> >
> > If you call Background() in a macro, then Asterisk will look for the
> > extensions to jump to in the CALLING Macro/context and NOT the Macro that
> > the Background() app was called in.
>
> I wouldn't call it a known problem.  It works precisely as it was designed
> to
> work.  It may not work the way that you want it to, but it works like a
> Macro:
> an independent set of instructions, with substitution, that acts as if it
> were
> invoked inline with the calling location.  That is why Background will
> match
> in the context of the calling location: it acts like it never left that
> original context (and, in a way, it really didn't).
>
> Subroutines are a different beast, and they are available with the Gosub/
> Return set of routines in app_stack.so.
>
> --
> Tilghman



See  this thread for info on  who  paid for ExternalIVR, who uses(used?) it
and some history.
http://www.asteriskguru.com/archives/image-vp255203.html


Thanks,
Steve Totaro
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Re: [asterisk-users] Asterisk as an IVR solution

[Tilghman Lesher]

On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
> It's a known problem.
>
> If you call Background() in a macro, then Asterisk will look for the
> extensions to jump to in the CALLING Macro/context and NOT the Macro that
> the Background() app was called in.

I wouldn't call it a known problem.  It works precisely as it was designed to
work.  It may not work the way that you want it to, but it works like a Macro:
an independent set of instructions, with substitution, that acts as if it were
invoked inline with the calling location.  That is why Background will match
in the context of the calling location: it acts like it never left that
original context (and, in a way, it really didn't).

Subroutines are a different beast, and they are available with the Gosub/
Return set of routines in app_stack.so.

-- 
Tilghman

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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Tilghman Lesher]

On Thursday 10 July 2008 18:48:15 Rob Hillis wrote:
> Matt Watson wrote:
> > That being said... i;m also quite pleased to see T.38 support being
> > worked on for Asterisk... its a pretty important area to further
> > develop IMHO.
>
> I absolutely agree.  It's been a notable omission for some time.
> Unfortunately getting it written isn't the major part of the battle.
> Due to the license incompatibilities between SpanDSP and Asterisk, this
> work is almost guaranteed /not/ to make it into the base Asterisk souce.
> (from what I recall, the problem is that Steve won't sign the disclaimer
> Digium requires for work to make it in to Asterisk - and I hasten to add
> that I don't blame him, given that if he did, his GPL work would end up
> as part of a fully commercial version of Asterisk!)

Actually, what has motivated this recent change is that Steve has released
SpanDSP under the LGPL, which is fully compatible with the Digium license.
So while his insistence on the GPL has previously kept it out of the core
Asterisk distribution, his recent change in licensing is what will make
putting it into Asterisk possible.

> Therefore, the /real/ battle is keeping the code maintained so that it
> will continue to work with Asterisk.

Not anymore, unless he returns to the GPL license.  And even then, the
version that he released as LGPL can still be independently maintained.

-- 
Tilghman

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Re: [asterisk-users] Diagnosing dropped calls...

[Steve Totaro]

Try dropping the IAX2 and only use SIP.  Don't ask "why?"  Just give it a
try and see if things improve for you.

Also when you assume, you make and "ass" out of you and "me" (just a little
joke, get it? ass-u-me.)

You could be hitting an overloaded router or whatever along the way, 10mbs
fiber does not mean low latency or lost packets.

Seriously though, if your business lives and dies by the phone system, get
T1 with SIP from your provider directly (point to point) with G729 or just
get a real ISDN or POTS lines.

And then you will still have "dropped" calls depending on your volume and
how vocal your users are.  Usually, once they perceive a problem, then even
if the other side of the call is on a cell and the cell drops the call, you
will get a complaint.  The only way to track those down are on a case by
case basis with ANI II codes 61-63
http://www.nanpa.com/number_resource_info/ani_ii_assignments.html

Thanks,
Steve Totaro

On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez <[EMAIL PROTECTED]>
wrote:

>My customer has a 10mpbs fiber connection to the Internet so we have
> always assumed that the connection is not really a problem.  We will
> look into it.  Thank you.
>
>
>
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Re: [asterisk-users] Should I remove the blank options?

[Tzafrir Cohen]

On Fri, Jul 11, 2008 at 11:36:49AM +1200, Lists wrote:
> Hi all,
> 
> I am very new to asterisk and I am just looking through the config files 
> to try and understand them a bit.
> In my zapata-auto.conf file I have
> ; Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2"
> ;;; line="5 WCTDM/1/0 FXOKS (In use)"
> signalling=fxo_ks
> callerid="Channel 5" <6005>
> mailbox=6005
> group=5
> context=from-internal
> channel => 5
> callerid=
> mailbox=
> group=
> context=default
> 
> Is there any reason I have two callerid mailbox group and context? 
> Should I remove the second lot.
> Also both contexts are defined one as from-internal and the other as 
> default will this cause a problem if left?

The format of zapata.conf (up until 1.4) required that workaround. When
the configuration is generated, it is generated for each channel
separately. If I set the context to, say, "from-internal" for this
channel, I cannot assume it won't be something else on another one.

One method to combat this is to always set all the "options" for all
channels. Even options I don't need for this channel: I might use them
on another one.

Or even worse: what happens if someone #include-s some arbitrary zaptel
configuration in zapata.conf after zapata-channels.conf (as is in, e.g.,
trixbox).

So I chose to try to "reset" the options. I can't just "save" the
configuration before configuring the channel and "restore" it later.
So I must revert to resetting to defualt values.

The proper fix: in Asterisk 1.6 zapata.conf supports separate contenxts
for channels. This allows me to safely apply some configurations just
for a single channel (or a group of them).

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

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Re: [asterisk-users] Asterisk as an IVR solution

[Douglas Garstang]

It's a known problem.

If you call Background() in a macro, then Asterisk will look for the extensions 
to jump to in the CALLING Macro/context and NOT the Macro that the Background() 
app was called in.

Eg:

[macro-DoSomething]
exten => s,1,Background(Prompt)
exten => 1,1,NoOP()

[context1]
exten => s,1,Macro(DoSomething)

If you press 1, Asterisk will look for an extension '1' in the context 
'context1', NOT the 'DoSomething' macro.

Doug.



- Original Message 
From: Al Baker <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, July 10, 2008 4:50:19 PM
Subject: Re: [asterisk-users] Asterisk as an IVR solution

Why can't you call Background() from a MACRO ?
Isn't is just an Application like any other ?
Curious minds want to know !

Quote "There's also the fact that you can't
> call Backgound() in a macro,"

Douglas Garstang wrote:
> Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the 
> input in seconds. If you try and use 0, it seems to drop back to a 
> default of 5s.
>
> - Original Message 
> From: MFH <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, July 10, 2008 12:37:31 PM
> Subject: Re: [asterisk-users] Asterisk as an IVR solution
>
> From what I can tell Read allows for a floating point input which uses
> ast_waitfordigit that accepts milliseconds as input.
>
> Douglas Garstang wrote:
> > Admittedly I have not used the ExternalIVR app. Is it any good?
> >
> > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure,
> > it can do it, but boy it is UGLY. There's also the fact that you can't
> > call Backgound() in a macro, which forces you to use Read() which
> > won't accept a timeout of <1s. There's no DTMF background detection
> > while playing SayDigits so you have to roll your own by calling an
> > external AGI and concatenating sound files. Yuck. By the time you code
> > in logic for handling timeouts and incorrect responses to menu's with
> > all the gotos and what-not, it turns into a god aweful mess.
> >
> > Sure, you can do it.
> >
> > Doug.
> >
> >
> >
> > - Original Message 
> > From: Steve Totaro <[EMAIL PROTECTED] 
> >
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> >  >
> > Sent: Thursday, July 10, 2008 10:37:55 AM
> > Subject: Re: [asterisk-users] Asterisk as an IVR solution
> >
> >
> >
> > On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED] 
> 
> > >> wrote:
> >
> >Hi.
> >
> >We are building an application that will provide users with the
> >ability to call in and report an absence. The caller will have to
> >validate themselves and the call tree will be dynamic, based on
> >data in a MySQL database. We will have many customers, each
> >calling a separate phone number, each having a different call
> >tree. New customers will be added regularly and we do not want a
> >solution that requires extensive programming each time (the call
> >trees are different in subtle ways from each other).
> >
> >Is Asterisk a great solution for this? If not do you know what
> >would? If so, we need someone to help us set it up, can you
> >suggest someone?
> >
> >Thanks in advance. Best.
> >
> >Mark
> >
> >
> > Asterisk certainly is a great solution for this.  If you find you need
> > or want extra flexibility,  the external IVR app. 
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
> >
> > Thanks,
> > Steve Totaro
> >
> > 
> >
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Re: [asterisk-users] changing inbuilt sound messages

[Tzafrir Cohen]

On Fri, Jul 11, 2008 at 09:56:29AM +1200, Lists wrote:
> I only did the 420 because thats what the original files looked like?
> r-- -w- ---
> Should I change this to 644?

Yes!

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Re: [asterisk-users] Asterisk as an IVR solution

[Al Baker]

Yes , you could easily do this with asterisk.
If you have formal specs for this project, I would be interested in exactly
what you are trying to do. Email me off-line.

Steve Totaro wrote:
>
>
> On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED] 
> > wrote:
>
> Hi.
>
> We are building an application that will provide users with the
> ability to call in and report an absence. The caller will have to
> validate themselves and the call tree will be dynamic, based on
> data in a MySQL database. We will have many customers, each
> calling a separate phone number, each having a different call
> tree. New customers will be added regularly and we do not want a
> solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what
> would? If so, we need someone to help us set it up, can you
> suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>
>
> Asterisk certainly is a great solution for this.  If you find you need 
> or want extra flexibility,  the external IVR app.  
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
>
> Thanks,
> Steve Totaro
> 
>
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Re: [asterisk-users] Asterisk as an IVR solution

[Al Baker]

Why can't you call Background() from a MACRO ?
Isn't is just an Application like any other ?
Curious minds want to know !

Quote "There's also the fact that you can't
 > call Backgound() in a macro,"

Douglas Garstang wrote:
> Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the 
> input in seconds. If you try and use 0, it seems to drop back to a 
> default of 5s.
>
> - Original Message 
> From: MFH <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, July 10, 2008 12:37:31 PM
> Subject: Re: [asterisk-users] Asterisk as an IVR solution
>
> From what I can tell Read allows for a floating point input which uses
> ast_waitfordigit that accepts milliseconds as input.
>
> Douglas Garstang wrote:
> > Admittedly I have not used the ExternalIVR app. Is it any good?
> >
> > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure,
> > it can do it, but boy it is UGLY. There's also the fact that you can't
> > call Backgound() in a macro, which forces you to use Read() which
> > won't accept a timeout of <1s. There's no DTMF background detection
> > while playing SayDigits so you have to roll your own by calling an
> > external AGI and concatenating sound files. Yuck. By the time you code
> > in logic for handling timeouts and incorrect responses to menu's with
> > all the gotos and what-not, it turns into a god aweful mess.
> >
> > Sure, you can do it.
> >
> > Doug.
> >
> >
> >
> > - Original Message 
> > From: Steve Totaro <[EMAIL PROTECTED] 
> >
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> >  >
> > Sent: Thursday, July 10, 2008 10:37:55 AM
> > Subject: Re: [asterisk-users] Asterisk as an IVR solution
> >
> >
> >
> > On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED] 
> 
> > >> wrote:
> >
> >Hi.
> >
> >We are building an application that will provide users with the
> >ability to call in and report an absence. The caller will have to
> >validate themselves and the call tree will be dynamic, based on
> >data in a MySQL database. We will have many customers, each
> >calling a separate phone number, each having a different call
> >tree. New customers will be added regularly and we do not want a
> >solution that requires extensive programming each time (the call
> >trees are different in subtle ways from each other).
> >
> >Is Asterisk a great solution for this? If not do you know what
> >would? If so, we need someone to help us set it up, can you
> >suggest someone?
> >
> >Thanks in advance. Best.
> >
> >Mark
> >
> >
> > Asterisk certainly is a great solution for this.  If you find you need
> > or want extra flexibility,  the external IVR app. 
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
> >
> > Thanks,
> > Steve Totaro
> >
> > 
> >
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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Rob Hillis]

Matt Watson wrote:
> That being said... i;m also quite pleased to see T.38 support being 
> worked on for Asterisk... its a pretty important area to further 
> develop IMHO.

I absolutely agree.  It's been a notable omission for some time.  
Unfortunately getting it written isn't the major part of the battle.  
Due to the license incompatibilities between SpanDSP and Asterisk, this 
work is almost guaranteed /not/ to make it into the base Asterisk souce. 
(from what I recall, the problem is that Steve won't sign the disclaimer 
Digium requires for work to make it in to Asterisk - and I hasten to add 
that I don't blame him, given that if he did, his GPL work would end up 
as part of a fully commercial version of Asterisk!)

Therefore, the /real/ battle is keeping the code maintained so that it 
will continue to work with Asterisk.

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[asterisk-users] Should I remove the blank options?

[Lists]

Hi all,

I am very new to asterisk and I am just looking through the config files 
to try and understand them a bit.
In my zapata-auto.conf file I have
; Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2"
;;; line="5 WCTDM/1/0 FXOKS (In use)"
signalling=fxo_ks
callerid="Channel 5" <6005>
mailbox=6005
group=5
context=from-internal
channel => 5
callerid=
mailbox=
group=
context=default

Is there any reason I have two callerid mailbox group and context? 
Should I remove the second lot.
Also both contexts are defined one as from-internal and the other as 
default will this cause a problem if left?

Thanks
Kate

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Re: [asterisk-users] Diagnosing dropped calls...

[Carlos Chavez]

My customer has a 10mpbs fiber connection to the Internet so we have
always assumed that the connection is not really a problem.  We will
look into it.  Thank you.

On Thu, 2008-07-10 at 17:49 -0500, John Faubion wrote:
> > -Original Message-
> > Subject: [asterisk-users] Diagnosing dropped calls...
> > 
> > I have a system that is driving me nuts.  My customer 
> > is running Asterisk 1.4.20.1 on a CentOS 5.2 server.  It is a 
> > purely SIP and IAX2 service with no cards installed and it 
> > uses ztdummy from Zaptel 1.4.11.
> > They use Teliax for calls to the USA and Protel for calls in Mexico.
> > 
> > The problem is that users complain that their calls get dropped.
> > Sometimes every few minutes and sometimes after a very long 
> > call over two hours so there is no clear pattern.  There is a 
> 
> I just went through this same scenario. The customer originally using Teliax
> over AT&T DSL. Incoming calls would sometimes have a delay of 1-2 seconds
> before connecting and some would just drop at random intervals. We would
> generally see in the logs that our connection to Teliax would often show
> lagged or unresponsive around the time of the drops. Testing with AT&T
> revealed that the customer was 18,034 feet from the CO so we decided to move
> them to a cable modem from Charter. This was done as it was the only other
> internet provider short of a dedicated T1. Things improved but did not
> completely subside. The delay on answering was becoming the major complaint.
> We decided to try a different provider so an account with Junction Networks
> was added. This eliminated the delay immediately but did not eliminate the
> dropped calls. One of the techs at Teliax told us that the delay was an
> issue caused by Asterisk 1.4 trying to talk with Asterisk 1.4 server. We
> tried their work around but it did not eliminate the delay issue though it
> did reduce the issue. We also had Charter monitor the service for nearly two
> weeks in which time they replaced the modem 4 times and claimed to have
> repaired connectors, replaced repeater nodes and changed other connections
> to no avail. The customer lives and dies by their telephone so we moved them
> to Cbeyond SIP Connect. All of the issues are now gone. The is no delay,
> they some times spend hours at a time on training calls with no issues. I
> also did a bit of testing with the Junction Network account after moving to
> Cbeyond. This was done since we still had money in the account anyway. Again
> no more issues with dropped calls. The key is in an Internet connection that
> is either very clean or supports QoS. The Cbeyond connection seems to have
> both. Obviously we don't have QoS back to Junction but with the cleaner
> connection it doesn't seem to matter. Also if you trying to use a cable
> modem, make sure your using a business class service and not a residential
> service. 
> 
> Also use mtr (mytraceroute) to look at the latency between your server and
> the proxy. Under Charter and AT&T, both of which were business class
> services, the mtr would sometimes show up to 20% packet loss at various
> systems between us and the proxy. Also try different proxies. For us the
> best one was typically either Atlanta or Denver. Keep in mind that I'm not
> bashing Teliax or Junction. I use both Teliax at home and Junction in our
> office, over a cable modem provided by Time Warner and the only issue I have
> is the occasional delay on answer with Teliax. However that one is the home
> line and the delay seems to work great for telemarketers. 
> 
> John
> 
> 
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Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Diagnosing dropped calls...

[John Faubion]

> -Original Message-
> Subject: [asterisk-users] Diagnosing dropped calls...
> 
>   I have a system that is driving me nuts.  My customer 
> is running Asterisk 1.4.20.1 on a CentOS 5.2 server.  It is a 
> purely SIP and IAX2 service with no cards installed and it 
> uses ztdummy from Zaptel 1.4.11.
> They use Teliax for calls to the USA and Protel for calls in Mexico.
> 
>   The problem is that users complain that their calls get dropped.
> Sometimes every few minutes and sometimes after a very long 
> call over two hours so there is no clear pattern.  There is a 

I just went through this same scenario. The customer originally using Teliax
over AT&T DSL. Incoming calls would sometimes have a delay of 1-2 seconds
before connecting and some would just drop at random intervals. We would
generally see in the logs that our connection to Teliax would often show
lagged or unresponsive around the time of the drops. Testing with AT&T
revealed that the customer was 18,034 feet from the CO so we decided to move
them to a cable modem from Charter. This was done as it was the only other
internet provider short of a dedicated T1. Things improved but did not
completely subside. The delay on answering was becoming the major complaint.
We decided to try a different provider so an account with Junction Networks
was added. This eliminated the delay immediately but did not eliminate the
dropped calls. One of the techs at Teliax told us that the delay was an
issue caused by Asterisk 1.4 trying to talk with Asterisk 1.4 server. We
tried their work around but it did not eliminate the delay issue though it
did reduce the issue. We also had Charter monitor the service for nearly two
weeks in which time they replaced the modem 4 times and claimed to have
repaired connectors, replaced repeater nodes and changed other connections
to no avail. The customer lives and dies by their telephone so we moved them
to Cbeyond SIP Connect. All of the issues are now gone. The is no delay,
they some times spend hours at a time on training calls with no issues. I
also did a bit of testing with the Junction Network account after moving to
Cbeyond. This was done since we still had money in the account anyway. Again
no more issues with dropped calls. The key is in an Internet connection that
is either very clean or supports QoS. The Cbeyond connection seems to have
both. Obviously we don't have QoS back to Junction but with the cleaner
connection it doesn't seem to matter. Also if you trying to use a cable
modem, make sure your using a business class service and not a residential
service. 

Also use mtr (mytraceroute) to look at the latency between your server and
the proxy. Under Charter and AT&T, both of which were business class
services, the mtr would sometimes show up to 20% packet loss at various
systems between us and the proxy. Also try different proxies. For us the
best one was typically either Atlanta or Denver. Keep in mind that I'm not
bashing Teliax or Junction. I use both Teliax at home and Junction in our
office, over a cable modem provided by Time Warner and the only issue I have
is the occasional delay on answer with Teliax. However that one is the home
line and the delay seems to work great for telemarketers. 

John


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Re: [asterisk-users] changing inbuilt sound messages

[Lists]

I only did the 420 because thats what the original files looked like?
r-- -w- ---
Should I change this to 644?

Kate

Tzafrir Cohen wrote:

On Thu, Jul 10, 2008 at 10:18:21AM +0200, Giorgio Incantalupo wrote:
  

Lists wrote:


Hi all,

I am wanting to change the sound files from the standard ones to a New 
Zealand voice pack.
I have copied the files into the /var/lib/asterisk/sounds directory and 
chowned them to asterisk:asterisk and chmod 420 
  


420? r-- -w- --- ?

Why not stick with the standard 644 or maybe 664?

Anyway, the Asterisk user does not need to write access to the standard
sound files. Only read-access.

  
to match the existing 
files but the system is still using the original files.
The original files seem to be wav files while the NZ voice pack ones are 
gsm files.
  


Only gsm? Any higher quality originals from which to produce other
formats?

  

How do I get the system to use the new gsm files?
  

Hi,
try to delete old .wav ones.
Why not using a sounds/nz subfolder and set language to nz?



en-nz?

  
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Re: [asterisk-users] (announce) asterisk T.38 gateway

[marek cervenka]

> marek cervenka wrote:
>> hi,
>>
>> there is T.38 fax gateway for asterisk
>> http://bugs.digium.com/view.php?id=12931
>>
>> please test it and report bugs
>>
>> for people from
>> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
>> if you still want donate t.38 development please contact me at cervajs at
>> fpf.slu.cz
>>
> And you will, of course, pass on 99% of the money to those who did 99%
> of the work, won't you? :-)

if you want, it's no problem (sponsors please CC: steveu at coppice.org)

but by now nobody respond ... (surprisingly)
contract for primary developer(dafe) is exhausted. that's the reason for 
bounty request

SpanDSP is good piece of software and BIG credit goes to you Steve
thanks!

---
Marek Cervenka
===


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[asterisk-users] Diagnosing dropped calls...

[Carlos Chavez]

I have a system that is driving me nuts.  My customer is running
Asterisk 1.4.20.1 on a CentOS 5.2 server.  It is a purely SIP and IAX2
service with no cards installed and it uses ztdummy from Zaptel 1.4.11.
They use Teliax for calls to the USA and Protel for calls in Mexico.

The problem is that users complain that their calls get dropped.
Sometimes every few minutes and sometimes after a very long call over
two hours so there is no clear pattern.  There is a very strange
behavior when some of the calls are dropped.  Some of these users call a
conference bridge, when their calls get dropped they will immediately
dial back and be connected to the same conference without having to put
in their PIN numbers, as if the call just dropped on their phones but
Asterisk kept the channel open and then reconnected the phone.  Some
others redial to the same conference and when they log in they realize
that their old call is still active and it gets dropped after a few
minutes.

I am having a big problem trying to diagnose why the calls get dropped.
I do not know if it is the phones, Asterisk or the VoIP provider.  What
tools would you use to determine the cause of the disconnection?

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


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Re: [asterisk-users] Asterisk as an IVR solution

[Douglas Garstang]

Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the input in 
seconds. If you try and use 0, it seems to drop back to a default of 5s.


- Original Message 
From: MFH <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, July 10, 2008 12:37:31 PM
Subject: Re: [asterisk-users] Asterisk as an IVR solution

>From what I can tell Read allows for a floating point input which uses 
ast_waitfordigit that accepts milliseconds as input.

Douglas Garstang wrote:
> Admittedly I have not used the ExternalIVR app. Is it any good?
>
> I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, 
> it can do it, but boy it is UGLY. There's also the fact that you can't 
> call Backgound() in a macro, which forces you to use Read() which 
> won't accept a timeout of <1s. There's no DTMF background detection 
> while playing SayDigits so you have to roll your own by calling an 
> external AGI and concatenating sound files. Yuck. By the time you code 
> in logic for handling timeouts and incorrect responses to menu's with 
> all the gotos and what-not, it turns into a god aweful mess.
>
> Sure, you can do it.
>
> Doug.
>
>
>
> - Original Message 
> From: Steve Totaro <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, July 10, 2008 10:37:55 AM
> Subject: Re: [asterisk-users] Asterisk as an IVR solution
>
>
>
> On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED] 
> > wrote:
>
> Hi.
>
> We are building an application that will provide users with the
> ability to call in and report an absence. The caller will have to
> validate themselves and the call tree will be dynamic, based on
> data in a MySQL database. We will have many customers, each
> calling a separate phone number, each having a different call
> tree. New customers will be added regularly and we do not want a
> solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what
> would? If so, we need someone to help us set it up, can you
> suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>
>
> Asterisk certainly is a great solution for this.  If you find you need 
> or want extra flexibility,  the external IVR app.  
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
>
> Thanks,
> Steve Totaro
>
> 
>
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Re: [asterisk-users] Asterisk as an IVR solution

[MFH]

 From what I can tell Read allows for a floating point input which uses 
ast_waitfordigit that accepts milliseconds as input.

Douglas Garstang wrote:
> Admittedly I have not used the ExternalIVR app. Is it any good?
>
> I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, 
> it can do it, but boy it is UGLY. There's also the fact that you can't 
> call Backgound() in a macro, which forces you to use Read() which 
> won't accept a timeout of <1s. There's no DTMF background detection 
> while playing SayDigits so you have to roll your own by calling an 
> external AGI and concatenating sound files. Yuck. By the time you code 
> in logic for handling timeouts and incorrect responses to menu's with 
> all the gotos and what-not, it turns into a god aweful mess.
>
> Sure, you can do it.
>
> Doug.
>
>
>
> - Original Message 
> From: Steve Totaro <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, July 10, 2008 10:37:55 AM
> Subject: Re: [asterisk-users] Asterisk as an IVR solution
>
>
>
> On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED] 
> > wrote:
>
> Hi.
>
> We are building an application that will provide users with the
> ability to call in and report an absence. The caller will have to
> validate themselves and the call tree will be dynamic, based on
> data in a MySQL database. We will have many customers, each
> calling a separate phone number, each having a different call
> tree. New customers will be added regularly and we do not want a
> solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what
> would? If so, we need someone to help us set it up, can you
> suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>
>
> Asterisk certainly is a great solution for this.  If you find you need 
> or want extra flexibility,  the external IVR app.  
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR
>
> Thanks,
> Steve Totaro
>
> 
>
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Re: [asterisk-users] Asterisk as an IVR solution

[Douglas Garstang]

Admittedly I have not used the ExternalIVR app. Is it any good?

I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do 
it, but boy it is UGLY. There's also the fact that you can't call Backgound() 
in a macro, which forces you to use Read() which won't accept a timeout of <1s. 
There's no DTMF background detection while playing SayDigits so you have to 
roll your own by calling an external AGI and concatenating sound files. Yuck. 
By the time you code in logic for handling timeouts and incorrect responses to 
menu's with all the gotos and what-not, it turns into a god aweful mess.

Sure, you can do it.

Doug.




- Original Message 
From: Steve Totaro <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, July 10, 2008 10:37:55 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution




On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED]> wrote:

Hi.

We are building an application that will provide users with the ability to call 
in and report an absence. The caller will have to validate themselves and the 
call tree will be dynamic, based on data in a MySQL database. We will have many 
customers, each calling a separate phone number, each having a different call 
tree. New customers will be added regularly and we do not want a solution that 
requires extensive programming each time (the call trees are different in 
subtle ways from each other).

Is Asterisk a great solution for this? If not do you know what would? If so, we 
need someone to help us set it up, can you suggest someone?

Thanks in advance. Best.

Mark

Asterisk certainly is a great solution for this.  If you find you need or want 
extra flexibility,  the external IVR app.  
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR

Thanks,
Steve Totaro



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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Matt Watson]

I'd probably be a little pissed if I were Steve Underwood if somebody
pocketed over 10k $USD for taking credit for a product that my free library
did the bulk of the work for.

I don;t think i'd feel that the entire bounty should be mine - after all
there would of been nothing stopping me from doing it myself... but credit
should be given where credit is due.   Even if its just something as a sign
of appreciation.

Given that spandsp is GPL'd Steve obviously never intended to make a ton of
money off of it... but I;m sure he'd love to receive something for his work,
or use some of that money to further develop spandsp.

That being said... i;m also quite pleased to see T.38 support being worked
on for Asterisk... its a pretty important area to further develop IMHO.

--
Matt Watson
http://www.mattgwatson.ca


On Thu, Jul 10, 2008 at 11:54 AM, Steve Totaro <
[EMAIL PROTECTED]> wrote:

>
>
> On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro <
> [EMAIL PROTECTED]> wrote:
>
>>
>>
>> On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood <[EMAIL PROTECTED]>
>> wrote:
>>
>>> Vinícius Fontes wrote:
>>> > When people release software under the GPL license, like Steve
>>> Underwood did with libunicall, spandsp and so on, they were supposed to know
>>> that other people has the right to use their code.
>>> >
>>> The problem is that almost any licence term which tries to limit the
>>> obnoxious behaviour of other people has too many unpleasant side
>>> effects. GPL 2.0 is the best compromise I've found, so that is what I
>>> used for everything unless recently. To make my stuff licence compatible
>>> with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This
>>> is having undesirable consequences, though. Its really a tough issue,
>>> and GPL 2.0 showed immense foresight in just accepting the non-existence
>>> of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat.
>>>
>>> Most of the time I just want to give up producing anything at all.
>>>
>>> Steve
>>>
>>
>> So are you angry that he may gain monetarily from your your work, or is it
>> hurt pride that he is basically taking credit for it?
>>
>> The answer to that should guide you in how you release your work in the
>> future.
>>
>> Thanks,
>> Steve Totaro
>>
>>
> I also want to add that if someone asked me to name the top five names that
> came to mind when thinking of Asterisk, Jim Dixon, Mark Spencer, Steve
> Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave
> anybody out ;)
>
> Thanks,
> Steve
>
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Re: [asterisk-users] Asterisk as an IVR solution

[Leotis buchanan]

Hey,

I am doing a similar project , which we will be integrating mysql db and a
ivr, maybe we can work on this together since we will be sharing components.
This should save us both some time.



On Thu, Jul 10, 2008 at 12:25 PM, Mark Carpenter <[EMAIL PROTECTED]> wrote:

> Hi.
> We are building an application that will provide users with the ability to
> call in and report an absence. The caller will have to validate themselves
> and the call tree will be dynamic, based on data in a MySQL database. We
> will have many customers, each calling a separate phone number, each having
> a different call tree. New customers will be added regularly and we do not
> want a solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what would? If
> so, we need someone to help us set it up, can you suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>
>
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-- 
Leotis Buchanan
Manager/Electronic Design Systems Engineer
Exterbox.com
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Re: [asterisk-users] Asterisk as an IVR solution

[Leotis buchanan]

Yes,asterisk can do that

On Thu, Jul 10, 2008 at 12:25 PM, Mark Carpenter <[EMAIL PROTECTED]> wrote:

> Hi.
> We are building an application that will provide users with the ability to
> call in and report an absence. The caller will have to validate themselves
> and the call tree will be dynamic, based on data in a MySQL database. We
> will have many customers, each calling a separate phone number, each having
> a different call tree. New customers will be added regularly and we do not
> want a solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what would? If
> so, we need someone to help us set it up, can you suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>
>
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>



-- 
Leotis Buchanan
Manager/Electronic Design Systems Engineer
Exterbox.com
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Re: [asterisk-users] Asterisk as an IVR solution

[Steve Totaro]

On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter <[EMAIL PROTECTED]> wrote:

> Hi.
> We are building an application that will provide users with the ability to
> call in and report an absence. The caller will have to validate themselves
> and the call tree will be dynamic, based on data in a MySQL database. We
> will have many customers, each calling a separate phone number, each having
> a different call tree. New customers will be added regularly and we do not
> want a solution that requires extensive programming each time (the call
> trees are different in subtle ways from each other).
>
> Is Asterisk a great solution for this? If not do you know what would? If
> so, we need someone to help us set it up, can you suggest someone?
>
> Thanks in advance. Best.
>
> Mark
>

Asterisk certainly is a great solution for this.  If you find you need or
want extra flexibility,  the external IVR app.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExternalIVR

Thanks,
Steve Totaro
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[asterisk-users] Asterisk as an IVR solution

[Mark Carpenter]

Hi.
We are building an application that will provide users with the ability to
call in and report an absence. The caller will have to validate themselves
and the call tree will be dynamic, based on data in a MySQL database. We
will have many customers, each calling a separate phone number, each having
a different call tree. New customers will be added regularly and we do not
want a solution that requires extensive programming each time (the call
trees are different in subtle ways from each other).

Is Asterisk a great solution for this? If not do you know what would? If so,
we need someone to help us set it up, can you suggest someone?

Thanks in advance. Best.

Mark
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[asterisk-users] Tracking Call Time While in Dial()

[Douglas Garstang]

So, I've been asked if this is possible.

Someone wants to actively monitor the duration of a call, while the call is 
still in progress. Obviously, in Asterisk, once the Dial() application starts, 
you lose dial plan control until after the call has ended, successful or 
otherwise.

Anyone know if that kind of thing is possible?

Doug.


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[asterisk-users] Festival issues

[Balu Raman]

I have used this [EMAIL PROTECTED] for a while now, 2 years.
Only recently, I am trying Festival and on invoking festival --server I get
these errors :

/usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version
`CXXABI_1.2' not found (required by /usr/share/festival/bin/festival)
/usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version
`GLIBCPP_3.2' not found (required by /usr/share/festival/bin/festival)

I have libstdc++.so.6 on the system and symbolically linking did not fix it
either.
The festival package is asteriskathome-festival-1.96.zip.

Thanks for your help.
- balu raman
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Re: [asterisk-users] astrundir not used

[Tilghman Lesher]

On Tuesday 08 July 2008 11:53:47 Cyril SCETBON wrote:
> hi,
>
> I'im using asterisk 4.1.21 and astrundir is configured as followed in
> /etc/asterisk/asterisk.conf :
>
> [global]
> astetcdir => /etc/asterisk
> astmoddir => /usr/lib/asterisk/modules
> astvarlibdir => /var/lib/asterisk
> astagidir => /usr/share/asterisk/agi-bin
> astspooldir => /var/spool/asterisk
> astrundir => /var/run/asterisk
> astlogdir => /var/log/asterisk
>
> when I start asterisk it creates his pid file and the ctl socket in
> /var/run and not in /var/run/asterisk
>
> How can I fix it ? Is it a known issue ? I did not get this error with
> asterisk 1.4.10

Those settings should actually be in the [directories] context, not the
[global] context.  Change that, and it should work just fine.

-- 
Tilghman

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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Steve Totaro]

On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro <
[EMAIL PROTECTED]> wrote:

>
>
> On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood <[EMAIL PROTECTED]>
> wrote:
>
>> Vinícius Fontes wrote:
>> > When people release software under the GPL license, like Steve Underwood
>> did with libunicall, spandsp and so on, they were supposed to know that
>> other people has the right to use their code.
>> >
>> The problem is that almost any licence term which tries to limit the
>> obnoxious behaviour of other people has too many unpleasant side
>> effects. GPL 2.0 is the best compromise I've found, so that is what I
>> used for everything unless recently. To make my stuff licence compatible
>> with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This
>> is having undesirable consequences, though. Its really a tough issue,
>> and GPL 2.0 showed immense foresight in just accepting the non-existence
>> of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat.
>>
>> Most of the time I just want to give up producing anything at all.
>>
>> Steve
>>
>
> So are you angry that he may gain monetarily from your your work, or is it
> hurt pride that he is basically taking credit for it?
>
> The answer to that should guide you in how you release your work in the
> future.
>
> Thanks,
> Steve Totaro
>
>
I also want to add that if someone asked me to name the top five names that
came to mind when thinking of Asterisk, Jim Dixon, Mark Spencer, Steve
Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave
anybody out ;)

Thanks,
Steve
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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Steve Totaro]

On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood <[EMAIL PROTECTED]>
wrote:

> Vinícius Fontes wrote:
> > When people release software under the GPL license, like Steve Underwood
> did with libunicall, spandsp and so on, they were supposed to know that
> other people has the right to use their code.
> >
> The problem is that almost any licence term which tries to limit the
> obnoxious behaviour of other people has too many unpleasant side
> effects. GPL 2.0 is the best compromise I've found, so that is what I
> used for everything unless recently. To make my stuff licence compatible
> with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This
> is having undesirable consequences, though. Its really a tough issue,
> and GPL 2.0 showed immense foresight in just accepting the non-existence
> of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat.
>
> Most of the time I just want to give up producing anything at all.
>
> Steve
>

So are you angry that he may gain monetarily from your your work, or is it
hurt pride that he is basically taking credit for it?

The answer to that should guide you in how you release your work in the
future.

Thanks,
Steve Totaro
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Re: [asterisk-users] Simple Call Screener

[Jared Smith]

On Wed, 2008-07-09 at 17:54 -0400, Ryan M. Colbert wrote:
> I'm trying to build a simple accept/reject screening app for inbound
> calls that * forwards to my cell phone.  Basically I want * to
> announce the caller ID and then let me press 1 to accept the call or 2
> to reject the call and send the outside party to voicemail.

While you can certainly do it by using a dial macro, a simpler method is
to check out the "p" and "P" options to the Dial() application.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Asterisk conference call with a HuntGroup

[Alexander Olekhnovich]

I'm sorry to be more concrete I want to make conference with all the hunt
group and caller by just dialing the HuntGroup.

On Thu, Jul 10, 2008 at 4:18 PM, Alexander Olekhnovich <
[EMAIL PROTECTED]> wrote:

> For example, here is a dial plan to call a huntgroup (111,222,333)
>
> [reach-hunt]
> exten => _X.,1,Dial(SIP/111&SIP/222&SIP/333|timeout|G(extra-context^s^1))
> exten => _X.,2,Hangup()
>
> Here in the dial plan if one of the numbers will answer the call it will be
> transferred to extension 2, and the caller to 1 of the extra-context. Other
> channels will be dropped. But I want to transfer to some extension to
> process somehow them.
> 1. Is it possible using standard features?
> 2. Is it possible by Asterisk design at all?
>
>
> On Thu, Jul 10, 2008 at 3:54 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>
>> Alexander Olekhnovich wrote:
>> > I just think because of the Asterisk design it can not be implemented.
>> >
>> > On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich
>> > <[EMAIL PROTECTED] > wrote:
>> >
>> > Hi,
>> >
>> > I'm interested if it's possible to configure Asterisk the
>> > following way: user calls a huntgroup, and then when one of the
>> > hunts answers the call, other hunts are not hung up, but Asterisk
>> > transfers the callees to some extensions, or something else.
>> >
>>
>> Perhaps if you were a little more clear in precisely what you want,
>> others may be able to help.  What exactly are you trying to achieve?
>>
>>
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>>
>
>
>
> --
> Best Regards
> Alexander Olekhnovich




-- 
Best Regards
Alexander Olekhnovich
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[asterisk-users] Asterisk hangup not working on inbound calls

[Giorgio Incantalupo]

Hi,
I have an Asterisk 1.2.18 box with a TDM400P card.
If I make a call and then I hangup the phone, the call ends correctly 
but if I receive a call and I hangup the phone the other party does not 
get the hangup signal from Asterisk.

Is there anybody who can explain this strange behaviour?

Thank you.

Giorgio Incantalupo.

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Re: [asterisk-users] Simple Call Screener

[Mark G. Thomas]

Hi Ryan,

On Wed, Jul 09, 2008 at 05:54:28PM -0400, Ryan M. Colbert wrote:
> I'm trying to build a simple accept/reject screening app for inbound calls 
> that * forwards to my cell phone.  Basically I want * to announce the caller 
> ID and then let me press 1 to accept the call or 2 to reject the call and 
> send the outside party to voicemail.

I'm doing something similar, ringing a Zaptel port and calling a cellphone,
as per below. I  had success using Read() to get the accept/reject.

Note that you need to Set(MACRO_RESULT=CONTINUE) if you do NOT want to accept
the call on the cellphone, and hence want the diaplan to CONTINUE.

In my case, I don't need to announce the caller ID since the cellphone
simply displays it, and I'm letting the human or analog voicemail on the 
zaptel port take the call if the cellphone user doesn't press 1.

> I've been messing around with variation of the script below... can anyone 
> tell me what I'm doing wrong?  It's got to be something obvious that I've 
> overlooked.
> 
> Thanks!!!
> 
> [main]
> exten => s,1,Answer
> exten => s,n,Ringing
> exten => s,n,Wait(1)
> exten => s,n,Dial(SIP/[EMAIL PROTECTED],120,gM(screen))
> exten => s,n,PlayBack(vm-goodbye)
> exten => s,n,Hangup
> 
> [macro-screen]
> exten => s,1,Wait(1)
> ;exten => s,n,SayDigits(${CALLERID(num)})
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=30)
> exten => s,n,Background(accept-reject)
> 
> exten => 1,1,Set(MACRO_RESULT=CONTINUE)
> exten => 2,1,PlayBack(vm-goodbye)
> exten => 2,2,Hangup
> 
> exten => s,6,Wait(10)
> exten => i,1,Goto(TT_VO,s,1)
...

[inbound]
exten => 211212,1,Playtones(ring)
exten => 211212,n,Dial(${PTNR}&local/[EMAIL PROTECTED],,t)

[internals]
exten => 101,1,Dial(${MARKCELL},30,tM(screen))

[macro-screen]
exten => s,1,Wait(0.5)
exten => s,n,Read(ACCEPT,inbound,1,,1,20)
exten => s,n,GotoIf($["${ACCEPT}" = "1"]?yes:no)
exten => s,n(yes),Background(connecting)
exten => s,n,Goto(end)
exten => s,n(no),Set(MACRO_RESULT=CONTINUE)
exten => s,n(end),NoOp

-- 
Mark G. Thomas ([EMAIL PROTECTED])

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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Steve Underwood]

Vinícius Fontes wrote:
> When people release software under the GPL license, like Steve Underwood did 
> with libunicall, spandsp and so on, they were supposed to know that other 
> people has the right to use their code.
>   
The problem is that almost any licence term which tries to limit the 
obnoxious behaviour of other people has too many unpleasant side 
effects. GPL 2.0 is the best compromise I've found, so that is what I 
used for everything unless recently. To make my stuff licence compatible 
with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This 
is having undesirable consequences, though. Its really a tough issue, 
and GPL 2.0 showed immense foresight in just accepting the non-existence 
of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat.

Most of the time I just want to give up producing anything at all.

Steve


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Re: [asterisk-users] Zap Bridged Channels

[Cosmin Prund]

I've had a similar problem with a A101DX + A202DX. I was trying to bridge from 
my A101 to my A202 to get faxes over my E1 line. I've done an number of things, 
I'm not exactly sure which one helped, but it now works very nice for fax:

 

(1) Using "zap show channel N" on the CLI I noticed that echo canceling was on 
even those I was bridging two Zap devices. I disabled the HWEC on my A202 card 
and it's now ok (no echo cancel on the A202 card). This was an option for me 
because I'm only doing fax on the analog card. I don't think this had a lot to 
do with the final "fix" of the problem.

(2) I emailed Sangoma and they told me there's a newer version of the drivers 
that tweek the echo cancel algorithm to make it better suited for fax. The 
driver was beta at the time so I didn't try it. You might want to contact 
Sangoma yourself!

(3) I fixed my "zttest" timing! When I tested I had really bad timing (94,00 
worst and < 99,00 average). The docs and the wiki say that's bad timing but I 
had absolutely no problems with voice quality. None! And I've only done one 
thing to fix my timing: "/etc/init.d/irqbalance restart". I have no idea why 
that makes a difference but it does and I've now got 99,95 average timing from 
zttest with the worst being over 99,00. You might want to try this yourself 
since you also seem to have the "X" (pci express) version of the card.

 

--

Cosmin Prund

 

 

De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Jeremy Mann
Trimis: Wednesday, July 09, 2008 10:28 PM
Către: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subiect: [asterisk-users] Zap Bridged Channels

 

I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for 
modem connectivity.

 

I have Zap/8 as a Fax Machine

 

Zap/5 is my outside line.  When a call rings in on Zap/5 it immediately calls 
Zap/8 and bridges the channels.  I see it doing a native bridge on the two.  I 
have echo cancel off on native bridge, but I can never get fax connectivity, it 
just tries to negotiate forever then eventually hangs up.

 

Anything special to getting this to work?  

 

Below is an example of CLI output when the Fax Machine tries to call out, it 
does the same thing, never get the two machines to complete the call and send 
the fax.  I've also included the CLI output of channel 5's properties, it does 
show the EC as off.  I noticed it says "Fax Handled: no", is there something I 
need to enable in Zapata.conf or zaptel.conf?

 

Would txgain/rxgain be the issue?

 

CLI Output 

-- Starting simple switch on 'Zap/8-1'

-- Executing [EMAIL PROTECTED]:1] Answer("Zap/8-1", "") in new stack

-- Executing [EMAIL PROTECTED]:2] Dial("Zap/8-1", "Zap/5") in new stack

-- Called 5

-- Zap/5-1 is ringing

-- Zap/5-1 is ringing

-- Zap/5-1 answered Zap/8-1

-- Native bridging Zap/8-1 and Zap/5-1

 

localhost*CLI> zap show channel 5

Channel: 5CLI>

File Descriptor: 27

Span: 2

Extension:

Dialing: no

Context: from-internal-fax

Caller ID:

Calling TON: 0

Caller ID name:

Destroy: 0

InAlarm: 0

Signalling Type: FXO Kewlstart

Radio: 0

Owner: Zap/5-1

Real: Zap/5-1

Callwait: 

Threeway: 

Confno: -1

Propagated Conference: -1

Real in conference: 0

DSP: yes

Relax DTMF: yes

Dialing/CallwaitCAS: 0/0

Default law: ulaw

Fax Handled: no

Pulse phone: no

Echo Cancellation: 128 taps unless TDM bridged, currently OFF

Master Channel: 8

Actual Confinfo: Num/8, Mode/0x0009

Actual Confmute: No

Hookstate (FXS only): Onhook

 

Zapata.conf -

 

[channels]

context=default

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=no

relaxdtmf=yes

rxgain=2.0

txgain=2.0

group=1

callgroup=1

pickupgroup=1

immediate=no

context=from-internal-fax

group=1

signalling = fxo_ks

channel => 5

context=from-zaptel-fax

group=3

signalling = fxs_ks

channel => 8

 



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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Steve Underwood]

Hi Mark,

I have no issue with people using something in new and interesting ways. 
Adding a small wrapper around a large library, and asking for a bounty 
is in a rather different category from that.

At least this is honest work. You should see some of the sleazy ways 
people have made money from my code.

In the present case, there is no real issue. I think only one entry on 
that T.38 bounty page was ever going to pay anything, anyway.

Regards,
Steve


Mark Hamilton wrote:
> I agree. In that case people who use includes in their scripts for 
> which they got paid should pay a portion of their pay to the writer of 
> each include they use.
>
>
>
>  Original Message 
> Subject: Re: [asterisk-users] (announce) asterisk T.38 gateway
> From: Rob Hillis <[EMAIL PROTECTED] .org>
> Date: Thu, July 10, 2008 8:52 am
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion<[EMAIL PROTECTED]
> .com>
>
> Steve Underwood wrote:
> > marek cervenka wrote:
> >
> >> hi,
> >>
> >> there is T.38 fax gateway for asterisk
> >> http://bugs.digium.com/view.php?id=12931
> >>
> >> please test it and report bugs
> >>
> >> for people from
> >> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
> >> if you still want donate t.38 development please contact me at
> cervajs at
> >> fpf.slu.cz 
> >>
> >>
> > And you will, of course, pass on 99% of the money to those who
> did 99%
> > of the work, won't you? :-)
> >
> > This is the problem with bounties. They favour those who easily
> string
> > together existing functionality, rather than those who do the heavy
> > lifting. I know several of bounties that have been paid to
> people who
> > wrote just a few lines of code to string together some of the
> > functionality I provided.
> >
>
> Undoubtedly I will get myself flamed for suggesting this, but
> perhaps it
> may pay for you to be a little more attentive to the bounties that
> are
> on offer. The bounty offered may utilise a lot of your code, but a
> library without an interface (be it a user interface or a technical
> interface) is just a blob of code.
>
> If developing the end user code doesn't interest you, that's fine
> - but
> don't take it out on people who take your work that extra step to
> make
> it do what someone else is willing to offer money for.
>


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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Vinícius Fontes]

When people release software under the GPL license, like Steve Underwood did 
with libunicall, spandsp and so on, they were supposed to know that other 
people has the right to use their code.

Atenciosamente,

Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
 
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

- "Mark Hamilton" <[EMAIL PROTECTED]> escreveu:

> I agree. In that case people who use includes in their scripts for
> which they got paid should pay a portion of their pay to the writer of
> each include they use.
> 
> 
> 
> 
> 
>  Original Message 
> Subject: Re: [asterisk-users] (announce) asterisk T.38 gateway
> From: Rob Hillis < [EMAIL PROTECTED] .org>
> Date: Thu, July 10, 2008 8:52 am
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion< [EMAIL PROTECTED] .com>
> 
> Steve Underwood wrote:
> > marek cervenka wrote:
> >
> >> hi,
> >>
> >> there is T.38 fax gateway for asterisk
> >> http://bugs.digium.com/view.php?id=12931
> >>
> >> please test it and report bugs
> >>
> >> for people from
> >> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
> >> if you still want donate t.38 development please contact me at
> cervajs at
> >> fpf.slu.cz
> >>
> >>
> > And you will, of course, pass on 99% of the money to those who did
> 99%
> > of the work, won't you? :-)
> >
> > This is the problem with bounties. They favour those who easily
> string
> > together existing functionality, rather than those who do the heavy
> > lifting. I know several of bounties that have been paid to people
> who
> > wrote just a few lines of code to string together some of the
> > functionality I provided.
> >
> 
> Undoubtedly I will get myself flamed for suggesting this, but perhaps
> it
> may pay for you to be a little more attentive to the bounties that are
> on offer. The bounty offered may utilise a lot of your code, but a
> library without an interface (be it a user interface or a technical
> interface) is just a blob of code.
> 
> If developing the end user code doesn't interest you, that's fine -
> but
> don't take it out on people who take your work that extra step to make
> it do what someone else is willing to offer money for.
> 
> 
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Re: [asterisk-users] res_odbc.conf and odbc show

[Vieri]

Replying to myself.
A "reload" isn't enough in 1.2.27. I needed to restart asterisk.



  

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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Mark Hamilton]

I agree. In that case people who use includes in their scripts for which they got paid should pay a portion of their pay to the writer of each include they use. 

 Original Message 
Subject: Re: [asterisk-users] (announce) asterisk T.38 gateway
From: Rob Hillis <[EMAIL PROTECTED].org>
Date: Thu, July 10, 2008 8:52 am
To: Asterisk Users Mailing List - Non-Commercial
Discussion<[EMAIL PROTECTED].com>

Steve Underwood wrote:
> marek cervenka wrote:
>   
>> hi,
>>
>> there is T.38 fax gateway for asterisk
>> http://bugs.digium.com/view.php?id=12931
>>
>> please test it and report bugs
>>
>> for people from
>> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
>> if you still want donate t.38 development please contact me at cervajs at 
>> fpf.slu.cz
>>   
>> 
> And you will, of course, pass on 99% of the money to those who did 99% 
> of the work, won't you? :-)
>
> This is the problem with bounties. They favour those who easily string 
> together existing functionality, rather than those who do the heavy 
> lifting. I know several of bounties that have been paid to people who 
> wrote just a few lines of code to string together some of the 
> functionality I provided.
>   

Undoubtedly I will get myself flamed for suggesting this, but perhaps it 
may pay for you to be a little more attentive to the bounties that are 
on offer.  The bounty offered may utilise a lot of your code, but a 
library without an interface (be it a user interface or a technical 
interface) is just a blob of code.

If developing the end user code doesn't interest you, that's fine - but 
don't take it out on people who take your work that extra step to make 
it do what someone else is willing to offer money for.


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Re: [asterisk-users] Friday June 11th: SIP love/hate

[randulo]

Obviously I meant JULY 11th.

On Thu, Jul 10, 2008 at 3:17 PM, randulo <[EMAIL PROTECTED]> wrote:
> Tomorrow at 12 Noon EDT, we'll exchange some views and notes about
> SIP, the protocol you usually can not avoid even if yiou wanted to.
>
> - Past, present and future of SIP
>
> - SIP greatest strengths and weaknesses.
>
> - What's the state of chan_sip in asterisk code
>
> - anything else anyone wants to add about motorcycles and SIP ;)
>
> To listen or participate:
>
> http://bit.ly/voip
>
> a little before 12 Noon EDT (9AM PDT, 11 Central, 4PM GMT)
> phone sip:[EMAIL PROTECTED] or PSTN: Call (724) 444-7444
>
> Enter 22622# then 1# or your PIN if you registered at Talkshoe.
> There are several guest PIN available if you grab my attention on the
> IRC channel below.
> The PIN lets me know who's speaking.
>
> IRC is on Freenode.net #voip-users-conference
>
> Forums, blogs, etc: http://bit.ly/voipusers
>
> Recordings: http://bit.ly/archives
>
>
> If http://bit.ly is down, just see http://voipusersconference.org for
> the info :)
>

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[asterisk-users] Why it keeps display the G729 codec during the call running on the consol

[bilal ghayyad]

Hi All;

I do not know why when I select the codec to be G729, it keeps display it on 
the Asterisk CLI during the call as following:

G729 
G729
G729
...
...

Any advise how to let this stop?
Regards
Bilal


  

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[asterisk-users] Friday June 11th: SIP love/hate

[randulo]

Tomorrow at 12 Noon EDT, we'll exchange some views and notes about
SIP, the protocol you usually can not avoid even if yiou wanted to.

- Past, present and future of SIP

- SIP greatest strengths and weaknesses.

- What's the state of chan_sip in asterisk code

- anything else anyone wants to add about motorcycles and SIP ;)

To listen or participate:

http://bit.ly/voip

a little before 12 Noon EDT (9AM PDT, 11 Central, 4PM GMT)
phone sip:[EMAIL PROTECTED] or PSTN: Call (724) 444-7444

Enter 22622# then 1# or your PIN if you registered at Talkshoe.
There are several guest PIN available if you grab my attention on the
IRC channel below.
The PIN lets me know who's speaking.

IRC is on Freenode.net #voip-users-conference

Forums, blogs, etc: http://bit.ly/voipusers

Recordings: http://bit.ly/archives


If http://bit.ly is down, just see http://voipusersconference.org for
the info :)

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Re: [asterisk-users] Asterisk conference call with a HuntGroup

[Alexander Olekhnovich]

For example, here is a dial plan to call a huntgroup (111,222,333)

[reach-hunt]
exten => _X.,1,Dial(SIP/111&SIP/222&SIP/333|timeout|G(extra-context^s^1))
exten => _X.,2,Hangup()

Here in the dial plan if one of the numbers will answer the call it will be
transferred to extension 2, and the caller to 1 of the extra-context. Other
channels will be dropped. But I want to transfer to some extension to
process somehow them.
1. Is it possible using standard features?
2. Is it possible by Asterisk design at all?

On Thu, Jul 10, 2008 at 3:54 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:

> Alexander Olekhnovich wrote:
> > I just think because of the Asterisk design it can not be implemented.
> >
> > On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich
> > <[EMAIL PROTECTED] > wrote:
> >
> > Hi,
> >
> > I'm interested if it's possible to configure Asterisk the
> > following way: user calls a huntgroup, and then when one of the
> > hunts answers the call, other hunts are not hung up, but Asterisk
> > transfers the callees to some extensions, or something else.
> >
>
> Perhaps if you were a little more clear in precisely what you want,
> others may be able to help.  What exactly are you trying to achieve?
>
>
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-- 
Best Regards
Alexander Olekhnovich
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Re: [asterisk-users] Asterisk conference call with a HuntGroup

[Rob Hillis]

Alexander Olekhnovich wrote:
> I just think because of the Asterisk design it can not be implemented.
>
> On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich 
> <[EMAIL PROTECTED] > wrote:
>
> Hi,
>
> I'm interested if it's possible to configure Asterisk the
> following way: user calls a huntgroup, and then when one of the
> hunts answers the call, other hunts are not hung up, but Asterisk
> transfers the callees to some extensions, or something else.
>

Perhaps if you were a little more clear in precisely what you want, 
others may be able to help.  What exactly are you trying to achieve?


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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Rob Hillis]

Steve Underwood wrote:
> marek cervenka wrote:
>   
>> hi,
>>
>> there is T.38 fax gateway for asterisk
>> http://bugs.digium.com/view.php?id=12931
>>
>> please test it and report bugs
>>
>> for people from
>> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
>> if you still want donate t.38 development please contact me at cervajs at 
>> fpf.slu.cz
>>   
>> 
> And you will, of course, pass on 99% of the money to those who did 99% 
> of the work, won't you? :-)
>
> This is the problem with bounties. They favour those who easily string 
> together existing functionality, rather than those who do the heavy 
> lifting. I know several of bounties that have been paid to people who 
> wrote just a few lines of code to string together some of the 
> functionality I provided.
>   

Undoubtedly I will get myself flamed for suggesting this, but perhaps it 
may pay for you to be a little more attentive to the bounties that are 
on offer.  The bounty offered may utilise a lot of your code, but a 
library without an interface (be it a user interface or a technical 
interface) is just a blob of code.

If developing the end user code doesn't interest you, that's fine - but 
don't take it out on people who take your work that extra step to make 
it do what someone else is willing to offer money for.


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[asterisk-users] RTP packets dropped

[Vinícius Fontes]

As RTP packets have a sequential number, is there some logging/debugging option 
in Asterisk to monitor how many packets have been lost on a SIP call?

Atenciosamente,

Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
 
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

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Re: [asterisk-users] asterisk 1.4.21.1 seg fault

[Sean Bright]

Jerry Geis wrote:
>>
>> What should I do now?
>>
> silly me it is 1.4.21.1 not 1.2.21.1
> 

If you haven't already, I'd suggest reporting an issue in mantis.

http://bugs.digium.com/

-- 
Sean Bright
[EMAIL PROTECTED]

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Re: [asterisk-users] Asterisk conference call with a HuntGroup

[Alexander Olekhnovich]

I just think because of the Asterisk design it can not be implemented.

On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich <
[EMAIL PROTECTED]> wrote:

> Hi,
>
> I'm interested if it's possible to configure Asterisk the following way:
> user calls a huntgroup, and then when one of the hunts answers the call,
> other hunts are not hung up, but Asterisk transfers the callees to some
> extensions, or something else.
>
> --
> Best Regards
> Alexander Olekhnovich




-- 
Best Regards
Alexander Olekhnovich
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[asterisk-users] Asterisk conference call with a HuntGroup

[Alexander Olekhnovich]

Hi,

I'm interested if it's possible to configure Asterisk the following way:
user calls a huntgroup, and then when one of the hunts answers the call,
other hunts are not hung up, but Asterisk transfers the callees to some
extensions, or something else.

-- 
Best Regards
Alexander Olekhnovich
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Re: [asterisk-users] (announce) asterisk T.38 gateway

[Steve Underwood]

marek cervenka wrote:
> hi,
>
> there is T.38 fax gateway for asterisk
> http://bugs.digium.com/view.php?id=12931
>
> please test it and report bugs
>
> for people from
> http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
> if you still want donate t.38 development please contact me at cervajs at 
> fpf.slu.cz
>   
And you will, of course, pass on 99% of the money to those who did 99% 
of the work, won't you? :-)

This is the problem with bounties. They favour those who easily string 
together existing functionality, rather than those who do the heavy 
lifting. I know several of bounties that have been paid to people who 
wrote just a few lines of code to string together some of the 
functionality I provided.

Steve


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[asterisk-users] callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable

[Remco Barendse]

Hi list,

My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get 
this error message on the asterisk console:

== Starting post polarity CID detection on channel 4
 -- Starting simple switch on 'Zap/4-1'
[Jul  8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: 
Couldn't detect start-character. CID parsing might be unreliable

A long time ago my CallerID used to work with the same settings. I don't 
really need CallerID but it would be nice to have it working. I am located 
in The Netherlands.

Any suggestions?

This is in my /etc/zaptel.conf :
fxoks=1
fxsks=4 
loadzone=nl 
defaultzone=nl 
This is in my /etc/asterisk/zapata.conf :
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

callerid=202
signalling=fxo_ks
group=1
context=intern-all
channel=>1

signalling=fxs_ks
immediate=yes 
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callwaiting=no 
callwaitingcallerid=no
adsi=no 
group=2 
context=inbound-analog
channel=>4


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Re: [asterisk-users] Simple Call Screener

[MFH]

This is what I use.  The Read does have a default timeout but you should 
be able to put your own.

extensions.conf:

   exten => 
s,n(dial),Dial(SIP/sipura2_1&SIP/sipura1_1&SIP/sipura2_2&SIP/spa942_3&SIP/aastra480_3,20,mtTM(screen))
   exten => s,n(vmail),Voicemail([EMAIL PROTECTED])


[macro-screen]

   exten => s,1,Wait(0.2)
   exten => s,n,Read(ACCEPT,screen-callee-options,1)
   exten => s,n,GotoIf($[${ACCEPT} = 1]?ok:cont)
   exten => s,n(ok),Noop
   exten => s,s+2(cont),Set(MACRO_RESULT=CONTINUE)




> I'm trying to build a simple accept/reject screening app for inbound calls 
> that * forwards to my cell phone.  Basically I want * to announce the caller 
> ID and then let me press 1 to accept the call or 2 to reject the call and 
> send the outside party to voicemail.
>
> I've been messing around with variation of the script below... can anyone 
> tell me what I'm doing wrong?  It's got to be something obvious that I've 
> overlooked.
>
> Thanks!!!
>
> [main]
> exten => s,1,Answer
> exten => s,n,Ringing
> exten => s,n,Wait(1)
> exten => s,n,Dial(SIP/[EMAIL PROTECTED],120,gM(screen))
> exten => s,n,PlayBack(vm-goodbye)
> exten => s,n,Hangup
>
> [macro-screen]
> exten => s,1,Wait(1)
> ;exten => s,n,SayDigits(${CALLERID(num)})
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=30)
> exten => s,n,Background(accept-reject)
>
> exten => 1,1,Set(MACRO_RESULT=CONTINUE)
> exten => 2,1,PlayBack(vm-goodbye)
> exten => 2,2,Hangup
>
> exten => s,6,Wait(10)
> exten => i,1,Goto(TT_VO,s,1)




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[asterisk-users] res_odbc.conf and odbc show

[Vieri]

I have a mixed PBX system with both Asterisk 1.4.21 and 1.2.27 (moving to 
1.2.28).
For now I need to keep a few boxes in 1.2 and not migrate them all to 1.4.
However, I would like to have func_odbc and res_odbc on all servers.

On 1.4.21, native func_odbc seems to work fine.

On 1.2.27, the func_odbc backport is giving me an error (I know that this 
backport is not "officially supported" but the issue I'm reporting is related 
to res_odbc).

"odbc show" does not display anything in 1.2.27. I'm expecting something like 
in 1.4.21:
# asterisk -rx "odbc show"
Name: astdb_cluster
DSN: astdb_cluster
Pooled: no
Connected: yes

Reference URLs for func_odbc are:

http://www.asterisk.org/func_odbc
http://svncommunity.digium.com/view/func_odbc/1.2/

The error I'm getting is:

Jul 10 12:07:04 VERBOSE[30281] logger.c: -- Executing 
Set("SIP/4053-b4410638"
, "ODBC_ASTDB_CLUSTER(voip2|CF/4053)=NULL") in new stack
Jul 10 12:07:04 DEBUG[30281] pbx.c: Function result is 'voip2'
Jul 10 12:07:04 DEBUG[30281] pbx.c: Function result is 'CF/4053'
Jul 10 12:07:04 ERROR[30281] func_odbc.c: Unable to load ODBC write class (check
 res_odbc.conf)


func_odbc.conf:

[ASTDB_CLUSTER]
; readhandle=astdb_cluster
; writehandle=astdb_cluster
dsn=astdb_cluster
readsql=SELECT value FROM astdb_cluster WHERE field='${SQL_ESC(${ARG1})}' and 
host='${SQL_ESC(${ARG2})}'
writesql=INSERT INTO astdb_cluster (host,field,value) VALUES 
('${SQL_ESC(${ARG1})}','${SQL_ESC(${ARG2})}',${VALUE})


res_odbc.conf:

[astdb_cluster]
enabled => yes
dsn => astdb_cluster
pre-connect => yes


/etc/unixODBC/odbc.ini:

[astdb_cluster]
Description = MySQL ODBC Driver ASTDB
Driver  = MySQL
Socket  = /var/run/mysqld/mysqld.sock
Server  = localhost
User= xx
Password= xx
Database= asteriskcluster
Option  = 3

("isql astdb_cluster" works fine even in * 1.2.27 but "asterisk -rx "odbc 
show"" doesn't)

Thanks

Vieri




  

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[asterisk-users] Why is the h extension being called ?

[Dovid B]

Call Flow:
1) Extension 10 calls out
2) Extension 10 transfers the call to extension 20
3) Extension 20 picks up the call.

Right when 10 transfers the call to 20 the h extension is invoked for extension 
10.  Why is this ? Is there any way to have the h extension not called on a 
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Re: [asterisk-users] changing inbuilt sound messages

[Tzafrir Cohen]

On Thu, Jul 10, 2008 at 10:18:21AM +0200, Giorgio Incantalupo wrote:
> Lists wrote:
> > Hi all,
> >
> > I am wanting to change the sound files from the standard ones to a New 
> > Zealand voice pack.
> > I have copied the files into the /var/lib/asterisk/sounds directory and 
> > chowned them to asterisk:asterisk and chmod 420 

420? r-- -w- --- ?

Why not stick with the standard 644 or maybe 664?

Anyway, the Asterisk user does not need to write access to the standard
sound files. Only read-access.

> > to match the existing 
> > files but the system is still using the original files.
> > The original files seem to be wav files while the NZ voice pack ones are 
> > gsm files.

Only gsm? Any higher quality originals from which to produce other
formats?

> >
> > How do I get the system to use the new gsm files?
> Hi,
> try to delete old .wav ones.
> Why not using a sounds/nz subfolder and set language to nz?

en-nz?

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

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Re: [asterisk-users] changing inbuilt sound messages

[Giorgio Incantalupo]

Hi,
try to delete old .wav ones.
Why not using a sounds/nz subfolder and set language to nz?

Giorgio


Lists wrote:
> Hi all,
>
> I am wanting to change the sound files from the standard ones to a New 
> Zealand voice pack.
> I have copied the files into the /var/lib/asterisk/sounds directory and 
> chowned them to asterisk:asterisk and chmod 420 to match the existing 
> files but the system is still using the original files.
> The original files seem to be wav files while the NZ voice pack ones are 
> gsm files.
>
> How do I get the system to use the new gsm files?
>
> Thanks
> Kate
>
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>   


-- 

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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