Re: [asterisk-users] New tutorial: storing audio recordings per day

[Lenz Emilitri]

Thank you! I updated the tutorial as well.
l.

2009/5/25 Atis Lezdins 

> On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri 
> wrote:
> > Hi everyone,
> > after doing the same thing multiple times and struggling to remember how
> it
> > was done, I have prepared a small tutorial that explains how to save
> > monitored files in different folders per day. This is quite useful
> > becausethe resultingfile system is way more manageable than having maybe
> > 100,000 files all saved in the same folder.
> > You can find the tutorial here:
> >
> > http://astrecipes.net/index.php?n=387
> >
> > As always, comments and suggestions are welcome.
> > l.
> > PS. I am also working on some scripts to "normalize" existing recordings
> > all-in-one-directory... if anybody is interested, please contact me.
>
> Actually You don't have to create folders in advance, as Asterisk will
> automatically create them when needed. Just make sure that Asterisk
> process is owner of parent directory.
>
>
> Set(__call_day=${STRFTIME(|${TIMEZONE}|%Y/%m/%d)});
> Set(MONITOR_FILENAME=${MONITOR_DIR}/${call_day}/call-${UNIQUEID});
> Monitor(ulaw,${MONITOR_FILENAME},b);
>
> Regards,
> Atis
>
> --
> Atis Lezdins,
> VoIP Project Manager / Developer,
> IQ Labs Inc,
> a...@iq-labs.net
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Cell Phone: +1 800 7300689
> Work phone: +1 800 7502835
>
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Re: [asterisk-users] Placing a MWI on-off call...

[eric weaver]

On Mon, May 25, 2009 at 9:52 PM, Olivier  wrote:

>
>
> 2009/5/25 eric weaver 
>
>> My grateful thanks to whoever can guide me in implementing this...
>>
>> I have a need to place calls via Asterisk Manager Protocol to a legacy PBX
>
> How are both boxes connected ?
>
The Asterisk box speaks via an Adtran TA-904 using PRI protocol to a T1 port
of the legacy PBX.

>
>
>
>> and twiddle its MWI lights.
>
> Which manage the phones you're talking about ?
>
The legacy PBX manages the phones.  This is for an external voicemail.
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Re: [asterisk-users] Placing a MWI on-off call...

[Olivier]

2009/5/25 eric weaver 

> My grateful thanks to whoever can guide me in implementing this...
>
> I have a need to place calls via Asterisk Manager Protocol to a legacy PBX

How are both boxes connected ?


> and twiddle its MWI lights.

Which manage the phones you're talking about ?

>
>
> By some means I get notified that an MWI light has to be changed.
> Via AMP:
> 1.  Send an ORIGINATE command honking up an incoming port on the PBX via a
> TA804.
>   a.  I presume that the TA804 should be a SIP channel.  yes/no?
>   b. The local extension I have made a pseudo extension that answers and
> plays a silent file.  Sensible?
>
> 2. loop calling Play_DTMF and maybe silence between, to the SIP channel to
> do the digits.
> 3. Need to sense fast-busy or stutter-dial-tone at this point.  Any
> pointers?
> 4. Hangup said SIP channel.
>
>
> Thanks in advance, etc.
>
>
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Re: [asterisk-users] howto store local exchange prefixes ?

[Anthony Messina]

 
 
original message-
From: "Jimmy Godbout" s...@inbox.com
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Date: Mon, 25 May 2009 18:01:11 -0800
-
 
 
> Check on www.localcallingguide.com. You'll find all npanxx that are local
to 
> your exchange.
> 
> Jimmy
>> -Original Message-
>> From: seandar...@gmail.com
>> Sent: Mon, 25 May 2009 21:39:30 -0400
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] howto store local exchange prefixes ?
>> 
>> Barry L. Kline wrote:
>>> -BEGIN PGP SIGNED MESSAGE-
>>> Hash: SHA1
>>> 
>>> sean darcy wrote:
>>> 
 I've looked at the Berkeley DB. That works pretty well, if the
 exchanges
 are all stored. But it looks like the exchanges have to be entered 1 by
 1 from the CLI. And can only be reviewed, corrected, or deleted from
 the
 CLI. I haven't found any simple frontend for the DB.
>>> 
>>> I do this be writing a dialplan which adds those entries. The first
>>> entry checks to see if the DB has been initialized and if so, skips to
>>> the lookup. Otherwise it loads each into the database before the
>>> lookup. It's very easy to write a quick script to generate the dialplan
>>> code.
>>> 
>>> Barry
>> 
>> Maybe I've not explained this correctly. I know, or can look up, the 40+
>> local exchanges that are local. I can parse the dial EXTEN to determine
>> the exchange. I can check the exchange against a DB. I want to determine
>> which exchanges are "local". I do not want to store an exchange dialed
>> by a user.
>> 
>> How can I store a lot of 3 digit numbers which I then can check against
>> an EXTEN to determine a local number?

in addition to localcallingguide, if your pstn connection is from at&t, you
can take a look at the script i made to grab only the "local" calls
(incurring no local-toll or long distance charges) which are"band a" and
"band b."

https://messinet.com/trac/telephony-tools/wiki/LocalCallingAreaGrabber

-- 

Anthony - http://messinet.com - http://messinet.com/~amessina/gallery



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Re: [asterisk-users] howto store local exchange prefixes ?

[Jimmy Godbout]

Check on www.localcallingguide.com. You'll find all npanxx that are local to 
your exchange.

Jimmy
> -Original Message-
> From: seandar...@gmail.com
> Sent: Mon, 25 May 2009 21:39:30 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] howto store local exchange prefixes ?
> 
> Barry L. Kline wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>> 
>> sean darcy wrote:
>> 
>>> I've looked at the Berkeley DB. That works pretty well, if the
>>> exchanges
>>> are all stored. But it looks like the exchanges have to be entered 1 by
>>> 1 from the CLI. And can only be reviewed, corrected, or deleted from
>>> the
>>> CLI. I haven't found any simple frontend for the DB.
>> 
>> I do this be writing a dialplan which adds those entries.  The first
>> entry checks to see if the DB has been initialized and if so, skips to
>> the lookup.  Otherwise it loads each into the database before the
>> lookup.  It's very easy to write a quick script to generate the dialplan
>> code.
>> 
>> Barry
> 
> Maybe I've not explained this correctly. I know, or can look up, the 40+
> local exchanges that are local. I can parse the dial EXTEN to determine
> the exchange. I can check the exchange against a DB. I want to determine
> which exchanges are "local". I do not want to store an exchange dialed
> by a user.
> 
> How can I store a lot of 3 digit numbers which I then can check against
> an EXTEN to determine a local number?
> 
> sean
> 
> 
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Re: [asterisk-users] howto store local exchange prefixes ?

[Michelle Dupuis]

I created a mysql table and lookup script for this.  One one server were we
could not use mysql, we created an array of exchanges and compared to those.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Monday, May 25, 2009 9:40 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] howto store local exchange prefixes ?

Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> sean darcy wrote:
> 
>> I've looked at the Berkeley DB. That works pretty well, if the 
>> exchanges are all stored. But it looks like the exchanges have to be 
>> entered 1 by
>> 1 from the CLI. And can only be reviewed, corrected, or deleted from 
>> the CLI. I haven't found any simple frontend for the DB.
> 
> I do this be writing a dialplan which adds those entries.  The first 
> entry checks to see if the DB has been initialized and if so, skips to 
> the lookup.  Otherwise it loads each into the database before the 
> lookup.  It's very easy to write a quick script to generate the 
> dialplan code.
> 
> Barry

Maybe I've not explained this correctly. I know, or can look up, the 40+
local exchanges that are local. I can parse the dial EXTEN to determine the
exchange. I can check the exchange against a DB. I want to determine which
exchanges are "local". I do not want to store an exchange dialed by a user.

How can I store a lot of 3 digit numbers which I then can check against an
EXTEN to determine a local number?

sean


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Re: [asterisk-users] howto store local exchange prefixes ?

[sean darcy]

Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> sean darcy wrote:
> 
>> I've looked at the Berkeley DB. That works pretty well, if the exchanges 
>> are all stored. But it looks like the exchanges have to be entered 1 by 
>> 1 from the CLI. And can only be reviewed, corrected, or deleted from the 
>> CLI. I haven't found any simple frontend for the DB.
> 
> I do this be writing a dialplan which adds those entries.  The first
> entry checks to see if the DB has been initialized and if so, skips to
> the lookup.  Otherwise it loads each into the database before the
> lookup.  It's very easy to write a quick script to generate the dialplan
> code.
> 
> Barry

Maybe I've not explained this correctly. I know, or can look up, the 40+ 
local exchanges that are local. I can parse the dial EXTEN to determine 
the exchange. I can check the exchange against a DB. I want to determine 
which exchanges are "local". I do not want to store an exchange dialed 
by a user.

How can I store a lot of 3 digit numbers which I then can check against 
an EXTEN to determine a local number?

sean


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Re: [asterisk-users] FritzBox 7270

[Philipp von Klitzing]

Hi!

> case-2
> Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your
> own Asterisk. Incoming VOIP-calls are answered by your own Asterisk. 

- the Fritz!Box usually doesn't "answer" unless you set it up for 
voicemail or fax

- define "forwarded": Do you mean "normally" an anlog phone would answer, 
but based on some condition (which?) you now want this call temporarily 
to go to your Asterisk box? That's what I meant with "divert is a 
misleading term", in the ISDN world diverting means a slightly different 
thing: "Don't call me nor my PBX, call him!"

> In case-2 Fritz has on the PSTN/ISDN-line an ordinairy DSL-modem,
> But also a FXO interface. While on the lan-side there is an VOIP-pbx, and
> a FXS interface for a local phone.
> 
> AFAICS, case-1 is do-able, but you don't gain anything with it.
> case-2 would give you two places for incoming calls (voip & PSTN)
> But i wonder if the HW would allow that (no fxo)
> 
> If case-2 is feasable, i'll dash-off for an 7270

Go run! :-) There is ISDN, S0 and also FXO in that box.

Philipp


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Re: [asterisk-users] Asterisk, SQL Database Update

[C. Savinovich]

Nothing is difficult my friend.   If you dedicate a few cups of coffee to
it,  a couple of days, and do some good googling, you will get it done
yourself.

 

Good luck!

 

CS

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T
Sent: Monday, May 25, 2009 3:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk, SQL Database Update

 

Thanks for your helpful reply.
 
I am not so good in coding.
 
simply all i need is as follow
 
When a call comes, goes into an IVR, and then depending on the entry option
it will connect to a remote SQL Database, to check the pre-existed data,
and in the end of the IVR the caller will enter an option that will need to
be written in the SQL Database.
 
Can you please give me a general scenrio how this will be achieved.
and which files that i will need to modify.
 
Thanks a lot.
 

 
> Date: Sun, 24 May 2009 22:15:31 +0200
> From: philipp.kemp...@amooma.de
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk, SQL Database Update
> 
> Torintino T schrieb:
> > Is there any method in Asterisk to enable the updating process
> > into another SQL database through entering IVR options during the call.
> 
> Depending on what you are trying to do there are various solutions:
> Channel Event Logging (CEL) - http://www.asterisk.org/node/48358
> AGI
> System()
> ODBC_*() functions
> 
> 
> Philipp Kempgen
> -- 
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
> -- 
> 
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Re: [asterisk-users] Channels configuration with DAHDI

[Daniel Bareiro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

El domingo 24 de mayo del 2009 a las 19:38:30 -0300,
Daniel Bareiro escribió:

> Now it would remain to find the cause of why I cannot call from a SIP
> extension to an analog telephone. Perhaps it is by something related
> to the contexts in the mentioned configuration files?

I forgot to copy the output that I obtain in the CLI when I call to a
SIP extension:

[May 25 19:22:57] NOTICE[4813]: chan_sip.c:14721 handle_request_invite:
Call from '201' to extension '1010' rejected because extension not
found.


Thanks for your reply.

Regards,
Daniel

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Re: [asterisk-users] howto store local exchange prefixes ?

[Jonathan Thurman]

On Mon, May 25, 2009 at 2:58 PM, John Novack
wrote:

>
>
> sean darcy wrote:
> > The local telco is now going 10 digit dialing even for local (free)
> > calls which used to be 7 digit. For a while no problem, everyone will
> > continue to dial 7 digits, and I'll add the area code. But pretty soon
> > everyone will become used to 10 digits.
> >
> >
> Lucky you.
> Other states require 11 digits for all calls, regardless, and yet others
> require 10 digit for local and 11 digit for toll, they way the NANP was
> SUPPOSED to evolve, until the inmates took over the asylum and each
> state ( in the US ) PUC sets the numbering plan and splits vs overlays.
>
> John Novack
>

> > There are about 40 3 digit local exchanges. I'd like to store the
> > exchanges in a database, and use the dialplan to check them. I can
> > figure that out.
> >
>

Very lucky, we have 700 prefixes to check that are 10 digits on one some our
trunks and 11 on others, and some that don't care either way!

Right now I have a script that parses the prefixes and creates the dial plan
in an #include file.  Since the prefixes don't change the frequently, it
seems to work.  Assume that everything is 11 digits, then using a dialing
macro, find an open trunk and strip the '1' if needed.  Now my users never
have to dial a 11, but it works if they do.

I would welcome some ideas for a more elegant solution!

So if cell phones never require 11 digits...

The company line about NANP and consistancy:
*"We don't care.**We don't have to.**We're the phone company."*-Jonathan
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Re: [asterisk-users] howto store local exchange prefixes ?

[John Novack]

sean darcy wrote:
> The local telco is now going 10 digit dialing even for local (free) 
> calls which used to be 7 digit. For a while no problem, everyone will 
> continue to dial 7 digits, and I'll add the area code. But pretty soon 
> everyone will become used to 10 digits.
>
>   
Lucky you.
Other states require 11 digits for all calls, regardless, and yet others 
require 10 digit for local and 11 digit for toll, they way the NANP was 
SUPPOSED to evolve, until the inmates took over the asylum and each 
state ( in the US ) PUC sets the numbering plan and splits vs overlays.

John Novack

> There are about 40 3 digit local exchanges. I'd like to store the 
> exchanges in a database, and use the dialplan to check them. I can 
> figure that out.
>
> I've looked at the Berkeley DB. That works pretty well, if the exchanges 
> are all stored. But it looks like the exchanges have to be entered 1 by 
> 1 from the CLI. And can only be reviewed, corrected, or deleted from the 
> CLI. I haven't found any simple frontend for the DB.
>
> I'd also consider sqlite3, but from the sqlite3 .conf.sample, it's only 
> for CDR. In any event, I couldn't find a simple frontend. I'd prefer not 
> to go into mysql etc for such a simple project.
>
> sean
>
>
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>   

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Re: [asterisk-users] FritzBox 7270

[Hans Witvliet]

On Mon, 2009-05-25 at 22:19 +0200, Philipp von Klitzing wrote:
> Hi!
> 
> > looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
> > one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the
> > hardware of Fritz is capable of it) 
> 
> "Divert"-ing is a misleading term in this case. As I said, use the new 
> firmware and register Asterisk to the Fritz!Box as "Internet telephone" 
> and the OP is ready to go.
> 
> http://www.ip-phone-forum.de/showpost.php?p=1298093&postcount=202
> http://www.ip-phone-forum.de/showthread.php?t=184818&;
> 

Well, to avoid further confusion, lets check with the O.P. ;-)

So Manoj, which case were you refering to:

case-1:
Incoming VOIP-calls are answered by Fritz, and then forwarded to your
own Asterisk.
(And for outgoing calls, Asterisk uses Fritz as an VOIP-gateway)

case-2
Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your own
Asterisk.
Incoming VOIP-calls are answered by your own Asterisk.


In case-1, Fritz has an ordinairy DSL-modem, and on the lan-side there
is an VOIP-pbx, and a FXS interface for a local phone.

In case-2 Fritz has on the PSTN/ISDN-line an ordinairy DSL-modem,
But also a FXO interface. While on the lan-side there is an VOIP-pbx,
and a FXS interface for a local phone.


AFAICS, case-1 is do-able, but you don't gain anything with it.
case-2 would give you two places for incoming calls (voip & PSTN)
But i wonder if the HW would allow that (no fxo)

If case-2 is feasable, i'll dash-off for an 7270


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Re: [asterisk-users] Asterisk, SQL Database Update

[Pascal Bruno]

If you have asterisk addons installed you can use the mysql  
applications to make queries. I find it to be very easy if you know  
how to do select and insert queries and understand the basic mechanism  
of the dialplan. Other than that, you may want to hire someone to do it.

Sent from my iPod

On May 25, 2009, at 5:03 PM, Edwin Quijada  
 wrote:

>
>
>>
>> Thanks for your helpful reply.
>>
>>
>>
>> I am not so good in coding.
>>
>>
>>
>> simply all i need is as follow
>>
>>
>>
>> When a call comes, goes into an IVR, and then depending on the  
>> entry option
>>
>> it will connect to a remote SQL Database, to check the pre-existed  
>> data,
>>
>> and in the end of the IVR the caller will enter an option that will  
>> need to be written in the SQL Database.
>>
>>
>>
>> Can you please give me a general scenrio how this will be achieved.
>>
>> and which files that i will need to modify.
>>
>
> I think that if you are not good coding you will have a few problems.
> Maybe, the best solution 4u is hire external to do that. It is simple
> but just in dialplan it is so difficult with AGI it is so easy but
> you dont want coding.
>
>
>
> *---*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-809-849-8087
> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera  
> de lo comun"
> *---*
>
> _
> Windows Live Hotmail now works up to 70% faster.
> http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster_112008
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Re: [asterisk-users] howto store local exchange prefixes ?

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

sean darcy wrote:

> I've looked at the Berkeley DB. That works pretty well, if the exchanges 
> are all stored. But it looks like the exchanges have to be entered 1 by 
> 1 from the CLI. And can only be reviewed, corrected, or deleted from the 
> CLI. I haven't found any simple frontend for the DB.

I do this be writing a dialplan which adds those entries.  The first
entry checks to see if the DB has been initialized and if so, skips to
the lookup.  Otherwise it loads each into the database before the
lookup.  It's very easy to write a quick script to generate the dialplan
code.

Barry
-BEGIN PGP SIGNATURE-
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[asterisk-users] Placing a MWI on-off call...

[eric weaver]

My grateful thanks to whoever can guide me in implementing this...

I have a need to place calls via Asterisk Manager Protocol to a legacy PBX
and twiddle its MWI lights.

By some means I get notified that an MWI light has to be changed.
Via AMP:
1.  Send an ORIGINATE command honking up an incoming port on the PBX via a
TA804.
  a.  I presume that the TA804 should be a SIP channel.  yes/no?
  b. The local extension I have made a pseudo extension that answers and
plays a silent file.  Sensible?

2. loop calling Play_DTMF and maybe silence between, to the SIP channel to
do the digits.
3. Need to sense fast-busy or stutter-dial-tone at this point.  Any
pointers?
4. Hangup said SIP channel.


Thanks in advance, etc.
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[asterisk-users] howto store local exchange prefixes ?

[sean darcy]

The local telco is now going 10 digit dialing even for local (free) 
calls which used to be 7 digit. For a while no problem, everyone will 
continue to dial 7 digits, and I'll add the area code. But pretty soon 
everyone will become used to 10 digits.

There are about 40 3 digit local exchanges. I'd like to store the 
exchanges in a database, and use the dialplan to check them. I can 
figure that out.

I've looked at the Berkeley DB. That works pretty well, if the exchanges 
are all stored. But it looks like the exchanges have to be entered 1 by 
1 from the CLI. And can only be reviewed, corrected, or deleted from the 
CLI. I haven't found any simple frontend for the DB.

I'd also consider sqlite3, but from the sqlite3 .conf.sample, it's only 
for CDR. In any event, I couldn't find a simple frontend. I'd prefer not 
to go into mysql etc for such a simple project.

sean


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Re: [asterisk-users] Asterisk, SQL Database Update

[Edwin Quijada]

>
> Thanks for your helpful reply.
>
>
>
> I am not so good in coding.
>
>
>
> simply all i need is as follow
>
>
>
> When a call comes, goes into an IVR, and then depending on the entry option
>
> it will connect to a remote SQL Database, to check the pre-existed data,
>
> and in the end of the IVR the caller will enter an option that will need to 
> be written in the SQL Database.
>
>
>
> Can you please give me a general scenrio how this will be achieved.
>
> and which files that i will need to modify.
>

I think that if you are not good coding you will have a few problems.
Maybe, the best solution 4u is hire external to do that. It is simple 
but just in dialplan it is so difficult with AGI it is so easy but
you dont want coding.
 
 
 
*---*
*-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] Asterisk, SQL Database Update

[Edwin Quijada]

Perl and AGI 
Piece of cake.!!!



*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087

* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun"
*---*







> From: torinti...@hotmail.com
> To: asterisk-users@lists.digium.com
> Date: Mon, 25 May 2009 22:28:54 +0300
> Subject: Re: [asterisk-users] Asterisk, SQL Database Update
>
>
>
>
>
>
>
>
> Thanks for your helpful reply.
>
>
>
> I am not so good in coding.
>
>
>
> simply all i need is as follow
>
>
>
> When a call comes, goes into an IVR, and then depending on the entry option
>
> it will connect to a remote SQL Database, to check the pre-existed data,
>
> and in the end of the IVR the caller will enter an option that will need to 
> be written in the SQL Database.
>
>
>
> Can you please give me a general scenrio how this will be achieved.
>
> and which files that i will need to modify.
>
>
>
> Thanks a lot.
>
>
>
>
>
>> Date: Sun, 24 May 2009 22:15:31 +0200
>> From: philipp.kemp...@amooma.de
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk, SQL Database Update
>>
>> Torintino T schrieb:
>>> Is there any method in Asterisk to enable the updating process
>>> into another SQL database through entering IVR options during the call.
>>
>> Depending on what you are trying to do there are various solutions:
>> Channel Event Logging (CEL) - http://www.asterisk.org/node/48358
>> AGI
>> System()
>> ODBC_*() functions
>>
>>
>> Philipp Kempgen
>> --
>> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
>> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
>> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
>> Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
>> --
>>
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>
> 
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> More than mail–Windows Live™ goes way beyond your inbox.
> More than messages
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Re: [asterisk-users] FritzBox 7270

[Philipp von Klitzing]

Hi!

> looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
> one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the
> hardware of Fritz is capable of it) 

"Divert"-ing is a misleading term in this case. As I said, use the new 
firmware and register Asterisk to the Fritz!Box as "Internet telephone" 
and the OP is ready to go.

http://www.ip-phone-forum.de/showpost.php?p=1298093&postcount=202
http://www.ip-phone-forum.de/showthread.php?t=184818&;

Philipp


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Re: [asterisk-users] Asterisk, SQL Database Update

[Steve Edwards]

Un-top-posting...

>> Torintino T schrieb:
>>
>>> Is there any method in Asterisk to enable the updating process into 
>>> another SQL database through entering IVR options during the call.

>> Date: Sun, 24 May 2009 22:15:31 +0200
>> From: philipp.kemp...@amooma.de
>>
>> Depending on what you are trying to do there are various solutions:
>> Channel Event Logging (CEL) - http://www.asterisk.org/node/48358
>> AGI
>> System()
>> ODBC_*() functions

On Mon, 25 May 2009, Torintino T wrote:

> Thanks for your helpful reply.
>
> I am not so good in coding.
>
> simply all i need is as follow
>
> When a call comes, goes into an IVR, and then depending on the entry 
> option it will connect to a remote SQL Database, to check the 
> pre-existed data, and in the end of the IVR the caller will enter an 
> option that will need to be written in the SQL Database.
>
> Can you please give me a general scenrio how this will be achieved. and 
> which files that i will need to modify.

1) Try the database functions in dialplan (extensions.[ael|conf).

2) If it starts to get ugly, code it up as an AGI in the language of your 
choice -- I prefer C.

3) Hire somebody to whack it out.

I think all database operations are ugly in dialplan, but I admit to most 
of my biases :)

Thanks in advance,

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

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Re: [asterisk-users] Asterisk, SQL Database Update

[Torintino T]

Thanks for your helpful reply.

 

I am not so good in coding.

 

simply all i need is as follow

 

When a call comes, goes into an IVR, and then depending on the entry option

it will connect to a remote SQL Database, to check the pre-existed data,

and in the end of the IVR the caller will enter an option that will need to be 
written in the SQL Database.

 

Can you please give me a general scenrio how this will be achieved.

and which files that i will need to modify.

 

Thanks a lot.

 


 
> Date: Sun, 24 May 2009 22:15:31 +0200
> From: philipp.kemp...@amooma.de
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk, SQL Database Update
> 
> Torintino T schrieb:
> > Is there any method in Asterisk to enable the updating process
> > into another SQL database through entering IVR options during the call.
> 
> Depending on what you are trying to do there are various solutions:
> Channel Event Logging (CEL) - http://www.asterisk.org/node/48358
> AGI
> System()
> ODBC_*() functions
> 
> 
> Philipp Kempgen
> -- 
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
> -- 
> 
> ___
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] FritzBox 7270

[Hans Witvliet]

On Mon, 2009-05-25 at 17:07 +0200, Ngo-Vi Hoai-Anh wrote:
> is installing asterisk directly  on FritzBox an option for you?  If yes 
> I 'v found an interesting link
> 
> http://www.ip-phone-forum.de/showthread.php?t=146132
> 
> 
> 
> Manoj Panicker - FOES schrieb:
> > Hi ,
> > Any idea as how to divert the Incoming PSTN calls on the FritzBox 
> > to one of the Numbers in the Asterisk domian? and vice versa.
> >  
> > I want ot use the FritzBox as the bridge between the PSTN and Astrisk
> >  

looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
one would need chan_dahdi.so or chan_misdn.so/chan_capi.so
(If the hardware of Fritz is capable of it)

hw

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Re: [asterisk-users] How to stop a background music

[Steve Edwards]

Un-top-posting...

> On Fri, May 22, 2009 at 7:37 PM, Steve Edwards 
> wrote:
>
>> On Fri, 22 May 2009, Noel R. Morais wrote:
>>
>>> But I need a way to actively stop it. Without waiting for user hit a 
>>> DTMF or the background timeout.
>>
>> What event would trigger your desire to stop the background()?

On Mon, 25 May 2009, Noel R. Morais wrote:

> I'm planning to play a background music, make some background process 
> and after that I will play another music or "transfer" the call to 
> another end point.
>
> I'm gonna see how difficult is to write a function like "StopBackground" 
> to do that. Any hints?

I think you are "barking up the wrong tree."

The purpose of background() is to play a file until a key is pressed or 
until the file is finished playing.

You will not execute the next step in your dialplan until the file is 
finished* so how will you create a background process?

What you described sounds more like an AGI, "music on hold," "parking," or 
dumping the caller into a conference and then transferring to the other 
end point.

Based on what you have described, I would write an AGI that played the 
file while waiting for whatever your triggering event is, set a channel 
variable (the name of the end point?) and returned to the dialplan -- but 
I tend to see AGIs as the solution to most non-trivial dial plan problems 
:)

*or the exten is entered or a timeout.

Thanks in advance,

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

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[asterisk-users] DNS issues again

[Joseph L. Casale]

I have a caching name server setup on one of our units but after a prolonged net
outage the internal phones stopped working as well. In searching the bug tracker
I see the bug is still not fixed even though it was thought to be (using 
1.6.0.8).

Some suggestions where to set srvlookup=yes but I fail to see how that would
help internal extensions? There is a pstn line and tdm card in this server and
the dial plan has provisions for this line to be used in and out.

Is it just the lookup for the remote peers that causes the internal peers to
fail as well? Would commenting out the remote peers in sip.conf fix this?

Thanks!
jlc

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Re: [asterisk-users] New tutorial: storing audio recordings per day

[Atis Lezdins]

On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri  wrote:
> Hi everyone,
> after doing the same thing multiple times and struggling to remember how it
> was done, I have prepared a small tutorial that explains how to save
> monitored files in different folders per day. This is quite useful
> becausethe resultingfile system is way more manageable than having maybe
> 100,000 files all saved in the same folder.
> You can find the tutorial here:
>
> http://astrecipes.net/index.php?n=387
>
> As always, comments and suggestions are welcome.
> l.
> PS. I am also working on some scripts to "normalize" existing recordings
> all-in-one-directory... if anybody is interested, please contact me.

Actually You don't have to create folders in advance, as Asterisk will
automatically create them when needed. Just make sure that Asterisk
process is owner of parent directory.


Set(__call_day=${STRFTIME(|${TIMEZONE}|%Y/%m/%d)});
Set(MONITOR_FILENAME=${MONITOR_DIR}/${call_day}/call-${UNIQUEID});
Monitor(ulaw,${MONITOR_FILENAME},b);

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] New tutorial: storing audio recordings per day

[Lenz Emilitri]

Hi everyone,
after doing the same thing multiple times and struggling to remember how it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
100,000 files all saved in the same folder.
You can find the tutorial here:

http://astrecipes.net/index.php?n=387

As always, comments and suggestions are welcome.
l.
PS. I am also working on some scripts to "normalize" existing recordings
all-in-one-directory... if anybody is interested, please contact me.

-- 
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] How to stop a background music

[Noel R. Morais]

I'm planning to play a background music, make some background process and
after that I will play another music or "transfer" the call to another end
point.

I'm gonna see how difficult is to write a function like "StopBackground" to
do that. Any hints?

Thanks in advance,

Noel

On Fri, May 22, 2009 at 7:37 PM, Steve Edwards wrote:

> On Fri, 22 May 2009, Noel R. Morais wrote:
>
> > But I need a way to actively stop it. Without waiting for user hit a DTMF
> or
> > the background timeout.
>
> What event would trigger your desire to stop the background()?
>
> Thanks in advance,
> 
> Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] FritzBox 7270

[Ngo-Vi Hoai-Anh]

is installing asterisk directly  on FritzBox an option for you?  If yes 
I 'v found an interesting link

http://www.ip-phone-forum.de/showthread.php?t=146132



Manoj Panicker - FOES schrieb:
> Hi ,
> Any idea as how to divert the Incoming PSTN calls on the FritzBox 
> to one of the Numbers in the Asterisk domian? and vice versa.
>  
> I want ot use the FritzBox as the bridge between the PSTN and Astrisk
>  
> Thanks
> Manoj
>
> 
> *From:* Manoj Panicker - FOES
> *Sent:* 24 May 2009 12:39
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* RE: [asterisk-users] FritzBox 7270
>
> Kare,
> Thanks much appreciated. It connected as soon as I created a SIP 
> account. However I must try and figure out as how to get this box use 
> IAX2.
> The vendors are not very helpful.
>  
> Thanks for your help.
>  
> Regards
> Manoj
>
> 
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *IT-Connect
> *Sent:* 20 May 2009 16:59
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] FritzBox 7270
>
> I only tried to connect my 7270 Fritz Box over a sip account on asterisk!
>
> There are some points, you have to note:
> - you have to select "Using internet number"
> - in the area select "other Provider"
> - in field "internet number" your asterisk number
> - in field "user name" your number too
> - your password
> - field "registrar" a name of your choice
> - other fields are blank
>
> I hope, these are the same options on you Fritz Box, because my gui is 
> in German!
>
> regards, Kare
>
> Manoj Panicker - FOES schrieb:
>>
>> Dear Users
>> Good day, need a help on connecting the FritzBox with my 
>> Asterisk Server. Both are in LAN and from the Asterisk Server I can 
>> ping the FritzBox. However the Username I gave in the box is somehow 
>> is not geeting registered in the Asterisk application. The usetname I 
>> configured in the box is of IAX2 type, is that the reason?
>>
>> Any information on how to connect the FritzBoz 7270 with Asterisk 
>> will be appreciated. I did not seem to get much help from the net. 
>> Can somebody help?
>>
>> Thanks
>> Manoj
>>
>> 
>>
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>
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>   Tel.+49 941 705 49 01
>   Fax   +49 941 378 70 39
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>
> Diese Email und etwaige Anhaenge sind nur fuer den angegebenen Empfaenger 
> bestimmt. Sie kann privilegierte und vertrauliche Informationen enthalten. 
> Wenn der Leser dieser Email nicht der bestimmungsgemaesse Empfaenger oder ein 
> authorisierter Vertreter ist, werden Sie hiermit darauf aufmerksam gemacht, 
> dass jede Weitergabe dieser Email streng verboten ist. Wenn Sie diese Email 
> faelschlicherweise erhalten haben, unterrichten Sie uns bitte sofort und 
> loeschen Sie die Email und etwaige Anhaenge von Ihrem Rechner.
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Re: [asterisk-users] Problem running Dahdi

[Mike]

I did run make install, probably 3-4 times before I ended up asking that
question in the mailing list.

Here is the required output: to the first one, "could not find module
dahdi".

To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi

As for the other questions:

What do you do? 
I simply try starting /etc/init.d/dahdi restart

What would you expect to happen?
No Red warnings, for one.  I have another system that I configured awhile
ago, and that starts fine.  I understand I have no hardware loaded, but all
modules load with a green OK.

What actually happens?

FATAL: Module Dahdi not found

[snip] all modules listed as not found [/snip]

Error: missing /dev/dahdi!

What system is this on?
 - What versions of dahdi-linux and dahdi-tools?
Latest as found on asterisk.org, that would be 
DAHDI Linux 2.1.0.4
DAHDI Tools 2.1.0.2

 - What distribution? What version?
CentOS, 5.3.  I tried updating all packages before trying again, same
result.

 - What kernel version?
2.6.18-128.1.10.el5


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Monday, May 25, 2009 9:53
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problem running Dahdi
> 
> On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote:
> > Sorry, it seems to have disappeared from my original email!
> >
> > FATAL: Module Dahdi not found
> >
> > [snip] all modules listed as not found [/snip]
> >
> > Error: missing /dev/dahdi!
> 
> Your description makes me suspect you have not run 'make install' in
> dahdi-linux.
> 
> What is the output of:
> 
>   modinfo dahdi
>   find /lib/modules -name dahdi
> 
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] FritzBox 7270

[Philipp von Klitzing]

Hi!

> You mean, you want to use the 7270 as an "isdn-ata" ? perhaps i'm
> wrong, but afaics the pbx-part in any DSL-modem works only on the
> ip-stream (wan/lan). 

Both 7270 and 7170 now have a new firmware that comes with a SIP proxy. 
It is not fully featured yet (no transfer, NAT/Dyndns issues). Apart from 
this you can very well use it as a SIP client for both WAN or LAN SIP 
registrars already for looong time.

BTW: The 7170 is close to end-of-maintenance, so I am not sure if the new 
SIP proxy will receive much attention to fix the bigger issues.

And there's also the D-Link HorstBox PRO, but its ISDN implementation 
isn't the best.

Philipp von Klitzing


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Re: [asterisk-users] Problem running Dahdi

[Tzafrir Cohen]

On Mon, May 25, 2009 at 08:14:44AM -0400, Mike wrote:
> Hi,
> 
>  
> 
> I've been building a new Asterisk server to replace my previous one, all
> using the latest 1.4.x downloads from asterisk.org.
> 
> I can't even get to the point where dahdi works. I have libpri, dahdi and
> dahdi-tools compiled and installed.  I have no hardware on that new server,
> but I am installing Dahdi for timing source for meetme.
> 
>  
> 
> Everything compiles seemingly well, but this is what I get when I try to
> start dahdi from the startup scripts (/etc/init.d/dahdi restart).

What do you do?

What would you expect to happen?

What actually happens?

What system is this on?

 - What versions of dahdi-linux and dahdi-tools?
 - What distribution? What version?
 - What kernel version?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Problem running Dahdi

[Tzafrir Cohen]

On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote:
> Sorry, it seems to have disappeared from my original email!
> 
> FATAL: Module Dahdi not found
> 
> [snip] all modules listed as not found [/snip]
> 
> Error: missing /dev/dahdi!

Your description makes me suspect you have not run 'make install' in
dahdi-linux.

What is the output of:

  modinfo dahdi
  find /lib/modules -name dahdi

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Problem running Dahdi

[Mike]

Sorry, it seems to have disappeared from my original email!

FATAL: Module Dahdi not found

[snip] all modules listed as not found [/snip]

Error: missing /dev/dahdi!


and /dev/dahdi is indeed absent.  How do I make sure it's created as part of
the install process?

Mike


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Steve Howes
> Sent: Monday, May 25, 2009 9:07
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Problem running Dahdi
> 
> 
> On 25 May 2009, at 13:14, Mike wrote:
> > Everything compiles seemingly well, but this is what I get when I
> > try to start dahdi from the startup scripts (/etc/init.d/dahdi
> > restart).
> 
> Define 'this'
> 
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Re: [asterisk-users] Problem running Dahdi

[Steve Howes]

On 25 May 2009, at 13:14, Mike wrote:
> Everything compiles seemingly well, but this is what I get when I  
> try to start dahdi from the startup scripts (/etc/init.d/dahdi  
> restart).

Define 'this'

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[asterisk-users] SIP Trunk groups

[Mariano Lecuona]

He all,

I have 2 GSM to Voip gateways and  probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as
truckgroup under dahdi but for sip trunks.

Is that possible?, have you ever done this before?

My Idea is:

sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3

gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3


[user]

exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${exten:1},30)
exten = _0.,n,Hangup

Thanks,

-- 
--
Mariano Lecuona
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Re: [asterisk-users] Problem running Dahdi

[Mike]

Hi,

 

I've been building a new Asterisk server to replace my previous one, all
using the latest 1.4.x downloads from asterisk.org.

I can't even get to the point where dahdi works. I have libpri, dahdi and
dahdi-tools compiled and installed.  I have no hardware on that new server,
but I am installing Dahdi for timing source for meetme.

 

Everything compiles seemingly well, but this is what I get when I try to
start dahdi from the startup scripts (/etc/init.d/dahdi restart).

 

Mike

 

Sorry for replying to my own post.  This is using CentOS 5.3 and it happened
on two servers (one VMWare, the other an HP box), which I failed to specify.

 

Mike

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[asterisk-users] Problem running Dahdi

[Mike]

Hi,

 

I've been building a new Asterisk server to replace my previous one, all
using the latest 1.4.x downloads from asterisk.org.

I can't even get to the point where dahdi works. I have libpri, dahdi and
dahdi-tools compiled and installed.  I have no hardware on that new server,
but I am installing Dahdi for timing source for meetme.

 

Everything compiles seemingly well, but this is what I get when I try to
start dahdi from the startup scripts (/etc/init.d/dahdi restart).

 

Mike

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[asterisk-users] Connected Number on incoming calls with mISDN

[Andreas Ruf]

Hello list,

we have a problem with ISDN-Phones (not cell, not analog) calling into
our Asterisk-Server. Assume we have the number (Germany) +49 123 4567 89
to be called from one ISDN-Phone. I will term (0)123 local dialling
code, 4567 base number of our ISDN-line and 89 the extension (of our
Asterisk-Server) to be actually called.

If a call is initiated from an ISDN-Phone to 0123 4567 89 and it gets
connected by asterisk, the display of the ISDN-Phone shows the called
Number like "0123 4567 4567 89".
The local dialling code 0123 and the first appearance of the base number
4567 are O.K.. The second displaying of the base number 4567 is not
O.K.. There should be only the extension 89 shown.
The Log attached below shows, that this comes from an empty/not set
"cad" and so chan_misdn sets variable "cad" to the value of variable
"dad" (which is 4567 89, base number + extension).

So our "dad" is always set to base number + extension (4567 89). And
variable "cad" being empty, is set to variable "dad". What possibilities
are there to get correct handling of this "misbehaviour"?
- Can we configure something to get "dad" set to extension only?
- Can we configure something to get "cad" set to extension only? Not
evaluating the value of "dad"?
- For chan_capi there is a variable termed "CONNECTEDNUMBER" in
Asterisk. With this variable the value of the "connected number" can be
explicitly set. Is there a similar variable in Asterisk for chan_misdn?
- Something else?

Many thanks in advance
Sincerely
Andreas Ruf



Our setup and Logs:

ISDN (Germany): BRI Point-to-Point (Anlagenanschluss)
ISDN-Card: BN4S0 ISDN card
Asterisk: 1.4.24.1 (+ integrated "chan_misdn")
mISDN: 1.1.9.1 (from www.misdn.org)
(FreePBX: 2.5.1.2)
(Debian/Lenny: 2.6.26)


misdn.conf:

[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
bridging=no
l1watcher_timeout=60
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=from-pstn
language=de
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=5
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
overlapdial=yes
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
immediate=no
presentation=-1
screen=-1
echotraining=no
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
max_incoming=-1
max_outgoing=-1

[isdn_provider]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
ports=1,2,3,4
; context where to go to when incoming Call on one of the above ports
context=from-pstn
msns=*
max_incoming=4
max_outgoing=4


misdn.log:

Wed May 20 12:31:33 2009: P[ 2]  I IND :NEW_CHANNEL oad:611xxx
dad:4567 pid:13 state:none
Wed May 20 12:31:33 2009: P[ 2]   --> channel:1 mode:TE cause:16
ocause:16 rad: cad:
Wed May 20 12:31:33 2009: P[ 2]   --> info_dad: onumplan:2 dnumplan:4
rnumplan:  cpnnumplan:0
Wed May 20 12:31:33 2009: P[ 2]   --> caps:Speech pi:0 keypad:
sending_complete:0
Wed May 20 12:31:33 2009: P[ 2]   --> screen:0 --> pres:0
Wed May 20 12:31:33 2009: P[ 2]   --> addr:0 l3id:40007 b_stid:0
layer_id:50010280
Wed May 20 12:31:33 2009: P[ 2]   --> facility:Fac_None
out_facility:Fac_None
Wed May 20 12:31:33 2009: P[ 2]   --> urate:0 rate:16 mode:0 user1:0
Wed May 20 12:31:33 2009: P[ 2]   --> bc:0x8acdf9c h:0 sh:0
Wed May 20 12:31:33 2009: P[ 2]   --> bc_state:BCHAN_CLEANED
Wed May 20 12:31:33 2009: P[ 2]  Chan not existing at the moment
bc->l3id:40007 bc:0x8acdf9c event:NEW_CHANNEL
 port:2 channel:1
Wed May 20 12:31:33 2009: P[ 2]  NO USERUESRINFO
Wed May 20 12:31:33 2009: P[ 2]  find_free_chan: req_chan:1
Wed May 20 12:31:33 2009: P[ 2]   --> found chan (preselected): 1
Wed May 20 12:31:33 2009: P[ 2]  set_chan_in_stack: 1
Wed May 20 12:31:33 2009: P[ 2]  $$$ Setting up bc with stid :10010200
Wed May 20 12:31:33 2009: P[ 2]  setup_bc: with dsp
Wed May 20 12:31:33 2009: P[ 2]   --> Channel is 1
Wed May 20 12:31:33 2009: P[ 2]   --> TRANSPARENT Mode
Wed May 20 12:31:33 2009: P[ 2]  $$$ Bchan Activated addr 50010202
Wed May 20 12:31:33 2009: P[ 2]  BC_STATE_CHANGE: l3id:40007
from:BCHAN_CLEANED to:BCHAN_ACTIVATED
Wed May 20 12:31:33 2009: P[ 2]  lib Got Prim: Addr 42000203 prim 30582
dinfo 40007
Wed May 20 12:31:33 2009: P[ 2]  I IND :SETUP oad:611xxx dad:4567
pid:13 state:none
Wed May 20 12:31:33 2009: P[ 2]   --> channel:1 mode:TE cause:16
ocause:16 rad: cad:
Wed May 20 12:31:33 2009: P[ 2]   --> info_dad: onumplan:2 dnumplan:4
rnumplan:  cpnnumplan:0
Wed May 20 12:31:33 2009: P[ 2]   --> caps:Speech pi:0 keypad:
sending_complete:0
Wed May 20 12:31:33 2009: P[ 2]   --> screen:0 --> pres:0
Wed May 20 12:31:33 2009: P[ 2]   --> addr:50010202 l3id:40007
b_stid:10010200 layer_id:50010280
Wed May 20 12:31:33 2009: P[ 2]   --> facility:Fac_None
out_facility:Fac_None
Wed May 20 12:31:33 200

Re: [asterisk-users] FritzBox 7270

[Hans Witvliet]

On Sun, 2009-05-24 at 16:21 +0400, Manoj Panicker - FOES wrote:
> Hi ,
> Any idea as how to divert the Incoming PSTN calls on the FritzBox
> to one of the Numbers in the Asterisk domian? and vice versa.
>  
> I want ot use the FritzBox as the bridge between the PSTN and Astrisk
>  
> Thanks
> Manoj
> 
> 
> __

You mean, you want to use the 7270 as an "isdn-ata" ?
perhaps i'm wrong, but afaics the pbx-part in any DSL-modem works only
on the ip-stream (wan/lan).

Would be nice, but as far as i know, such a thing, an isdn-ata, does not
exist in any appliance. (Perhaps a business opportunaty for someone)


hw

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