[asterisk-users] help please (D 300 JCT)

[Tahar .H]

i need to know how can i configure a D 300JCT with asterisk, i want to
connect two PBX where each one have this card on it,i really need your help
as soon as
possible.

i already done some file configuration system.conf and in chan_dahdi.conf
and i have installed the DAHDI and the LibPri modules.

-- 
*
HARAZ Tahar

*
*Engineering Student at the National Institute for Posts and
Telecommunications (INPT)
*
*
Phone: +212 6 78030050
E-mail: harazta...@gmail.com 
*
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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Philipp Kempgen]

Steve Totaro schrieb:

> I would not run MySQL on the local box.  I would simple use Asterisk's
> csv CDRs and then use some script to import the CSVs into a database
> residing on another server using some sort of script.  Depending on
> your needs, you could probably run that during low call volume.  I
> also think that you adapt the free queue_log to database script by
> Queuemetrics to do what you want on the fly.

We're using a custom script for the queue_log -> db import
in Gemeinschaft as well. But I'm not really happy with that.
You need to run such a script at least once a minute to get
real-time statistics for the GUI etc. Everything could be
much nicer if Asterisk wrote the queue log into the database
directly.
As an alternative solution you can use a named pipe. But
Asterisk is not prepared to handle the broken pipe error
which occurs if your script should ever fail to read from
the pipe.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Philipp Kempgen]

Al Baker schrieb:
> Are you saying the * server does NOT TRY to re-establish the BD connection ?

The MySQL Realtime driver _does_ reconnect.
(Search for mysql_reconnect() in res_config_mysql.c)

> If  NOT, what happens to you CDR records ?

Same thing with cdr_addon_mysql.c - it tries to reconnect.
When there is no connection it writes the CDRs to a file and as
soon as it successfully reconnects stores them in the database.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Steve Totaro]

On Tue, May 6, 2008 at 11:42 AM, Anthony Francis <[EMAIL PROTECTED]> wrote:
>
>
>
>  Tilghman Lesher wrote:
>  > On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>  >
>  >> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>  >>
>  >>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>  >>>
>   5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
>  
>  > On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>  >
>  >> Another issue that we need to fix with the MYSQL driver is that
>  >> we're
>  >> lacking a connection pool. Everything seems to be handled over one
>  >> connection to Mysql, which causes issues.
>  >>
>  > That's not true.  The MYSQL app generally uses multiple connections,
>  > one
>  > for each channel.  The only way one might use only a single
>  > connection is
>  > by using a global variable to store a single connection id, but that
>  > method
>  > is not documented anywhere, AFAIK.
>  >
>   You talk about the Mysql APP, but is this the case with the Realtime
>   driver as well?
>  
>  >>> No, the native Realtime driver uses a single connection.  The ODBC
>  >>> Realtime
>  >>> driver generally uses a single connection but can be configured to
>  >>> use a
>  >>> separate connection for each query.
>  >>>
>  >> So, we're back to where we started. A developer that can help us with
>  >> a connection
>  >> pool or a separate connection for each query would be a Nice Thing (TM).
>  >>
>  >
>  > What issues are you specifically seeing that merit using multiple
>  > connections?
>  >
>  >
>  I can specify an issue that would merit multiple connections, if the
>  link to your db goes away Asterisk likes to freeze writing CDRs.
>  I have a few remote * servers that this happens to. My solution so far
>  has been to record CDR's to a local DB and then have a
>  perl script that attempts to move them over to my transaction DB. I
>  would suggest this solution to anyone who depends on their CDR records.
>
>  --
>  Thank you and have any kind of day you want,
>
>  Anthony Francis
>  Rockynet VOIP
>

I would not run MySQL on the local box.  I would simple use Asterisk's
csv CDRs and then use some script to import the CSVs into a database
residing on another server using some sort of script.  Depending on
your needs, you could probably run that during low call volume.  I
also think that you adapt the free queue_log to database script by
Queuemetrics to do what you want on the fly.

Thanks,
Steve Totaro

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Al Baker]

Are you saying the * server does NOT TRY to re-establish the BD connection ?

Does your whole * SERVER freeze ?

If  NOT, what happens to you CDR records ?

Anthony Francis wrote:
> Tilghman Lesher wrote:
>   
>> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>>   
>> 
>>> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>>> 
>>>   
 On Monday 05 May 2008 11:24, Johansson Olle E wrote:
   
 
> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> 
>   
>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>>   
>> 
>>> Another issue that we need to fix with the MYSQL driver is that
>>> we're
>>> lacking a connection pool. Everything seems to be handled over one
>>> connection to Mysql, which causes issues.
>>> 
>>>   
>> That's not true.  The MYSQL app generally uses multiple connections,
>> one
>> for each channel.  The only way one might use only a single
>> connection is
>> by using a global variable to store a single connection id, but that
>> method
>> is not documented anywhere, AFAIK.
>>   
>> 
> You talk about the Mysql APP, but is this the case with the Realtime
> driver as well?
> 
>   
 No, the native Realtime driver uses a single connection.  The ODBC
 Realtime
 driver generally uses a single connection but can be configured to
 use a
 separate connection for each query.
   
 
>>> So, we're back to where we started. A developer that can help us with
>>> a connection
>>> pool or a separate connection for each query would be a Nice Thing (TM).
>>> 
>>>   
>> What issues are you specifically seeing that merit using multiple
>> connections?
>>
>>   
>> 
> I can specify an issue that would merit multiple connections, if the 
> link to your db goes away Asterisk likes to freeze writing CDRs.
> I have a few remote * servers that this happens to. My solution so far 
> has been to record CDR's to a local DB and then have a
> perl script that attempts to move them over to my transaction DB. I 
> would suggest this solution to anyone who depends on their CDR records.
>
>   

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Anthony Francis]

Tilghman Lesher wrote:
> On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
>   
>> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
>> 
>>> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>>>   
 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
 
> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>   
>> Another issue that we need to fix with the MYSQL driver is that
>> we're
>> lacking a connection pool. Everything seems to be handled over one
>> connection to Mysql, which causes issues.
>> 
> That's not true.  The MYSQL app generally uses multiple connections,
> one
> for each channel.  The only way one might use only a single
> connection is
> by using a global variable to store a single connection id, but that
> method
> is not documented anywhere, AFAIK.
>   
 You talk about the Mysql APP, but is this the case with the Realtime
 driver as well?
 
>>> No, the native Realtime driver uses a single connection.  The ODBC
>>> Realtime
>>> driver generally uses a single connection but can be configured to
>>> use a
>>> separate connection for each query.
>>>   
>> So, we're back to where we started. A developer that can help us with
>> a connection
>> pool or a separate connection for each query would be a Nice Thing (TM).
>> 
>
> What issues are you specifically seeing that merit using multiple
> connections?
>
>   
I can specify an issue that would merit multiple connections, if the 
link to your db goes away Asterisk likes to freeze writing CDRs.
I have a few remote * servers that this happens to. My solution so far 
has been to record CDR's to a local DB and then have a
perl script that attempts to move them over to my transaction DB. I 
would suggest this solution to anyone who depends on their CDR records.

-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP


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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Tilghman Lesher]

On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote:
> 5 maj 2008 kl. 19.58 skrev Tilghman Lesher:
> > On Monday 05 May 2008 11:24, Johansson Olle E wrote:
> >> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> >>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>  Another issue that we need to fix with the MYSQL driver is that
>  we're
>  lacking a connection pool. Everything seems to be handled over one
>  connection to Mysql, which causes issues.
> >>>
> >>> That's not true.  The MYSQL app generally uses multiple connections,
> >>> one
> >>> for each channel.  The only way one might use only a single
> >>> connection is
> >>> by using a global variable to store a single connection id, but that
> >>> method
> >>> is not documented anywhere, AFAIK.
> >>
> >> You talk about the Mysql APP, but is this the case with the Realtime
> >> driver as well?
> >
> > No, the native Realtime driver uses a single connection.  The ODBC
> > Realtime
> > driver generally uses a single connection but can be configured to
> > use a
> > separate connection for each query.
>
> So, we're back to where we started. A developer that can help us with
> a connection
> pool or a separate connection for each query would be a Nice Thing (TM).

What issues are you specifically seeing that merit using multiple
connections?

-- 
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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Johansson Olle E]

5 maj 2008 kl. 19.58 skrev Tilghman Lesher:

> On Monday 05 May 2008 11:24, Johansson Olle E wrote:
>> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
>>> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
 Another issue that we need to fix with the MYSQL driver is that  
 we're
 lacking a connection pool. Everything seems to be handled over one
 connection to Mysql, which causes issues.
>>>
>>> That's not true.  The MYSQL app generally uses multiple connections,
>>> one
>>> for each channel.  The only way one might use only a single
>>> connection is
>>> by using a global variable to store a single connection id, but that
>>> method
>>> is not documented anywhere, AFAIK.
>>
>> You talk about the Mysql APP, but is this the case with the Realtime
>> driver as well?
>
> No, the native Realtime driver uses a single connection.  The ODBC  
> Realtime
> driver generally uses a single connection but can be configured to  
> use a
> separate connection for each query.

So, we're back to where we started. A developer that can help us with  
a connection
pool or a separate connection for each query would be a Nice Thing (TM).

/O

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Andrew Kohlsmith (lists)]

On May 5, 2008 01:58:42 pm Tilghman Lesher wrote:
> > Hmm. Haven't found any Digium Stockholm office to discuss with ;-)
> That hasn't stopped any of the Canadian employees.  :-)

That's because nothing stops Canadians, short of Hockey Night in Canada  :-)

-A.

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Tilghman Lesher]

On Monday 05 May 2008 11:24, Johansson Olle E wrote:
> 5 maj 2008 kl. 17.51 skrev Tilghman Lesher:
> > On Monday 05 May 2008 09:45, Johansson Olle E wrote:
> >> Another issue that we need to fix with the MYSQL driver is that we're
> >> lacking a connection pool. Everything seems to be handled over one
> >> connection to Mysql, which causes issues.
> >
> > That's not true.  The MYSQL app generally uses multiple connections,
> > one
> > for each channel.  The only way one might use only a single
> > connection is
> > by using a global variable to store a single connection id, but that
> > method
> > is not documented anywhere, AFAIK.
>
> You talk about the Mysql APP, but is this the case with the Realtime
> driver as well?

No, the native Realtime driver uses a single connection.  The ODBC Realtime
driver generally uses a single connection but can be configured to use a
separate connection for each query.

> >> Any MySQL developers out there that can help us fix this? We need
> >> someone that has been developing towards the Mysql api.
> >>
> >> And please do not always refer to "Digium developers" when you have
> >> problems in Asterisk. There are developers that are not employees of
> >> Digium...
> >
> > There used to be more outside developers, but Digium is a great
> > place to work... ;-)
>
> Hmm. Haven't found any Digium Stockholm office to discuss with ;-)

That hasn't stopped any of the Canadian employees.  :-)

-- 
Tilghman

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Johansson Olle E]

5 maj 2008 kl. 17.51 skrev Tilghman Lesher:

> On Monday 05 May 2008 09:45, Johansson Olle E wrote:
>> Another issue that we need to fix with the MYSQL driver is that we're
>> lacking a connection pool. Everything seems to be handled over one
>> connection to Mysql, which causes issues.
>
> That's not true.  The MYSQL app generally uses multiple connections,  
> one
> for each channel.  The only way one might use only a single  
> connection is
> by using a global variable to store a single connection id, but that  
> method
> is not documented anywhere, AFAIK.
You talk about the Mysql APP, but is this the case with the Realtime  
driver as well?

>
>
>> Any MySQL developers out there that can help us fix this? We need
>> someone that has been developing towards the Mysql api.
>>
>> And please do not always refer to "Digium developers" when you have
>> problems in Asterisk. There are developers that are not employees of
>> Digium...
>
> There used to be more outside developers, but Digium is a great  
> place to
> work... ;-)

Hmm. Haven't found any Digium Stockholm office to discuss with ;-)

/O

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Tilghman Lesher]

On Monday 05 May 2008 09:45, Johansson Olle E wrote:
> Another issue that we need to fix with the MYSQL driver is that we're
> lacking a connection pool. Everything seems to be handled over one
> connection to Mysql, which causes issues.

That's not true.  The MYSQL app generally uses multiple connections, one
for each channel.  The only way one might use only a single connection is
by using a global variable to store a single connection id, but that method
is not documented anywhere, AFAIK.

> Any MySQL developers out there that can help us fix this? We need
> someone that has been developing towards the Mysql api.
>
> And please do not always refer to "Digium developers" when you have
> problems in Asterisk. There are developers that are not employees of
> Digium...

There used to be more outside developers, but Digium is a great place to
work... ;-)

-- 
Tilghman

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Tilghman Lesher]

On Monday 05 May 2008 10:16, Al Baker wrote:
> I must be overlooking it, I pulled up the electronic version and
> searched for and read every instance where ODBC was mentioned and I
> could not find a single place where it said
> ODBC was to be the only or even the best method.  If so I would never
> ever have gone down this road :(

While there are some things that you can do with the MYSQL app that you
can't do with func_odbc, most of what people want to do (single row queries,
updates, and inserts) can be done with func_odbc.  Additionally, it's well
supported, the author (me) is still around, and we're still considering
enhancements.  The MYSQL app is somewhat less certain.  I don't know if
the author is still around, it is largely in maintenance mode, and, due to a
matter of licensing, the app is not in the Asterisk core distribution.

ODBC is not going to be the only supported method of accessing a database;
some native drivers exist, and they will continue to exist.  The reason why
ODBC is so important to us is that it allows us to interface with many varied
databases out there for which the native API is not immediately available.
Additionally, it significantly lightens our maintenance workload not to have
to maintain drivers for some databases which our users may never use.  And
most of our database integration has been for the specific purpose of
realtime.  Our func_odbc module is about the most generic of database
interfaces, and its usage is completely user-defined, unlike realtime or the
other database interface (voicemail storage).

-- 
Tilghman

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Al Baker]

I must be overlooking it, I pulled up the electronic version and 
searched for and read every instance where ODBC was mentioned and I 
could not find a single place where it said
ODBC was to be the only or even the best method.  If so I would never 
ever have gone down this road :(

Quote "And according to the O'Reilly book ODBC is the way to go."

Roderick A. Anderson wrote:
> Steve Totaro wrote:
>   
>> A quote from Tilghman Lesher from a previous post.
>>
>> "That's fine, but I have had the most horrid results using any distribution-
>> supplied ODBC drivers.  The best results are obtained by source-compiling
>> the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
>> PsqlODBC.  UnixODBC is fairly safe to use from distribution channels, 
>> however."
>> 
>
> And according to the O'Reilly book ODBC is the way to go.
>
> Though they use PostgreSQL for their examples and Asterisk is installed 
> on a CentOS system the instructions are really good.  Getting it to work 
> with MySQL should be pretty simple and I'm sure on-line resources for 
> doing this are be out there.
>
> 
> Personally I never use MySQL except in cases where I am under extreme 
> duress.  Therefore I tried and tossed trixbox, AsteriskNOW, and 
> freeePBX.  Yes I know I can get around the database engine issue but 
> that is what a distribution should be for: no hacking (or at least 
> not-too-much) required.
>
> It is now CentOS 5, Asterisk from source, PostgreSQL (on another system) 
> and hand edited (for now anyway) *.conf files.  Maybe AsteriskGUI later.
> 
>
>
> Rod
>   

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Roderick A. Anderson]

Steve Totaro wrote:
> A quote from Tilghman Lesher from a previous post.
> 
> "That's fine, but I have had the most horrid results using any distribution-
> supplied ODBC drivers.  The best results are obtained by source-compiling
> the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
> PsqlODBC.  UnixODBC is fairly safe to use from distribution channels, 
> however."

And according to the O'Reilly book ODBC is the way to go.

Though they use PostgreSQL for their examples and Asterisk is installed 
on a CentOS system the instructions are really good.  Getting it to work 
with MySQL should be pretty simple and I'm sure on-line resources for 
doing this are be out there.


Personally I never use MySQL except in cases where I am under extreme 
duress.  Therefore I tried and tossed trixbox, AsteriskNOW, and 
freeePBX.  Yes I know I can get around the database engine issue but 
that is what a distribution should be for: no hacking (or at least 
not-too-much) required.

It is now CentOS 5, Asterisk from source, PostgreSQL (on another system) 
and hand edited (for now anyway) *.conf files.  Maybe AsteriskGUI later.



Rod
-- 
> 
> Thanks,
> Steve Totaro
> 
> On Mon, May 5, 2008 at 10:06 AM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
>> User intervention is required Al.  You need to open a bug report on
>>  mantis and anyone can edit the wiki (this is the nature of a wiki).
>>  http://bugs.digium.com/main_page.php
>>
>>  Anyways, I am fairly certain that the UnixODBC connector is the
>>  "preferred" way of connecting to MySQL.
>>
>>  Thanks,
>>  Steve Totaro
>>
>>
>>
>>  On Mon, May 5, 2008 at 9:29 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>>  > I looked all over VOIP-INFO and ATFOT and could not find anything that
>>  >  said or even suggested not using the mysql driver.(except NOT to have
>>  >  BOTH drivers loaded at the same time). I could easily be missing
>>  >  something. But the apparent BUG I am seeing is at such a Basic and
>>  >  Simple Level of functionality that either DIGIUM ought to fix it ASAP or
>>  >  update VOIP-INFO pages and their own documentation to say "Broke - No
>>  >  Workie and We Are No Gonna Fixie" :)
>>  >
>>  >
>>  >
>>  >  Steve Totaro wrote:
>>  >  > On Mon, May 5, 2008 at 4:21 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>>  >  >
>>  >  >>  I would appreciate any and all advice on what appears to be a BUG 
>> (or a
>>  >  >> brainfart on my part) with the MySQL add-on for Asterisk this is of 
>> FEDORA 8
>>  >  >> fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
>>  >  >>
>>  >  >>  It appears that the interface "eats" the first field requested from a
>>  >  >> table. If only One Field is Requested from the Table , that field is 
>> eaten
>>  >  >> ENTIRELY by Asterisk. If several fields are requested, the First 
>> Field Is
>>  >  >> Eaten and the remaining filed are returned, but place in the WRONG 
>> Variable
>>  >  >> since the 1tst fileld data was eaten. In the DIALPLAN below I have 
>> tried 3
>>  >  >> Different ways to approach this.
>>  >  >>
>>  >  >>  Extension  – Get only ONE (1) field from Table
>>  >  >>
>>  >  >>  Extension  – Get THREE(3) fields from the Table and Quote Them.
>>  >  >>
>>  >  >>  Extension  - Get THREE(3) fields from the Table
>>  >  >>
>>  >  >>  I have show the Output from the Asterisk CL for each, which clearly 
>> show
>>  >  >> that SOMETHING is not
>>  >  >>  right. Maybe the Software, maybe the person using the software :)
>>  >  >>
>>  >  >>  Here is the Table in the Database.
>>  >  >>
>>  >  >>  mysql> select * from agent;
>>  >  >>
>>  >  >>  +--+-+++-+
>>  >  >>
>>  >  >>  | id  | cust_id  | status |phone |tlce |
>>  >  >>
>>  >  >>  +--+-+++-+
>>  >  >>  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
>>  >  >>
>>  >  >>  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
>>  >  >>
>>  >  >>  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
>>  >  >>
>>  >  >>  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
>>  >  >>  +--+-+++-+
>>  >  >>
>>  >  >>  4 rows in set (0.00 sec)
>>  >  >>
>>  >  >>
>>  >  >>  Here is the DIALPLAN
>>  >  >>
>>  >  >>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>>  >  >>
>>  >  >>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
>>  >  >> status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
>>  >  >>
>>  >  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus 
>> mytlce)
>>  >  >>
>>  >  >>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} 
>> MYTLCE is
>>  >  >> ${mytlce})
>>  >  >>
>>  >  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. 
>> ${resultid}
>>  >  >> CONNID is ${connid})
>>  >  >>
>>  >  >>  exten => ,n,MYSQL(Clear ${resu

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Johansson Olle E]

Another issue that we need to fix with the MYSQL driver is that we're  
lacking a connection pool. Everything seems to be handled over one  
connection to Mysql, which causes issues.

Any MySQL developers out there that can help us fix this? We need  
someone that has been developing towards the Mysql api.

And please do not always refer to "Digium developers" when you have  
problems in Asterisk. There are developers that are not employees of  
Digium...

Cheers,
/Olle

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Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Steve Totaro]

A quote from Tilghman Lesher from a previous post.

"That's fine, but I have had the most horrid results using any distribution-
supplied ODBC drivers.  The best results are obtained by source-compiling
the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
PsqlODBC.  UnixODBC is fairly safe to use from distribution channels, however."

Thanks,
Steve Totaro

On Mon, May 5, 2008 at 10:06 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> User intervention is required Al.  You need to open a bug report on
>  mantis and anyone can edit the wiki (this is the nature of a wiki).
>  http://bugs.digium.com/main_page.php
>
>  Anyways, I am fairly certain that the UnixODBC connector is the
>  "preferred" way of connecting to MySQL.
>
>  Thanks,
>  Steve Totaro
>
>
>
>  On Mon, May 5, 2008 at 9:29 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>  > I looked all over VOIP-INFO and ATFOT and could not find anything that
>  >  said or even suggested not using the mysql driver.(except NOT to have
>  >  BOTH drivers loaded at the same time). I could easily be missing
>  >  something. But the apparent BUG I am seeing is at such a Basic and
>  >  Simple Level of functionality that either DIGIUM ought to fix it ASAP or
>  >  update VOIP-INFO pages and their own documentation to say "Broke - No
>  >  Workie and We Are No Gonna Fixie" :)
>  >
>  >
>  >
>  >  Steve Totaro wrote:
>  >  > On Mon, May 5, 2008 at 4:21 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>  >  >
>  >  >>  I would appreciate any and all advice on what appears to be a BUG (or 
> a
>  >  >> brainfart on my part) with the MySQL add-on for Asterisk this is of 
> FEDORA 8
>  >  >> fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
>  >  >>
>  >  >>  It appears that the interface "eats" the first field requested from a
>  >  >> table. If only One Field is Requested from the Table , that field is 
> eaten
>  >  >> ENTIRELY by Asterisk. If several fields are requested, the First Field 
> Is
>  >  >> Eaten and the remaining filed are returned, but place in the WRONG 
> Variable
>  >  >> since the 1tst fileld data was eaten. In the DIALPLAN below I have 
> tried 3
>  >  >> Different ways to approach this.
>  >  >>
>  >  >>  Extension  – Get only ONE (1) field from Table
>  >  >>
>  >  >>  Extension  – Get THREE(3) fields from the Table and Quote Them.
>  >  >>
>  >  >>  Extension  - Get THREE(3) fields from the Table
>  >  >>
>  >  >>  I have show the Output from the Asterisk CL for each, which clearly 
> show
>  >  >> that SOMETHING is not
>  >  >>  right. Maybe the Software, maybe the person using the software :)
>  >  >>
>  >  >>  Here is the Table in the Database.
>  >  >>
>  >  >>  mysql> select * from agent;
>  >  >>
>  >  >>  +--+-+++-+
>  >  >>
>  >  >>  | id  | cust_id  | status |phone |tlce |
>  >  >>
>  >  >>  +--+-+++-+
>  >  >>  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
>  >  >>
>  >  >>  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
>  >  >>
>  >  >>  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
>  >  >>
>  >  >>  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
>  >  >>  +--+-+++-+
>  >  >>
>  >  >>  4 rows in set (0.00 sec)
>  >  >>
>  >  >>
>  >  >>  Here is the DIALPLAN
>  >  >>
>  >  >>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>  >  >>
>  >  >>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
>  >  >> status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
>  >  >>
>  >  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus 
> mytlce)
>  >  >>
>  >  >>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} 
> MYTLCE is
>  >  >> ${mytlce})
>  >  >>
>  >  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. 
> ${resultid}
>  >  >> CONNID is ${connid})
>  >  >>
>  >  >>  exten => ,n,MYSQL(Clear ${resultid})
>  >  >>
>  >  >>  exten => ,n,MYSQL(Disconnect ${connid})
>  >  >>
>  >  >>  exten => ,n,HANGUP
>  >  >>
>  >  >>
>  >  >>
>  >  >>  exten => ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
>  >  >> 'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
>  >  >>
>  >  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus 
> mytlce)
>  >  >>
>  >  >>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} 
> MYTLCE is
>  >  >> ${mytlce})
>  >  >>
>  >  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. 
> ${resultid}
>  >  >> CONNID is ${connid})
>  >  >>
>  >  >>  exten => ,n,MYSQL(Clear ${resultid})
>  >  >>
>  >  >>  exten => ,n,MYSQL(Disconnect ${connid})
>  >  >>
>  >  >>  exten => ,n,HANGUP
>  >  >>
>  >  >>
>  >  >>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>  >  >>
>  >  >>  exten =>

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Steve Totaro]

User intervention is required Al.  You need to open a bug report on
mantis and anyone can edit the wiki (this is the nature of a wiki).
http://bugs.digium.com/main_page.php

Anyways, I am fairly certain that the UnixODBC connector is the
"preferred" way of connecting to MySQL.

Thanks,
Steve Totaro

On Mon, May 5, 2008 at 9:29 AM, Al Baker <[EMAIL PROTECTED]> wrote:
> I looked all over VOIP-INFO and ATFOT and could not find anything that
>  said or even suggested not using the mysql driver.(except NOT to have
>  BOTH drivers loaded at the same time). I could easily be missing
>  something. But the apparent BUG I am seeing is at such a Basic and
>  Simple Level of functionality that either DIGIUM ought to fix it ASAP or
>  update VOIP-INFO pages and their own documentation to say "Broke - No
>  Workie and We Are No Gonna Fixie" :)
>
>
>
>  Steve Totaro wrote:
>  > On Mon, May 5, 2008 at 4:21 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>  >
>  >>  I would appreciate any and all advice on what appears to be a BUG (or a
>  >> brainfart on my part) with the MySQL add-on for Asterisk this is of 
> FEDORA 8
>  >> fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
>  >>
>  >>  It appears that the interface "eats" the first field requested from a
>  >> table. If only One Field is Requested from the Table , that field is eaten
>  >> ENTIRELY by Asterisk. If several fields are requested, the First Field Is
>  >> Eaten and the remaining filed are returned, but place in the WRONG 
> Variable
>  >> since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
>  >> Different ways to approach this.
>  >>
>  >>  Extension  – Get only ONE (1) field from Table
>  >>
>  >>  Extension  – Get THREE(3) fields from the Table and Quote Them.
>  >>
>  >>  Extension  - Get THREE(3) fields from the Table
>  >>
>  >>  I have show the Output from the Asterisk CL for each, which clearly show
>  >> that SOMETHING is not
>  >>  right. Maybe the Software, maybe the person using the software :)
>  >>
>  >>  Here is the Table in the Database.
>  >>
>  >>  mysql> select * from agent;
>  >>
>  >>  +--+-+++-+
>  >>
>  >>  | id  | cust_id  | status |phone |tlce |
>  >>
>  >>  +--+-+++-+
>  >>  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
>  >>
>  >>  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
>  >>
>  >>  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
>  >>
>  >>  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
>  >>  +--+-+++-+
>  >>
>  >>  4 rows in set (0.00 sec)
>  >>
>  >>
>  >>  Here is the DIALPLAN
>  >>
>  >>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>  >>
>  >>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
>  >> status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
>  >>
>  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>  >>
>  >>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE 
> is
>  >> ${mytlce})
>  >>
>  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>  >> CONNID is ${connid})
>  >>
>  >>  exten => ,n,MYSQL(Clear ${resultid})
>  >>
>  >>  exten => ,n,MYSQL(Disconnect ${connid})
>  >>
>  >>  exten => ,n,HANGUP
>  >>
>  >>
>  >>
>  >>  exten => ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
>  >> 'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
>  >>
>  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>  >>
>  >>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE 
> is
>  >> ${mytlce})
>  >>
>  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>  >> CONNID is ${connid})
>  >>
>  >>  exten => ,n,MYSQL(Clear ${resultid})
>  >>
>  >>  exten => ,n,MYSQL(Disconnect ${connid})
>  >>
>  >>  exten => ,n,HANGUP
>  >>
>  >>
>  >>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>  >>
>  >>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
>  >> agent\ where\ phone=\'1234567890\')
>  >>
>  >>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid)
>  >>
>  >>  exten => ,n,NoOp(CUSTID is ${custid})
>  >>
>  >>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>  >> CONNID is ${connid})
>  >>
>  >>  exten => ,n,MYSQL(Clear ${resultid})
>  >>
>  >>  exten => ,n,MYSQL(Disconnect ${connid})
>  >>
>  >>  exten => ,n,HANGUP
>  >>
>  >>
>  >>
>  >>
>  >>  Here is the Asterisk CLI Output
>  >>
>  >>  dial 
>  >>
>  >>  == Console is full duplex
>  >>
>  >>  *CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
>  >> localhost ivr ivrxxx dtc") in new stack
>  >>
>  >>  -- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Al Baker]

I looked all over VOIP-INFO and ATFOT and could not find anything that 
said or even suggested not using the mysql driver.(except NOT to have 
BOTH drivers loaded at the same time). I could easily be missing 
something. But the apparent BUG I am seeing is at such a Basic and 
Simple Level of functionality that either DIGIUM ought to fix it ASAP or 
update VOIP-INFO pages and their own documentation to say "Broke - No 
Workie and We Are No Gonna Fixie" :)

Steve Totaro wrote:
> On Mon, May 5, 2008 at 4:21 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>   
>>  I would appreciate any and all advice on what appears to be a BUG (or a
>> brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
>> fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
>>
>>  It appears that the interface "eats" the first field requested from a
>> table. If only One Field is Requested from the Table , that field is eaten
>> ENTIRELY by Asterisk. If several fields are requested, the First Field Is
>> Eaten and the remaining filed are returned, but place in the WRONG Variable
>> since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
>> Different ways to approach this.
>>
>>  Extension  – Get only ONE (1) field from Table
>>
>>  Extension  – Get THREE(3) fields from the Table and Quote Them.
>>
>>  Extension  - Get THREE(3) fields from the Table
>>
>>  I have show the Output from the Asterisk CL for each, which clearly show
>> that SOMETHING is not
>>  right. Maybe the Software, maybe the person using the software :)
>>
>>  Here is the Table in the Database.
>>
>>  mysql> select * from agent;
>>
>>  +--+-+++-+
>>
>>  | id  | cust_id  | status |phone |tlce |
>>
>>  +--+-+++-+
>>  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
>>
>>  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
>>
>>  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
>>
>>  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
>>  +--+-+++-+
>>
>>  4 rows in set (0.00 sec)
>>
>>
>>  Here is the DIALPLAN
>>
>>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>>
>>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
>> status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
>>
>>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>>
>>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
>> ${mytlce})
>>
>>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>> CONNID is ${connid})
>>
>>  exten => ,n,MYSQL(Clear ${resultid})
>>
>>  exten => ,n,MYSQL(Disconnect ${connid})
>>
>>  exten => ,n,HANGUP
>>
>>
>>
>>  exten => ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
>> 'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
>>
>>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>>
>>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
>> ${mytlce})
>>
>>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>> CONNID is ${connid})
>>
>>  exten => ,n,MYSQL(Clear ${resultid})
>>
>>  exten => ,n,MYSQL(Disconnect ${connid})
>>
>>  exten => ,n,HANGUP
>>
>>
>>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>>
>>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
>> agent\ where\ phone=\'1234567890\')
>>
>>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid)
>>
>>  exten => ,n,NoOp(CUSTID is ${custid})
>>
>>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
>> CONNID is ${connid})
>>
>>  exten => ,n,MYSQL(Clear ${resultid})
>>
>>  exten => ,n,MYSQL(Disconnect ${connid})
>>
>>  exten => ,n,HANGUP
>>
>>
>>
>>
>>  Here is the Asterisk CLI Output
>>
>>  dial 
>>
>>  == Console is full duplex
>>
>>  *CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
>> localhost ivr ivrxxx dtc") in new stack
>>
>>  -- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp", "Query resultid 5 SELECT
>> cust_id from agent where phone='1234567890'") in new stack
>>
>>  -- Executing [EMAIL PROTECTED]:3] MYSQL("OSS/dsp", "Fetch fetchid 6 
>> custid") in
>> new stack
>>
>>  -- Executing [EMAIL PROTECTED]:4] NoOp("OSS/dsp", "CUSTID is ") in new stack
>>
>>  -- Executing [EMAIL PROTECTED]:5] NoOp("OSS/dsp", "FETCHID is 1 RESULUT ID 
>> is
>> .. 6 CONNID is 5") in new stack
>>
>>  -- Executing [EMAIL PROTECTED]:6] MYSQL("OSS/dsp", "Clear 6") in new stack
>>
>>  -- Executing [EMAIL PROTECTED]:7] MYSQL("OSS/dsp", "Disconnect 5") in new 
>> stack
>>
>>  -- Executing [EMAIL PROTECTED]:8] Hangup("OSS/dsp", "") in new stack
>>
>>  == Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'
>>
>>  << Hangup on con

Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Steve Totaro]

On Mon, May 5, 2008 at 4:21 AM, Al Baker <[EMAIL PROTECTED]> wrote:
>
>  I would appreciate any and all advice on what appears to be a BUG (or a
> brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
> fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
>
>  It appears that the interface "eats" the first field requested from a
> table. If only One Field is Requested from the Table , that field is eaten
> ENTIRELY by Asterisk. If several fields are requested, the First Field Is
> Eaten and the remaining filed are returned, but place in the WRONG Variable
> since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
> Different ways to approach this.
>
>  Extension  – Get only ONE (1) field from Table
>
>  Extension  – Get THREE(3) fields from the Table and Quote Them.
>
>  Extension  - Get THREE(3) fields from the Table
>
>  I have show the Output from the Asterisk CL for each, which clearly show
> that SOMETHING is not
>  right. Maybe the Software, maybe the person using the software :)
>
>  Here is the Table in the Database.
>
>  mysql> select * from agent;
>
>  +--+-+++-+
>
>  | id  | cust_id  | status |phone |tlce |
>
>  +--+-+++-+
>  | 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
>
>  | 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
>
>  | 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
>
>  | 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
>  +--+-+++-+
>
>  4 rows in set (0.00 sec)
>
>
>  Here is the DIALPLAN
>
>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>
>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
> status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
>
>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>
>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
> ${mytlce})
>
>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
> CONNID is ${connid})
>
>  exten => ,n,MYSQL(Clear ${resultid})
>
>  exten => ,n,MYSQL(Disconnect ${connid})
>
>  exten => ,n,HANGUP
>
>
>
>  exten => ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
> 'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
>
>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
>
>  exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
> ${mytlce})
>
>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
> CONNID is ${connid})
>
>  exten => ,n,MYSQL(Clear ${resultid})
>
>  exten => ,n,MYSQL(Disconnect ${connid})
>
>  exten => ,n,HANGUP
>
>
>  exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
>
>  exten => ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
> agent\ where\ phone=\'1234567890\')
>
>  exten => ,n,MYSQL(Fetch fetchid ${resultid} custid)
>
>  exten => ,n,NoOp(CUSTID is ${custid})
>
>  exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
> CONNID is ${connid})
>
>  exten => ,n,MYSQL(Clear ${resultid})
>
>  exten => ,n,MYSQL(Disconnect ${connid})
>
>  exten => ,n,HANGUP
>
>
>
>
>  Here is the Asterisk CLI Output
>
>  dial 
>
>  == Console is full duplex
>
>  *CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
> localhost ivr ivrxxx dtc") in new stack
>
>  -- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp", "Query resultid 5 SELECT
> cust_id from agent where phone='1234567890'") in new stack
>
>  -- Executing [EMAIL PROTECTED]:3] MYSQL("OSS/dsp", "Fetch fetchid 6 custid") 
> in
> new stack
>
>  -- Executing [EMAIL PROTECTED]:4] NoOp("OSS/dsp", "CUSTID is ") in new stack
>
>  -- Executing [EMAIL PROTECTED]:5] NoOp("OSS/dsp", "FETCHID is 1 RESULUT ID is
> .. 6 CONNID is 5") in new stack
>
>  -- Executing [EMAIL PROTECTED]:6] MYSQL("OSS/dsp", "Clear 6") in new stack
>
>  -- Executing [EMAIL PROTECTED]:7] MYSQL("OSS/dsp", "Disconnect 5") in new 
> stack
>
>  -- Executing [EMAIL PROTECTED]:8] Hangup("OSS/dsp", "") in new stack
>
>  == Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'
>
>  << Hangup on console
>
>  *CLI> dial 
>
>  == Console is full duplex
>
>  *CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
> localhost ivr ivrxxx dtc") in new stack
>
>  -- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp", "Query resultid 5 SELECT
> cust_id, status, tlce from agent where phone='1234567890'") in new stack
>
>  -- Executing [EMAIL PROTECTED]:3] MYSQL("OSS/dsp", "Fetch fetchid 6 custid
> mystatus mytlce") in new stack
>
>  -- Executing [EMAIL PROTECTED]:4] NoOp("OSS/dsp", "CUSTID is free MYSTATUS is
> 2008-04-17 02:32:02 MYTLCE is ") in new stack
>
>  -- Executing [EMAIL PROTECTED]:5] NoOp("OS

[asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data

[Al Baker]

I would appreciate any and
all advice on what
appears to be a BUG (or a brainfart on my part) with the MySQL add-on
for Asterisk this is of FEDORA 8 fully patched with Asterisk Addons
1-4-6 with the Asterisk 1.4.18.1


It appears that the interface “eats” the
first field requested from
a table. If only One Field is Requested
from the Table , that field is
eaten ENTIRELY by Asterisk. If several fields
are requested, the First
Field Is Eaten and the remaining filed are
returned, but place in the
WRONG Variable since the 1tst fileld data
was eaten. In the DIALPLAN
below I have tried 3 Different ways to approach this. 

Extension  – Get only ONE (1) field from Table


Extension  – Get THREE(3) fields from
the Table
and Quote
Them.

Extension  - Get THREE(3) fields from the Table 

I have show the Output from the Asterisk CL for each, which clearly
show that SOMETHING is not 
right. Maybe the Software, maybe the person using the software :)


Here is the Table in the Database.


mysql> select * from agent;


+--+-+++-+


| id          | cust_id  | status |    phone |    tlce |


+--+-+++-+

| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 | 

| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 | 

| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 | 

| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 | 
+--+-+++-+


4 rows in set (0.00 sec)



Here is the DIALPLAN 

exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)


exten => ,n,MYSQL(Query resultid ${connid}
SELECT\ cust_id\, \ status\,\ tlce\ from\
agent\ where\ phone=\'1234567890\')


exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)


exten => ,n,NoOp(CUSTID is ${custid}
MYSTATUS is ${mystatus} MYTLCE is ${mytlce})


exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid} CONNID is ${connid})


exten => ,n,MYSQL(Clear ${resultid})


exten => ,n,MYSQL(Disconnect ${connid})


exten => ,n,HANGUP




exten => ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')


exten => ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus
mytlce)


exten => ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus}
MYTLCE is ${mytlce})


exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid} CONNID is ${connid})


exten => ,n,MYSQL(Clear ${resultid})


exten => ,n,MYSQL(Disconnect ${connid})


exten => ,n,HANGUP



exten => ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)


exten => ,n,MYSQL(Query resultid ${connid}
SELECT\ 'cust_id'\ from\ agent\ where\ phone=\'1234567890\')


exten => ,n,MYSQL(Fetch fetchid
${resultid} custid)


exten => ,n,NoOp(CUSTID is ${custid})


exten => ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid} CONNID is ${connid})


exten => ,n,MYSQL(Clear ${resultid})


exten => ,n,MYSQL(Disconnect ${connid})


exten => ,n,HANGUP





Here is the Asterisk CLI Output


dial 


== Console is full duplex


*CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
localhost ivr ivrxxx dtc") in new stack


-- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp", "Query
resultid 5
SELECT cust_id from agent where phone='1234567890'") in new
stack


-- Executing [EMAIL PROTECTED]:3] MYSQL("OSS/dsp", "Fetch
fetchid 6 custid") in new stack


-- Executing [EMAIL PROTECTED]:4] NoOp("OSS/dsp",
"CUSTID is ") in new stack


-- Executing [EMAIL PROTECTED]:5] NoOp("OSS/dsp", "FETCHID
is 1 RESULUT ID is .. 6 CONNID is 5") in new stack


-- Executing [EMAIL PROTECTED]:6] MYSQL("OSS/dsp", "Clear 6") in new stack


-- Executing [EMAIL PROTECTED]:7] MYSQL("OSS/dsp", "Disconnect 5") in new
stack


-- Executing [EMAIL PROTECTED]:8] Hangup("OSS/dsp", "") in new stack


== Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'


<< Hangup on console 

*CLI> dial 


== Console is full duplex


*CLI> -- Executing [EMAIL PROTECTED]:1] MYSQL("OSS/dsp", "Connect connid
localhost ivr ivrxxx dtc") in new stack


-- Executing [EMAIL PROTECTED]:2] MYSQL("OSS/dsp", "Query
resultid 5
SELECT cust_id, status, tlce from agent where phone='1234567890'")
in
new stack


-- Executing [EMAIL PROTECTED]:3] MYSQL("OSS/dsp",
"Fetch fetchid 6 custid mystatus mytlce") in new stack


-- Executing [EMAIL PROTECTED]:4] NoOp("OSS/dsp", "CUSTID
is free MYSTATUS is 2008-04-17 02:32:02 MYTLCE is ") in new
stack


-- Executing [EMAIL PROTECTED]:5] NoOp("OSS/dsp", "FETCHID
is 1 RESULUT ID is .. 6 CONNID is 5") in new stack


-- Executing [EMAIL PROTECTED]:6] MYSQL("OSS/dsp", "Clear 6") in new stack


-- Executing [EMAIL PROTECTED]:7] MYSQL("OSS/dsp", "Disconnect 5") in new
stack


-- Executing [EMAIL PROTECTED]:8] Hangup("OSS/dsp", "") in new stack


== Spawn ex

Re: [asterisk-users] help please

[Tzafrir Cohen]

On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote:
> hello, I have a A400P01 PCI from OpenVox.
> 
> I have installed some extension and a VoipBuste account to callo out of my 
> LAN.
> 
> How can I receive and send calls from a nd to outside by my analog line???
> 
> I want to receive dthe calls from 20100 extension.
> 
> Here you have my config files, thanks for all.

A few things unrelated to your issue that may help you to get more
effetive answers from this list:

1. Please give more descriptive subject lines.
The subject of your first message ("asterisk on Debian") was good.
The subject of your more recent messages are rather poor: "please help
me" gives no hint as to what the problem is.

2. You have already started a thread, and another list member has asked
you for some details. The files attached to this message appear to be
replies to that message. If they are, please follow-up the same thread.

3. You did not write what is actually wrong:

"I do XYZ. I expect it to cause ABC but instead I get DEF"

See also the document on how to ask questions effectively:
http://www.catb.org/~esr/faqs/smart-questions.html

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

[Josu Lazkano Lete]

hello, I have a A400P01 PCI from OpenVox.

I have installed some extension and a VoipBuste account to callo out of my LAN.

How can I receive and send calls from a nd to outside by my analog line???

I want to receive dthe calls from 20100 extension.

Here you have my config files, thanks for all.fxsks=1
loadzone=es
defaultzone=es[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[miprimerejemplo]
exten => 2,1,Dial(SIP/2,30,Ttm)
exten => 2,2,Hangup
exten => 2,102,Voicemail(2)
exten => 2,103,Hangup

exten => 20100,1,Dial(SIP/20100,30,Ttm)
exten => 20100,2,Hangup
exten => 20100,102,Voicemail(20100)
exten => 20100,103,Hangup

exten => 20200,1,Dial(SIP/20200,30,Ttm)
exten => 20200,2,Hangup
exten => 202000,102,Voicemail(20200)
exten => 20200,103,Hangup

exten => 20300,1,Dial(SIP/20300,30,Ttm)
exten => 20300,2,Hangup
exten => 203000,102,Voicemail(20300)
exten => 20300,103,Hangup

exten => 20400,1,Dial(SIP/20400,30,Ttm)
exten => 20400,2,Hangup
exten => 204000,102,Voicemail(20400)
exten => 20400,103,Hangup

exten => 3,1,VoicemailMain

exten => _9,1,Dial(SIP/[EMAIL PROTECTED])
exten => _9,2,Hangup[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[2]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
[EMAIL PROTECTED]

[20100]
type=friend
secret=some
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=miprimerejemplo
[EMAIL PROTECTED] 

[20200]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
[EMAIL PROTECTED]

[20300]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
[EMAIL PROTECTED]

[20400]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=miprimerejemplo
[EMAIL PROTECTED]

[VoipBuster]
type=peer
host=sip.voipbuster.com
username=somesi3
fromuser=somesi3
secret=some[channels]
language=es
context=incoming
switchtype=euroisdn
usercallid=yes
hidecallerid=no
musiconhold=default
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
inmediate=no
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbriged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxs_ks
context=incoming
channel=4___
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[asterisk-users] help please

[Josu Lazkano Lete]

hello, I have a A400P01 PCI from OpenVox.

I have installed some extension and a VoipBuste account to callo out of my LAN.

How can I receive and send calls from a nd to outside by my analog line???

I want to receive dthe calls from 20100 extension.

Here you have my config files, thanks for all.

asterisk.rar
Description: Binary data
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[asterisk-users] help please

[Josu Lazkano Lete]

hello, I have a A400P01 PCI from OpenVox.

I have installed some extension and a VoipBuste account to callo out of my LAN.

How can I receive and send calls from a nd to outside by my analog line???

I want to receive dthe calls from 20100 extension.

Here you have my config files, thanks for all.

zaptel.conf
Description: Binary data


extensions.conf
Description: Binary data


sip.conf
Description: Binary data


zapata.conf
Description: Binary data
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Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Naija Man]

-- Forwarded message --From: "William Piper" <
[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>Date: Sat, 30 Sep 2006 22:08:23 -0400
Subject: Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LANSure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try & see what happens.

 
bp 
On 9/28/06, Wolfgang_Borgon <[EMAIL PROTECTED]
> wrote:
David,Yes, I've also forwarded port 4569 to the server.Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as theclient isn't going out past the LAN, it shouldn'tmatter... unless there's something else going on thatI don't know about.ThanksWolfgang--- David J Carter <
[EMAIL PROTECTED]> wrote:> Wolfgang wrote:  ->> I've already sunk several hours into this without
> any> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a> softphone on XP ProSP2 to a Asterisk server on Linux> FC4 over a LAN through a Netgear router. The server> will then go out to a PSTN termination service.
>> Thus far, the PSTN termination connection works fine> -- I've opened up 4569 with iptables, and forwarded> 4569 to the server IP.  I am not, however, having> any> luck connecting the softphone to the server.
>> I can telnet, ftp, and http to the server, but not> IAX2. Iaxping times out, registration by Idefisk and> Firefly also times out.>> The server fails to see the client as well.

>> Here's a portion of my iax.conf:>> [client]> type=friend> username=client> secret=**> host=
192.168.1.40> context=clientcon
>> and extensions.conf:>> [clientcon]> exten => 2278,1,Dial(IAX2/client)>>>==> You say you have 4569 configured in iptables, what
> about the netgear router?>> Have you port forwarded 4569 there?>> Dave>In your iax.conf, instead of...host=
192.168.1.40Use...host=dynamic-Buki - Da Naija Man
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Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[William Piper]

Sure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try & see what happens.
 
bp 
On 9/28/06, Wolfgang_Borgon <[EMAIL PROTECTED]> wrote:
David,Yes, I've also forwarded port 4569 to the server.Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as theclient isn't going out past the LAN, it shouldn'tmatter... unless there's something else going on thatI don't know about.ThanksWolfgang--- David J Carter <
[EMAIL PROTECTED]> wrote:> Wolfgang wrote:  ->> I've already sunk several hours into this without> any> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a> softphone on XP ProSP2 to a Asterisk server on Linux> FC4 over a LAN through a Netgear router. The server> will then go out to a PSTN termination service.
>> Thus far, the PSTN termination connection works fine> -- I've opened up 4569 with iptables, and forwarded> 4569 to the server IP.  I am not, however, having> any> luck connecting the softphone to the server.
>> I can telnet, ftp, and http to the server, but not> IAX2. Iaxping times out, registration by Idefisk and> Firefly also times out.>> The server fails to see the client as well.
>> Here's a portion of my iax.conf:>> [client]> type=friend> username=client> secret=**> host=192.168.1.40> context=clientcon
>> and extensions.conf:>> [clientcon]> exten => 2278,1,Dial(IAX2/client)>>>==> You say you have 4569 configured in iptables, what
> about the netgear router?>> Have you port forwarded 4569 there?>> Dave>> ___> --Bandwidth and Colocation sponsored by 
Easynews.com> -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com>
http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:>>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Tim Panton]

On 28 Sep 2006, at 21:39, Wolfgang_Borgon wrote:


David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang

--- David J Carter <[EMAIL PROTECTED]> wrote:


Wolfgang wrote:  -

I've already sunk several hours into this without
any
real progress, so I'd really appreciate any help  My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN termination service.

Thus far, the PSTN termination connection works fine
-- I've opened up 4569 with iptables, and forwarded
4569 to the server IP.  I am not, however, having
any
luck connecting the softphone to the server.

I can telnet, ftp, and http to the server, but not
IAX2. Iaxping times out, registration by Idefisk and
Firefly also times out.

The server fails to see the client as well.

Here's a portion of my iax.conf:

[client]
type=friend
username=client
secret=**
host=192.168.1.40
context=clientcon

and extensions.conf:

[clientcon]
exten => 2278,1,Dial(IAX2/client)




==

You say you have 4569 configured in iptables, what
about the netgear router?

Have you port forwarded 4569 there?

Dave

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You'll need to run ethereal or another packet sniffer on the network
to see what is happening.

In general if your asterisk is going to _register_ (or qualify)
your external IAX provider you don't need to port-forward 4569,
the registration will setup a suitable nat/port mapping
and the qualify (or re-register) will keep it alive.
(But that isn't your problem)

T.


Tim Panton

www.mexuar.com



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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang_Borgon]

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang

--- David J Carter <[EMAIL PROTECTED]> wrote:

> Wolfgang wrote:  -
> 
> I've already sunk several hours into this without
> any
> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a
> softphone on XP ProSP2 to a Asterisk server on Linux
> FC4 over a LAN through a Netgear router. The server
> will then go out to a PSTN termination service.
> 
> Thus far, the PSTN termination connection works fine
> -- I've opened up 4569 with iptables, and forwarded
> 4569 to the server IP.  I am not, however, having
> any
> luck connecting the softphone to the server.
> 
> I can telnet, ftp, and http to the server, but not
> IAX2. Iaxping times out, registration by Idefisk and
> Firefly also times out.  
> 
> The server fails to see the client as well.  
> 
> Here's a portion of my iax.conf:
> 
> [client]
> type=friend
> username=client
> secret=**
> host=192.168.1.40
> context=clientcon
> 
> and extensions.conf:
> 
> [clientcon]
> exten => 2278,1,Dial(IAX2/client)
> 
> 
>
==
> You say you have 4569 configured in iptables, what
> about the netgear router?
> 
> Have you port forwarded 4569 there?
> 
> Dave
> 
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>
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>   
>
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> 




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[asterisk-users] Help please ==> Wrong password

[Noc Phibee]

Hi

i have a small problems with my asterisk connected to phonesystems :

Now i have this message:

<-- SIP read from 62.39.136.151:5060:
SIP/2.0 403 Cant accept register from myself
Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060
From: ;tag=as42b95c05
To: 
;tag=e3fe971527b049ab0c1e91db33fcbf5f.cf8c

Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
Server: PSN Sip Proxy (1.1.3 (PRX3-EXTERNAL))
Content-Length: 0
Warning: 392 62.39.136.151:5060 "Noisy feedback tells:  pid=11434 
req_src_ip=62.39.136.151 req_src_port=5060 
in_uri=sip:sip3.phonesystems.net out_uri=sip:sip3.phonesystems.net 
via_cnt==2"



--- (9 headers 0 lines)---
Aug 30 17:12:50 WARNING[15568]: chan_sip.c:10010 handle_response: 
Forbidden - wrong password on authentication for REGISTER




but my login/password are correct into sip.conf

the configuration have changed in asterisk 1.2.11 ?

thanks for your help

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Re: [Asterisk-Users] HELP PLEASE: What A Pain RSA

[OTR Comm]

Hello Kevin,


> If your keys have passphrases, then you must run the 'init keys' CLI
> command to enter the passphrase before the keys can be loaded.
> 
> It's much simpler to just not use passphrases and instead protect your
> private key every other way that you can :-)

Yeah I rebuilt my keys without a passphrase, but I still can't get the
two boxes to register.

On one box called pbx1 (xxx.xxx.xxx.2), I have in iax.conf:

[general]
...
register => pbx1:[EMAIL PROTECTED]
...
[pbx1too]
type=friend
host=dynamic
auth=rsa
inkeys=pbx1too
outkeys=pbx1
username=pbx1too
context=outgoing

Where the keys for pbx1 are pbx1.pub and pbx1.key, and a copy of
pbx1too.pub is in /var/lib/asterisk/keys/

And then on another box called pbx1too (xxx.xxx.xxx.3), in the iax.conf:

[general]
...
register => pbx1too:[EMAIL PROTECTED]
...
[pbx1]
type=friend
host=dynamic
auth=rsa
inkeys=pbx1
outkeys=pbx1too
username=pbx1
context=incoming


Where the keys for pbx1too are pbx1too.pub and pbx1too.key, and a copy
of pbx1.pub is in /var/lib/asterisk/keys/

I can't see anything wrong with this, but I have been at it for more
hours than I want to think about right now :)

Hope you can!


Got another problem involving D-channels that go away and calls get
dropped, then the D-channels come back and the B-channels get rebuilt. 
Only on PRI -> SIP calls, but, one problem at a time.

Thanks for looking,

Murrah Boswell
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Re: [Asterisk-Users] HELP PLEASE: What A Pain RSA

[Kevin P. Fleming]

OTR Comm wrote:


I did give the key a passphrase, could this be the problem?  Do I have
to put the passphrase in some config file?


If your keys have passphrases, then you must run the 'init keys' CLI 
command to enter the passphrase before the keys can be loaded.


It's much simpler to just not use passphrases and instead protect your 
private key every other way that you can :-)

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[Asterisk-Users] HELP PLEASE: What A Pain RSA

[OTR Comm]

Hello all,

I am having a terrible time with RSA registry.  I have genetrated a key
for 'pbx1too' with astgenkey and put it in /var/lib/asterisk/keys/, but
when I reload iax2 I get an error:

authenticate: Unable to find private key 'pbx1too'

Where else would asterisk be looking for the key?

I also put 'pbx1too.pub' in /var/lib/asterisk/keys/

I did give the key a passphrase, could this be the problem?  Do I have
to put the passphrase in some config file?

Thanks,
Murrah Boswell
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang Borgon]

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang

--- David J Carter <[EMAIL PROTECTED]> wrote:

> Wolfgang wrote:  -
> 
> I've already sunk several hours into this without
> any
> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a
> softphone on XP ProSP2 to a Asterisk server on Linux
> FC4 over a LAN through a Netgear router. The server
> will then go out to a PSTN termination service.
> 
> Thus far, the PSTN termination connection works fine
> -- I've opened up 4569 with iptables, and forwarded
> 4569 to the server IP.  I am not, however, having
> any
> luck connecting the softphone to the server.
> 
> I can telnet, ftp, and http to the server, but not
> IAX2. Iaxping times out, registration by Idefisk and
> Firefly also times out.  
> 
> The server fails to see the client as well.  
> 
> Here's a portion of my iax.conf:
> 
> [client]
> type=friend
> username=client
> secret=**
> host=192.168.1.40
> context=clientcon
> 
> and extensions.conf:
> 
> [clientcon]
> exten => 2278,1,Dial(IAX2/client)
> 
> 
>
==
> You say you have 4569 configured in iptables, what
> about the netgear router?
> 
> Have you port forwarded 4569 there?
> 
> Dave
> 
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang Borgon]

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang

--- David J Carter <[EMAIL PROTECTED]> wrote:

> Wolfgang wrote:  -
> 
> I've already sunk several hours into this without
> any
> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a
> softphone on XP ProSP2 to a Asterisk server on Linux
> FC4 over a LAN through a Netgear router. The server
> will then go out to a PSTN termination service.
> 
> Thus far, the PSTN termination connection works fine
> -- I've opened up 4569 with iptables, and forwarded
> 4569 to the server IP.  I am not, however, having
> any
> luck connecting the softphone to the server.
> 
> I can telnet, ftp, and http to the server, but not
> IAX2. Iaxping times out, registration by Idefisk and
> Firefly also times out.  
> 
> The server fails to see the client as well.  
> 
> Here's a portion of my iax.conf:
> 
> [client]
> type=friend
> username=client
> secret=**
> host=192.168.1.40
> context=clientcon
> 
> and extensions.conf:
> 
> [clientcon]
> exten => 2278,1,Dial(IAX2/client)
> 
> 
>
==
> You say you have 4569 configured in iptables, what
> about the netgear router?
> 
> Have you port forwarded 4569 there?
> 
> Dave
> 
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[David J Carter]

>>David:

>>Also port 1:2 is a good idea to forward to the server as well..

Only needed for SIP. 4569 is all that is required for IAX2.

> David,
> Yes, I've also forwarded port 4569 to the server.
> Since the router is forwarding to the server, I cannot
> forward it to the client as well -- however, as the
> client isn't going out past the LAN, it shouldn't
> matter... unless there's something else going on that
> I don't know about.
> Thanks
> Wolfgang
>
You might try: -

[2278]
type=friend
secret=**
host=dynamic
context=clientcon
>
> --- David J Carter <[EMAIL PROTECTED]> wrote:
>
>> Wolfgang wrote:  -
>>
>> I've already sunk several hours into this without
>> any
>> real progress, so I'd really appreciate any help  My
>> task is simple -- establish a connection between a
>> softphone on XP ProSP2 to a Asterisk server on Linux
>> FC4 over a LAN through a Netgear router. The server
>> will then go out to a PSTN termination service.
>>
>> Thus far, the PSTN termination connection works fine
>> -- I've opened up 4569 with iptables, and forwarded
>> 4569 to the server IP.  I am not, however, having
>> any
>> luck connecting the softphone to the server.
>>
>> I can telnet, ftp, and http to the server, but not
>> IAX2. Iaxping times out, registration by Idefisk and
>> Firefly also times out.
>>
>> The server fails to see the client as well.
>>
>> Here's a portion of my iax.conf:
>>
>> [client]
>> type=friend
>> username=client
>> secret=**
>> host=192.168.1.40
>> context=clientcon
>>
>> and extensions.conf:
>>
>> [clientcon]
>> exten => 2278,1,Dial(IAX2/client)
>>
>>
>>
> ==
>> You say you have 4569 configured in iptables, what
>> about the netgear router?
>>
>> Have you port forwarded 4569 there?
>>
>> Dave
>>
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>> --
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>> Asterisk-Users@lists.digium.com
>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>
>>
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>>
>
>
>
>
>
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[pbx]

David:

Also port 1:2 is a good idea to forward to the server as well..


> David,
> Yes, I've also forwarded port 4569 to the server.
> Since the router is forwarding to the server, I cannot
> forward it to the client as well -- however, as the
> client isn't going out past the LAN, it shouldn't
> matter... unless there's something else going on that
> I don't know about.
> Thanks
> Wolfgang
>
>
> --- David J Carter <[EMAIL PROTECTED]> wrote:
>
>> Wolfgang wrote:  -
>>
>> I've already sunk several hours into this without
>> any
>> real progress, so I'd really appreciate any help  My
>> task is simple -- establish a connection between a
>> softphone on XP ProSP2 to a Asterisk server on Linux
>> FC4 over a LAN through a Netgear router. The server
>> will then go out to a PSTN termination service.
>>
>> Thus far, the PSTN termination connection works fine
>> -- I've opened up 4569 with iptables, and forwarded
>> 4569 to the server IP.  I am not, however, having
>> any
>> luck connecting the softphone to the server.
>>
>> I can telnet, ftp, and http to the server, but not
>> IAX2. Iaxping times out, registration by Idefisk and
>> Firefly also times out.
>>
>> The server fails to see the client as well.
>>
>> Here's a portion of my iax.conf:
>>
>> [client]
>> type=friend
>> username=client
>> secret=**
>> host=192.168.1.40
>> context=clientcon
>>
>> and extensions.conf:
>>
>> [clientcon]
>> exten => 2278,1,Dial(IAX2/client)
>>
>>
>>
> ==
>> You say you have 4569 configured in iptables, what
>> about the netgear router?
>>
>> Have you port forwarded 4569 there?
>>
>> Dave
>>
>> ___
>> --Bandwidth and Colocation sponsored by Easynews.com
>> --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>
>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
>
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang Borgon]

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang


--- David J Carter <[EMAIL PROTECTED]> wrote:

> Wolfgang wrote:  -
> 
> I've already sunk several hours into this without
> any
> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a
> softphone on XP ProSP2 to a Asterisk server on Linux
> FC4 over a LAN through a Netgear router. The server
> will then go out to a PSTN termination service.
> 
> Thus far, the PSTN termination connection works fine
> -- I've opened up 4569 with iptables, and forwarded
> 4569 to the server IP.  I am not, however, having
> any
> luck connecting the softphone to the server.
> 
> I can telnet, ftp, and http to the server, but not
> IAX2. Iaxping times out, registration by Idefisk and
> Firefly also times out.  
> 
> The server fails to see the client as well.  
> 
> Here's a portion of my iax.conf:
> 
> [client]
> type=friend
> username=client
> secret=**
> host=192.168.1.40
> context=clientcon
> 
> and extensions.conf:
> 
> [clientcon]
> exten => 2278,1,Dial(IAX2/client)
> 
> 
>
==
> You say you have 4569 configured in iptables, what
> about the netgear router?
> 
> Have you port forwarded 4569 there?
> 
> Dave
> 
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> --
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> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[David J Carter]

Wolfgang wrote:  -

I've already sunk several hours into this without any
real progress, so I'd really appreciate any help  My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN termination service.

Thus far, the PSTN termination connection works fine
-- I've opened up 4569 with iptables, and forwarded
4569 to the server IP.  I am not, however, having any
luck connecting the softphone to the server.

I can telnet, ftp, and http to the server, but not
IAX2. Iaxping times out, registration by Idefisk and
Firefly also times out.  

The server fails to see the client as well.  

Here's a portion of my iax.conf:

[client]
type=friend
username=client
secret=**
host=192.168.1.40
context=clientcon

and extensions.conf:

[clientcon]
exten => 2278,1,Dial(IAX2/client)


==
You say you have 4569 configured in iptables, what about the netgear router?

Have you port forwarded 4569 there?

Dave

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[Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang Borgon]

I've already sunk several hours into this without any
real progress, so I'd really appreciate any help  My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN termination service.

Thus far, the PSTN termination connection works fine
-- I've opened up 4569 with iptables, and forwarded
4569 to the server IP.  I am not, however, having any
luck connecting the softphone to the server.

I can telnet, ftp, and http to the server, but not
IAX2. Iaxping times out, registration by Idefisk and
Firefly also times out.  

The server fails to see the client as well.  

Here's a portion of my iax.conf:

[client]
type=friend
username=client
secret=**
host=192.168.1.40
context=clientcon

and extensions.conf:

[clientcon]
exten => 2278,1,Dial(IAX2/client)


Here's the output of 'iax show peers':

Name/UsernameHost Mask
Port  Status
voxee/#   66.246.246.52   (S)  255.255.255.255
 4569  Unmonitored
client/client  192.168.1.40 (S) 
255.255.255.255  0 Unmonitored
demo/asterisk216.207.245.47  (S)  255.255.255.255 
4569  Unmonitored

Note the port listed at 0.


Debug reponse to 'dial [EMAIL PROTECTED]':

-- Executing Dial("ALSA/default", "IAX2/client")
in new stack
-- Called client
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
   Timestamp: 00016ms  SCall: 1  DCall: 0
[192.168.1.40:0]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ilbc|ulaw|alaw|gsm)
   CALLING PRESNTN : 67
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   USERNAME: client
   FORMAT  : 2
   CAPABILITY  : 64526
   ADSICPE : 0
   DATE TIME   : 2005-10-10  00:04:14

Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
   Timestamp: 00016ms  SCall: 1  DCall: 0
[192.168.1.40:0]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ilbc|ulaw|alaw|gsm)
   CALLING PRESNTN : 67
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   USERNAME: client
   FORMAT  : 2
   CAPABILITY  : 64526
   ADSICPE : 0
   DATE TIME   : 2005-10-10  00:04:14

-- IAX2/client-1 is circuit-busy
Oct 10 00:04:19 NOTICE[3615]: chan_iax2.c:2754
auto_congest: Auto-congesting cal l due to slow
response
-- Hungup 'IAX2/client-1'
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'ALSA/default' status
is 'CONGESTION'

















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[Asterisk-Users] Help please

[Carlos Trujillo]

Hello, i`m carlos, i`m just begining to use Asterisk at Home, so i have learned to configure a several extensions, but now i have a FXO target and i wanna to connect to PSTN, but i dunno how to do. i`d like to receive support from you...thanks

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[Asterisk-Users] help please

[asterisk]

what i have at present is asterisk installed on 
fedora core 3, with a dev kit with 1 fxo and 1fxs module and a TE110p 
card,  using the sample files i can dial asterisk using the dev kit and get 
the asterisk welcome and congratulations message
i have got the zaptel.conf conigured as 
follows
fxoks=1fxsks=4
span=1,1,0,ccs,hdb3,crc4bchan=5-19,21-35dchan=20
loadzone=nldefaultzone=nl
the zapata.conf as follows
[channels]language=nlcontext=defaultusecallerid=yeshidecallerid=nocallwaiting=yescallwaitinguserid=yesthreewaycalling=yesechocancel=yesechocancelwhenbridged=norxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nocallerid=206388230busydetect=nocallprogress=nomusiconhold=default
signalling=fxo_kschannel => 1
signalling fxs_kschannel => 4
what i want to be able to do is this
have 5 incomming numbers over the pri(e1) lines and also dialing out over whichever 
line is available on the pri(e1) 
line.
i would like to know 
1/ how do you setup your zapata.conf file for 2 
cards??? as i have 1 dev kit and 1 te110p card
2/ how do i get asterisk to detect what number has 
ben dialled by the outside caller and route it to the appropriate extension, if 
no answer after 10 seconds ring all the other phones and if no answer after that 
go to voicemail? and allow other users to pick up someone elses phone from their 
extension.
user10031204161092 (full international 
number)0204161092 (number as dialled from the netherlands)4161092 
(number as dialled from amsterdam)
user20031204161091 (full international 
number)0204161091 (number as dialled from the netherlands)4161091 
(number as dialled from amsterdam)
user30031204161093 (full international 
number)0204161093 (number as dialled from the netherlands)4161091 
(number as dialled from amsterdam)
user40031204161094 (full international 
number)0204161094 (number as dialled from the netherlands)4161091 
(number as dialled from amsterdam)
3/ how do i record calls and set them to a file 
in the format  .mp3
I know its a lot to ask but once i get it up and 
running i intend on providing a step by step method of how i installed it from 
installing fdc3 to up and running so will help many others too so please 
help.
 
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RE: [Asterisk-Users] Help Please Multiple Users for Broadvoice

[Kerry Garrison]

http://geekgazette.com 
has a "how to" for Broadvoice
-Kerry
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of mr. 
barkerSent: Saturday, May 14, 2005 6:03 AMTo: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Help Please Multiple Users for Broadvoice


I would like to be able to have 
multiple users (the wife and kids) to be able to access the Broadvoice account 
at the same.  No complaining that way from them J.
 
I seen someones configuration in the 
group here but now I can’t find it (lost my glasses). If someone could post 
theirs’s or the shortcut that would be great.
 
Thanks for your 
help.
 
“Dad she’s on the phone again 
!”
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[Asterisk-Users] Help Please Multiple Users for Broadvoice

[mr. barker]

I would like to be able to have multiple users (the wife and
kids) to be able to access the Broadvoice account at the same.  No
complaining that way from them J.

 

I seen someones configuration in the group here but now I
can’t find it (lost my glasses). If someone could post theirs’s or
the shortcut that would be great.

 

Thanks for your help.

 

“Dad she’s on the phone again !”






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Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[Tim Burt]

I made the Vonage mistake too.  Cost me a hundred bucks!

They will hit you with a Disconnect Fee, if you don't return their
equipment in the original box.

Sux, if you ask me.

Vonage does offer an attractive flat rate for 500 minutes.

You could, purchase a digium card, and plug the Vonage POTS line into the
digium card, just like you would a POTS line.  Kinda a workaround, but I
tried it and it works.

Or you can buy the softfone addon, and pay more per month to Vonage and
settle for SIP...

Or better yet, try:

http://connect.voicepulse.com

You get free incoming calls.  A low monthly rate, and a cheap per minute
rate for outgoing.

Best part of all, is you get trunking..  4 simultaneous calls in and out. 
Which is what I think you said you were looking for.

> Check the list archive.  This thread just happened a couple days ago.
>
> A couple that I remember as supporting IAX:
>
> voipjet.com
> nufone.net
>
> You also might try opbx.com
>
> JD Austin wrote:
>> Im a newbie to this list (joined today).
>> Other than Broadvoice, what voip providers work well with Asterisk?
>> I'd like a service that will allow trunking so that I can have more than
>> one outbound/inbound call if possible.
>>
>> JD
>>
>> Kerry Garrison wrote:
>>
>>> You arent going to make this happen as you describe. Vonage is not a
>>> good
>>> service to use with Asterisk. To quote from the Wiki:
>>>
>>> Vonage service is locked to the ATA they send you. It is not possible
>>> to
>>> connect Asterisk (or any other SIP UA) directly to your main Vonage
>>> service.
>>> http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
>>>
>>> If you want to use Asterisk, you will need a different provider.
>>> -Kerry
>>>
>>>
>>>
>>> -Original Message-
>>> From: [EMAIL PROTECTED]
>>> [mailto:[EMAIL PROTECTED] On Behalf Of J W
>>> Sent: Tuesday, March 22, 2005 11:54 AM
>>> To: asterisk-users@lists.digium.com
>>> Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
>>>
>>> I just installed Asterisk on my server and I have Vonage softphone.
>>>
>>> I need my Asterisk server to receive calls through the Vonage
>>> Softphone DID
>>> and make outgoing calls through the Vonage ATA using an X100p to
>>> connect to
>>> it. Can someone help me out on configuring this? I really need this
>>> for my
>>> business and would greatly appreciate the help.
>>>
>>> _
>>> Express yourself instantly with MSN Messenger! Download today - it's
>>> FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
>>>
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Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[Kris Edwards]

Check the list archive.  This thread just happened a couple days ago.

A couple that I remember as supporting IAX:

voipjet.com
nufone.net

You also might try opbx.com

JD Austin wrote:
> Im a newbie to this list (joined today).
> Other than Broadvoice, what voip providers work well with Asterisk?
> I'd like a service that will allow trunking so that I can have more than
> one outbound/inbound call if possible.
> 
> JD
> 
> Kerry Garrison wrote:
> 
>> You arent going to make this happen as you describe. Vonage is not a good
>> service to use with Asterisk. To quote from the Wiki:
>>
>> Vonage service is locked to the ATA they send you. It is not possible to
>> connect Asterisk (or any other SIP UA) directly to your main Vonage
>> service.
>> http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
>>
>> If you want to use Asterisk, you will need a different provider.
>> -Kerry
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of J W
>> Sent: Tuesday, March 22, 2005 11:54 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
>>
>> I just installed Asterisk on my server and I have Vonage softphone.
>>
>> I need my Asterisk server to receive calls through the Vonage
>> Softphone DID
>> and make outgoing calls through the Vonage ATA using an X100p to
>> connect to
>> it. Can someone help me out on configuring this? I really need this
>> for my
>> business and would greatly appreciate the help.
>>
>> _
>> Express yourself instantly with MSN Messenger! Download today - it's
>> FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
>>
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RE: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[Kerry Garrison]

Good point, I forgot about using the softphone add-on account. 
-Kerry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Tuesday, March 22, 2005 1:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage



Kerry Garrison wrote:

>You arent going to make this happen as you describe. Vonage is not a good
service to use with Asterisk. To quote from the Wiki:
>
>Vonage service is locked to the ATA they send you. It is not possible to
connect Asterisk (or any other SIP UA) directly to your main Vonage service.
>http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
>  
>
IOf you would read his original posting, you will see he wants to connect
with Asterisk via his softphone account, and make outgoing calls through his
ATA and X100 card.
All should be doable.
Someone  posted a working config to the list within the past 10 days or so,
so a search through the recent archives should bring a working result.
The key here is the MAIN  Vonage number is locked to the ATA, but a
softphone add on account from Vonage should work.

John Novack


>If you want to use Asterisk, you will need a different provider.
>-Kerry
>
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of J W
>Sent: Tuesday, March 22, 2005 11:54 AM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
>
>I just installed Asterisk on my server and I have Vonage softphone.
>
>I need my Asterisk server to receive calls through the Vonage Softphone 
>DID and make outgoing calls through the Vonage ATA using an X100p to 
>connect to it. Can someone help me out on configuring this? I really 
>need this for my business and would greatly appreciate the help.
>
>_
>Express yourself instantly with MSN Messenger! Download today - it's FREE! 
>http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
>
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>
>
>
>  
>
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Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[JD Austin]

Im a newbie to this list (joined today).
Other than Broadvoice, what voip providers work well with Asterisk?
I'd like a service that will allow trunking so that I can have more than 
one outbound/inbound call if possible.

JD
Kerry Garrison wrote:
You arent going to make this happen as you describe. Vonage is not a good
service to use with Asterisk. To quote from the Wiki:
Vonage service is locked to the ATA they send you. It is not possible to
connect Asterisk (or any other SIP UA) directly to your main Vonage service.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
If you want to use Asterisk, you will need a different provider.
-Kerry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J W
Sent: Tuesday, March 22, 2005 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I just installed Asterisk on my server and I have Vonage softphone.
I need my Asterisk server to receive calls through the Vonage Softphone DID
and make outgoing calls through the Vonage ATA using an X100p to connect to
it. Can someone help me out on configuring this? I really need this for my
business and would greatly appreciate the help.
_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[Russell Handorf]

I was only able to get the softphone account to make inbound calls on 
one sip.conf config, or outbound calls on another sip.conf config. I 
didnt investigate the issue completely, but from what I could tell, they 
wouldnt allow multiple SIP sessions from the same IP address. I didnt 
try running 2 asterisk instances (one for in and one for out) that would 
talk to each other, however I would believe that Vonage does account 
state/session checks.

Net2Phone now does authentication on 3 credentials; account number, pin, 
and MAC address.

Kerry Garrison wrote:
You arent going to make this happen as you describe. Vonage is not a good
service to use with Asterisk. To quote from the Wiki:
Vonage service is locked to the ATA they send you. It is not possible to
connect Asterisk (or any other SIP UA) directly to your main Vonage service.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
If you want to use Asterisk, you will need a different provider.
-Kerry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J W
Sent: Tuesday, March 22, 2005 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I just installed Asterisk on my server and I have Vonage softphone.
I need my Asterisk server to receive calls through the Vonage Softphone DID
and make outgoing calls through the Vonage ATA using an X100p to connect to
it. Can someone help me out on configuring this? I really need this for my
business and would greatly appreciate the help.
_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[John Novack]

Kerry Garrison wrote:
You arent going to make this happen as you describe. Vonage is not a good 
service to use with Asterisk. To quote from the Wiki:
Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage
 

IOf you would read his original posting, you will see he wants to 
connect with Asterisk via his softphone account, and make outgoing calls 
through his ATA and X100 card.
All should be doable.
Someone  posted a working config to the list within the past 10 days or 
so, so a search through the recent archives should bring a working result.
The key here is the MAIN  Vonage number is locked to the ATA, but a 
softphone add on account from Vonage should work.

John Novack

If you want to use Asterisk, you will need a different provider.
-Kerry

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J W
Sent: Tuesday, March 22, 2005 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I just installed Asterisk on my server and I have Vonage softphone.
I need my Asterisk server to receive calls through the Vonage Softphone DID
and make outgoing calls through the Vonage ATA using an X100p to connect to
it. Can someone help me out on configuring this? I really need this for my
business and would greatly appreciate the help.
_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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RE: [Asterisk-Users] Help please for newb on Asterisk to Vonage

[Kerry Garrison]

You arent going to make this happen as you describe. Vonage is not a good
service to use with Asterisk. To quote from the Wiki:

Vonage service is locked to the ATA they send you. It is not possible to
connect Asterisk (or any other SIP UA) directly to your main Vonage service.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage

If you want to use Asterisk, you will need a different provider.
-Kerry



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J W
Sent: Tuesday, March 22, 2005 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage

I just installed Asterisk on my server and I have Vonage softphone.

I need my Asterisk server to receive calls through the Vonage Softphone DID
and make outgoing calls through the Vonage ATA using an X100p to connect to
it. Can someone help me out on configuring this? I really need this for my
business and would greatly appreciate the help.

_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Asterisk-Users] Help please for newb on Asterisk to Vonage

[J W]

I just installed Asterisk on my server and I have Vonage softphone.
I need my Asterisk server to receive calls through the Vonage Softphone DID 
and make outgoing calls through the Vonage ATA using an X100p to connect to 
it. Can someone help me out on configuring this? I really need this for my 
business and would greatly appreciate the help.

_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [Asterisk-Users] Help Please!!!!

[Erick Weber V.]

Thanks, I will begin my testing
Erick
- Original Message - 
From: "Race Vanderdecken" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Wednesday, February 16, 2005 8:18 PM
Subject: RE: [Asterisk-Users] Help Please


Greetings Mr. Weber,
Remember the rule in mathematics that is much easier to solve for one
variable.
You stateed you are having a problem with the 1088 extension. If look
like you are trying to make a call from the 404 extension to the 1088
extension.
1.
If you have 6 ATA's running shut 5 of them off.
Test each one separately.
Then turn one on at a time and see the problem can be traced to one ATA
2.
You are getting sent an authorization request from asterisk to the 1088
extension.
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Make sure you don't have any of the secret= or the md5secret= stuff set
in the sip.conf, until you can get each phone to talk in the open.
Then change, one, 1, uno, phone at a time.
3.
If you have a SIP phone that is not an ATA then set it up and try to
dial the 1088 and see if you get the same thing.
4.
Do a sip show users to make sure the 1088 is registered with asterisk.
5. Do the normal, things don't work dance, by unplugging the phone and
reconnecting a different phone to the ata. Change the power suplly with
another ata. Change the RJ45 patch cable. Try a different port in the
switch or wall. Swap one of the known working ATA and change it to the
1088 ata.
6.
Go to lunch and have a beer. Find a new job and settle down with a good
woman. Leave telecom and go into organic farming.
Race "The Tyrant" Vanderdecken
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Weber V.
Sent: Wednesday, February 16, 2005 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help Please
Importance: High
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem
is
that one of them is dropping calls an I can't figure out what is the
problem; I had made a SIP DEBUG PEER 1088 that is the peer with the
problem.
Any help will be appreciate
Thanks
Erick Weber
VoIP*CLI> sip debug peer 1088
SIP Debugging Enabled for IP: 201.133.170.82:5060
Peer RTP is at port 192.168.1.69:0
Peer RTP is at port 192.168.1.69:0
   -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack
We're at XXX.XXX.XXX.XXX port 17506
Answering/Requesting with root capability 256
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport
From: "Weber Automundo" ;tag=as4da46cda
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164
v=0
o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 17506 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
(NAT) to 201.133.170.82:5060
   -- Called 1088
   -- SIP/1088-ec82 is ringing
Found RTP audio format 18
Found RTP audio format 101
Peer RTP is at port 192.168.1.2:0
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100(G729A), peer -
audio=0x100(G729A)/video=0x0(EMPTY),
combined - 0x100(G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
list_route: hop: 
set_destination: Parsing
 for address/port to
send to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
From: "Weber Automundo" ;tag=as4da46cda
To: ;tag=939809556
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 201.133.170.82:5060
   -- SIP/1088-ec82 answered SIP/404-cbc9
   -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
   -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
Using latest request as basis request
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Content-Length: 0
to 201.133.170.82:5060
Scheduling destruction of call '[

RE: [Asterisk-Users] Help Please!!!!

[Race Vanderdecken]

Greetings Mr. Weber,

Remember the rule in mathematics that is much easier to solve for one
variable.

You stateed you are having a problem with the 1088 extension. If look
like you are trying to make a call from the 404 extension to the 1088
extension.

1.
If you have 6 ATA's running shut 5 of them off.
Test each one separately.
Then turn one on at a time and see the problem can be traced to one ATA

2.
You are getting sent an authorization request from asterisk to the 1088
extension.

WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"

Make sure you don't have any of the secret= or the md5secret= stuff set
in the sip.conf, until you can get each phone to talk in the open. 
Then change, one, 1, uno, phone at a time.

3. 
If you have a SIP phone that is not an ATA then set it up and try to
dial the 1088 and see if you get the same thing.

4. 
Do a sip show users to make sure the 1088 is registered with asterisk.

5. Do the normal, things don't work dance, by unplugging the phone and
reconnecting a different phone to the ata. Change the power suplly with
another ata. Change the RJ45 patch cable. Try a different port in the
switch or wall. Swap one of the known working ATA and change it to the
1088 ata.

6. 
Go to lunch and have a beer. Find a new job and settle down with a good
woman. Leave telecom and go into organic farming.

Race "The Tyrant" Vanderdecken
[EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Weber V.
Sent: Wednesday, February 16, 2005 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help Please
Importance: High

I have a asterisk server with 6 Cisco ATA connected in SIP. My problem
is 
that one of them is dropping calls an I can't figure out what is the 
problem; I had made a SIP DEBUG PEER 1088 that is the peer with the
problem.

Any help will be appreciate

Thanks

Erick Weber


VoIP*CLI> sip debug peer 1088
SIP Debugging Enabled for IP: 201.133.170.82:5060
Peer RTP is at port 192.168.1.69:0
Peer RTP is at port 192.168.1.69:0
-- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack
We're at XXX.XXX.XXX.XXX port 17506
Answering/Requesting with root capability 256
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport
From: "Weber Automundo" ;tag=as4da46cda
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 17506 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
 (NAT) to 201.133.170.82:5060
-- Called 1088
-- SIP/1088-ec82 is ringing
Found RTP audio format 18
Found RTP audio format 101
Peer RTP is at port 192.168.1.2:0
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100(G729A), peer -
audio=0x100(G729A)/video=0x0(EMPTY), 
combined - 0x100(G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
list_route: hop: 
set_destination: Parsing 
 for address/port to

send to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
From: "Weber Automundo" ;tag=as4da46cda
To: ;tag=939809556
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 201.133.170.82:5060
-- SIP/1088-ec82 answered SIP/404-cbc9
-- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
-- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
Using latest request as basis request
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Content-Length: 0


 to 201.133.170.82:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Using latest request as basis request
Sending to 192.168.1.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-I

[Asterisk-Users] Help Please!!!!

[Erick Weber V.]

I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is 
that one of them is dropping calls an I can't figure out what is the 
problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem.

Any help will be appreciate
Thanks
Erick Weber
VoIP*CLI> sip debug peer 1088
SIP Debugging Enabled for IP: 201.133.170.82:5060
Peer RTP is at port 192.168.1.69:0
Peer RTP is at port 192.168.1.69:0
   -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack
We're at XXX.XXX.XXX.XXX port 17506
Answering/Requesting with root capability 256
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport
From: "Weber Automundo" ;tag=as4da46cda
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164
v=0
o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 17506 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
(NAT) to 201.133.170.82:5060
   -- Called 1088
   -- SIP/1088-ec82 is ringing
Found RTP audio format 18
Found RTP audio format 101
Peer RTP is at port 192.168.1.2:0
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), 
combined - 0x100(G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
list_route: hop: 
set_destination: Parsing 
 for address/port to 
send to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
From: "Weber Automundo" ;tag=as4da46cda
To: ;tag=939809556
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(NAT) to 201.133.170.82:5060
   -- SIP/1088-ec82 answered SIP/404-cbc9
   -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
   -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
Using latest request as basis request
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Content-Length: 0
to 201.133.170.82:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Using latest request as basis request
Sending to 192.168.1.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: ;tag=3858230914
To: ;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: ;expires=120
Date: Wed, 16 Feb 2005 00:43:46 GMT
Content-Length: 0
to 201.133.170.82:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:201.133.170.82 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7
From: "asterisk" ;tag=as59adf4c2
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 201.133.170.82:5060
Destroying call '[EMAIL PROTECTED]'
set_destination: Parsing 
 for address/port to 
send to
set_destination: set destination to 192.168.1.2, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2bdff4fa;rport
From: "Weber Automundo" ;tag=as4da46cda
To: ;tag=939809556
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

(NAT) to 201.133.170.82:5060
 == Spawn extension (hi, 1088, 1) exited non-zero on 'SIP/404-cbc9'
   -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 
192.168.1.2
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:201.133.170.82 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9

Re: [Asterisk-Users] Help please

[Walker Haddock]

On Wed, Nov 19, 2003 at 09:52:13AM -0700, Michael Welter wrote:
> Hi,
> 
> My PBX/key system failed when building power was switched off and then 
> back on.  The service rep says the PBX unit is so old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline sets.  There are 
I have a client running 9 incomming PRI lines with their 768K internet all on one T-1. 
 Works very well.  I have another client with 4 pots lines.  Works OK, but you have 
some problems with call detection.

> voice mailboxes on each line.
Voice mail is excellent with Asterisk.  Very flexible.

> 
> Switching to Qwest's Centrex system would cost about $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex increment would be 
> about $480/yr.
With Asterisk you would be looking at an initial fixed price investment and about 20% 
of that per year for support.  The cost for pots lines vs. T1 for the PBX would be 
about the same.  You would need to check with LECs and CLECs to determine the most 
economical way to deliver the PSTN services to your office.
> 
> Purchasing a new PBX would run anywhere from $4K to $9K.
> 
> My questions are:  should I be considering Asterisk?  What type of 
> telephone set could I use with Asterisk?  Would I be able to conference 
> outside parties into a call?  Would I have voice mail?
Absolutely consider Asterisk.
You can use standard POTs style phones, some ADSI phones and SIP phones.  The Cisco 
7960 is very nice.

You can set up Meetme conferences and you can conference many incoming calls to the 
same connection.  If you have VoIP channels you could conference them in as well.  The 
client I mentioned above using the T1 PRI does depositions on the conferences.  We've 
had conferences up for 7 hours!

Yes, Voicemail is very good and flexible.
> 
Walker
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
***
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Re: [Asterisk-Users] Help please

[Steven Critchfield]

On Wed, 2003-11-19 at 10:52, Michael Welter wrote:
> Hi,
> 
> My PBX/key system failed when building power was switched off and then 
> back on.  The service rep says the PBX unit is so old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline sets.  There are 
> voice mailboxes on each line.
> 
> Switching to Qwest's Centrex system would cost about $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex increment would be 
> about $480/yr.
> 
> Purchasing a new PBX would run anywhere from $4K to $9K.
> 
> My questions are:  should I be considering Asterisk?  What type of 
> telephone set could I use with Asterisk?  Would I be able to conference 
> outside parties into a call?  Would I have voice mail?

Sorry for the last empty message, don't know if I was able to delete it
from my outbox quick enough.

Anyways, we discussed pricing recently. Your price for lines is pretty
nice. You probably couldn't justify the extra cost of switching to T1 or
PRI as the loop itself would cost a little more than you currently are
paying. 

So now you have to figure out how to deal with these lines. You will
have to decide whether or not you want IP phones, or analog phones.
Either way, you probably are looking at a channel bank and a T1 card to
get the 8 lines into a single asterisk machine.

8x12 would fit on a Zhone, but you have to know that the zhone doesn't
pass callerid on the FXO ports. You could go with just about any modular
channel bank that supports FXO ports. Look at the Adtrans or Adit lines.

So for an analog system you are looking at
$500 PC
$500 T100P card
$500 or so for a channel bank of of ebay.
$30x12 for phones.

or approximately $1860 if you get lucky on ebay.

On a purely money view point, if you don't for see growth in the number
of lines into your office, you won't see the break even of asterisk for
about 4 years over the Centrex lines. Of course you can assign value to
having your own machine in house, and the VoIP possibilities, and you
can claim break even in a shorter term.

Don't take this as downing asterisk, I love it. I just wanted to cut to
the dollars since you have placed them on the table.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Help please

[Steven Critchfield]

On Wed, 2003-11-19 at 10:52, Michael Welter wrote:
> Hi,
> 
> My PBX/key system failed when building power was switched off and then 
> back on.  The service rep says the PBX unit is so old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline sets.  There are 
> voice mailboxes on each line.
> 
> Switching to Qwest's Centrex system would cost about $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex increment would be 
> about $480/yr.
> 
> Purchasing a new PBX would run anywhere from $4K to $9K.
> 
> My questions are:  should I be considering Asterisk?  What type of 
> telephone set could I use with Asterisk?  Would I be able to conference 
> outside parties into a call?  Would I have voice mail?
> 
> Thanks for your help,
> Michael Welter
> 
> 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Help please

[C M]

in my opinion, asterisk is the best solution. it is
relatively very cheap, asterisk is free and u can use
any soft phones to do calls. you can find cheap digium
cards, around $400+. you can set up any normal phone
to FXS cards. yu can do conference call and voicemails
in asterisk.

this link may be helpful to read though there are lots
of it out there
http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb

--- Michael Welter <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> My PBX/key system failed when building power was
> switched off and then 
> back on.  The service rep says the PBX unit is so
> old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline
> sets.  There are 
> voice mailboxes on each line.
> 
> Switching to Qwest's Centrex system would cost about
> $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex
> increment would be 
> about $480/yr.
> 
> Purchasing a new PBX would run anywhere from $4K to
> $9K.
> 
> My questions are:  should I be considering Asterisk?
>  What type of 
> telephone set could I use with Asterisk?  Would I be
> able to conference 
> outside parties into a call?  Would I have voice
> mail?
> 
> Thanks for your help,
> Michael Welter
> 
> 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
http://lists.digium.com/mailman/listinfo/asterisk-users


=
Designs

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Re: [Asterisk-Users] Help please

[C M]

in my opinion, asterisk is the best solution. it is
relatively very cheap, asterisk is free and u can use
any soft phones to do calls. you can find cheap digium
cards, around $400+. you can set up any normal phone
to FXS cards. yu can do conference call and voicemails
in asterisk.

this link may be helpful to read though there are lots
of it out there
http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb

--- Michael Welter <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> My PBX/key system failed when building power was
> switched off and then 
> back on.  The service rep says the PBX unit is so
> old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline
> sets.  There are 
> voice mailboxes on each line.
> 
> Switching to Qwest's Centrex system would cost about
> $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex
> increment would be 
> about $480/yr.
> 
> Purchasing a new PBX would run anywhere from $4K to
> $9K.
> 
> My questions are:  should I be considering Asterisk?
>  What type of 
> telephone set could I use with Asterisk?  Would I be
> able to conference 
> outside parties into a call?  Would I have voice
> mail?
> 
> Thanks for your help,
> Michael Welter
> 
> 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
http://lists.digium.com/mailman/listinfo/asterisk-users


=
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Protect your identity with Yahoo! Mail AddressGuard
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___
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Re: [Asterisk-Users] Help please

[C M]

in my opinion, asterisk is the best solution. it is
relatively very cheap, asterisk is free and u can use
any soft phones to do calls. you can find cheap digium
cards, around $400+. you can set up any normal phone
to FXS cards. yu can do conference call and voicemails
in asterisk.

this link may be helpful to read though there are lots
of it out there
http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb

--- Michael Welter <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> My PBX/key system failed when building power was
> switched off and then 
> back on.  The service rep says the PBX unit is so
> old that it is not 
> repairable.
> 
> The unit has 8 incoming POTS lines and 12 multiline
> sets.  There are 
> voice mailboxes on each line.
> 
> Switching to Qwest's Centrex system would cost about
> $3800/yr.  My 8 
> business lines now cost $3320/yr, so the Centrex
> increment would be 
> about $480/yr.
> 
> Purchasing a new PBX would run anywhere from $4K to
> $9K.
> 
> My questions are:  should I be considering Asterisk?
>  What type of 
> telephone set could I use with Asterisk?  Would I be
> able to conference 
> outside parties into a call?  Would I have voice
> mail?
> 
> Thanks for your help,
> Michael Welter
> 
> 
> 
> 
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[Asterisk-Users] Help please

[Michael Welter]

Hi,

My PBX/key system failed when building power was switched off and then 
back on.  The service rep says the PBX unit is so old that it is not 
repairable.

The unit has 8 incoming POTS lines and 12 multiline sets.  There are 
voice mailboxes on each line.

Switching to Qwest's Centrex system would cost about $3800/yr.  My 8 
business lines now cost $3320/yr, so the Centrex increment would be 
about $480/yr.

Purchasing a new PBX would run anywhere from $4K to $9K.

My questions are:  should I be considering Asterisk?  What type of 
telephone set could I use with Asterisk?  Would I be able to conference 
outside parties into a call?  Would I have voice mail?

Thanks for your help,
Michael Welter


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Re: [Asterisk-Users] help please with single t1 configuration

[Brian West]

did you happen to run ztcfg after you setup the configs?

On Sat, 9 Aug 2003, Barry Porch wrote:

> I am attempting to set up an Asterisk box which I am only concerned with
> getting a single T1 working.  I have this T1 connected to my PBX and I
> am looking at using Asterisk as a conference bridge.
>
> Here is my zaptel.conf:
>
> span=1,1,0,d4,ami
> e&m=1-24
> loadzone=us
> defaultzone=us
>
> Here is my Zapata.conf:
>
> [channels]
> language=en
> context=default
> signalling=em_w
> group=1
> channel=1-24
>
> As you can see I am trying to set this up as a D4/AMI E&M T1.  I also
> have "modprobe zaptel" and "modprobe wct1xxp" in my rc.local.
>
> When I try to run Asterisk it complains that it cannot access channel 1
> and dies.
>
> I can run zttool which "sees" the T100 card that I have installed.
> However, when I run "ztcfg -" it sells me that 0 channels are
> configured.
>
> What am I missing here?
>
> Thanks!
>
> Barry
>
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RE: [Asterisk-Users] help please with single t1 configuration

[Adams, Gavin]

> -Original Message-
> From: Barry Porch [mailto:[EMAIL PROTECTED]
> 
> Yes, I run ztcfg -vvv and it tells me that there are 0 channels.

Barry, are you modifying the zaptel.conf file in /etc or /etc/asterisk.
If the latter, try ztcfg -vvv -c /etc/asterisk/zaptel.conf. I had a
similar issue last week when first starting with Asterisk.

Regards,

--- Gavin
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RE: [Asterisk-Users] help please with single t1 configuration

[Barry Porch]

Yes, I run ztcfg -vvv and it tells me that there are 0 channels.

-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED] 
Sent: Saturday, August 09, 2003 6:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] help please with single t1 configuration

did you happen to run ztcfg after you setup the configs?

On Sat, 9 Aug 2003, Barry Porch wrote:

> I am attempting to set up an Asterisk box which I am only concerned
with
> getting a single T1 working.  I have this T1 connected to my PBX and I
> am looking at using Asterisk as a conference bridge.
>
> Here is my zaptel.conf:
>
> span=1,1,0,d4,ami
> e&m=1-24
> loadzone=us
> defaultzone=us
>
> Here is my Zapata.conf:
>
> [channels]
> language=en
> context=default
> signalling=em_w
> group=1
> channel=1-24
>
> As you can see I am trying to set this up as a D4/AMI E&M T1.  I also
> have "modprobe zaptel" and "modprobe wct1xxp" in my rc.local.
>
> When I try to run Asterisk it complains that it cannot access channel
1
> and dies.
>
> I can run zttool which "sees" the T100 card that I have installed.
> However, when I run "ztcfg -" it sells me that 0 channels are
> configured.
>
> What am I missing here?
>
> Thanks!
>
> Barry
>
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[Asterisk-Users] help please with single t1 configuration

[Barry Porch]

I am attempting to set up an Asterisk box which I am only
concerned with getting a single T1 working.  I have this T1 connected to my PBX and I
am looking at using Asterisk as a conference bridge.

 

Here is my zaptel.conf:

 

span=1,1,0,d4,ami

e&m=1-24

loadzone=us

defaultzone=us

 

Here is my Zapata.conf:

 

[channels]

language=en

context=default

signalling=em_w

group=1

channel=1-24

 

As you can see I am trying to set this up as a D4/AMI
E&M T1.  I also have “modprobe zaptel” and “modprobe wct1xxp” in my rc.local.

 

When I try to run Asterisk it complains that it cannot
access channel 1 and dies.

 

I can run zttool which “sees”
the T100 card that I have installed. 
However, when I run “ztcfg –” it sells me that 0 channels are configured.

 

What am I missing here?

 

Thanks!

 

Barry