[asterisk-users] users.conf in 1.4

[clive.chan\(atn\)]

Hi all, 

Can some one tell me how actually the users.conf working with SIP.conf,
IAX.conf, ZAPATA.conf ? 

How can I add the SIP user to this file and do I relate this user profile
(in the users.conf) with the sip.conf?

I am a bit confused with this:-(

Can some one help??

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Re: [asterisk-users] Teliax Quality of Service

[Brian Capouch]

Stephen Bosch wrote:

> 
> PSTN service still sets the standard.
> 

With infrastructure paid for under a gracious guaranteed-profit monopoly 
by ratepayers, now being used as a weapon to stifle competition from 
VoIP, cable, and other emerging technologies.

b.

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Re: [asterisk-users] Teliax Quality of Service

[Mark Coccimiglio]

Steve Totaro wrote:

>What if a train derails and slices through the main fiber connections.  
>OK, so you have XO, Global Crossing, Verizon, and UCN all for 
>redundancy.  Well guess what?  They are all most likely running over 
>those strands of fiber.  You better have a VSAT connection too!
>  
>
That's why I lease a few servers in a data center on other side of the 
country.  Setup in a "hot stand-by" state.  Its that peace-of-mind you 
can't buy any way else.  So it costs a few hundred dollars a month 
(actually less then $500).  It  kicks in to take-up capacity when my 
main servers gets "real busy" or go off-line for maintenance.  Its 
instant and automatic.  Ok sure it took a lot of planning to get it 
right, but that's what I get paid to do. 

Single point of failure should NEVER completely disable your company.  
Yes outages happen and backhoe's cut fibre all the time.  From within 
this stuff can make one's life rather difficult, but from the outside it 
should be almost unnoticed. When was the last time you noticed an outage 
at Google, Microsoft or the DoD?  Do you think they don't happen? 

Its not that difficult or all that expensive if planned and implemented 
properly. 

Mark C.

-- 
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realize what a foul beast humanity really is.


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[asterisk-users] ISDN30 card for UK : sanity check

[Rory Campbell-Lange]

We will be connecting our Asterisk server to ISDN 30 and intend using
the Sangoma A101 card. The install location is in London (UK).

Sangoma card at Voipon
http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA

I would be grateful to hear if this is the right choice of card. Usage
reports would be helpful.

Regards
Rory

-- 
Rory Campbell-Lange
Campbell-Lange Workshop Ltd.
<[EMAIL PROTECTED]>


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Re: [asterisk-users] ATA phones ring when they register

[Mr Shunz]

On 8/7/07, Vieri <[EMAIL PROTECTED]> wrote:
>
> --- Mr Shunz <[EMAIL PROTECTED]> wrote:
>
> > "Caller ID Scheme" as
> >
> > ETSI-FSK Prior to Ringing with DTAS...
>
> Thank you Daniele.
>
> That seems to work.
> I tested it on analog phones without a display.

Yeah, we had them without display too ...
But, if i recall correctly, we have a customer who used
to have a philips cordless and could see the correct CID.

> I had previously experimented with different schemes
> because I needed some of our phones  to correctly
> display the caller's ID. I noticed that I needed two
> different caller id schemes for two different types of
> analog phones w/display. Since one can only define two
> profiles in the Grandstream device, I'm crossing my
> fingers and hoping that I can get both the caller id
> on screen and the no-ring-when-register feature.

I must admit it was a real pain at first configuring it ...
it's mostly trial and error ;)

>
> Thanks for your help.
>

you're welcome


-- 
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[EMAIL PROTECTED]  ..o
Linux User #415108   ooo

8N1:
8 bit di dati,
1 bit di stop,
Nessuna Pieta`

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[asterisk-users] test the email-list

[zhu lizhong]

test only. good luck!
james.zhu
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Re: [asterisk-users] Free sitting

[Olivier]

Gordon,

What you described is exactly Follow-me feature : users are always logged
and can be reached somewhere.

By the way, do you introduce special settings so that ringing tones are
different ?
Let me explain this :

If Alice dials its extension and PIN code using Bob's hardphones, Bob and
Alice can both be called with the same phone.
Is it possible to have different ringing for Alice and Bob's incoming calls
?
Maybe an SDP option inside INVITE SIP message would do the trick ?
Maybe hardphone settings would read INVITE fields (Contact info ?) to
segregate calls ?
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Re: [asterisk-users] Free sitting

[Gordon Henderson]

On Tue, 7 Aug 2007, Olivier wrote:

> Gordon,
>
> What you described is exactly Follow-me feature : users are always logged
> and can be reached somewhere.

I've heard of some variants of this feature - that's the beauty (and 
down-side!) of a programmable system - it's open to different people's 
interpretations... (And why I think some of these features shouldn't be 
hard-coded into the system when they are implementable in the dialplan or 
AGI)

> By the way, do you introduce special settings so that ringing tones are
> different ?
> Let me explain this :
>
> If Alice dials its extension and PIN code using Bob's hardphones, Bob and
> Alice can both be called with the same phone.
> Is it possible to have different ringing for Alice and Bob's incoming calls
> ?

The simple answer is "I don't know"..

> Maybe an SDP option inside INVITE SIP message would do the trick ?
> Maybe hardphone settings would read INVITE fields (Contact info ?) to
> segregate calls ?

A simple way might be to change the caller-id on follow-me calls - change 
the name part into the number and change the number into a special number 
that the phone recognises as a separate ring-tone, but you lose 
information here, and need a phone that can display both name and number 
at the same time, and connect numbers to different ring-tones - then you 
end up going down the route of requiring a certian phone for a certian 
service - which might be acceptable to some people, but defeats the whole 
generic "any SIP phone will do" type ideas.

I have experimented with sending text message to phones (for other 
purposes - eg. the print the speed-dial numbers to the display when they 
get set), but again, different phones handle this differently, and some 
you need to push a key-sequence to get the message, by which time it might 
be too late!

Gordon

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Re: [asterisk-users] CDR/MySQL basic config

[Adrian Marsh]

Hmm.. This is what I get:

[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql> use asteriskcdrdb ;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Database changed
mysql> select Host from user where User = 'asteriskcdruser' ;
ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
mysql>


Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: 07 August 2007 02:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR/MySQL basic config

Adrian,

What host/ip did you specify when you created the user?

#> mysql --user=root --password

#mysql> use mysql;

#mysql> select Host from user where User = 'asteriskcdruser'
(this line is case sensitive)

Does it return 127.0.0.1 or localhost.  Make cdr_mysql reflect that.

You should also check out cdr_odbc, asterisk can connect through an
ODBC connection which in turn is a connection to the MySQL database.
There seems to be more suport for the ODBC driver.

Hope this helps some



On 8/6/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install.  The
> add-ons pack has been installed for a while, so now I'm trying to add
> the Mysql config.
>
> I've created a mysql database, added the grants for a user acces, and
> can run a mysql -u asteriskcdruser -p and can connect to the database.
>
> I've been using this as a guide:
>
http://www.757.org/~joat/wiki/index.php/Asterisk#Viewing_CDR_Data_with_A
> sterisk:_CDR_Analyzer
>
> I've created cdr_mysql.conf:
>
> [global]
> hostname=localhost
> dbname=asteriskcdrdb
> table=cdr
> password=password
> user=asteriskcdruser
> port=3306
> sock=/tmp/mysql.sock
> userfield=1
>
> But when I start asterisk (1.4 on my test machine), I get:
>
>   == Parsing '/etc/asterisk/cdr_mysql.conf': Found
> [Aug  6 21:01:14] ERROR[32512]: cdr_addon_mysql.c:436 my_load_module:
> Failed to connect to mysql database asteriskcdrdb on localhost.
> cdr_addon_mysql.so => (MySQL CDR Backend)
> [Aug  6 21:01:14] ERROR[32512]: res_config_mysql.c:627
mysql_reconnect:
> MySQL RealTime: Failed to connect database server  on  (err 2002).
Check
> debug for more info.
> [Aug  6 21:01:14] WARNING[32512]: res_config_mysql.c:474 load_module:
> MySQL RealTime: Couldn't establish connection. Check debug.
> [Aug  6 21:01:14] NOTICE[32512]: config.c:1171
> ast_config_engine_register: Registered Config Engine mysql
> MySQL RealTime driver loaded.
> res_config_mysql.so => (MySQL RealTime Configuration Driver)
>
>
> I'm also looking as to what CDR viewers there are available, and which
> people think are best.  I want to view/report on the calls made within
> A*k.
>
> Thanks,
>
> Adrian
>
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-- 
***
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IAXTEL: 17002871718
[EMAIL PROTECTED]
http://www.shift8.biz

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Re: [asterisk-users] Free sitting

[Olivier]

2007/8/7, Gordon Henderson <[EMAIL PROTECTED]>:
>
> On Tue, 7 Aug 2007, Olivier wrote:
>
> > Gordon,
> >
> > What you described is exactly Follow-me feature : users are always
> logged
> > and can be reached somewhere.
>
> I've heard of some variants of this feature - that's the beauty (and
> down-side!) of a programmable system - it's open to different people's
> interpretations... (And why I think some of these features shouldn't be
> hard-coded into the system when they are implementable in the dialplan or
> AGI)
>
> > By the way, do you introduce special settings so that ringing tones are
> > different ?
> > Let me explain this :
> >
> > If Alice dials its extension and PIN code using Bob's hardphones, Bob
> and
> > Alice can both be called with the same phone.
> > Is it possible to have different ringing for Alice and Bob's incoming
> calls
> > ?
>
> The simple answer is "I don't know"..
>
> > Maybe an SDP option inside INVITE SIP message would do the trick ?
> > Maybe hardphone settings would read INVITE fields (Contact info ?) to
> > segregate calls ?
>
> A simple way might be to change the caller-id on follow-me calls - change
> the name part into the number and change the number into a special number
> that the phone recognises as a separate ring-tone, but you lose
> information here, and need a phone that can display both name and number
> at the same time, and connect numbers to different ring-tones - then you
> end up going down the route of requiring a certian phone for a certian
> service - which might be acceptable to some people, but defeats the whole
> generic "any SIP phone will do" type ideas.


Could you elaborate ?

I know some hardphone (eg Thomson ST2030) can set ring-tone according
Caller's presence inside phone's directory.

In this case,  Asterisk would have to :
- fake original caller-name and set it to "call for Alice",
- replace original caller-id with Alice extension (eg 4111 instead of +44
812 41 54 66)
so that  hardphone  gets  everything it needs to :
- recognize from caller-id that the calls comes from Alice (though it's a
call FOR Alice)
- and then uses Alice ringing tone instead of Bob's tone.

Is this roughly correct ?

How many phones behaves like this ?
As you said, it would be sad to loose SIP portability.

It would be nice to use SIP protocol to drive such behaviour.


I have experimented with sending text message to phones (for other
> purposes - eg. the print the speed-dial numbers to the display when they
> get set), but again, different phones handle this differently, and some
> you need to push a key-sequence to get the message, by which time it might
> be too late!
>
> Gordon
>
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Re: [asterisk-users] Free sitting

[Steve Langstaff]

Have a look at your SIP phones' support for the Alert-Info header (and
Asterisk's support for it, come to that).
 


I know some hardphone (eg Thomson ST2030) can set ring-tone
according Caller's presence inside phone's directory.

In this case,  Asterisk would have to :
- fake original caller-name and set it to "call for Alice", 
- replace original caller-id with Alice extension (eg 4111
instead of +44  812 41 54 66)
so that  hardphone  gets  everything it needs to :
- recognize from caller-id that the calls comes from Alice
(though it's a call FOR Alice) 
- and then uses Alice ringing tone instead of Bob's tone.

Is this roughly correct ?

How many phones behaves like this ?
As you said, it would be sad to loose SIP portability.

It would be nice to use SIP protocol to drive such behaviour. 



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[asterisk-users] .call file and logging

[Vieri]

I am writing a cron script to check if certain
extensions are online and if they aren't then Asterisk
creates a couple of .call files to notify another set
of extensions or external numbers.

It works fine except for logging information.

What I'm doing in the script is setting a "fake"
caller ID (as it's generated by Asterisk, not by a
user) and calling out real users.

So the user's extension is specified in the "Channel:
" field. When the user picks the phone up, asterisk
drops into the custom_NOTIFY context which plays a
menu.

My problem is that when I check the logs in /var/log
or in the MySQL CDR database, I can't always
demonstrate that Asterisk actually called a specific
number (in the code below, the number I need to log is
$alerts).

If I use a SIP extension in the Channel field then the
logging works for me because I can see that the
SIP/EXTEN was used (see below).

However, if I use a Zap extension then only the Zap
channel number is logged but the extension's number
isn't (in the example below, 7022 does not appear in
the logs).

Any suggestions as to how I can solve this?
Maybe by changing the "Extension:" line or setting
variables. A quick simple example would be
appreciated.

Thanks,

Vieri

Code snippet:

$ftime = time();
$fname = "/tmp/asterisk_".$ftime.".call";
$fname_call =
"/var/spool/asterisk/outgoing/asterisk_".$ftime.".call";
$fd = fopen($fname, 'w');
fwrite($fd, "Channel: ".$alerts."\n");
fwrite($fd, "Callerid: IT <7021>\n");
fwrite($fd, "Set: FHMNUM=".$FAILURES."\n");
fwrite($fd, "MaxRetries: 2\n");
fwrite($fd, "RetryTime: 20\n");
fwrite($fd, "WaitTime: 40\n");
fwrite($fd, "Context: custom-NOTIFY\n");
fwrite($fd, "Extension: s\n");
fwrite($fd, "Priority: 1\n");
fclose($fd);
chown($fname,"asterisk");
chgrp($fname,"asterisk");
rename($fname,$fname_call);

# cat cdr_custom.conf
;
; Mappings for custom config file
;
[mappings]
Master.csv =>
"${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"

If $alerts is "Zap/g1/7022" Then:
# tail /var/log/asterisk/cdr-csv/Master.csv
"","7021","s","custom-NOTIFY","""IT""
<7021>","Zap/2-1","","Hangup","","2007-08-07
13:30:19","2007-08-07 13:30:19","2007-08-07
13:30:26",7,7,"ANSWERED","DOCUMENTATION"

If $alerts is "SIP/4053" Then:
# tail /var/log/asterisk/cdr-csv/Master.csv
"","7021","s","custom-NOTIFY","""IT""
<7021>","SIP/4053-0829b6a0","","Hangup","","2007-08-07
12:58:02","2007-08-07 12:58:02","2007-08-07
12:58:15",13,13,"ANSWERED","DOCUMENTATION"

(This is an Asterisk/FreePBX system)



   

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Re: [asterisk-users] Free sitting

[Gordon Henderson]

On Tue, 7 Aug 2007, Olivier wrote:

> 2007/8/7, Gordon Henderson <[EMAIL PROTECTED]>:
>>
>> On Tue, 7 Aug 2007, Olivier wrote:
>>
>>> Gordon,
>>>
>>> What you described is exactly Follow-me feature : users are always
>> logged
>>> and can be reached somewhere.
>>
>> I've heard of some variants of this feature - that's the beauty (and
>> down-side!) of a programmable system - it's open to different people's
>> interpretations... (And why I think some of these features shouldn't be
>> hard-coded into the system when they are implementable in the dialplan or
>> AGI)
>>
>>> By the way, do you introduce special settings so that ringing tones are
>>> different ?
>>> Let me explain this :
>>>
>>> If Alice dials its extension and PIN code using Bob's hardphones, Bob
>> and
>>> Alice can both be called with the same phone.
>>> Is it possible to have different ringing for Alice and Bob's incoming
>> calls
>>> ?
>>
>> The simple answer is "I don't know"..
>>
>>> Maybe an SDP option inside INVITE SIP message would do the trick ?
>>> Maybe hardphone settings would read INVITE fields (Contact info ?) to
>>> segregate calls ?
>>
>> A simple way might be to change the caller-id on follow-me calls - change
>> the name part into the number and change the number into a special number
>> that the phone recognises as a separate ring-tone, but you lose
>> information here, and need a phone that can display both name and number
>> at the same time, and connect numbers to different ring-tones - then you
>> end up going down the route of requiring a certian phone for a certian
>> service - which might be acceptable to some people, but defeats the whole
>> generic "any SIP phone will do" type ideas.
>
>
> Could you elaborate ?
>
> I know some hardphone (eg Thomson ST2030) can set ring-tone according
> Caller's presence inside phone's directory.

The Grandstream GXP2000's can store 4 ring-tones. 3 of these can be 
matched to an incoming callerId, so for 3 different numbers you can have 3 
different ring tones, with everything else using the default ring tone.

Another option I've just thought of (after having a look at the config 
screen one one of my GXP2000s) might be to use a different account on each 
phone for the follow-me feature, so if you had extenstions 100 through 199 
which were real people, and extensions 200-299 mapped to the 2nd account 
on each phone, then ring that on a follow-me and assign that account on 
the phone with a different ring tone - that would preserve all caller-id 
information, but it would them depend on having phones with multiple 
account support.

So you "own" extension 123. You sit at the phone which is extension 150 
and dial in the follow-me codes. Someone dials your extension, 123, the 
system recognises you've got follow-me set, and diverts it to the 
follow-me extension plus 100 - ie. 250 which is the 2nd account on phone 
150 which then activates a different ring-tone... (and on the Grandstreams 
you'd need a different LED flash for the differnet account being rung)

More work to setup the system and phones, but ...

> In this case,  Asterisk would have to :
> - fake original caller-name and set it to "call for Alice",
> - replace original caller-id with Alice extension (eg 4111 instead of +44
> 812 41 54 66)
> so that  hardphone  gets  everything it needs to :
> - recognize from caller-id that the calls comes from Alice (though it's a
> call FOR Alice)
> - and then uses Alice ringing tone instead of Bob's tone.
>
> Is this roughly correct ?

Yes, but messy :)

> How many phones behaves like this ?

I'd suspect all phones which have multi-account support, so they could 
have at least one ring-tone per account.

The GXP2000's can match on 3 incoming numbers in addition to the default.

The Snom 300 I have appears to have 4 lines and 9 different ring tones you 
can assign to each line, as well as 4 categories of ring groups in the 
address book, so I guess the other models in the range have this, or more.

> As you said, it would be sad to loose SIP portability.
>
> It would be nice to use SIP protocol to drive such behaviour.

Indeed...

I think the 2-account system might be workable though, but more work to 
setup on the phone side of things...

And I like that enough to implement it for GXP2000 (And Snom customers, I 
think!) At least as an option, anyway.

Gordon

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Re: [asterisk-users] Free sitting

[Olivier]

That's exactly what I was after.
Thanks

Maybe a bit of SIP MESSAGE and it would be perfect.
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Re: [asterisk-users] TAE to RJ11 connector (hope not OT)

[Anselm Martin Hoffmeister]

Am Montag, den 06.08.2007, 18:09 +0200 schrieb gincantalupo:
> Hi,
> I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on 
> it...only a TAE connector.
> I'd like to create an adapter so I need to know which TAE pins to 
> connect to RJ 11 pins.
> Is there anybody who knows where I can find a schema of that adapter?
> Single connector pinout may help too.

Have a look at
http://de.wikipedia.org/wiki/TAE

You need the "La" and "Lb" wires. They are usually top-left and
middle-left, watching the device from the plug side.

BR
Anselm


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Re: [asterisk-users] CDR/MySQL basic config

[Alessandro Russo]

Hi,
first step is correct

> Hmm.. This is what I get:
>
> [EMAIL PROTECTED] ~]# mysql -u root -p
> Enter password:
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 187143 to server version: 4.1.20
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
> You make an errore here : mysql> use asteriskcdrdb
>
users' information are stored in mysql db

mysql> use mysql;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Database changed
mysql

mysql> select Host from user where User = 'asteriskcdruser' ;
+---+
| Host  |
+---+
| localhost |
+---+
1 row in set (0.00 sec)

mysql>

Are you sure that user 'asteriskcdruser' has the privileges to insert record
in DB "asteriskcdrdb"?
If not...allow 'asteriskcdruser' to insert record ^_^

mysql> grant insert on asteriskcdrdb.* to
[EMAIL PROTECTED] by 'asteriskcdruser';
mysql> exit

Reload asterisk and try



On 8/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
>
>
> Hmm.. This is what I get:
>
> [EMAIL PROTECTED] ~]# mysql -u root -p
> Enter password:
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 187143 to server version: 4.1.20
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
> mysql> use asteriskcdrdb ;
> Reading table information for completion of table and column names
> You can turn off this feature to get a quicker startup with -A
>
> Database changed
> mysql> select Host from user where User = 'asteriskcdruser' ;
> ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
> mysql>
>
>
> Adrian Marsh
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Forrest
> Beck
> Sent: 07 August 2007 02:59
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] CDR/MySQL basic config
>
> Adrian,
>
> What host/ip did you specify when you created the user?
>
> #> mysql --user=root --password
>
> #mysql> use mysql;
>
> #mysql> select Host from user where User = 'asteriskcdruser'
> (this line is case sensitive)
>
> Does it return 127.0.0.1 or localhost.  Make cdr_mysql reflect that.
>
> You should also check out cdr_odbc, asterisk can connect through an
> ODBC connection which in turn is a connection to the MySQL database.
> There seems to be more suport for the ODBC driver.
>
> Hope this helps some
>
>
>
> On 8/6/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install.  The
> > add-ons pack has been installed for a while, so now I'm trying to add
> > the Mysql config.
> >
> > I've created a mysql database, added the grants for a user acces, and
> > can run a mysql -u asteriskcdruser -p and can connect to the database.
> >
> > I've been using this as a guide:
> >
> http://www.757.org/~joat/wiki/index.php/Asterisk#Viewing_CDR_Data_with_A
> > sterisk:_CDR_Analyzer
> >
> > I've created cdr_mysql.conf:
> >
> > [global]
> > hostname=localhost
> > dbname=asteriskcdrdb
> > table=cdr
> > password=password
> > user=asteriskcdruser
> > port=3306
> > sock=/tmp/mysql.sock
> > userfield=1
> >
> > But when I start asterisk (1.4 on my test machine), I get:
> >
> >   == Parsing '/etc/asterisk/cdr_mysql.conf': Found
> > [Aug  6 21:01:14] ERROR[32512]: cdr_addon_mysql.c:436 my_load_module:
> > Failed to connect to mysql database asteriskcdrdb on localhost.
> > cdr_addon_mysql.so => (MySQL CDR Backend)
> > [Aug  6 21:01:14] ERROR[32512]: res_config_mysql.c:627
> mysql_reconnect:
> > MySQL RealTime: Failed to connect database server  on  (err 2002).
> Check
> > debug for more info.
> > [Aug  6 21:01:14] WARNING[32512]: res_config_mysql.c:474 load_module:
> > MySQL RealTime: Couldn't establish connection. Check debug.
> > [Aug  6 21:01:14] NOTICE[32512]: config.c:1171
> > ast_config_engine_register: Registered Config Engine mysql
> > MySQL RealTime driver loaded.
> > res_config_mysql.so => (MySQL RealTime Configuration Driver)
> >
> >
> > I'm also looking as to what CDR viewers there are available, and which
> > people think are best.  I want to view/report on the calls made within
> > A*k.
> >
> > Thanks,
> >
> > Adrian
> >
> > ___
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> >
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> >
>
>
> --
> ***
> Forrest Beck
> IAXTEL: 17002871718
> [EMAIL PROTECTED]
> http://www.shift8.biz
>
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Re: [asterisk-users] A102d samgoma's card

[Christian Victor]

fateme fatah schrieb:
> Please every that work with A102d say how about is it?Is it really difficult 
> to install card for me new in asterisk?
> Best regards.

It is not more difficult to install than any other E1 card for Asterisk.
In fact in my opinion it's one of the easier to install.

Christian

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[asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Nasir Iqbal]

Hi List,

Me setup for faxing is

Asterisk (TxFAX)  => ATA => FAX Machine

And SIP setting is

Codec uLaw
dtmfmode inband


but I am facing a problem

when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
Machine Print two pages (Enlarging the page) but shows it received one
page.


Please help me


Thanks


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[asterisk-users] TE207P Question

[Jeremy Mann]

I need help on my zaptel.conf and Zapata.conf for a TE207P

I'd like Span 1 to receive a PRI from the phone company(US PRI).

I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the 
phone company)

Essentially my asterisk box is a man in the middle intercepting calls from the 
PRI passing certain DID to the Nortel, also intercepting calls from the Nortel 
passing them via IAX to other asterisk boxes as necessary.

Do I just need to make both PRI signaling?  See below:

/etc/zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

/etc/asterisk/Zapata.conf
Group=1
Signaling=pri_cpe
Switchtype=national
Context=from-pri
Channel=1-23
Group=2
Context=from-nortel
Channel=25-47


Thanks for any help.



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[asterisk-users] Which spandsp & unicall version to use with 1.2?

[Patrick]

Hi all,

Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?

Would that be the ones listed below?

http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/

http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/
http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/

Thanks!
Patrick



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Re: [asterisk-users] Teliax Quality of Service

[John Novack]

Brian Capouch wrote:
> Stephen Bosch wrote:
>   
>> PSTN service still sets the standard.
>>
>> 
>
> With infrastructure paid for under a gracious guaranteed-profit monopoly by 
> ratepayers, 
At least in the US, this hasn't been done for many a year.
There is no LEGAL monopoly. There is, in many areas, little to no 
competition though. As long as the ILEC owns and controls the outside 
plant, competition is difficult. CLECs that survive, for the most part, 
have their own outside plant. As a practical matter, due to the heavy 
capital investment, this means smaller towns.

Regardless, the PSTN IS the standard, and until VOIP reaches that level 
of reliability and quality, there will be many business failures

John Novack

-- 
Dog is my co-pilot


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[asterisk-users] caller ID strangeness

[Jerry Geis]

when executing a NOOP(caller id ${CALLERIDNUM}) in the dialplan
I am getting odd caller id results from a SIP connection. The SIP 
Connection is to
a nortel cs 1000.

*4145664222;phonecontext=+1

notice the extra stuff after the number

I am using asterisk 1.2.17
Is there a caller ID issue?

Jerry
*

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Re: [asterisk-users] Call Center SoftPhone with Auto Answer

[Thiago Maluf]

Ola Joao,
tem um modo do Asterisk fazer isso sim.
Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações.
Abs!

Hi List,
The asterisk have one way to do it.
just put one script to discovery if this user is online or offline.
case is offline play one music. if not, call the user.
understand?
thiago!



2007/8/6, Joao Pereira <[EMAIL PROTECTED]>:
>
> Hello
> I need a Softphone with auto answer where users can't turn it off.
> Does someone knows a softphone where users can't turn the auto answer off?
> Or is there any way Asterisk could force the clients to answer the phone?
>
> Thanks
> Regards
> Joao Pereira
>
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-- 

THIAGO MALUF RESENDE
Consultor Voip e Programador WEB (Voip Developer and Web Developer)
Tel: +55 21 86042100
e-mail: [EMAIL PROTECTED]
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Re: [asterisk-users] Which spandsp & unicall version to use with 1.2?

[Tzafrir Cohen]

On Tue, Aug 07, 2007 at 04:01:54PM +0200, Patrick wrote:
> Hi all,
> 
> Anyone have an idea which version of spandsp, libunicall, libmfcr2,
> libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
> latest asterisk 1.2?
> 
> Would that be the ones listed below?
> 
> http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
> http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/
> 
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/
> http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/

Nither. Use spandsp 0.0.3 for asterisk 1.2 .

-- 
   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] OT, I'm looking for SIP/register-enabled softphone

[Kate Kretz]

Can You please advice me free softphone which supports SIP registrations ?

Cheers,
Kate
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Re: [asterisk-users] CDR/MySQL basic config

[Adrian Marsh]

Hi Alessandro,

 

Thanks for that.. I'm pretty sure about the user. I used Webmin to
confirm the user configs, but I ran your commands anyway:

 

 

mysql> use mysql ;

Reading table information for completion of table and column names

You can turn off this feature to get a quicker startup with -A

 

Database changed

mysql> select Host from user where User = 'asteriskcdruser' ;

+---+

| Host  |

+---+

| localhost |

+---+

1 row in set (0.00 sec)

 

mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED]
identified by 'asteriskcdruser';

Query OK, 0 rows affected (0.00 sec)

 

But I still get the failure:

 

[Aug  7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module:
Failed to connect to mysql database asteriskcdrdb on localhost.

cdr_addon_mysql.so => (MySQL CDR Backend)

[Aug  7 15:14:10] ERROR[29103]: res_config_mysql.c:627 mysql_reconnect:
MySQL RealTime: Failed to connect database server  on  (err 2002). Check
debug for more info.

[Aug  7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module:
MySQL RealTime: Couldn't establish connection. Check debug.

[Aug  7 15:14:10] NOTICE[29103]: config.c:1171
ast_config_engine_register: Registered Config Engine mysql

MySQL RealTime driver loaded.

res_config_mysql.so => (MySQL RealTime Configuration Driver)

 

This box also das Cacti installed on it, which makes use of the MySql
server as well (and all is ok there).

 

 

Adrian Marsh

  



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alessandro
Russo
Sent: 07 August 2007 14:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR/MySQL basic config

 

Hi, 
first step is correct 

Hmm.. This is what I get:

[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20

Type 'help;' or '\h' for help. Type '\c' to clear the buffer. 

You make an errore here : mysql> use asteriskcdrdb 

users' information are stored in mysql db

mysql> use mysql;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A

Database changed 
mysql   

mysql> select Host from user where User = 'asteriskcdruser' ;
+---+
| Host  |
+---+
| localhost |
+---+
1 row in set (0.00 sec)

mysql> 

Are you sure that user 'asteriskcdruser' has the privileges to insert
record in DB "asteriskcdrdb"?
If not...allow 'asteriskcdruser' to insert record ^_^

mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED]
identified by 'asteriskcdruser';
mysql> exit

Reload asterisk and try 




On 8/7/07, Adrian Marsh < [EMAIL PROTECTED]> wrote:


Hmm.. This is what I get:

[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql> use asteriskcdrdb ;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A 

Database changed
mysql> select Host from user where User = 'asteriskcdruser' ;
ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
mysql>


Adrian Marsh
-Original Message- 
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] ] On Behalf Of Forrest
Beck
Sent: 07 August 2007 02:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR/MySQL basic config

Adrian,

What host/ip did you specify when you created the user? 

#> mysql --user=root --password

#mysql> use mysql;

#mysql> select Host from user where User = 'asteriskcdruser'
(this line is case sensitive)

Does it return 127.0.0.1 or localhost.  Make cdr_mysql reflect that.

You should also check out cdr_odbc, asterisk can connect through an
ODBC connection which in turn is a connection to the MySQL database.
There seems to be more suport for the ODBC driver. 

Hope this helps some



On 8/6/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install.  The
> add-ons pack has been installed for a while, so now I'm trying to add
> the Mysql config.
>
> I've created a mysql database, added the grants for a user acces, and
> can run a mysql -u asteriskcdruser -p and can connect to the database.

>
> I've been using this as a guide:
>
http://www.757.org/~joat/wiki/index.php/Asterisk#Viewing_CDR_Data_with_A

> sterisk:_CDR_Analyzer
>
> I've created cdr_mysql.conf:
>
> [global]
> hostname=localhost
> dbname=asteriskcdrdb
> table=cdr
> password=password
> user=asteriskcdruser 
> port=3306
> sock=/tmp/mysql.sock
> userfield=1
>
> But when I start asterisk (1.4 on my test machine), I get:
>
>   == P

Re: [asterisk-users] TE207P Question

[Tzafrir Cohen]

On Tue, Aug 07, 2007 at 09:01:56AM -0500, Jeremy Mann wrote:
> I need help on my zaptel.conf and Zapata.conf for a TE207P
> 
> I'd like Span 1 to receive a PRI from the phone company(US PRI).
> 
> I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as 
> the phone company)
> 
> Essentially my asterisk box is a man in the middle intercepting calls from 
> the PRI passing certain DID to the Nortel, also intercepting calls from the 
> Nortel passing them via IAX to other asterisk boxes as necessary.
> 
> Do I just need to make both PRI signaling?  See below:
> 
> /etc/zaptel.conf
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> span=2,1,0,esf,b8zs
> bchan=25-47
> dchan=48
> 
> /etc/asterisk/Zapata.conf

/etc/asterisk/zapata.conf

(non-capital Z)

> Group=1
> Signaling=pri_cpe
> Switchtype=national
> Context=from-pri
> Channel=1-23
> Group=2

signalling = pri_net

> Context=from-nortel
> Channel=25-47

Again here: parameter names should be in small caps.

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

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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Rizwan Hisham]

try using codec gsm or g729

On 8/7/07, Nasir Iqbal <[EMAIL PROTECTED]> wrote:
>
> Hi List,
>
> Me setup for faxing is
>
> Asterisk (TxFAX)  => ATA => FAX Machine
>
> And SIP setting is
>
> Codec uLaw
> dtmfmode inband
>
>
> but I am facing a problem
>
> when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
> Machine Print two pages (Enlarging the page) but shows it received one
> page.
>
>
> Please help me
>
>
> Thanks
>
>
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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] Teliax Quality of Service

[Anthony Francis]

Douglas Garstang wrote:
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of SIP
>> Sent: Monday, August 06, 2007 8:56 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Teliax Quality of Service
>>
>> Steve Totaro wrote:
>> 
>>> Anthony Francis wrote:
>>>
>>>   
 Tim Panton wrote:


 
> On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
>
>
>
>
>   
>> I don't think creating a network without a single point of
>> 
> failure
>   
>> is unreasonable.
>>
>>
>>
>> 
> It's impossible. I can't think of a single example where this
> actually exists.
>
> Getting even close is hideously expensive.
>
> Tim, speaking for himself :-)
>
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>
>
>
>   
 In fact, the only people who would say something like this are
 
> folks
>   
>> who
>> 
 have never PHYSICALLY implemented a network, they simply don't
 understand the limitations involved.
 
>
> I worked for a CLEC in Montana, not Silicon Valley, not Manhatten, but
> rather PODUNK, Montana. We had redundant multi-homed servers, connected
> to multiple switches, running OSPF. A failure in any component (server,
> network, cable) would cause a failover to a backup component in about 6
> seconds. We had multiple upstream providers. The servers where divided
> between multiple racks, split between different power plants. We did
> just about everything we could to make the setup redundant.
>
> The CPE equipment at any single location might fail, and that wasn't
> redundant, but at least if that failed, it would not affect any other
> customers. CPE equipment included POE enabled phones, a UPS, a POE
> switch and power being delivered from our plant.
>
> Yes, all the equipment was located at the same physical location. In
> hindsight, we could have multi-homed our collocations. Why can't service
> providers multi home their edge systems to accept incoming calls from
> two physical locations? If a service provider did this, they would have
> two completely independent facilities, potentially thousands of miles
> apart, connected to different upstream providers. I can't think of
> anything short of nuclear war that would destroy their ability to accept
> calls. If they did least cost routing, it wouldn't even matter if their
> providers failed. China gets hit by a meteor and NO provider can deliver
> calls to China? Fine... at least you can still call everywhere else.
>
> Maybe it still had some holes, but jeez, at least we tried to deliver
> high quality service.
>
>
>
>
>
>
>
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There is no one here not doing best-effort redundancy, what the first 
gentleman had said was a network with NO single points of failure. 
Clearly that is a pipe dream. To the person with six second failover, 
that 6 seconds would have dropped calls and dialing out issues resulting 
in complaints. You would then tell your customer that you got it working 
immediately and often they don't care, they are still angry about the 
dropped call. MY point is, VOIP is good, great even, but anyone 
expecting a less than 20 year old tech to be more reliable than a tech 
that has been around for over a hundred (PSTN) needs to spend some more 
time thinking about that.

Anthony

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Re: [asterisk-users] TE207P Question

[Jared Smith]

On Tue, 2007-08-07 at 09:01 -0500, Jeremy Mann wrote:
> I’d like Span 1 to receive a PRI from the phone company(US PRI).

> I’d like Span 2 to interface with a Nortel Phone system as a
> PRI(acting as the phone company)

[snip]

> Do I just need to make both PRI signaling?  See below:

Your config files are close to being correct, but you're missing one
little detail.  On the first span, you'll want to set your signalling
(yes, you have to spell it that way) in zapata.conf to pri_cpe, as
you're acting as the CPE (customer premises equipment) end of the PRI
connection.  On your second span, you want to set signalling=pri_net, so
that Asterisk acts as the network (telco) side of the PRI connection.

As an added note, you may want to change the timing setting on your
second span= line in zaptel.conf.  If you're acting as the telco, you
might want to send timing to your Nortel, depending on how the Nortel is
configured.  (The way you currently have your spans configured, you're
telling Zaptel to get it's timing from both the telco and the Nortel,
and the card can only sync to one timing source at a time.)


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Teliax Quality of Service

[Anthony Francis]

Mark Coccimiglio wrote:
> Steve Totaro wrote:
>
>   
>> What if a train derails and slices through the main fiber connections.  
>> OK, so you have XO, Global Crossing, Verizon, and UCN all for 
>> redundancy.  Well guess what?  They are all most likely running over 
>> those strands of fiber.  You better have a VSAT connection too!
>>  
>>
>> 
> That's why I lease a few servers in a data center on other side of the 
> country.  Setup in a "hot stand-by" state.  Its that peace-of-mind you 
> can't buy any way else.  So it costs a few hundred dollars a month 
> (actually less then $500).  It  kicks in to take-up capacity when my 
> main servers gets "real busy" or go off-line for maintenance.  Its 
> instant and automatic.  Ok sure it took a lot of planning to get it 
> right, but that's what I get paid to do. 
>
> Single point of failure should NEVER completely disable your company.  
> Yes outages happen and backhoe's cut fibre all the time.  From within 
> this stuff can make one's life rather difficult, but from the outside it 
> should be almost unnoticed. When was the last time you noticed an outage 
> at Google, Microsoft or the DoD?  Do you think they don't happen? 
>
> Its not that difficult or all that expensive if planned and implemented 
> properly. 
>
> Mark C.
>
>   
What are you talking about google was completely unavailable to most of 
the country for four hours about a month ago.

Anthony

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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Jared Smith]

On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
> try using codec gsm or g729

No, please don't.  I'll be the first do admit I don't know much about
faxing, but I *do* know that you don't want to try to send faxes over a
highly-compressed codec such as gsm or g.729.  It will probably only
work over ulaw or alaw unless you're using something like T.38 (which
uses udptl to transmit the fax data).


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] how to specify a channel inside txfax command

[gincantalupo]

Hi,
I've compiled rxfax and txfax on my Asterisk box.
I've made a small extensions.conf for test so that when I call a number 
with my Idefisk softphone I activate txfax command.
My goal is to try to send a fax using an analog line first and then an 
ISDN line but I do not know how to specify the channel inside txfax command.
Sorry if it may sound a stoopid question but found no docs on internet 
and "show application txfax" shows nothing useful.
What am I doing wrong?

TIA

Giorgio

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Re: [asterisk-users] TE207P Question

[Jeremy Mann]

So would the timing be 0?

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Tuesday, August 07, 2007 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE207P Question

As an added note, you may want to change the timing setting on your
second span= line in zaptel.conf.  If you're acting as the telco, you
might want to send timing to your Nortel, depending on how the Nortel is
configured.  (The way you currently have your spans configured, you're
telling Zaptel to get it's timing from both the telco and the Nortel,
and the card can only sync to one timing source at a time.)

This e-mail, facsimile, or letter and any files or attachments transmitted with 
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prohibited without proper authorization. If you are not the intended recipient, 
any disclosure, copying, printing, or use of this information is strictly 
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you have received this information in error, please notify Texas Health 
Management Group immediately at 1-817-310-4999. Texas Health Management Group, 
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to this information.

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[asterisk-users] OT - Callto:// tags inside web pages

[Olivier]

Hi,

Where can I find relevant information concerning callto:// tags ?

Is it standardized or browser specific ?
How within your browser, can you specify the software and parameters to used
when clicking on such callto:// tags ?
I couldn't find much googling or reading Preferences tab in Firefox.

Regards
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Re: [asterisk-users] Login info from Active directory

[Eric Chamberlain]

Active Directory relies on Kerberos for authentication.  Kerberos uses
tickets and does not centrally track presence.



You would either have to parse all the Domain Controller logs or use
some other method to determine when a user is logged in, such as setting
a flag with login/logout scripts.



--

Eric Chamberlain, CISSP

Chief Technical Officer

Voxilla - http://voxilla.com/



  _

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Monday, August 06, 2007 10:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Login info from Active directory



Hello,

Is it easy to retrieve user presence from Asterisk dialplan according
Active directory data ?
I mean how do you know a user is logged reading data from Active
directory ?

best regards



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[asterisk-users] Intermittent busy tone detection on loopback setup

[Manish Sapariya]

Hi List,

I am using asterisk to test another asterisk/voip software, by
generating user agent on asterisk which can be used to place
calls to unit under test. I have also tests in which I place
calls over PSTN lines and receive them back on PSTN line and
verify the results.

Everything is fine, except some times asterisk sees busy tone
on callee side when the caller side plays the audio file, for
pstn call. To understand the problem I ran simple test by using
a loopback cable from fxs to fxo, repeatedly. What I found is that
the received file indeed was so distorted and had some busy tone
like sounds that asterisk identified it as busy tone and hanged
up the line.

There are following questions that I am trying to answer:
- Why the busy like distortion is generated on the PSTN line?
- When I recorded the wav files to see the actual quality of
the voice I found them not very good. My assumption was that
the loopback setup should give the best result possible. Did
I misunderstand anything?
- Will PSTN hardware, I have digium cards, cause such intermittent
noise generation when use repeatedly to place calls.

Any ideas?

Thanks and Regards,
Manish



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Re: [asterisk-users] TE207P Question

[Jared Smith]

On Tue, 2007-08-07 at 09:48 -0500, Jeremy Mann wrote:
> So would the timing be 0?

Yes, that's correct if you're supplying timing to the device on the
other end.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Use of context=... in [default] section of sip.conf

[Filipe Brandenburger]

Hi,

If I have [myprovider] section with context=something. When I do an
outgoing call by using Dial(SIP/myprovider/464646)", does context=...
affect anything? As I understand it, it only affects incoming calls, but
I might be wrong.

Another thing, the setting of context=... on [default] section will
affect all [provider] sections without context=..., right? What if I
don't specify any context on [default], what would be the default
context? What if there's no context or an invalid context on a section,
what would happen to incoming calls that match that section?

If I want to receive calls on the URI sip:[EMAIL PROTECTED]
from anywhere on the net, I should set context=... on [default] section,
right? That's it, or would I also need to set something like
autocreatepeer? I read it's insecure to set it, but I wonder if there's
another way to do it...

Thanks,
Filipe


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Re: [asterisk-users] CDR/MySQL basic config

[Alessandro Russo]

Hi, try to login as asteriskcdruser to mysql

# mysql -u asteriskcdruser -p
Enter password: password
Welcome to the MySQL monitor.  Commands end with ; or \g.
Your MySQL connection id is 12
Server version: 5.0.32-Debian_7etch1-log Debian etch distribution

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql>


Can you login with asteriskcdruser?
If you cannot login there are some problems with privileges or...I don't
know :(


On 8/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
>
>  Hi Alessandro,
>
>
>
> Thanks for that.. I'm pretty sure about the user. I used Webmin to confirm
> the user configs, but I ran your commands anyway:
>
>
>
>
>
> mysql> use mysql ;
>
> Reading table information for completion of table and column names
>
> You can turn off this feature to get a quicker startup with -A
>
>
>
> Database changed
>
> mysql> select Host from user where User = 'asteriskcdruser' ;
>
> +---+
>
> | Host  |
>
> +---+
>
> | localhost |
>
> +---+
>
> 1 row in set (0.00 sec)
>
>
>
> mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> 'asteriskcdruser';
>
> Query OK, 0 rows affected (0.00 sec)
>
>
>
> But I still get the failure:
>
>
>
> [Aug  7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module:
> Failed to connect to mysql database asteriskcdrdb on localhost.
>
> cdr_addon_mysql.so => (MySQL CDR Backend)
>
> [Aug  7 15:14:10] ERROR[29103]: res_config_mysql.c:627 mysql_reconnect:
> MySQL RealTime: Failed to connect database server  on  (err 2002). Check
> debug for more info.
>
> [Aug  7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module:
> MySQL RealTime: Couldn't establish connection. Check debug.
>
> [Aug  7 15:14:10] NOTICE[29103]: config.c:1171 ast_config_engine_register:
> Registered Config Engine mysql
>
> MySQL RealTime driver loaded.
>
> res_config_mysql.so => (MySQL RealTime Configuration Driver)
>
>
>
> This box also das Cacti installed on it, which makes use of the MySql
> server as well (and all is ok there).
>
>
>
>
>
> Adrian Marsh
>
>
>   --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Alessandro Russo
> *Sent:* 07 August 2007 14:13
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] CDR/MySQL basic config
>
>
>
> Hi,
> first step is correct
>
> Hmm.. This is what I get:
>
> [EMAIL PROTECTED] ~]# mysql -u root -p
> Enter password:
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 187143 to server version: 4.1.20
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
>  You make an errore here : mysql> use asteriskcdrdb
>
> users' information are stored in mysql db
>
> mysql> use mysql;
> Reading table information for completion of table and column names
> You can turn off this feature to get a quicker startup with -A
>
> Database changed
> mysql
>
> mysql> select Host from user where User = 'asteriskcdruser' ;
> +---+
> | Host  |
> +---+
> | localhost |
> +---+
> 1 row in set (0.00 sec)
>
> mysql>
>
> Are you sure that user 'asteriskcdruser' has the privileges to insert
> record in DB "asteriskcdrdb"?
> If not...allow 'asteriskcdruser' to insert record ^_^
>
> mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> 'asteriskcdruser';
> mysql> exit
>
> Reload asterisk and try
>
>
>  On 8/7/07, *Adrian Marsh* < [EMAIL PROTECTED]> wrote:
>
>
> Hmm.. This is what I get:
>
> [EMAIL PROTECTED] ~]# mysql -u root -p
> Enter password:
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 187143 to server version: 4.1.20
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
> mysql> use asteriskcdrdb ;
> Reading table information for completion of table and column names
> You can turn off this feature to get a quicker startup with -A
>
> Database changed
> mysql> select Host from user where User = 'asteriskcdruser' ;
> ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
> mysql>
>
>
> Adrian Marsh
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] ] On Behalf Of Forrest
> Beck
> Sent: 07 August 2007 02:59
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] CDR/MySQL basic config
>
> Adrian,
>
> What host/ip did you specify when you created the user?
>
> #> mysql --user=root --password
>
> #mysql> use mysql;
>
> #mysql> select Host from user where User = 'asteriskcdruser'
> (this line is case sensitive)
>
> Does it return 127.0.0.1 or localhost.  Make cdr_mysql reflect that.
>
> You should also check out cdr_odbc, asterisk can connect through an
> ODBC connection which in turn is a connection t

Re: [asterisk-users] caller ID strangeness

[Vieri]

--- Jerry Geis <[EMAIL PROTECTED]> wrote:

> when executing a NOOP(caller id ${CALLERIDNUM})
>
> I am using asterisk 1.2.17

I use CALLERID(num) or CALLERID(all) in 1.2+.
I don't know if that can help.



   

Building a website is a piece of cake. Yahoo! Small Business gives you all the 
tools to get online.
http://smallbusiness.yahoo.com/webhosting 

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Re: [asterisk-users] ISDN30 card for UK : sanity check

[Gavin Henry]

Very good. Sangoma cards are great. Get the a101d though. Nice wee review:

http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/

Voipon are great guys too. We resell for them.

On 07/08/07, Rory Campbell-Lange <[EMAIL PROTECTED]> wrote:
> We will be connecting our Asterisk server to ISDN 30 and intend using
> the Sangoma A101 card. The install location is in London (UK).
>
> Sangoma card at Voipon
> http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA
>
> I would be grateful to hear if this is the right choice of card. Usage
> reports would be helpful.
>
> Regards
> Rory
>
> --
> Rory Campbell-Lange
> Campbell-Lange Workshop Ltd.
> <[EMAIL PROTECTED]>
> 
>
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Re: [asterisk-users] Use of context=... in [default] section of sip.conf

[Jared Smith]

On Tue, 2007-08-07 at 11:18 -0400, Filipe Brandenburger wrote:
> If I have [myprovider] section with context=something. When I do an
> outgoing call by using Dial(SIP/myprovider/464646)", does context=...
> affect anything? As I understand it, it only affects incoming calls, but
> I might be wrong.

That's correct.  The context is only there to tell Asterisk where in the
dialplan to send *incoming* calls.

> Another thing, the setting of context=... on [default] section will
> affect all [provider] sections without context=..., right? What if I
> don't specify any context on [default], what would be the default
> context? 

My guess would be the [default] context, but I could be wrong.

> What if there's no context or an invalid context on a section,
> what would happen to incoming calls that match that section?

The calls would most likely be rejected by Asterisk.



-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] OT, I'm looking for SIP/register-enabled softphone

[Tzafrir Cohen]

On Tue, Aug 07, 2007 at 08:16:35PM +0600, Kate Kretz wrote:
> Can You please advice me free softphone which supports SIP registrations ?

twinkle? ekiga?

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

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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Alexander Lopez]

That is wrong on so many levels.

 

You may want to take the time to install hylafax+iaxmodem, it offers
error correction and has many more features that offset the time
required to install...

 

Alex

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Tuesday, August 07, 2007 10:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] Prblem with Page Hight While Faxing over
uLaw

 

try using codec gsm or g729

On 8/7/07, Nasir Iqbal <[EMAIL PROTECTED] > wrote:

Hi List,

Me setup for faxing is

Asterisk (TxFAX)  => ATA => FAX Machine 

And SIP setting is

Codec uLaw
dtmfmode inband


but I am facing a problem

when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
Machine Print two pages (Enlarging the page) but shows it received one 
page.


Please help me


Thanks


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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com 

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Re: [asterisk-users] CDR/MySQL basic config

[Jaswinder Singh]

sock=/tmp/mysql.sock

Is this path for socket correct ?
In some distro it is /var/lib/mysql/mysql.sock . Type "locate mysql.sock" in
shell .  Also remove  uncomment port=3306  if using socket to connect .

On 07/08/07, Alessandro Russo <[EMAIL PROTECTED]> wrote:
>
> Hi, try to login as asteriskcdruser to mysql
>
> 
> # mysql -u asteriskcdruser -p
> Enter password: password
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 12
> Server version: 5.0.32-Debian_7etch1-log Debian etch distribution
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
> mysql>
>
>
> 
> Can you login with asteriskcdruser?
> If you cannot login there are some problems with privileges or...I don't
> know :(
>
>
> On 8/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> >
> >  Hi Alessandro,
> >
> >
> >
> > Thanks for that.. I'm pretty sure about the user. I used Webmin to
> > confirm the user configs, but I ran your commands anyway:
> >
> >
> >
> >
> >
> > mysql> use mysql ;
> >
> > Reading table information for completion of table and column names
> >
> > You can turn off this feature to get a quicker startup with -A
> >
> >
> >
> > Database changed
> >
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> >
> > +---+
> >
> > | Host  |
> >
> > +---+
> >
> > | localhost |
> >
> > +---+
> >
> > 1 row in set (0.00 sec)
> >
> >
> >
> > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > 'asteriskcdruser';
> >
> > Query OK, 0 rows affected (0.00 sec)
> >
> >
> >
> > But I still get the failure:
> >
> >
> >
> > [Aug  7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module:
> > Failed to connect to mysql database asteriskcdrdb on localhost.
> >
> > cdr_addon_mysql.so => (MySQL CDR Backend)
> >
> > [Aug  7 15:14:10] ERROR[29103]: res_config_mysql.c:627 mysql_reconnect:
> > MySQL RealTime: Failed to connect database server  on  (err 2002). Check
> > debug for more info.
> >
> > [Aug  7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module:
> > MySQL RealTime: Couldn't establish connection. Check debug.
> >
> > [Aug  7 15:14:10] NOTICE[29103]: config.c:1171
> > ast_config_engine_register: Registered Config Engine mysql
> >
> > MySQL RealTime driver loaded.
> >
> > res_config_mysql.so => (MySQL RealTime Configuration Driver)
> >
> >
> >
> > This box also das Cacti installed on it, which makes use of the MySql
> > server as well (and all is ok there).
> >
> >
> >
> >
> >
> > Adrian Marsh
> >
> >
> >   --
> >
> > *From:* [EMAIL PROTECTED] [mailto:
> > [EMAIL PROTECTED] *On Behalf Of *Alessandro Russo
> > *Sent:* 07 August 2007 14:13
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Subject:* Re: [asterisk-users] CDR/MySQL basic config
> >
> >
> >
> > Hi,
> > first step is correct
> >
> > Hmm.. This is what I get:
> >
> > [EMAIL PROTECTED] ~]# mysql -u root -p
> > Enter password:
> > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > Your MySQL connection id is 187143 to server version: 4.1.20
> >
> > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> >
> >  You make an errore here : mysql> use asteriskcdrdb
> >
> > users' information are stored in mysql db
> >
> > mysql> use mysql;
> > Reading table information for completion of table and column names
> > You can turn off this feature to get a quicker startup with -A
> >
> > Database changed
> > mysql
> >
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> > +---+
> > | Host  |
> > +---+
> > | localhost |
> > +---+
> > 1 row in set (0.00 sec)
> >
> > mysql>
> >
> > Are you sure that user 'asteriskcdruser' has the privileges to insert
> > record in DB "asteriskcdrdb"?
> > If not...allow 'asteriskcdruser' to insert record ^_^
> >
> > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > 'asteriskcdruser';
> > mysql> exit
> >
> > Reload asterisk and try
> >
> >
> >  On 8/7/07, *Adrian Marsh* < [EMAIL PROTECTED]> wrote:
> >
> >
> > Hmm.. This is what I get:
> >
> > [EMAIL PROTECTED] ~]# mysql -u root -p
> > Enter password:
> > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > Your MySQL connection id is 187143 to server version: 4.1.20
> >
> > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> >
> > mysql> use asteriskcdrdb ;
> > Reading table information for completion of table and column names
> > You can turn off this feature to get a quicker startup with -A
> >
> > Database changed
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> > ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
> > mysql>
> >
> >
> > Adrian Marsh
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [m

Re: [asterisk-users] Use of context=... in [default] section of sip.conf

[Jaswinder Singh]

When you make calls then context=xxx of the peer you are using ( your
extension ) will matter , the context=yyy line of your trunk wont matter .
If you dont specify a context= for  a peer then it is considered to be in
[default] context  .

On 07/08/07, Jared Smith <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2007-08-07 at 11:18 -0400, Filipe Brandenburger wrote:
> > If I have [myprovider] section with context=something. When I do an
> > outgoing call by using Dial(SIP/myprovider/464646)", does context=...
> > affect anything? As I understand it, it only affects incoming calls, but
> > I might be wrong.
>
> That's correct.  The context is only there to tell Asterisk where in the
> dialplan to send *incoming* calls.
>
> > Another thing, the setting of context=... on [default] section will
> > affect all [provider] sections without context=..., right? What if I
> > don't specify any context on [default], what would be the default
> > context?
>
> My guess would be the [default] context, but I could be wrong.
>
> > What if there's no context or an invalid context on a section,
> > what would happen to incoming calls that match that section?
>
> The calls would most likely be rejected by Asterisk.
>
>
>
> --
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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Re: [asterisk-users] CDR/MySQL basic config

[Alessandro Russo]

Hi,
this is my cdr_mysql.conf

[global]
hostname=localhost
dbname=asterisk
table=cdr
password= mypassword
user=asteriskcdr
;port=3306
;sock=/tmp/mysql.sock
;userfield=1

as you can see, the last three are comment out, try to comment these rows,
your system could be able to retrieve the sock in same way...maybe.
My debian :)

/var/run/mysqld/mysqld.sock

bye

On 8/7/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
>
> sock=/tmp/mysql.sock
>
> Is this path for socket correct ?
> In some distro it is /var/lib/mysql/mysql.sock . Type "locate mysql.sock"
> in shell .  Also remove  uncomment port=3306  if using socket to connect .
>
> On 07/08/07, Alessandro Russo <[EMAIL PROTECTED]> wrote:
> >
> > Hi, try to login as asteriskcdruser to mysql
> >
> > 
> > # mysql -u asteriskcdruser -p
> > Enter password: password
> > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > Your MySQL connection id is 12
> > Server version: 5.0.32-Debian_7etch1-log Debian etch distribution
> >
> > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> >
> > mysql>
> >
> >
> > 
> > Can you login with asteriskcdruser?
> > If you cannot login there are some problems with privileges or...I don't
> > know :(
> >
> >
> > On 8/7/07, Adrian Marsh < [EMAIL PROTECTED]> wrote:
> > >
> > >  Hi Alessandro,
> > >
> > >
> > >
> > > Thanks for that.. I'm pretty sure about the user. I used Webmin to
> > > confirm the user configs, but I ran your commands anyway:
> > >
> > >
> > >
> > >
> > >
> > > mysql> use mysql ;
> > >
> > > Reading table information for completion of table and column names
> > >
> > > You can turn off this feature to get a quicker startup with -A
> > >
> > >
> > >
> > > Database changed
> > >
> > > mysql> select Host from user where User = 'asteriskcdruser' ;
> > >
> > > +---+
> > >
> > > | Host  |
> > >
> > > +---+
> > >
> > > | localhost |
> > >
> > > +---+
> > >
> > > 1 row in set (0.00 sec)
> > >
> > >
> > >
> > > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > > 'asteriskcdruser';
> > >
> > > Query OK, 0 rows affected (0.00 sec)
> > >
> > >
> > >
> > > But I still get the failure:
> > >
> > >
> > >
> > > [Aug  7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module:
> > > Failed to connect to mysql database asteriskcdrdb on localhost.
> > >
> > > cdr_addon_mysql.so => (MySQL CDR Backend)
> > >
> > > [Aug  7 15:14:10] ERROR[29103]: res_config_mysql.c:627
> > > mysql_reconnect: MySQL RealTime: Failed to connect database server  on  
> > > (err
> > > 2002). Check debug for more info.
> > >
> > > [Aug  7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module:
> > > MySQL RealTime: Couldn't establish connection. Check debug.
> > >
> > > [Aug  7 15:14:10] NOTICE[29103]: config.c:1171
> > > ast_config_engine_register: Registered Config Engine mysql
> > >
> > > MySQL RealTime driver loaded.
> > >
> > > res_config_mysql.so => (MySQL RealTime Configuration Driver)
> > >
> > >
> > >
> > > This box also das Cacti installed on it, which makes use of the MySql
> > > server as well (and all is ok there).
> > >
> > >
> > >
> > >
> > >
> > > Adrian Marsh
> > >
> > >
> > >   --
> > >
> > > *From:* [EMAIL PROTECTED] [mailto:
> > > [EMAIL PROTECTED] *On Behalf Of *Alessandro
> > > Russo
> > > *Sent:* 07 August 2007 14:13
> > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > > *Subject:* Re: [asterisk-users] CDR/MySQL basic config
> > >
> > >
> > >
> > > Hi,
> > > first step is correct
> > >
> > > Hmm.. This is what I get:
> > >
> > > [EMAIL PROTECTED] ~]# mysql -u root -p
> > > Enter password:
> > > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > > Your MySQL connection id is 187143 to server version: 4.1.20
> > >
> > > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> > >
> > >  You make an errore here : mysql> use asteriskcdrdb
> > >
> > > users' information are stored in mysql db
> > >
> > > mysql> use mysql;
> > > Reading table information for completion of table and column names
> > > You can turn off this feature to get a quicker startup with -A
> > >
> > > Database changed
> > > mysql
> > >
> > > mysql> select Host from user where User = 'asteriskcdruser' ;
> > > +---+
> > > | Host  |
> > > +---+
> > > | localhost |
> > > +---+
> > > 1 row in set (0.00 sec)
> > >
> > > mysql>
> > >
> > > Are you sure that user 'asteriskcdruser' has the privileges to insert
> > > record in DB "asteriskcdrdb"?
> > > If not...allow 'asteriskcdruser' to insert record ^_^
> > >
> > > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > > 'asteriskcdruser';
> > > mysql> exit
> > >
> > > Reload asterisk and try
> > >
> > >
> > >  On 8/7/07, *Adrian Marsh* < [

Re: [asterisk-users] Intermittent busy tone detection on loopback setup

[Steve Totaro]

Either your configs are wrong or the card is faulty. You should have 
crystal clear calls.

Thanks,
Steve

Manish Sapariya wrote:
> Hi List,
>
> I am using asterisk to test another asterisk/voip software, by
> generating user agent on asterisk which can be used to place
> calls to unit under test. I have also tests in which I place
> calls over PSTN lines and receive them back on PSTN line and
> verify the results.
>
> Everything is fine, except some times asterisk sees busy tone
> on callee side when the caller side plays the audio file, for
> pstn call. To understand the problem I ran simple test by using
> a loopback cable from fxs to fxo, repeatedly. What I found is that
> the received file indeed was so distorted and had some busy tone
> like sounds that asterisk identified it as busy tone and hanged
> up the line.
>
> There are following questions that I am trying to answer:
> - Why the busy like distortion is generated on the PSTN line?
> - When I recorded the wav files to see the actual quality of
> the voice I found them not very good. My assumption was that
> the loopback setup should give the best result possible. Did
> I misunderstand anything?
> - Will PSTN hardware, I have digium cards, cause such intermittent
> noise generation when use repeatedly to place calls.
>
> Any ideas?
>
> Thanks and Regards,
> Manish
>
>
>
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>
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Re: [asterisk-users] Teliax Quality of Service

[Steve Totaro]

Anthony Francis wrote:
> Douglas Garstang wrote:
>   
>>> -Original Message-
>>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>>> [EMAIL PROTECTED] On Behalf Of SIP
>>> Sent: Monday, August 06, 2007 8:56 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] Teliax Quality of Service
>>>
>>> Steve Totaro wrote:
>>> 
>>>   
 Anthony Francis wrote:

   
 
> Tim Panton wrote:
>
>
> 
>   
>> On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
>>
>>
>>
>>
>>   
>> 
>>> I don't think creating a network without a single point of
>>> 
>>>   
>> failure
>>   
>> 
>>> is unreasonable.
>>>
>>>
>>>
>>> 
>>>   
>> It's impossible. I can't think of a single example where this
>> actually exists.
>>
>> Getting even close is hideously expensive.
>>
>> Tim, speaking for himself :-)
>>
>> ___
>> --Bandwidth and Colocation Provided by
>>   
>> 
>> http://www.api-digital.com--
>>   
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>   
>> 
> In fact, the only people who would say something like this are
> 
>   
>> folks
>>   
>> 
>>> who
>>> 
>>>   
> have never PHYSICALLY implemented a network, they simply don't
> understand the limitations involved.
> 
>   
>> I worked for a CLEC in Montana, not Silicon Valley, not Manhatten, but
>> rather PODUNK, Montana. We had redundant multi-homed servers, connected
>> to multiple switches, running OSPF. A failure in any component (server,
>> network, cable) would cause a failover to a backup component in about 6
>> seconds. We had multiple upstream providers. The servers where divided
>> between multiple racks, split between different power plants. We did
>> just about everything we could to make the setup redundant.
>>
>> The CPE equipment at any single location might fail, and that wasn't
>> redundant, but at least if that failed, it would not affect any other
>> customers. CPE equipment included POE enabled phones, a UPS, a POE
>> switch and power being delivered from our plant.
>>
>> Yes, all the equipment was located at the same physical location. In
>> hindsight, we could have multi-homed our collocations. Why can't service
>> providers multi home their edge systems to accept incoming calls from
>> two physical locations? If a service provider did this, they would have
>> two completely independent facilities, potentially thousands of miles
>> apart, connected to different upstream providers. I can't think of
>> anything short of nuclear war that would destroy their ability to accept
>> calls. If they did least cost routing, it wouldn't even matter if their
>> providers failed. China gets hit by a meteor and NO provider can deliver
>> calls to China? Fine... at least you can still call everywhere else.
>>
>> Maybe it still had some holes, but jeez, at least we tried to deliver
>> high quality service.
>>
>>
>>
>>
>>
>>
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> There is no one here not doing best-effort redundancy, what the first 
> gentleman had said was a network with NO single points of failure. 
> Clearly that is a pipe dream. To the person with six second failover, 
> that 6 seconds would have dropped calls and dialing out issues resulting 
> in complaints. You would then tell your customer that you got it working 
> immediately and often they don't care, they are still angry about the 
> dropped call. MY point is, VOIP is good, great even, but anyone 
> expecting a less than 20 year old tech to be more reliable than a tech 
> that has been around for over a hundred (PSTN) needs to spend some more 
> time thinking about that.
>
> Anthony
>
>   

How about this for a single point of failure. Before the demarc we had a 
big rack labeled "Bell" so you know it was old. In the cabinet were many 
(what looked like) car batteries. These things held absolutely no charge 
and the Teco said it was against policy to replace them anymore, they 
also said I could not do it myself. EVERYTHING else was redundant 
network and power.

It is things like that that pop up and get you. We finally stuck a UPS 
between the the outlet and the cabinet but all of this was totally 
unexpected and there was significant downtime.

The other thing was, two strands of fiber was carryin

Re: [asterisk-users] ISDN30 card for UK : sanity check

[Rory Campbell-Lange]

Hi Gavin

Many thanks for the note. For what reason do you recommend the old a101
though?

Regards
Rory

On 07/08/07, Gavin Henry ([EMAIL PROTECTED]) wrote:
> Very good. Sangoma cards are great. Get the a101d though. Nice wee review:
> 
> http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/
> 
> Voipon are great guys too. We resell for them.
> 
> On 07/08/07, Rory Campbell-Lange <[EMAIL PROTECTED]> wrote:
> > We will be connecting our Asterisk server to ISDN 30 and intend using
> > the Sangoma A101 card. The install location is in London (UK).
> >
> > Sangoma card at Voipon
> > http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA
> >
> > I would be grateful to hear if this is the right choice of card. Usage
> > reports would be helpful.
-- 
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
<[EMAIL PROTECTED]>


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Re: [asterisk-users] A102d samgoma's card

[Steve Totaro]

Christian Victor wrote:
> fateme fatah schrieb:
>   
>> Please every that work with A102d say how about is it?Is it really difficult 
>> to install card for me new in asterisk?
>> Best regards.
>> 
>
> It is not more difficult to install than any other E1 card for Asterisk.
> In fact in my opinion it's one of the easier to install.
>
> Christian
>
>   

Very easy in my opinion.

It has a nice little "wizard" that guides you through the setup and even 
creates your zapata.conf and zaptel.conf.

Thanks,
Steve


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Re: [asterisk-users] Teliax Quality of Service

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Anthony Francis
> Sent: Tuesday, August 07, 2007 7:29 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Teliax Quality of Service
> 
> Douglas Garstang wrote:
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of SIP
> >> Sent: Monday, August 06, 2007 8:56 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Teliax Quality of Service
> >>
> >> Steve Totaro wrote:
> >>
> >>> Anthony Francis wrote:
> >>>
> >>>
>  Tim Panton wrote:
> 
> 
> 
> > On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
> >
> >
> >
> >
> >
> >> I don't think creating a network without a single point of
> >>
> > failure
> >
> >> is unreasonable.
> >>
> >>
> >>
> >>
> > It's impossible. I can't think of a single example where this
> > actually exists.
> >
> > Getting even close is hideously expensive.
> >
> > Tim, speaking for himself :-)
> >
> > ___
> > --Bandwidth and Colocation Provided by
> >
> > http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
>  In fact, the only people who would say something like this are
> 
> > folks
> >
> >> who
> >>
>  have never PHYSICALLY implemented a network, they simply don't
>  understand the limitations involved.
> 
> >
> > I worked for a CLEC in Montana, not Silicon Valley, not Manhatten,
but
> > rather PODUNK, Montana. We had redundant multi-homed servers,
connected
> > to multiple switches, running OSPF. A failure in any component
(server,
> > network, cable) would cause a failover to a backup component in
about 6
> > seconds. We had multiple upstream providers. The servers where
divided
> > between multiple racks, split between different power plants. We did
> > just about everything we could to make the setup redundant.
> >
> > The CPE equipment at any single location might fail, and that wasn't
> > redundant, but at least if that failed, it would not affect any
other
> > customers. CPE equipment included POE enabled phones, a UPS, a POE
> > switch and power being delivered from our plant.
> >
> > Yes, all the equipment was located at the same physical location. In
> > hindsight, we could have multi-homed our collocations. Why can't
service
> > providers multi home their edge systems to accept incoming calls
from
> > two physical locations? If a service provider did this, they would
have
> > two completely independent facilities, potentially thousands of
miles
> > apart, connected to different upstream providers. I can't think of
> > anything short of nuclear war that would destroy their ability to
accept
> > calls. If they did least cost routing, it wouldn't even matter if
their
> > providers failed. China gets hit by a meteor and NO provider can
deliver
> > calls to China? Fine... at least you can still call everywhere else.
> >
> > Maybe it still had some holes, but jeez, at least we tried to
deliver
> > high quality service.
> >
> >
> >
> >
> >
> >
> >
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> There is no one here not doing best-effort redundancy, what the first
> gentleman had said was a network with NO single points of failure.
> Clearly that is a pipe dream. To the person with six second failover,
> that 6 seconds would have dropped calls and dialing out issues
resulting
> in complaints. You would then tell your customer that you got it
working
> immediately and often they don't care, they are still angry about the
> dropped call. MY point is, VOIP is good, great even, but anyone
> expecting a less than 20 year old tech to be more reliable than a tech
> that has been around for over a hundred (PSTN) needs to spend some
more
> time thinking about that.

So you've never gotten a dropped call or dead air on a PSTN call? Put it
in a little perspective.


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Re: [asterisk-users] Teliax Quality of Service

[Stephen Bosch]

Brian Capouch wrote:
> Stephen Bosch wrote:
> 
>> PSTN service still sets the standard.
>>
> 
> With infrastructure paid for under a gracious guaranteed-profit monopoly 
> by ratepayers,

In a regulated marketplace with legislated minimum service levels.

In Canada, most of the phone systems were government-owned. It was a
good system, at least from the point of view of reliability. I don't
miss the surly (and often slow) service, but it's arguable whether
today's service -- in which everyone smiles nice and *pretends* to serve
you while ignoring you completely -- is any better. At least the bloody
stuff worked.

Communications infrastructure is a strategic, national asset, and only
really useful if it goes everywhere, even to the unprofitable pockets
like Podunk Corners, North Dakota. People forget this. In a totally free
marketplace, Podunk Corners waits years for service and gets tin cans
and string when it finally arrives.

> now being used as a weapon to stifle competition from 
> VoIP, cable, and other emerging technologies.

Is it? Maybe -- in some circumstances. The history of this makes for
some pretty distorted economics, if you ask me.

If you want an example of what happens when you don't have regulation to
build infrastructure, look at Africa. All wireless, all horribly
oversubscribed, and correspondingly unreliable. That's how you pay for
expensive equipment in a "free market".

-Stephen-

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Re: [asterisk-users] sip issue with one way audio

[Eric Lubow]

Jason,

   What type of phones are you using?  I originally started getting this
error when I got the Cisco 7961Gs (prior to dumping them and going with
all Polycoms).  It turned out to be some setting in the XML provisioning
boot file (although I can't remember which one).  Once I went to a
minimal config, the problem seemed to solve itself.  Eventually I
upgraded the SIP firmware and the problem disappeared regarless of the
config file.

Eric

On Mon, 2007-08-06 at 23:38 -0600, Al lists wrote:
> Nat?
> 
> 
> On 8/6/07, Jason Walker <[EMAIL PROTECTED]> wrote:
> I am getting this error
> [Aug  6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt:
> Maximum
> retries exceeded on transmission [EMAIL PROTECTED]
> for seqno
> 102 (Critical Response)
> [Aug  6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt:
> Hanging 
> up call [EMAIL PROTECTED] - no reply to our critical
> packet.
> 
> any Ideas?
> 
> Jason
> 
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Systems Administrator
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Re: [asterisk-users] Teliax Quality of Service

[Stephen Bosch]

Douglas Garstang wrote:
> So you've never gotten a dropped call or dead air on a PSTN call? Put it
> in a little perspective.

I can count on one hand the number of outages of this kind that I've had
on PSTN in my lifetime.

Your mileage may vary.

-Stephen-


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Re: [asterisk-users] Teliax Quality of Service

[Stephen Bosch]

Mark Coccimiglio wrote:
> Single point of failure should NEVER completely disable your company.  
> Yes outages happen and backhoe's cut fibre all the time.  From within 
> this stuff can make one's life rather difficult, but from the outside it 
> should be almost unnoticed. When was the last time you noticed an outage 
> at Google, Microsoft or the DoD?  Do you think they don't happen? 

Of course not -- but how many hundreds of millions have been invested in
their infrastructure?

-Stephen-


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Re: [asterisk-users] Sangoma PRI

[Matt]

Sangoma is working on it with me just for information here is what
appears in /var/log/messages when the PRI goes down:

Aug  7 06:11:48 EMSPBX kernel: wanpipe1: Critical: Echo Canceller Chip
Security Compromised: Disabling Driver!
Aug  7 06:11:48 EMSPBX kernel: wanpipe1: Please call Sangoma Tech
Support (www.sangoma.com)!
Aug  7 06:11:48 EMSPBX kernel: wanpipe1: Error: Card Critically Shutdown!
Aug  7 06:11:48 EMSPBX kernel: wanpipe1: T1 Front End unconfigation!
Aug  7 06:11:48 EMSPBX kernel: wanpipe1: AFT communications disabled!

On 8/5/07, Matt <[EMAIL PROTECTED]> wrote:
> Hi,
> I have a client who has a system with a Sangoma 1 port PRI card with
> echo canceling in it.For some reason, when the system comes up the
> PRI will stay up for about 4-5 hours, then drop.   "zap show status"
> shows everything as ok, but we can't make or receive any calls until
> the system is rebooted.   Just restarting asterisk does not fix the
> problem.
>
> I am going to call Verizon, however wanted to consult the list to see
> if anyone here had any ideas.  At this point, I am putting my finger
> on a Verizon issue, as in our lab the system did not have any issues
> keeping the PRI active and taking calls.
>
> Any thoughts?
>

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Re: [asterisk-users] low-level dump for PRI dchan debugging

[Andrew Joakimsen]

On 8/6/07, Erik Anderson <[EMAIL PROTECTED]> wrote:
> On 8/6/07, Anthony Francis <[EMAIL PROTECTED]> wrote:
> > Yeah you are sending the SABME's because you think you are the master,
> > they are not replaying with a UA because they think they are the master,
> > you should def be pri_cpe.
>
> Tried it...no go.

Usually, if that is the issue, you would get messages such as "We
think we're network, but so does the far end"


>
> > There is one other potential cause here, you may not have had the
> > sangoma install patch and rebuild zaptel. Not doing that can cause a D
> > channel lockout on your end, but the provider should be able to see the
> > the D is in lockout.
>
> I re-patched zaptel, compiled, and re-installed.  No difference.
>
> I think I'm just going to have to wait until tomorrow when I can get
> both Sangoma and the telco on the phone.
>
> -erik
>

In your wanpipe1.conf see if you have

TDMV_DCHAN  = 0

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Re: [asterisk-users] low-level dump for PRI dchan debugging

[Erik Anderson]

On 8/7/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> In your wanpipe1.conf see if you have
>
> TDMV_DCHAN  = 0

Nope.  I have it set to 24.

-erik

-- 
Erik Anderson
http://andersonfam.org

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Re: [asterisk-users] Teliax Quality of Service

[Mark Coccimiglio]

Stephen Bosch wrote:

>Of course not -- but how many hundreds of millions have been invested in
>their infrastructure?
>  
>
You missed the point. The standard formula I use is "5 days out"  or 
more precisely 2% of gross revenues each year.  For google its still a 
kings ransom, but for a small business it not too hard to implement.  
Case-in-point I have half a dozen cell phones I bought at the beginning 
of the summer (V3RAZR).  Costs me nothing (after rebates) to setup other 
then an afternoon at the cellophane store, but I like "toy shopping" 
anyway.  The residual cost is an additional $60/month.  I now have a 
practical backup to my phone system.  My regular phone stuff (office 
DiDs and 1-800 number) is setup to forward to the cells on a simple 
phone call.  Its not a "perfect" solution but I've planned "something" 
which is better then most.  Most people's way of handling an outage is 
to "go home" and let the techs fix it for tomorrow. 

-- 
As I slowly sip my coffee I feel my humanity start to slip back into me and 
realize what a foul beast humanity really is.


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[asterisk-users] Macro Overlap

[Nicholas Blasgen]

I've got 4 SIP phone lines with a call-limit of 2 for each.  I've written a
handy macro to allow my users to dial a phone number and the macro will
figure out the next available line to use by first checking if the GROUP()
is over 2 and then checking to see if ChanIsAvail() as a backup, and if it
can't use the line for either reason it goes to the next line.  The problem
is that there are enough situations that the Macro gets called twice without
much time seperation.  Both macros check the group() number, it comes back
as free, they check the line availability and it's open, and they try
dialing.  But because they both started at more or less the same instant,
they've both at the same stage in the macro and sometimes (maybe 10% of the
time) a macro will try dialing on a line that's already in use.

My question is this.  Is it possible to tell Asterisk to execute part of a
macro as a block without allowing any other commands to be processed during
that time?  Some way to LOCK the dialplan (as you'd do in SQL).  I want my
macro to be able to execute the part of the code that checks line status and
then sets the GROUP() without allowing any other dialplans from running
during that time.  Anyone know if this is a current feature?

-- 
/Nick
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Re: [asterisk-users] Macro Overlap

[Jared Smith]

On Tue, 2007-08-07 at 11:13 -0700, Nicholas Blasgen wrote:
> My question is this.  Is it possible to tell Asterisk to execute part
> of a macro as a block without allowing any other commands to be
> processed during that time?  

You'll want to check out the MacroExclusive() application.  It does
exactly what you're looking for.  If I remember correctly, it's new in
Asterisk 1.4.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Teliax Quality of Service

[Steve Totaro]

Mark Coccimiglio wrote:
> Stephen Bosch wrote:
>
>   
>> Of course not -- but how many hundreds of millions have been invested in
>> their infrastructure?
>>  
>>
>> 
> You missed the point. The standard formula I use is "5 days out"  or 
> more precisely 2% of gross revenues each year.  For google its still a 
> kings ransom, but for a small business it not too hard to implement.  
> Case-in-point I have half a dozen cell phones I bought at the beginning 
> of the summer (V3RAZR).  Costs me nothing (after rebates) to setup other 
> then an afternoon at the cellophane store, but I like "toy shopping" 
> anyway.  The residual cost is an additional $60/month.  I now have a 
> practical backup to my phone system.  My regular phone stuff (office 
> DiDs and 1-800 number) is setup to forward to the cells on a simple 
> phone call.  Its not a "perfect" solution but I've planned "something" 
> which is better then most.  Most people's way of handling an outage is 
> to "go home" and let the techs fix it for tomorrow. 
>
>   
Setup those cell phones to use chan_mobile and you have a very nice 
solution.  Unless the phones are assigned to people who use them as 
their own.  You could possibly add some lines on a family plan $10/mo 
extra w/T-Mobile and use those strictly as PSTN fialover lines.

Thanks,
Steve

Thanks,
Steve

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Re: [asterisk-users] Macro Overlap

[Mojo with Horan & Company, LLC]

set your own mutex using astdb?  It may just be atomic enough for you to 
get by.

Nicholas Blasgen wrote:
> I've got 4 SIP phone lines with a call-limit of 2 for each.  I've 
> written a handy macro to allow my users to dial a phone number and the 
> macro will figure out the next available line to use by first checking 
> if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a 
> backup, and if it can't use the line for either reason it goes to the 
> next line.  The problem is that there are enough situations that the 
> Macro gets called twice without much time seperation.  Both macros check 
> the group() number, it comes back as free, they check the line 
> availability and it's open, and they try dialing.  But because they both 
> started at more or less the same instant, they've both at the same stage 
> in the macro and sometimes (maybe 10% of the time) a macro will try 
> dialing on a line that's already in use.
>  
> My question is this.  Is it possible to tell Asterisk to execute part of 
> a macro as a block without allowing any other commands to be processed 
> during that time?  Some way to LOCK the dialplan (as you'd do in SQL).  
> I want my macro to be able to execute the part of the code that checks 
> line status and then sets the GROUP() without allowing any other 
> dialplans from running during that time.  Anyone know if this is a 
> current feature?
> 
> -- 
> /Nick
> 
> 
> 
> 
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Re: [asterisk-users] Macro Overlap

[Nicholas Blasgen]

I'm going to try it out, but I'm not very hopefull although it's exactly
what's needed.  My macro contains a Dial() command and my concern is that
the dialplan isn't considered done untill Dial() returns.  But I'm going to
try it.  Will report back shortly.

On 8/7/07, Jared Smith <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2007-08-07 at 11:13 -0700, Nicholas Blasgen wrote:
> > My question is this.  Is it possible to tell Asterisk to execute part
> > of a macro as a block without allowing any other commands to be
> > processed during that time?
>
> You'll want to check out the MacroExclusive() application.  It does
> exactly what you're looking for.  If I remember correctly, it's new in
> Asterisk 1.4.
>
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Re: [asterisk-users] Teliax Quality of Service

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen Bosch
> Sent: Tuesday, August 07, 2007 10:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Teliax Quality of Service
> 
> Brian Capouch wrote:
> > Stephen Bosch wrote:
> >
> >> PSTN service still sets the standard.
> >>
> >
> > With infrastructure paid for under a gracious guaranteed-profit
monopoly
> > by ratepayers,
> 
> In a regulated marketplace with legislated minimum service levels.
> 
> In Canada, most of the phone systems were government-owned. It was a
> good system, at least from the point of view of reliability. I don't
> miss the surly (and often slow) service, but it's arguable whether
> today's service -- in which everyone smiles nice and *pretends* to
serve
> you while ignoring you completely -- is any better. At least the
bloody
> stuff worked.
> 
> Communications infrastructure is a strategic, national asset, and only
> really useful if it goes everywhere, even to the unprofitable pockets
> like Podunk Corners, North Dakota. People forget this. In a totally
free
> marketplace, Podunk Corners waits years for service and gets tin cans
> and string when it finally arrives.

I disagree. There is more competition in smaller towns and rural areas.
It isn't cost effective for the bigger carriers to move in, so the small
ones do. They get state/federal subsidies. I'll bet you there's more
ISP's, and CLEC's per square inch in Montana than there is in the bay
area.


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Re: [asterisk-users] Teliax Quality of Service

[Mark Coccimiglio]

Steve Totaro wrote:

>Setup those cell phones to use chan_mobile and you have a very nice 
>solution.  Unless the phones are assigned to people who use them as 
>their own.  You could possibly add some lines on a family plan $10/mo 
>extra w/T-Mobile and use those strictly as PSTN fialover lines.
>
>Thanks,
>Steve
>  
>
Actually the chan_mobile is a really good idea.  I'll need to look into 
that.  I was just gonna have them on hand for "grab and go" use.  With a 
means to fall back in the event my PSTN or VoIP failure. 

Mark C

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Re: [asterisk-users] OT - Callto:// tags inside web pages

[Anselm Martin Hoffmeister]

Am Dienstag, den 07.08.2007, 16:51 +0200 schrieb Olivier:
> Hi,
> 
> Where can I find relevant information concerning callto:// tags ?
> 
> Is it standardized or browser specific ?
> How within your browser, can you specify the software and parameters
> to used when clicking on such callto:// tags ? 
> I couldn't find much googling or reading Preferences tab in Firefox.

AFAIK for SIP the "sip:" protocol would be what you want, "callto:" is
the Skype idea of phone URIs IIRC.

In windows you can assign protocol handlers for protocols like "sip:",
some softphones will do that automatically. Works pretty much the same
like file type associations.

BR
Anselm


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[asterisk-users] Outbound dialing

[Tim Johnson]

Hello all. I am just getting back into Asterisk and I am setting up my  
Linksys SPA3102. I have incoming calls working fine, as is the phone  
plugged into the unit. My problem is I cannot get the SPA3102 to dial  
a phone number automatically. I can call the extention of the PSTN and  
I get a second dialtone, and I can then manually dial. I'd like to be  
able to have Asterisk pass the number I dialed to the SPA and have it  
dialout. I've played with dialplans on the SPA I've found during my  
googling, but I think it might be something I am missing in my  
extentions.conf file. Any ideas?

Tim

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Re: [asterisk-users] Outbound dialing

[Nicholas Blasgen]

Not specific to the SPA3102, but just normal outbound dialing is as follows:

exten => _1NXXNXX,1,Dial(//${EXTEN})

or if you want to require people to dial 9, then:

exten => _91NXXNXX,1,Dial(//${EXTEN})

or if you're like me and you're used to a cell phone and don't like dialing
the 1:

exten => _NXXNXX,1,Dial(//1${EXTEN})


On 8/7/07, Tim Johnson <[EMAIL PROTECTED]> wrote:
>
> Hello all. I am just getting back into Asterisk and I am setting up my
> Linksys SPA3102. I have incoming calls working fine, as is the phone
> plugged into the unit. My problem is I cannot get the SPA3102 to dial
> a phone number automatically. I can call the extention of the PSTN and
> I get a second dialtone, and I can then manually dial. I'd like to be
> able to have Asterisk pass the number I dialed to the SPA and have it
> dialout. I've played with dialplans on the SPA I've found during my
> googling, but I think it might be something I am missing in my
> extentions.conf file. Any ideas?
>
> Tim
>
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-- 
/Nick
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Re: [asterisk-users] Teliax Quality of Service

[Andres Paglayan]

On Aug 6, 2007, at 10:42 AM, Stephen Bosch wrote:

> Eric "ManxPower" Wieling wrote:
>> Douglas Garstang wrote:
>>> Let's assume for a moment that it's impossible. That does not  
>>> mean adding additional servers and additional networking  
>>> equipment does not add value, or is a worthless endeavour.
>>
>> I agree with that.  At least two people that I know run ITSPs.  Each
>> time they have an outage (which is not very often) they DO learn from
>> the experience and work to avoid a future outage cause by the same  
>> issue.
>>
>> You would be surprised at how many little things can cause an outage.
>
> My own experience is that increasing "failover redundancy", which adds
> correspondingly increasing complexity, also increases the odds of  
> an outage.
>
> It is very rare that failover redundancy works as intended during an
> actual failover, no matter how many times you simulate it.
>
> I would rather have a simple network design where the cause of  
> failure,
> when it happens, is obvious and quickly corrected. For example, I  
> would
> rather have replacement parts on the shelf and be able to slap them in
> quickly than be running hot standbys and paying for the  
> electricity, and
> then have the thing break anyway when there's a failure.
>

I'll second that,
specially for smaller installations,

> -Stephen-
>
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Andres Paglayan

--"Harmony is more important than being right"
Bapak





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Re: [asterisk-users] Outbound dialing

[Steve Totaro]

Tim Johnson wrote:
> Hello all. I am just getting back into Asterisk and I am setting up my  
> Linksys SPA3102. I have incoming calls working fine, as is the phone  
> plugged into the unit. My problem is I cannot get the SPA3102 to dial  
> a phone number automatically. I can call the extention of the PSTN and  
> I get a second dialtone, and I can then manually dial. I'd like to be  
> able to have Asterisk pass the number I dialed to the SPA and have it  
> dialout. I've played with dialplans on the SPA I've found during my  
> googling, but I think it might be something I am missing in my  
> extentions.conf file. Any ideas?
>
> Tim
>
>   

Have a look here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF

Maybe you can use senddtmf to solve your problem.

Thanks,
Steve Totaro


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[asterisk-users] Switchtype

[Jeremy Mann]

In Zapata.conf, if my PRI is NI-2 configured, do I still use 
switchtype=national ?


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Re: [asterisk-users] Switchtype

[Erik Anderson]

On 8/7/07, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>
> In Zapata.conf, if my PRI is NI-2 configured, do I still use
> switchtype=national ?

Yup:

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf#ISDNPRISwitchConfiguration

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Re: [asterisk-users] Outbound dialing

[Drew Gibson]

Tim,

If the Asterisk stuff below doesn't fix it, try the docs at 
http://www.jmgtechnology.com.au/spa_3000_guide.pdf


Ensure you enable VoIP to PSTN gateway mode and that "PSTN Line" is 
registered with Asterisk. This is probably OK as you appear to get 
dialtone back from the SPA. If you are calling from the phone on "Line 
1", make all calls go through Asterisk. See above docs for details.


In case you are dialing from a phone on "Line 1", here is the "Line 1" 
dialplan from my home SPA3102...


(*xx|[3469]11|0|00|[29]x|1xxx[2-9]xx|2[01]x|50[01]|.)

I can't remember if that is default or if I tweaked it. Works in Ontario.

If that is OK, try increasing the gain "SPA to PSTN". If the gain is too 
low, the DTMF may not be recognised by the CO. I found this out whilst 
troubleshooting echo problems.


regards,

Drew


Nicholas Blasgen wrote:
Not specific to the SPA3102, but just normal outbound dialing is as 
follows:
 
exten => _1NXXNXX,1,Dial(//${EXTEN})
 
or if you want to require people to dial 9, then:
 
exten => _91NXXNXX,1,Dial(//${EXTEN})
 
or if you're like me and you're used to a cell phone and don't like 
dialing the 1:
 
exten => _NXXNXX,1,Dial(//1${EXTEN})


 
On 8/7/07, *Tim Johnson* <[EMAIL PROTECTED] 
> wrote:


Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a second dialtone, and I can then manually dial. I'd like to be
able to have Asterisk pass the number I dialed to the SPA and have it
dialout. I've played with dialplans on the SPA I've found during my
googling, but I think it might be something I am missing in my
extentions.conf file. Any ideas?

Tim

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--
/Nick


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--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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Re: [asterisk-users] OT - Callto:// tags inside web pages

[Olivier]

Replying to myself, I got this : http://en.wikipedia.org/wiki/URI_scheme

Anyway, I'm still wondering which is the best way to go, for standard and
usage compliance.

It seems that callto: was initialially used by netmeeting before being by
Skype software.
I could find a tab in XP Internet Option configuration panel, where you can
either select Skype or Netmeeting to be launched as default software
whenever a callto: tag is clicked.

If you had to design a web site, which scheme would you adopt ?

cheers
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Re: [asterisk-users] OT - Callto:// tags inside web pages

[mitcheloc]

Ollvier,

You could use the Firefox plug-in for Snap. It will auto detect
numbers on a webpage and make them dialable.

Cheers,
Mitchel

On 8/7/07, Olivier <[EMAIL PROTECTED]> wrote:
> Replying to myself, I got this :
> http://en.wikipedia.org/wiki/URI_scheme
>
> Anyway, I'm still wondering which is the best way to go, for standard and
> usage compliance.
>
> It seems that callto: was initialially used by netmeeting before being by
> Skype software.
> I could find a tab in XP Internet Option configuration panel, where you can
> either select Skype or Netmeeting to be launched as default software
> whenever a callto: tag is clicked.
>
> If you had to design a web site, which scheme would you adopt ?
>
> cheers
>
>
>
>
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-- 

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com

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Re: [asterisk-users] Teliax Quality of Service

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andres Paglayan
> Sent: Tuesday, August 07, 2007 1:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Teliax Quality of Service
> 
> 
> On Aug 6, 2007, at 10:42 AM, Stephen Bosch wrote:
> 
> > Eric "ManxPower" Wieling wrote:
> >> Douglas Garstang wrote:
> >>> Let's assume for a moment that it's impossible. That does not
> >>> mean adding additional servers and additional networking
> >>> equipment does not add value, or is a worthless endeavour.
> >>
> >> I agree with that.  At least two people that I know run ITSPs.
Each
> >> time they have an outage (which is not very often) they DO learn
from
> >> the experience and work to avoid a future outage cause by the same
> >> issue.
> >>
> >> You would be surprised at how many little things can cause an
outage.
> >
> > My own experience is that increasing "failover redundancy", which
adds
> > correspondingly increasing complexity, also increases the odds of
> > an outage.
> >
> > It is very rare that failover redundancy works as intended during an
> > actual failover, no matter how many times you simulate it.
> >
> > I would rather have a simple network design where the cause of
> > failure,
> > when it happens, is obvious and quickly corrected. For example, I
> > would
> > rather have replacement parts on the shelf and be able to slap them
in
> > quickly than be running hot standbys and paying for the
> > electricity, and
> > then have the thing break anyway when there's a failure.
> >
> 
> I'll second that,
> specially for smaller installations,

You must have the kind of customers that don't mind having no phone
service for a few hours.



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Re: [asterisk-users] OT - Callto:// tags inside web pages

[Dean Collins]

Have both Sip and Callto (I'd use skype)

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Tuesday, 7 August 2007 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT - Callto:// tags inside web pages

 

Replying to myself, I got this : http://en.wikipedia.org/wiki/URI_scheme

Anyway, I'm still wondering which is the best way to go, for standard
and usage compliance. 

It seems that callto: was initialially used by netmeeting before being
by Skype software.
I could find a tab in XP Internet Option configuration panel, where you
can either select Skype or Netmeeting to be launched as default software
whenever a callto: tag is clicked. 

If you had to design a web site, which scheme would you adopt ?

cheers




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[asterisk-users] ASA-2007-019: Remote crash vulnerability in Skinny channel driver

[The Asterisk Development Team]

   Asterisk Project Security Advisory - ASA-2007-019

   ++
   |  Product   | Asterisk  |
   |+---|
   |  Summary   | Remote crash vulnerability in Skinny channel  |
   || driver|
   |+---|
   | Nature of Advisory | Denial of Service |
   |+---|
   |   Susceptibility   | Remote Authenticated Sessions |
   |+---|
   |  Severity  | Moderate  |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | August 7, 2007|
   |+---|
   |Reported By | Wei Wang of McAfee AVERT Labs |
   |+---|
   | Posted On  | August 7, 2007|
   |+---|
   |  Last Updated On   | August 7, 2007|
   |+---|
   |  Advisory Contact  | Jason Parker <[EMAIL PROTECTED]> |
   |+---|
   |  CVE Name  |   |
   ++

   ++
   | Description | The Asterisk Skinny channel driver, chan_skinny, has a   |
   | | remotely exploitable crash vulnerability. A segfault can |
   | | occur when Asterisk receives a   |
   | | "CAPABILITIES_RES_MESSAGE" packet where the capabilities |
   | | count is greater than the total number of items in the   |
   | | capabilities_res_message array. Note that this requires  |
   | | an authenticated session.|
   ++

   ++
   | Resolution | Asterisk code has been modified to limit the incoming |
   || capabilities count.   |
   ||   |
   || Users with configured Skinny devices should upgrade to|
   || the appropriate version listed in the corrected in|
   || section of this advisory. |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| Not affected  |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| Not affected  |
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| All versions prior to |
   |  | | 1.4.10|
   |--+-+---|
   |Asterisk Business Edition |A.x.x| Not affected  |
   |--+-+---|
   |Asterisk Business Edition |B.x.x| Not affected  |
   |--+-+---|
   |   AsteriskNOW| pre-release | All versions prior to |
   |  | | beta7 |
   |--+-+---|
   | Asterisk App

Re: [asterisk-users] Login info from Active directory

[C F]

You can try using NetBIOS and see who is logged it at a machine, given
you know which machine a user loggs into, you also have to make sure
that NetBIOS is running on that machine/s. Or you could write a
logon/logoff script that will login/logout that user from Asterisk.


On 8/7/07, Olivier <[EMAIL PROTECTED]> wrote:
> Hello,
>
> Is it easy to retrieve user presence from Asterisk dialplan according Active
> directory data ?
> I mean how do you know a user is logged reading data from Active directory ?
>
> best regards
>
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Re: [asterisk-users] OT - Callto:// tags inside web pages

[Dean Collins]

Mitchel, he's not looking for a click to dial solution - he wants to
implement some form of click on his website so people can call him.

At the end of the day most people aren't going to have it configured
correctly etc and you should really use web page based softphone.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of mitcheloc
> Sent: Tuesday, 7 August 2007 5:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] OT - Callto:// tags inside web pages
> 
> Ollvier,
> 
> You could use the Firefox plug-in for Snap. It will auto detect
> numbers on a webpage and make them dialable.
> 
> Cheers,
> Mitchel
> 
> On 8/7/07, Olivier <[EMAIL PROTECTED]> wrote:
> > Replying to myself, I got this :
> > http://en.wikipedia.org/wiki/URI_scheme
> >
> > Anyway, I'm still wondering which is the best way to go, for
standard and
> > usage compliance.
> >
> > It seems that callto: was initialially used by netmeeting before
being by
> > Skype software.
> > I could find a tab in XP Internet Option configuration panel, where
you can
> > either select Skype or Netmeeting to be launched as default software
> > whenever a callto: tag is clicked.
> >
> > If you had to design a web site, which scheme would you adopt ?
> >
> > cheers
> >
> >
> >
> >
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> --
> 
> Mitchel Constantin
> Snap - A desktop user interface for Asterisk
> www.snapanumber.com
> 
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Re: [asterisk-users] test the email-list

[C F]

This is the postmaster at the list and I am notifying you that your
message failed.

On 8/7/07, zhu lizhong <[EMAIL PROTECTED]> wrote:
> test only. good luck!
> james.zhu
>
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[asterisk-users] Asterisk 1.2.24 and 1.4.10 released

[The Asterisk Development Team]

The Asterisk development team has released Asterisk versions 1.2.24 and 1.4.10.

Version 1.2.24 is the final 1.2 release that contains normal bug fixes.  The 1.2
branch will only be maintained with security fix releases from now until it is
completely deprecated.

Version 1.4.10 contains numerous bug fixes for things all over Asterisk, as well
as a fix for a security issue.  The security issue only affects users of
chan_skinny and is documented in ASA-2007-019.

http://downloads.digium.com/pub/asa/ASA-2007-019.pdf

Another set of noteworthy changes in version 1.4.10 include many fixes for the
IAX2 channel driver.  Special recognition goes out to the developers over at
Wimba (http://www.wimba.com/) for their dedication to tracking down numerous
complicated issues in the 1.4 version of chan_iax2.  Thank you very much Mihai,
Steve, and Pete!

These releases are available for download from the following location:

http://downloads.digium.com/pub/telephony/asterisk/

Thank you very much for your support!


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[asterisk-users] turn off music on hold for a single sip user

[Damon Estep]

Is there a clean way to disable music on hold for a specific user sip
user?

 

I have seen one example that creates a class called [none] that points
to an empty directory, which creates log errors that are annoying (but
not harmful?)

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Re: [asterisk-users] turn off music on hold for a single sip user

[Steve Totaro]

Damon Estep wrote:
>
> Is there a clean way to disable music on hold for a specific user sip 
> user?
>
>  
>
> I have seen one example that creates a class called [none] that points 
> to an empty directory, which creates log errors that are annoying (but 
> not harmful?)
>

How about using the same method but dial your recording extension with 
your phone on mute and leave it going for as long as you need.  It will 
then play a file with no audio and not throw any errors.

Thanks,
Steve Totaro


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Re: [asterisk-users] turn off music on hold for a single sip user

[voiplist]

You could set a variable in the users sip.conf details like:

setvar=PlayMOH=NO

or

setvar=PlayMOH=NO


Then in your extensions.conf setup a GoToIf() which reads the variable
PlayMOH and either sets the "m" or the "r" in the dial command..

This should work fine and I know it will work in 1.4.x but not sure
about earlier versions only because I am not sure how far back version
wise you can set a variable in the sip.conf. Maybe it's always been
possible, not sure.

Hope this helps.


Regards,
 Todd R.

--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com

On 8/7/07, Damon Estep <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Is there a clean way to disable music on hold for a specific user sip user?
>
>
>
> I have seen one example that creates a class called [none] that points to an
> empty directory, which creates log errors that are annoying (but not
> harmful?)
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[asterisk-users] Which spandsp & unicall version to use with 1.2?

[Patrick]

*** resent cause the first email never showed up on the list for me ***

Hi all,

Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?

Would that be the ones listed below?

http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/

http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/
http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/

Thanks!
Patrick



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Re: [asterisk-users] Which spandsp & unicall version to use with 1.2?

[randulo]

I received the original message at 7:01 AM today

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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Paul Hales]

And I thought my sense of humour was poor

PaulH

On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
> try using codec gsm or g729
> 
> On 8/7/07, Nasir Iqbal <[EMAIL PROTECTED]> wrote:
> Hi List,
> 
> Me setup for faxing is
> 
> Asterisk (TxFAX)  => ATA => FAX Machine 
> 
> And SIP setting is
> 
> Codec uLaw
> dtmfmode inband
> 
> 
> but I am facing a problem
> 
> when I send a FAX of one page from Asterisk to ATA+FAX Machine
> the FAX
> Machine Print two pages (Enlarging the page) but shows it
> received one 
> page.
> 
> 
> Please help me
> 
> 
> Thanks
> 
> 
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> 
> 
> -- 
> Best Regards
> Rizwan Hisham
> Software Engineer
> Axvoice Inc.
> www.axvoice.com 
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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Paul Hales]

On Tue, 2007-08-07 at 10:33 -0400, Jared Smith wrote:
> On Tue, 2007-08-07 at 19:22 +0500, Rizwan Hisham wrote:
> > try using codec gsm or g729
> 
> No, please don't.  I'll be the first do admit I don't know much about
> faxing, but I *do* know that you don't want to try to send faxes over a
> highly-compressed codec such as gsm or g.729.  It will probably only
> work over ulaw or alaw unless you're using something like T.38 (which
> uses udptl to transmit the fax data).
> 
> 
I have found that T38 (if properly supported) works quite well.

PaulH


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Re: [asterisk-users] Cant Play gsm file

[atik]

Hi, I am still haveing the problem

its not a NAT issue i can hear the music on hold or even Echo is
working. only problem is VoiceMail and Playback doesn't work with
asterisk sound file.It just wait forever in Playing a file. I Did rtp
debug , i see only got Packet no send send packet, but for music on
hold rtp debug is showing both got + send packet..

is there anyway to debug PlayBack() or VoiceMailMain()?

thanks
atik

On 8/6/07, atik <[EMAIL PROTECTED]> wrote:
> Hi,
>
> i am having problem on playing asterisk sound file on my new installed
> asterisk..
> i have the following extension , if i call from any SIP / IAX  phone
> playback or voicemail doesnt play anything  but when i dial 102, I
> hear the MP3 music ..
>
> exten => 99,1,Answer()
> exten => 99,2,Playback(prepaid-welcome)
> exten => 99,3,Hangup()
>
> exten => 101,1,VoiceMailMain()
>
> exten => 102,1,Answer()
> exten => 102,2,MusicOnHold(default)
>
> I have format_gsm.so, codec_gsm.so loaded and i am using
> asterisk-sounds-1.2.1  , asterisk-1.2.23 on debian 4.0.
>
> do i miss any audio library?
>
> thanks
> atik
>

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[asterisk-users] Looking for unified messaging expert

[Justin Newman]

Anyone in the bay area with strong unified messaging experience?

Respond off list at:  [EMAIL PROTECTED]

Justin


   

Yahoo! oneSearch: Finally, mobile search 
that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC___
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Re: [asterisk-users] Prblem with Page Hight While Faxing over uLaw

[Nasir Iqbal]

Hi,

can anybody help me?.

Hi List,

Thanks for all replies.

IAX Modem + HylaFAX
T.38 Modem + HylaFAX
T.38 (using Callweaver)

all is ok 

but please help me that

what is wrong with my setting?

I think there is speed difference between Asterisk and FAX Machine due
to  improper Negotiation (Hand Shaking). But how I can solve it?

I am currently using Asterisk 1.2

Thanks

On Tue, 2007-08-07 at 18:24 +0500, Nasir Iqbal wrote: 
> Hi List,
> 
> Me setup for faxing is
> 
> Asterisk (TxFAX)  => ATA => FAX Machine
> 
> And SIP setting is
> 
> Codec uLaw
> dtmfmode inband
> 
> 
> but I am facing a problem
> 
> when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
> Machine Print two pages (Enlarging the page) but shows it received one
> page.
> 
> 
> Please help me
> 
> 
> Thanks


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Re: [asterisk-users] OT, I'm looking for SIP/register-enabled softphone

[Kate Kretz]

sorry, I meant RFC 3856, "sip presence", not sip regitration

On 8/7/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Tue, Aug 07, 2007 at 08:16:35PM +0600, Kate Kretz wrote:
> > Can You please advice me free softphone which supports SIP registrations
> ?
>
> twinkle? ekiga?
>
> --
>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] E1 or analog line

[fateme fatah]

Hi:
I want to have conference call(meetme) service with asterisk and 30 users.Now 
do I use  1E1 or 30 analog lines with due attention to high price of E1 
line?And which interface card do I use?   
Best regards.

   
-
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[asterisk-users] Method for scripting options specified in make menuconfig

[arkda]

I've been digging around and I haven't found a way to do this, but I have a
feeling I'll feel like an idiot because it's something I'm over looking.

Normally if I need to specify an additional option (such as different
language sound files) or I'm building an Asterisk server with a lean
configuration and need to remove some modules I do so with 'make
menuconfig'. I've ran into a need however to install Asterisk entirely from
the command line, so I'm looking for the method of accomplishing what I've
normally done through 'make menuconfig' solely from the command line.

Anyone know how this is accomplished?

Thanks in advance.
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Re: [asterisk-users] E1 or analog line

[Erik Anderson]

On 8/7/07, fateme fatah <[EMAIL PROTECTED]> wrote:
> Hi:
> I want to have conference call(meetme) service with asterisk and 30
> users.Now do I use  1E1 or 30 analog lines with due attention to high price
> of E1 line?And which interface card do I use?

I'm not sure what analog prices are in your area, but I'd be fairly
certain that the costs for 30 analog lines would be *far* more
expensive than an E1 line. Cost aside, terminating and managing 30
analog lines will be a big pain.  Go with the E1 - it'll save you many
many headaches.  I've only worked with Sangoma interface cards, but
they've worked flawlessly for me so far.

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Re: [asterisk-users] Teliax Quality of Service

[Stephen Bosch]

Douglas Garstang wrote:
>>> Stephen Bosch wrote:
>> In Canada, most of the phone systems were government-owned. It was a
>> good system, at least from the point of view of reliability. I don't
>> miss the surly (and often slow) service, but it's arguable whether
>> today's service -- in which everyone smiles nice and *pretends* to
> serve
>> you while ignoring you completely -- is any better. At least the
> bloody
>> stuff worked.
>>
>> Communications infrastructure is a strategic, national asset, and only
>> really useful if it goes everywhere, even to the unprofitable pockets
>> like Podunk Corners, North Dakota. People forget this. In a totally
> free
>> marketplace, Podunk Corners waits years for service and gets tin cans
>> and string when it finally arrives.
> 
> I disagree. There is more competition in smaller towns and rural areas.
> It isn't cost effective for the bigger carriers to move in, so the small
> ones do. They get state/federal subsidies.
   

That's exactly my point. I said: "In a totally free marketplace, Podunk
Corners waits years for service..."

A subsidized marketplace is not a free marketplace. Whether you do it
with regulation and sanctioned monopoly or with subsidy, that is still a
market intervention. I can't see any other way that service to sparsely
populated areas would be financially viable.

> I'll bet you there's more
> ISP's, and CLEC's per square inch in Montana than there is in the bay
> area.

Oh, I believe you. But ultimately -- what do you mean by competition?
Who owns the cable plant? I have a hard time believing that there are
any areas in Montana with redundant last-mile infrastructure.

-Stephen-

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