[asterisk-users] Is HPEC compliant with B410P ?

[Olivier]

Hi,

As latest asterisk-libpri-dahdi is introducing dahdi support of B410P, can
we use High Performance Echo Canceling addon with B410P ?*

Regards
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[asterisk-users] Which installation policy is behind Asterisk doc delivered with source code ?

[Olivier]

Hi,

Testing latest 1.6.1, it occurred to me I had to add a couple of noload
statements in /etc/asterisk/modules.conf to remove ERROR messages, when
starting Asterisk.
(I don't imply those ERROR messages were fatal to Asterisk but as a general
rule, I tried to start Asterisk without any of those).

As I'm not very familiar with module concepts at the moment, I told myself
it would be helpful, if strictly following instructions included in README
files, I did get a system that starts without any ERROR message of any kind
and still could provide some basic telephony services.

So my question is :
- is there a policy fixing the target of README files ?
- for example, if someone installs Asterisk (according README files) on a
platform equipped with a Digium analog board, should this board be
automatically discovered, configured and ready to run ?

Regards
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[asterisk-users] func_odbc and hash problem

[Giedrius Augys]

Hello,

  Now I'm testing func_odbc and hash. My configurations are:

func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}

extensions.conf
exten => s,1,Ringing
exten => s,n,Wait(4)
exten => s,n,Answer
exten => s,n,Set(NUMERIS=37037210602)
exten => s,n,Set(HASH(RESULTATAS)=${ODBC_GETNUMBER(${NUMERIS})})
exten => s,n,Verbose(1, Number is  ${HASH(RESULTATAS, number)}.)
exten => s,n,Verbose(1, Realus 1  ${HASH(RESULTATAS, real_number1)}.)
exten => s,n,Verbose(1, Realus 2  ${HASH(RESULTATAS, real_number2)}.)
exten => s,n,Verbose(1, Statusas  ${HASH(RESULTATAS, status)}.)

And asterisk output is:

== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto("SIP/sip.call.lt-01993050",
"odbc-test,s,1") in new stack
-- Goto (odbc-test,s,1)
-- Executing [EMAIL PROTECTED]:1] Ringing("SIP/sip.call.lt-01993050", "") in
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/sip.call.lt-01993050", "4") in
new stack
-- Executing [EMAIL PROTECTED]:3] Answer("SIP/sip.call.lt-01993050", "") in
new stack
-- Executing [EMAIL PROTECTED]:4] Set("SIP/sip.call.lt-01993050",
"NUMERIS=111") in new stack
-- Executing [EMAIL PROTECTED]:5] Set("SIP/sip.call.lt-01993050",
"HASH(RESULTATAS)=111,222,333,2") in new stack
-- Executing [EMAIL PROTECTED]:6] Verbose("SIP/sip.call.lt-01993050", "1,
Number is  .") in new stack
  Number is  .
-- Executing [EMAIL PROTECTED]:7] Verbose("SIP/sip.call.lt-01993050", "1,
Realus 1  .") in new stack
  Realus 1  .
-- Executing [EMAIL PROTECTED]:8] Verbose("SIP/sip.call.lt-01993050", "1,
Realus 2  .") in new stack
  Realus 2  .
-- Executing [EMAIL PROTECTED]:9] Verbose("SIP/sip.call.lt-01993050", "1,
Statusas  .") in new stack
  Statusas  .
-- Auto fallthrough, channel 'SIP/sip.call.lt-01993050' status is
'UNKNOWN'


As I read documentation, function hash gives posibility to get values using
column name. But my test was unsuccessful. Maybe somebody can help...

Thanks

-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] func_odbc questions

[Giedrius Augys]

2008/12/1 Tim Panton <[EMAIL PROTECTED]>

>
> On 1 Dec 2008, at 13:38, Giedrius Augys wrote:
>
>
>
> 2008/12/1 Tilghman Lesher <[EMAIL PROTECTED]>
>
>> On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
>> >   I'm working with asterisk 1.6. And I have success using func_odbc with
>> > one row query results (SELECT source,destination from cc WHERE ... ):
>> > exten => s,1,Ringing
>> > exten => s,n,Wait(4)
>> > exten => s,n,Answer
>> > exten =>
>> >
>> s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(
>> >${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1}
>> > ${REALNUMBER1}, ${STATUSAS})
>> >
>> >
>> > But I don't know how to retrieve data, if query returns a lot of rows.
>> In
>> > documentation I read that need to use in config file:
>> > mode=multirow,
>> > and use function ODBC_FETCH. But how to get result-id variable and use
>> > ODBC_FETCH?
>>
>> The initial result in mode=multirow is not data at all, but a query_id
>> that
>> may be used with ODBC_FETCH to return the first row of data and every
>> subsequent row of data (up to the max number of rows, if any).
>>
>> > And another question is, if I execute not SELECT , but stored procedure,
>> > and this procedure will return two, three tables? Is it possible
>> retrieve
>> > these data from couple tables?
>>
>> If you're talking about a JOIN, then yes.  As long as the fields have
>> distinct
>> names, then you can retrieve each row in turn, same as any other query.
>>
>> --
>> Tilghman
>>
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>>
>
>
> Thanks for your reply, but I don't get it... Is there any documentation or
> simple examples how to use ODBC_FETCH and so on.
>
>
> For your own sanity (if nothing else) I'd wrap the stored procedure in a
> view, then
> get FUNC_ODBC to query that view.
>
> Tim.
>
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Thanks for your replies, I found how to do that :)
-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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[asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released

[Asterisk Team]

The Asterisk.org development team has released Asterisk versions 1.2.30.3, 
1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 
and 1.6.1-rc2.  These releases are available for immediate download from 
http://downloads.digium.com/.

This update for Asterisk includes a fix for a regression introduced in 
Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the Asterisk 1.6 
branch since release.  All releases with the exception of Asterisk 1.2.30.3 
also contain a vast assortment of bugfixes in these releases.  For a full 
list of changes, see the ChangeLogs:

http://svn.digium.com/view/asterisk/tags/1.2.30.3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.4.23-rc2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.0.2/ChangeLog?view=markup
http://svn.digium.com/view/asterisk/tags/1.6.1-beta3/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.0.1/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.6.1-rc2/ChangeLog?view=markup

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] OT: What do you guys think of this?

[Doug]

At 18:56 12/1/2008, Tilghman Lesher wrote:
 >On Monday 01 December 2008 06:21:33 pm Doug wrote:
 >> We tell our customers that they are not allowed to
 >> download copyrighted material.
 >
 >So your customers are only allowed to download public domain
 >material?  That kind of restricts the amount of information
 >available on the Internet.  Nitpick:  just about everything, including
 >this email, is copyrighted by somebody.  Forbidding the download
 >of copyrighted works is not only a draconian policy, but may actually
 >violate several copyright laws (you're interfering with a copyright
 >owner's right to distribute his/her/their works, and courts are
 >generally not very sympathetic with your position).

Oops!  Didn't mean to start a fire here.

I meant to say "illegal copyrighted material".  Also, if they
are using up hundreds of Internet connections, we can see
that.  It essentially causes a Denial of Service situation
for other users on that leg of our wireless network.  The system
supposedly has rate limiting, but seems to get overloaded when
someone goes completely nuts with BitTorrent.  We are working
on ways to limit the number of simultaneous connections.

When we get a copyright infringment notice from our upstream
provider, we are compelled to reprimand the user.  I don't
think we have sent a customer to the "shower" even if they
had several notices.

"Net Neutrality" is great in principle.  But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth.  It's dollars and cents.

Es tut mir leid für das Durcheinander meine Brüder!



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Re: [asterisk-users] Can asterisk work with a dynamic IP?

[Matt Gibson]

We're using it here on dynamic IP from our ISP. 

They provide reverse DNS, which we've simply setup a CNAME to. 

So, CPE390480Q239432098423.MYISP.COM is cnamed to PBX.MYBUSINESSDOMAIN.COM 

Did not have to change anything else for this to work. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ronald Wiplinger (Lists)
> Sent: Monday, December 01, 2008 8:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Can asterisk work with a dynamic IP?
> 
> I know I can setup asterisk without Internet at all and it works as
> local pbx.
> 
> Would an asterisk box work with a dynamic IP, with a dyndns name?
> What must I take care if I try that?
> 
> bye
> 
> Ronald
> 
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[asterisk-users] Parking calls

[Sebastian]

 

Hi,

 

How can I park a call from dialplan and get going??

 

Example:

 

 

1.   Answer

2.   While follow = false

3.   ParkCall

4.   Checksomthing à follow = true

5.   Endwhile

6.   UnParkCall

7.   Go on…..

 

The idea is let the call waiting while I do some things on the dialplan, is
it possible?? Maybe is not parking the solution??

 

Thanks

 

 

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Re: [asterisk-users] OT: What do you guys think of this?

[Alex Balashov]

BJ Weschke wrote:
> Alex Balashov wrote:
>> RE Kushner List Account wrote:
>>
>>   
>>> The question is, what are you actually paying for as a customer?  To 
>>> discriminate against bits just because they actually use what they are 
>>> paying for is beyond me.
>>>
>>> At least a bandwidth cap is easier to understand. You get what you pay for.
>>> 
>> Speaking as a former sysadmin of an ISP, I would say that the issue is 
>> the following:
>>
>> 1) There is a high correlation of network-disrupting levels of traffic 
>> and BitTorrent;
>>
>> 2) Unlike some "bursty" downloads (like your CentOS ISO from an FTP 
>> server), BitTorrent traffic has the tendency to be sustained at higher 
>> levels for longer periods since the architecture presumes that 
>> everyone's a client and everyone's a server and fragments are always 
>> moving around.  This is what tends to upset oversubscription assumptions 
>> that are otherwise functional, and are the only way that the ISP can 
>> possibly afford to give you the bandwidth for the price of 
>> consumer-grade broadband.
>>
>>
>> I would tend to agree with you that discriminating against types of 
>> services and/or traffic through rate-limiting buckets and deep packet 
>> inspection is worse than a blanket bandwidth cap.   However, you need to 
>> keep in mind the other side of the coin;  were it not for Torrent, there 
>> would not be a need for traffic policing (in the overwhelming 
>> preponderance of cases) either way, so it's considered unfair to punish 
>> everyone with a bandwidth cap on everything when in reality, it's not a 
>> problem if their applications *occasionally* burst to very high levels 
>> of throughput.  This is different from using up a lot of bandwidth 
>> continuously.
>>
>> My ISP doesn't care if I chug down a CentOS ISO tonight at close to my 
>> DSL line rate.  But if I downloaded them all day long, all day, every 
>> day, there would be a problem, but the way to solve that problem isn't 
>> by taking away others' freedom to download a CentOS ISO when they feel 
>> like it in principle.
>>
>>   
>  Have you checked the FTP and/or HTTP mirrors lately for the DVD iso of 
> CentOS? The only place I've been able to find them is on the Torrents 
> themselves.

OK, so maybe that's a bad example.  Shows how much I know - I'm a Debian 
guy.  :)  But it doesn't really undermine my point.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] OT: What do you guys think of this?

[BJ Weschke]

Alex Balashov wrote:
> RE Kushner List Account wrote:
>
>   
>> The question is, what are you actually paying for as a customer?  To 
>> discriminate against bits just because they actually use what they are 
>> paying for is beyond me.
>>
>> At least a bandwidth cap is easier to understand. You get what you pay for.
>> 
>
> Speaking as a former sysadmin of an ISP, I would say that the issue is 
> the following:
>
> 1) There is a high correlation of network-disrupting levels of traffic 
> and BitTorrent;
>
> 2) Unlike some "bursty" downloads (like your CentOS ISO from an FTP 
> server), BitTorrent traffic has the tendency to be sustained at higher 
> levels for longer periods since the architecture presumes that 
> everyone's a client and everyone's a server and fragments are always 
> moving around.  This is what tends to upset oversubscription assumptions 
> that are otherwise functional, and are the only way that the ISP can 
> possibly afford to give you the bandwidth for the price of 
> consumer-grade broadband.
>
>
> I would tend to agree with you that discriminating against types of 
> services and/or traffic through rate-limiting buckets and deep packet 
> inspection is worse than a blanket bandwidth cap.   However, you need to 
> keep in mind the other side of the coin;  were it not for Torrent, there 
> would not be a need for traffic policing (in the overwhelming 
> preponderance of cases) either way, so it's considered unfair to punish 
> everyone with a bandwidth cap on everything when in reality, it's not a 
> problem if their applications *occasionally* burst to very high levels 
> of throughput.  This is different from using up a lot of bandwidth 
> continuously.
>
> My ISP doesn't care if I chug down a CentOS ISO tonight at close to my 
> DSL line rate.  But if I downloaded them all day long, all day, every 
> day, there would be a problem, but the way to solve that problem isn't 
> by taking away others' freedom to download a CentOS ISO when they feel 
> like it in principle.
>
>   
 Have you checked the FTP and/or HTTP mirrors lately for the DVD iso of CentOS? 
The only place I've been able to find them is on the Torrents themselves.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/




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Re: [asterisk-users] OT: What do you guys think of this?

[Alex Balashov]

RE Kushner List Account wrote:

> The question is, what are you actually paying for as a customer?  To 
> discriminate against bits just because they actually use what they are 
> paying for is beyond me.
> 
> At least a bandwidth cap is easier to understand. You get what you pay for.

Speaking as a former sysadmin of an ISP, I would say that the issue is 
the following:

1) There is a high correlation of network-disrupting levels of traffic 
and BitTorrent;

2) Unlike some "bursty" downloads (like your CentOS ISO from an FTP 
server), BitTorrent traffic has the tendency to be sustained at higher 
levels for longer periods since the architecture presumes that 
everyone's a client and everyone's a server and fragments are always 
moving around.  This is what tends to upset oversubscription assumptions 
that are otherwise functional, and are the only way that the ISP can 
possibly afford to give you the bandwidth for the price of 
consumer-grade broadband.


I would tend to agree with you that discriminating against types of 
services and/or traffic through rate-limiting buckets and deep packet 
inspection is worse than a blanket bandwidth cap.   However, you need to 
keep in mind the other side of the coin;  were it not for Torrent, there 
would not be a need for traffic policing (in the overwhelming 
preponderance of cases) either way, so it's considered unfair to punish 
everyone with a bandwidth cap on everything when in reality, it's not a 
problem if their applications *occasionally* burst to very high levels 
of throughput.  This is different from using up a lot of bandwidth 
continuously.

My ISP doesn't care if I chug down a CentOS ISO tonight at close to my 
DSL line rate.  But if I downloaded them all day long, all day, every 
day, there would be a problem, but the way to solve that problem isn't 
by taking away others' freedom to download a CentOS ISO when they feel 
like it in principle.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] Can asterisk work with a dynamic IP?

[Ronald Wiplinger (Lists)]

I know I can setup asterisk without Internet at all and it works as
local pbx.

Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?

bye

Ronald

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Re: [asterisk-users] OT: What do you guys think of this?

[RE Kushner List Account]

Igor Widlinski wrote:
> Awesome, I always wanted to see this "law" in real life.
>
>   

Technically I didn't call him THAT guy. I was thinking of that recently 
elected Chicago street thug who speaks before large crowds at night.

Just spend ten seconds on YouTube and you'll see it's not my original 
thought. If your ISP lets you that is.

-Ron


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Re: [asterisk-users] OT: What do you guys think of this?

[Tilghman Lesher]

On Monday 01 December 2008 06:21:33 pm Doug wrote:
> We tell our customers that they are not allowed to
> download copyrighted material.

So your customers are only allowed to download public domain
material?  That kind of restricts the amount of information
available on the Internet.  Nitpick:  just about everything, including
this email, is copyrighted by somebody.  Forbidding the download
of copyrighted works is not only a draconian policy, but may actually
violate several copyright laws (you're interfering with a copyright
owner's right to distribute his/her/their works, and courts are
generally not very sympathetic with your position).

-- 
Tilghman

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Re: [asterisk-users] OT: What do you guys think of this?

[Igor Widlinski]

Awesome, I always wanted to see this "law" in real life.

Thank You!!

Alex Balashov wrote:
> RE Kushner List Account wrote:
>   
>> Doug wrote:
>> 
>>> Why the BitTorrent guys want to give themselves
>>> even a worse reputation is beyond me.  We tell
>>> our customers that they are not allowed to
>>> download copyrighted material.  But for other,
>>> legal BitTorrent transfers, we suggest that
>>> they use the scheduling feature of uTorrent to
>>> avoid high-traffic transfers during the day.
>>>
>>>   
>>>   
>> Sieg heil! Sieg heil! Sieg heil! Give me the lash! We love the lash!
>>
>> I'd give you the finger if you told me as an ISP that I can't download 
>> CentOS 5 on DVD whenever I wanted, or to watch a Netflix movie I paid 
>> for. How you remain in business acting like a jack booted thug is beyond me.
>> 
>
> Mein overenthusiastic Oberreichskommendant,
>
> That doesn't seem like an entirely fair assessment of Doug's stated 
> policy.  :-)
>
> You are also hereby reaffirming Godwin's Law:
>
> http://en.wikipedia.org/wiki/Godwin%27s_Law
>
>
> Mit nationalsozialistischen Grüß,
>
> -- Alex
>
>   


-- 
Igor Widlinski
Systems Administrator
Eigen Development Ltd.
#300 - 1807 West 10th Avenue
Vancouver BC, V6J 2A9

t. 604.736.1066 
f. 604.736.5669
e. [EMAIL PROTECTED]

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Re: [asterisk-users] OT: What do you guys think of this?

[RE Kushner List Account]

Alex Balashov wrote:
> RE Kushner List Account wrote:
>  
>> Doug wrote:
>>
>>> Why the BitTorrent guys want to give themselves
>>> even a worse reputation is beyond me.  We tell
>>> our customers that they are not allowed to
>>> download copyrighted material.  But for other,
>>> legal BitTorrent transfers, we suggest that
>>> they use the scheduling feature of uTorrent to
>>> avoid high-traffic transfers during the day.
>>>
>>> 
>> Sieg heil! Sieg heil! Sieg heil! Give me the lash! We love the lash!
>>
>> I'd give you the finger if you told me as an ISP that I can't 
>> download CentOS 5 on DVD whenever I wanted, or to watch a Netflix 
>> movie I paid for. How you remain in business acting like a jack 
>> booted thug is beyond me.
>> 
>
> Mein overenthusiastic Oberreichskommendant,
>
> That doesn't seem like an entirely fair assessment of Doug's stated 
> policy.  :-)
>   

The question is, what are you actually paying for as a customer?  To 
discriminate against bits just because they actually use what they are 
paying for is beyond me.

At least a bandwidth cap is easier to understand. You get what you pay for.

-Ron

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Re: [asterisk-users] OT: What do you guys think of this?

[Alex Balashov]

RE Kushner List Account wrote:
> Doug wrote:
>> Why the BitTorrent guys want to give themselves
>> even a worse reputation is beyond me.  We tell
>> our customers that they are not allowed to
>> download copyrighted material.  But for other,
>> legal BitTorrent transfers, we suggest that
>> they use the scheduling feature of uTorrent to
>> avoid high-traffic transfers during the day.
>>
>>   
> 
> Sieg heil! Sieg heil! Sieg heil! Give me the lash! We love the lash!
> 
> I'd give you the finger if you told me as an ISP that I can't download 
> CentOS 5 on DVD whenever I wanted, or to watch a Netflix movie I paid 
> for. How you remain in business acting like a jack booted thug is beyond me.

Mein overenthusiastic Oberreichskommendant,

That doesn't seem like an entirely fair assessment of Doug's stated 
policy.  :-)

You are also hereby reaffirming Godwin's Law:

http://en.wikipedia.org/wiki/Godwin%27s_Law


Mit nationalsozialistischen Grüß,

-- Alex

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] OT: What do you guys think of this?

[RE Kushner List Account]

Doug wrote:
> Why the BitTorrent guys want to give themselves
> even a worse reputation is beyond me.  We tell
> our customers that they are not allowed to
> download copyrighted material.  But for other,
> legal BitTorrent transfers, we suggest that
> they use the scheduling feature of uTorrent to
> avoid high-traffic transfers during the day.
>
>   

Sieg heil! Sieg heil! Sieg heil! Give me the lash! We love the lash!

I'd give you the finger if you told me as an ISP that I can't download 
CentOS 5 on DVD whenever I wanted, or to watch a Netflix movie I paid 
for. How you remain in business acting like a jack booted thug is beyond me.

-Ron


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Re: [asterisk-users] OT: What do you guys think of this?

[Doug]

At 12:34 12/1/2008, Alex Balashov wrote:
 >http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
 >
 >FUD?  Interesting?  Boring?  New news?  Old news?

Hmmm.  When our users are pounding the network
with BitTorrent traffic, we just shut them down
and wait for them to complain.  It's against our
Acceptable Use Policy, and causes all sorts of
VOIP headaches.

Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me.  We tell
our customers that they are not allowed to
download copyrighted material.  But for other,
legal BitTorrent transfers, we suggest that
they use the scheduling feature of uTorrent to
avoid high-traffic transfers during the day.




 >
 >--
 >Alex Balashov
 >Evariste Systems
 >Web: http://www.evaristesys.com/
 >Tel: (+1) (678) 954-0670
 >Direct : (+1) (678) 954-0671
 >Mobile : (+1) (706) 338-8599
 >
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[asterisk-users] Asternic Call Center and Asterisk 1.4 Queues

[JR Richardson]

Hi All,

I'm testing the Asterinic Call Center Queue Log Analizer.  Working ok
except for realtime monitoring.  The page updates queue summary and
calls waiting, but not Agent status.  When an agent is (busy) in
[asterisk> queue show], the 'state' of the agent in agent status on
the web page does not change, always shows 'not in use'.  The page
does update with 'Last In Call' info after hangup of a call.

Any ideas?

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] [OT] Re: Threading etc.

[Wilton Helm]

Thanks.  I have them going to their own folder now which is a great help.  It 
doesn't appear that OE can organize them in the folder, though.  I does a good 
job in a news folder, but doesn't seem to have that ability in a mail folder.  
Nor does it create a usable To field for replies.  I still have to change that 
every time.

Wilton
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Re: [asterisk-users] Inbound calls from Asterisk to Asterisk with SIP"Forbidden" from '"asterisk"

[Danny Nicholas]

You shouldn't "open text" your password.   Shouldn't IP on Asterisk 2 be
1.2.3.4?

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin
Sent: Monday, December 01, 2008 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound calls from Asterisk to Asterisk with
SIP"Forbidden" from '"asterisk"

 

Please help.

Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

Asterisk 2 sip.conf
  GNU nano 1.3.12  File: sip_custom.conf

[VoipDirect777821]
type=friend
host=141.122.139
username=VoipDirect777821
secret=wsPiOov8830
accountcode=5260477782
amaflags=billing
context=Incomming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

sip show peers shows both as registered.

this is the error when try and place a call from Asterisk 1 to Asterisk 2:

- Executing [EMAIL PROTECTED]:1] Dial("Console/dsp", 
"SIP/VoipDirect777821|60|") in
new stack
-- Called VoipDirect777821
[Dec  1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite:
Received response: "Forbidden" from '"asterisk"
;tag=as070b02e2'
-- SIP/VoipDirect777821-0876c360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("Console/dsp", "") in new stack
  == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp'
 << Hangup on console >>

I get the same error even if I include this on Asterisk 1:
register => VoipDirect777821:[EMAIL PROTECTED]

Please help 

 

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Re: [asterisk-users] Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"

[Rafael Canchola]

You can use next parameter:
Fromuser = VoipDirect777821



At 04:23 p.m. 01/12/2008, Shaun Wingrin wrote:

Please help.

Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

Asterisk 2 sip.conf
  GNU nano 1.3.12  File: sip_custom.conf

[VoipDirect777821]
type=friend
host=141.122.139
username=VoipDirect777821
secret=wsPiOov8830
accountcode=5260477782
amaflags=billing
context=Incomming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

sip show peers shows both as registered.

this is the error when try and place a call from Asterisk 1 to Asterisk 2:

- Executing [EMAIL PROTECTED]:1] Dial("Console/dsp", 
"SIP/VoipDirect777821|60|") in new stack

-- Called VoipDirect777821
[Dec  1 23:20:21] WARNING[25399]: chan_sip.c:12334 
handle_response_invite: Received response: "Forbidden" from 
'"asterisk" ;tag=as070b02e2'

-- SIP/VoipDirect777821-0876c360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("Console/dsp", "") in new stack
  == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp'
 << Hangup on console >>

I get the same error even if I include this on Asterisk 1:
register => VoipDirect777821:[EMAIL PROTECTED]

Please help

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--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 14
VoIP 523663801
d00d! cyberalph
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[asterisk-users] Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"

[Shaun Wingrin]

Please help.

Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

Asterisk 2 sip.conf
  GNU nano 1.3.12  File: sip_custom.conf

[VoipDirect777821]
type=friend
host=141.122.139
username=VoipDirect777821
secret=wsPiOov8830
accountcode=5260477782
amaflags=billing
context=Incomming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

sip show peers shows both as registered.

this is the error when try and place a call from Asterisk 1 to Asterisk 2:

- Executing [EMAIL PROTECTED]:1] Dial("Console/dsp", 
"SIP/VoipDirect777821|60|") in new stack
-- Called VoipDirect777821
[Dec  1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite: 
Received response: "Forbidden" from '"asterisk" ;tag=as070b02e2'
-- SIP/VoipDirect777821-0876c360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("Console/dsp", "") in new stack
  == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp'
 << Hangup on console >>

I get the same error even if I include this on Asterisk 1:
register => VoipDirect777821:[EMAIL PROTECTED]

Please help
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Tzafrir Cohen]

On Mon, Dec 01, 2008 at 01:43:17PM -0700, Wilton Helm wrote:
> >Try to use a decent mailer that does not break threading.
> 
> 
> This is an opportunity for me to ask a question regarding this mailing 
> list.  I've worked with several other groups using a variety of 
> communications techniques from Web based to news reader based, but 
> never anything like this.
> 
> Due to my lack of experience (and/or wrong tools) this looks totally 
> like chaos.  Asterisk messages come in with no apparent thread 
> ordering other than noticing the subject line.  They are all mixed up 
> with my general E-Main (and spam, which they exceed in volume).  I 
> can't reply to them, as the To line generated isn't viable.
> 
> Is there some organizational structure I am missing here?  I am using 
> XP and OE 6 for E-Mail, which may not be that friendly towards this 
> type of situation.  It does have a very good newsgroup reader that I 
> regularly use for another group I am part of, and I would be much 
> better served by that than by having it all land in my general in-box 
> at the rate of 100 a day, all mixed up with my personal and business 
> correspondence.

A. Give Thunderbird a shot. It does a great job with threads.

B. High-volume mailing list should go into their own folder. Don't let a
flame-war in asterisk-users distract you from your work.

You can easily set up a filter rule by the List-ID header. This sends
only messages sent from the mailing list server directly to that folder,
and not any follow-ups.

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

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[asterisk-users] [OT] Re: Threading etc.

[Philipp Kempgen]

Philipp Kempgen schrieb:

> And in most MUAs or servers you can define a filter / processing
> rule so you can store mails matching a certain rule (
> List-Id header contains "asterisk-users.lists.digium.com" or
> To contains "asterisk-users@lists.digium.com" or
> Sender contains "[EMAIL PROTECTED]")
> in a separate (IMAP or local) folder.

Looks like you can do that in Outlook too.
http://www.google.de/search?q=outlook+express+message+rules


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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[asterisk-users] "OE" (was: Re: Dahdi, b410p and looping from 1 port to another)

[Philipp Kempgen]

Tilghman Lesher schrieb:
> On Monday 01 December 2008 02:43:17 pm Wilton Helm wrote:

>> I am using
[...]
>> OE 6 for E-Mail

> Never heard of that mailer.

:-)

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] MySQL Error Message

[Barton Fisher]

> Barton Fisher wrote:
>> Can some tell me what this warnings means?
>> The dialplan works, but I get these warnings every once in a while:
>>
>
> I'm guessing that some times the caller-id is blank.  I got tired of
> those errors and did the following before the query:
>
> exten => s,1,GotoIf($["${CALLERID(num)}" = "" ]?2:3)
> exten => s,n,Set(CALLERID(all)=Restricted <0>)
> exten => s,1,Noop()
> exten => s,n,MYSQL(Connect connid localhost userid password database)
>
>
> Doug
>

Sorry, I removed those lines before I posted dialplan - I do check to see if 
there is some value before I write
Here's another case:

[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Identifier 0, 
identifier_type 2 not found in identifier list
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Invalid result 
identifier 0 passed in aMYSQL_clear
[Dec 1 13:48:15] VERBOSE[14017] logger.c: [5412254900-inn] [8006848429] 
[update-cnam] NAME Updated=[8006848429 800 Service ]
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Identifier 4, 
identifier_type 2 not found in identifier list
[Dec 1 13:48:15] WARNING[14017] app_addon_sql_mysql.c: Invalid result 
identifier 4 passed in aMYSQL_clear
[Dec 1 13:48:15] VERBOSE[14017] logger.c: [5412254900-inn] [8006848429] 
Database Marketing [5412254900] [8006848429] [800 Service ]

any ideas?

Bart 


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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Wilton Helm]

>Never heard of that mailer.  You might try using Kontact under Kubuntu, as it
>has reasonable defaults.


Outlook Express, the default mail handler on Windows XP.  I only have one Linux 
computer at present and it is dedicated to Asterisk and file backups and web 
serving, etc., so whatever I use has to work under Windows (I use a lot of aps 
and exchange files with customers that require Windows to be my normal OS).

My real question was more how this is supposed to work.  I've been reading 
posts and sending a few, and seem to be functioning OK, but I don't know if I'm 
doing anything right, and using the tools I'm using it is 10 to 1 more chaotic 
and less useful than anything else I've ever done, so I assume other know a 
better way.

Wilton
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[asterisk-users] Threading etc. (was: Re: Dahdi, b410p and looping from 1 port to another)

[Philipp Kempgen]

Wilton Helm schrieb:

> This is an opportunity for me to ask a question regarding this mailing list.  
> I've worked with several other groups using a variety of communications 
> techniques from Web based to news reader based, but never anything like this.
> 
> Due to my lack of experience (and/or wrong tools) this looks totally like 
> chaos.  Asterisk messages come in with no apparent thread ordering other than 
> noticing the subject line.  They are all mixed up with my general E-Main (and 
> spam, which they exceed in volume).  I can't reply to them, as the To line 
> generated isn't viable.
> 
> Is there some organizational structure I am missing here?  I am using XP and 
> OE 6 for E-Mail, which may not be that friendly towards this type of 
> situation.  It does have a very good newsgroup reader that I regularly use 
> for another group I am part of, and I would be much better served by that 
> than by having it all land in my general in-box at the rate of 100 a day, all 
> mixed up with my personal and business correspondence.

Most mail readers I know support a threaded view (they read the
References header).
And in most MUAs or servers you can define a filter / processing
rule so you can store mails matching a certain rule (
List-Id header contains "asterisk-users.lists.digium.com" or
To contains "asterisk-users@lists.digium.com" or
Sender contains "[EMAIL PROTECTED]")
in a separate (IMAP or local) folder.

Outlook is one of the worst mail clients because it gets every-
thing wrong every time.  :-)

In fact the difference between a newsgroup and an email list is
basically non-existent. Newsgroups scale much better but that's
it.

Hijacking treads instead of starting a new one for a new topic is
bad because for people who use a threaded view the messages show
up in the wrong thread. And it's bad for the archives as well.

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] MySQL Error Message

[Doug Lytle]

Barton Fisher wrote:
> Can some tell me what this warnings means?
> The dialplan works, but I get these warnings every once in a while:
>  

I'm guessing that some times the caller-id is blank.  I got tired of 
those errors and did the following before the query:

exten => s,1,GotoIf($["${CALLERID(num)}" = "" ]?2:3)
exten => s,n,Set(CALLERID(all)=Restricted <0>)
exten => s,1,Noop()
exten => s,n,MYSQL(Connect connid localhost userid password database)


Doug


-- 
 
Ben Franklin quote:

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


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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Tilghman Lesher]

On Monday 01 December 2008 02:43:17 pm Wilton Helm wrote:
> >Try to use a decent mailer that does not break threading.
>
> This is an opportunity for me to ask a question regarding this mailing
> list.  I've worked with several other groups using a variety of
> communications techniques from Web based to news reader based, but never
> anything like this.
>
> Due to my lack of experience (and/or wrong tools) this looks totally like
> chaos.  Asterisk messages come in with no apparent thread ordering other
> than noticing the subject line.  They are all mixed up with my general
> E-Main (and spam, which they exceed in volume).  I can't reply to them, as
> the To line generated isn't viable.
>
> Is there some organizational structure I am missing here?  I am using XP
> and OE 6 for E-Mail, which may not be that friendly towards this type of
> situation.  It does have a very good newsgroup reader that I regularly use
> for another group I am part of, and I would be much better served by that
> than by having it all land in my general in-box at the rate of 100 a day,
> all mixed up with my personal and business correspondence.

Never heard of that mailer.  You might try using Kontact under Kubuntu, as it
has reasonable defaults.

-- 
Tilghman

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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject

[Olivier]

2008/12/1 Andrew Thomas <[EMAIL PROTECTED]>

>  The DAHDI docs actually include the 'steps' that used to be in Zaptel's
> 'make b410p'.  These steps involve downloading and compiling mISDN.  Why
> re-invent the wheel?
>
in 1.6.1rc5, mISDN is optional ...

>
>
> Just a thought J.
>
I will compare both ASAP ...

>
>
> As for the 100ohm termination bit – it's simply changing a couple of dip
> switches on the b410p card (as described in the manual).
>
That's the point : why is it necessary to change to 100ohm termination ?

>
>
> Andy
>
>
>
>
>
>
>
> -Original Message-
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Olivier
> *Sent:* 01 December 2008 17:54
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p
> and looping from 1 port to another - Email found in subject -Email found in
> subject - Email found in subject
>
>
>
>
>
> 2008/12/1 Andrew Thomas <[EMAIL PROTECTED]>
>
> Apart from you were dialling out on your inbound context and vice-versa.
>
>
>
> The best advice I can give now is to change to mISDN – as this is proven to
> work with v1.4 and v1.6.
>
>   I wanted to try the Digium B410P card with Dahdi as those are now
> supported in 1.6.X.
> Hopefully, it would simplify software configuration as both asterisk and
> dahdi are supported by Asterisk community.
> As soon as I can get this card running with dahdi, I will give a try with
> mISDN.
>
> Fortunately, you still can use misdn with B410P w/ 1.6.X.
>
>
>
> Actually – have you tried putting the 100ohm termination on for your NT
> port?
>
>I didn't dare to try as I'm not familiar with concepts behind that.
>
>
>
> I need to do that with mISDN as it only allows ptmp for ISDN extensions.
>
>
>
> Cheers
>
> Andy
>
>
>
>
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[asterisk-users] MySQL Error Message

[Barton Fisher]

Can some tell me what this warnings means? 
The dialplan works, but I get these warnings every once in a while:

Log:

[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Identifier 0, 
identifier_type 2 not found in identifier list
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Invalid result 
identifier 0 passed in aMYSQL_clear
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Identifier 4, 
identifier_type 2 not found in identifier list
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Invalid result 
identifier 4 passed in aMYSQL_clear

DialPlan:

 exten => s,1,Noop()
 exten => s,n,MYSQL(Connect connid localhost userid password database)
 exten => s,n,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ cnam\ WHERE\ 
ani=\'${CALLERID(number)}\')
 exten => s,n,MYSQL(Fetch fetchid ${resultid} name)
 exten => s,n,MYSQL(Clear ${resultid})
 exten => s,n,MYSQL(Disconnect ${connid})

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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Olivier]

Setup is:

/etc/dahdi/system.conf:
span=1,0,0,ccs,ami
bchan=1,2
hardhdlc=3
span=2,1,0,ccs,ami
bchan=4,5
hardhdlc=6

/etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
context=isdntrunk
switchtype=euroisdn
group=1
immediate=no
signalling=bri_cpe
channel=>1-2
group=2
signalling=bri_net
channel=>4-5

/etc/asterisk/extensions.ael:
context isdntrunk {
_X. => {
NoOp(Call from ${CALLERID(num)});
Dial(SIP/7530);
Hangup();
};
};

context mylocal {
_10. => {
ChanIsAvail(DAHDI/g1);
NoOp(${AVAILCHAN});
NoOp(${AVAILSTATUS});
ChanIsAvail(DAHDI/g2);
NoOp(${AVAILCHAN});
NoOp(${AVAILSTATUS});
Dial(DAHDI/g2/${EXTEN:1});
Hangup();
};
_753[0356] => {
Dial(SIP/${EXTEN});
Hangup();
};
};


When dialing from 7533 to 10123456789 :

-- Executing [EMAIL PROTECTED]:1] ChanIsAvail("SIP/7533-0a0c7968",
"DAHDI/g1") in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp("SIP/7533-0a0c7968", "") in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp("SIP/7533-0a0c7968", "34") in new
stack
-- Executing [EMAIL PROTECTED]:4] ChanIsAvail("SIP/7533-0a0c7968",
"DAHDI/g2") in new stack
-- Hungup 'DAHDI/4-1'
-- Executing [EMAIL PROTECTED]:5] NoOp("SIP/7533-0a0c7968", "DAHDI/4-1")
in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp("SIP/7533-0a0c7968", "0") in new
stack
-- Executing [EMAIL PROTECTED]:7] Dial("SIP/7533-0a0c7968",
"DAHDI/g2/0123456") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0123456
[Dec  1 21:56:16] NOTICE[3948]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Dec  1 21:56:16] NOTICE[3948]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Dec  1 21:56:16] NOTICE[3949]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 2
  == Primary D-Channel on span 2 down
[Dec  1 21:56:17] WARNING[3949]: chan_dahdi.c:3156 pri_find_dchan: No
D-channels available!  Using Primary channel 6 as D-channel anyway!
[Dec  1 21:56:17] WARNING[3955]: app_dial.c:1067 wait_for_answer: Unable to
forward voice or dtmf
-- Hungup 'DAHDI/4-1'

and with pri set debug on span 2, my console if rapidly fulling with tens of
lines like:

-- Got SABME from cpe peer.
Sending Unnumbered Acknowledgement
q921.c:782 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:733 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Got SABME from cpe peer.
Sending Unnumbered Acknowledgement
q921.c:782 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED


 dahdi show status
Description  Alarms  IRQbpviol CRC4   Fra
Codi Options  LBO
B4XXP (PCI) Card 0 Span 1OK  0  0  0  CCS
AMI  YEL  0 db (CSU)/0-133 feet (DSX-1)
B4XXP (PCI) Card 0 Span 2OK  0  0  0  CCS
AMI  YEL  0 db (CSU)/0-133 feet (DSX-1)
B4XXP (PCI) Card 0 Span 3RED 0  0  0  CAS
Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
B4XXP (PCI) Card 0 Span 4RED 0  0  0  CAS
Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)

dahdi show version
DAHDI Version: 2.1.0-rc5 Echo Canceller:


# lsmod
Module  Size  Used by
dahdi_dummy 3236  0
xpp_usb13676  0
xpp   131296  1 xpp_usb
wcb4xxp77828  6
wctdm  32876  0
wcfxo   9888  0
wctdm24xxp108320  0
wcte11xp   20992  0
wct1xxp11520  0
wcte12xp   58560  0
wct4xxp   291680  0
dahdi 182792  22
dahdi_dummy,xpp,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp
firmware_class  6816  3 wctdm24xxp,wcte12xp,wct4xxp
crc_ccitt   2080  1 dahdi




Observations:
span1 channels (ie NT port) are always busy (cause 34) while span2 channels
(ie IE port) are available,
before hanging up, I can hear a weird noise in my phone handset.

Question:
1. Why in the first place, should spans 1 and 2 get different availability
as both are connected to each other ?
2. Is bri_net the protocol name for TE ports and bri_cpe for
NT-point-to-point ?
3. Is it normal that dahdi_dummy is not used ?
4. Any idea ?

Regards
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Wilton Helm]

>Try to use a decent mailer that does not break threading.


This is an opportunity for me to ask a question regarding this mailing list.  
I've worked with several other groups using a variety of communications 
techniques from Web based to news reader based, but never anything like this.

Due to my lack of experience (and/or wrong tools) this looks totally like 
chaos.  Asterisk messages come in with no apparent thread ordering other than 
noticing the subject line.  They are all mixed up with my general E-Main (and 
spam, which they exceed in volume).  I can't reply to them, as the To line 
generated isn't viable.

Is there some organizational structure I am missing here?  I am using XP and OE 
6 for E-Mail, which may not be that friendly towards this type of situation.  
It does have a very good newsgroup reader that I regularly use for another 
group I am part of, and I would be much better served by that than by having it 
all land in my general in-box at the rate of 100 a day, all mixed up with my 
personal and business correspondence.

Thanks,
Wilton
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Tzafrir Cohen]

SPAM SPAM SPAM SPAM[0]

Please:

1. Try to use a decent mailer that does not break threading.
2. Don't reply to messages you get with the extra word "[SPAM]" in the
subject. This breaks thing even worse, as even the subject line is
changed.

Sanity has been restored to the subject.

Back to your message:

On Mon, Dec 01, 2008 at 07:46:38PM -, Andrew Thomas wrote:
> The DAHDI docs actually include the 'steps' that used to be in Zaptel's
> 'make b410p'.  These steps involve downloading and compiling mISDN.  Why
> re-invent the wheel?

DAHDI's makefile no longer includes the legacy b410p target (nither in
-linux nor in -tools), and the READMEs have been updated accordingly.

> As for the 100ohm termination bit - it's simply changing a couple of dip
> switches on the b410p card (as described in the manual).

The termination is completely unrequired for a single ptp connection.


[0] SPAM (in uppercase) is a trademark of HORNEL. See
http://spam.com/legal/terms/

And yes, the subject line was beginning to look like:
http://www.detritus.org/spam/skit.html

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

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Re: [asterisk-users] CDR Desgin

[Philipp Kempgen]

JD schrieb:

> As to the idea of piping to a deamon via socket or dbus: how would 
> asterisk behave if the daemon froze or worse, it lagged?

Asterisk could write the CEL events to the database and either
on every event or after some kind of final event send a signal
to the socket, i.e. write a byte (or the uniqueid, ...) and
ignore pipe errors so it doesn't receive SIGPIPEs.

   Philipp Kempgen

-- 
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Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject

[Andrew Thomas]

The DAHDI docs actually include the 'steps' that used to be in Zaptel's
'make b410p'.  These steps involve downloading and compiling mISDN.  Why
re-invent the wheel?

 

Just a thought :-).

 

As for the 100ohm termination bit - it's simply changing a couple of dip
switches on the b410p card (as described in the manual).

 

Andy

 

 

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: 01 December 2008 17:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p
and looping from 1 port to another - Email found in subject -Email found
in subject - Email found in subject

 

 

2008/12/1 Andrew Thomas <[EMAIL PROTECTED]>

Apart from you were dialling out on your inbound context and
vice-versa.

 

The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.

I wanted to try the Digium B410P card with Dahdi as those are now
supported in 1.6.X.
Hopefully, it would simplify software configuration as both asterisk and
dahdi are supported by Asterisk community.
As soon as I can get this card running with dahdi, I will give a try
with mISDN.

Fortunately, you still can use misdn with B410P w/ 1.6.X.

 

Actually - have you tried putting the 100ohm termination
on for your NT port?

I didn't dare to try as I'm not familiar with concepts behind that.

 

I need to do that with mISDN as it only allows ptmp for
ISDN extensions.

 

Cheers

Andy

 


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Re: [asterisk-users] Mitel 3300

[Ron Joffe]

We have one connected.

What's your question ?

On Monday 01 December 2008 13:49, Mark Bergen wrote:
> Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice
> together?
>
> Mark Bergen
> Information Systems Manager
> Number TEN Architectural Group
> Winnipeg - 204.942.0981
> Victoria - 250.360.2106
> www.numberten.com


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Re: [asterisk-users] CDR Desgin

[JD]

Daniel Hazelbaker wrote:
> On Dec 1, 2008, at 9:07 AM, JD wrote:
>
>   
>> Steve Murphy wrote:
>> 
>>> Freddi--
>>>
>>> Very interesting. Brian Degenhardt had some code we just gave some
>>> thought
>>> to, wherein we determine if the last channel involved in a linkedID  
>>> set
>>> has been closed. If so, then the entire set is finished. We can use  
>>> this
>>> facility to get you a closing attribute, that could be added to the  
>>> last
>>> CDR emmitted for that set; OR, we could just emit another CDR with  
>>> type
>>> CLOSE or FINAL or something, that signals the end of the chain.
>>>
>>> murf
>>>
>>>   
>> Just thinking out loud: how about a feature wherein, after the FINAL  
>> is
>> sent, asterisk can
>>  1. create a temp text file with just those entries, and
>>  2. launch a user-made script.
>>
>> cdr_manager.conf
>>  [general]
>>  legparsecmd=/usr/local/bin/my_parser.pl
>>
>> wherein the linkedID is passed as the first parameter and the text  
>> file
>> name&path as the second
>>
>> Ignore this suggestion if it horribly complicates things.
>> 
>
> Hmm.. While I normally like having this kind of "instant  
> notification", I could see this as a very big problem for larger  
> installations.  Most OS's are not so great at launching new tasks, and  
> on a heavily loaded system that could easily be a number of tasks  
> launched every second, each doing a lot of database queries.  Perhaps  
> a different approach would be to have a field that can be set to show  
> that the record(s) have been parsed into whatever standard CDR format  
> you want.  This may or may not make more sense as a separate table  
> with just a list of linkedid's that have been parsed.
>
> Daniel

I agree with your and Phillips take on that. It would definitely have to 
be optional; perhaps disabled by default.

And the launcher would need to be generic. That way, it could be a 
compiled C program, etc.

Even if this were implemented, I would hope that the optional 'parsecmd' 
or whatever it's called would
merely add to the function, not turn the main CDR logging off. If 
nothing else, I'd want the main CDR text file still made for debugging 
purposes.

As to the idea of piping to a deamon via socket or dbus: how would 
asterisk behave if the daemon froze or worse, it lagged? I'd hate to 
create a dependency.

(This, by the way, is the core problem I always had with some of the 
OBDC stuff for asterisk. It doesn't handle partial failure well.)

John

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Re: [asterisk-users] CDR Design

[Freddi Hansen]

> On Dec 1, 2008, at 9:07 AM, JD wrote:
>
>> Steve Murphy wrote:
>>>
>>> Freddi--
>>>
>>> Very interesting. Brian Degenhardt had some code we just gave some
>>> thought
>>> to, wherein we determine if the last channel involved in a linkedID set
>>> has been closed. If so, then the entire set is finished. We can use 
>>> this
>>> facility to get you a closing attribute, that could be added to the 
>>> last
>>> CDR emmitted for that set; OR, we could just emit another CDR with type
>>> CLOSE or FINAL or something, that signals the end of the chain.
>>>
>>> murf
>>>
>> Just thinking out loud: how about a feature wherein, after the FINAL is
>> sent, asterisk can
>>  1. create a temp text file with just those entries, and
>>  2. launch a user-made script.
>>
>> cdr_manager.conf
>>  [general]
>>  legparsecmd=/usr/local/bin/my_parser.pl
>>
>> wherein the linkedID is passed as the first parameter and the text file
>> name&path as the second
>>
>> Ignore this suggestion if it horribly complicates things.
>
> Hmm.. While I normally like having this kind of "instant 
> notification", I could see this as a very big problem for larger 
> installations.  Most OS's are not so great at launching new tasks, and 
> on a heavily loaded system that could easily be a number of tasks 
> launched every second, each doing a lot of database queries.  Perhaps 
> a different approach would be to have a field that can be set to show 
> that the record(s) have been parsed into whatever standard CDR format 
> you want.  This may or may not make more sense as a separate table 
> with just a list of linkedid's that have been parsed.
>
> Daniel
I think that e.g. a socket would be preferable.  You do not have the 
load of launching new a task for each cdr and the process listening on 
the socket would normally run at lower priority than Astrisk itself. If 
gives the freedom to send unprocessed 'leg-information' to a cdr backend 
process on another server if  heavy loaded enviroment or just process it 
local if it's a small installation.
To me it looks new cdr spec from murf would be a big step forward for 
Asterisk.

Freddi
   



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Re: [asterisk-users] OT: What do you guys think of this?

[Christopher Dobbs]

Sounds possible, but as a user of uTorrent, I have yet to see this "feature"
It may simply be that I havnt looked hard enough.

I can say, that I still have to have a tcp port routed for uTorrent to work
properly.

I may post an update, If I notice a change in this behavour.

--Christopher Dobbs


On Mon, Dec 1, 2008 at 10:34 AM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
>
> FUD?  Interesting?  Boring?  New news?  Old news?
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
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[asterisk-users] Mitel 3300

[Mark Bergen]

Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice together?

Mark Bergen
Information Systems Manager
Number TEN Architectural Group
Winnipeg - 204.942.0981
Victoria - 250.360.2106
www.numberten.com
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[asterisk-users] OT: What do you guys think of this?

[Alex Balashov]

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/

FUD?  Interesting?  Boring?  New news?  Old news?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789

[Olivier]

Thanks for replying !
I've added a link here
http://www.voip-info.org/wiki/view/Channels+and+Groups to
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels.


2008/12/1 Mark Michelson <[EMAIL PROTECTED]>

> Olivier wrote:
> > Hello,
> >
> > Groups in asterisk are summarized here
> > (http://www.voip-info.org/wiki/view/Channels+and+Groups).
> > Is there any difference between DAHDI/G1/0123456789 and
> > DAHDI/g1/0123456789 (as I've been advised in another thread, to switch
> > from one notation to the other and I can't see the reason behind that) ?
> >
> >
> > Regards
> >
>
> Using 'g' means that channels will be used starting with the lowest
> available
> number and then counting up. Starting with 'G' means to start at the
> highest
> channel number and count down. I believe that incoming calls will use the
> lowest
> numbered channels first, so if your outgoing calls start with the higher
> numbers
> it decreases the chance for "glare" on calls.
>
> Mark Michelson
>
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Re: [asterisk-users] CPU Usage

[Wilton Helm]

>When system is busy, asterisk uses 99.9% cpu.
>I want asterisk to use more 100% cpu to process more calls.


Am I in a time warp?  Is this April 1 in disguise?

The difference between 99.9% and 100% is not discernable.  The other .1% is 
probably used determining that asterisk is using the rest.  Even if it were, 
this would be an OS issue.

Wilton
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Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789

[Mark Michelson]

Olivier wrote:
> Hello,
> 
> Groups in asterisk are summarized here 
> (http://www.voip-info.org/wiki/view/Channels+and+Groups).
> Is there any difference between DAHDI/G1/0123456789 and 
> DAHDI/g1/0123456789 (as I've been advised in another thread, to switch 
> from one notation to the other and I can't see the reason behind that) ?
> 
> 
> Regards
> 

Using 'g' means that channels will be used starting with the lowest available 
number and then counting up. Starting with 'G' means to start at the highest 
channel number and count down. I believe that incoming calls will use the 
lowest 
numbered channels first, so if your outgoing calls start with the higher 
numbers 
it decreases the chance for "glare" on calls.

Mark Michelson

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Re: [asterisk-users] CDR Desgin

[Daniel Hazelbaker]

On Dec 1, 2008, at 9:07 AM, JD wrote:

> Steve Murphy wrote:
>>
>> Freddi--
>>
>> Very interesting. Brian Degenhardt had some code we just gave some
>> thought
>> to, wherein we determine if the last channel involved in a linkedID  
>> set
>> has been closed. If so, then the entire set is finished. We can use  
>> this
>> facility to get you a closing attribute, that could be added to the  
>> last
>> CDR emmitted for that set; OR, we could just emit another CDR with  
>> type
>> CLOSE or FINAL or something, that signals the end of the chain.
>>
>> murf
>>
> Just thinking out loud: how about a feature wherein, after the FINAL  
> is
> sent, asterisk can
>  1. create a temp text file with just those entries, and
>  2. launch a user-made script.
>
> cdr_manager.conf
>  [general]
>  legparsecmd=/usr/local/bin/my_parser.pl
>
> wherein the linkedID is passed as the first parameter and the text  
> file
> name&path as the second
>
> Ignore this suggestion if it horribly complicates things.

Hmm.. While I normally like having this kind of "instant  
notification", I could see this as a very big problem for larger  
installations.  Most OS's are not so great at launching new tasks, and  
on a heavily loaded system that could easily be a number of tasks  
launched every second, each doing a lot of database queries.  Perhaps  
a different approach would be to have a field that can be set to show  
that the record(s) have been parsed into whatever standard CDR format  
you want.  This may or may not make more sense as a separate table  
with just a list of linkedid's that have been parsed.

Daniel

>
>
> John


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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject

[Olivier]

2008/12/1 Andrew Thomas <[EMAIL PROTECTED]>

>   Apart from you were dialling out on your inbound context and vice-versa.
>
>
>
> The best advice I can give now is to change to mISDN – as this is proven to
> work with v1.4 and v1.6.
>
> I wanted to try the Digium B410P card with Dahdi as those are now supported
in 1.6.X.
Hopefully, it would simplify software configuration as both asterisk and
dahdi are supported by Asterisk community.
As soon as I can get this card running with dahdi, I will give a try with
mISDN.

Fortunately, you still can use misdn with B410P w/ 1.6.X.

>
>
> Actually – have you tried putting the 100ohm termination on for your NT
> port?
>
> I didn't dare to try as I'm not familiar with concepts behind that.

>
>
> I need to do that with mISDN as it only allows ptmp for ISDN extensions.
>
>
>
> Cheers
>
> Andy
>
>
>
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Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789

[Dave Fullerton]

Olivier wrote:
> Hello,
> 
> Groups in asterisk are summarized here (
> http://www.voip-info.org/wiki/view/Channels+and+Groups).
> Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
> (as I've been advised in another thread, to switch from one notation to the
> other and I can't see the reason behind that) ?
> 
> 
> Regards

Assuming nothing has changed from Zaptel to DAHDI, the difference can 
found here:

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

Basically, the lowercase g chooses the lowest number available channel 
from the group where the uppercase G chooses the highest number 
available channel. This is used to reduce glare on analog or T1 
(non-PRI) channels that are part of a hunt group.

-Dave

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Re: [asterisk-users] CDR Design

[Steve Murphy]

On Mon, 2008-12-01 at 10:55 -0600, JD wrote:
> Steve Murphy wrote:
> > [...]
> 
> I love it! You will have it done later today, correct? (joke.)
> 
> Just a non-technical/social suggestion: don't call this CDR. Call it 
> "Enhanced CEL" or something like that.
> 
> Why?: because otherwise you will forever get arguments about it. 
> Traditionally, a CDR is one line per call; all inclusive. And as you 
> know, that is a horrible standard for todays complex systems; but it is 
> what it is.
> 
> So, perhaps Asterisk should not build native CDR at all. It should build 
> your "Enhanced CEL". A separate perl/ruby/etc script could be included 
> in the Asterisk distribution that build the "CDRs" after the fact. Or 
> multiple CDR scripts based on the flavor-of-the-day of what CDR means.
> 
> Just my thoughts. I very much look forward to your code. This will make 
> my life so much easier.
> 
> John

John--

Point well taken, but if I call it anything but 
CDR, people will go "Huh?".

CEL is per-event, but CDR's are event-groups. I 
think we are safe to keep calling it CDR, because 
I perceive that at least some of the big-name
pbx vendors are generating much more than simple 
line-per-call records and they still call them CDR's.

Using a db to make sense of CEL events for billing 
purposes is extraordinarily difficult, as the strings 
of events are reported as they occur, and with
multiple calls going on simultaneously, the events are
interlaced. You can pull out single threads by sorting 
on the linkedid; but still you have strings of events 
among multiple channels that will not be obviously easy to 
decode, especially by concocting SQL queries. Thus, the 
need to provide some sense via CDR's.

And, yes, I agree with you. I will try to make it so 
a CEL->CDR generator could be run either 'real time'
inside asterisk (via make menuselect choice), (and 
using your selection of existing backends), AND, I 
can try to provide a stand-alone option that could 
be run on CEL events that were logged to one of the 
CEL backends; probably just one of the plain-text
ones.

murf

-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789

[Olivier]

Hello,

Groups in asterisk are summarized here (
http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
(as I've been advised in another thread, to switch from one notation to the
other and I can't see the reason behind that) ?


Regards
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Olivier]

2008/12/1 Tzafrir Cohen <[EMAIL PROTECTED]>

> On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
> > Hi,
> >
> > Hardware is b410p with :
> > - span1 jumpers set to NT (no 100 ohm termination)
> > - span2 jumpers set to TE (no 100 ohm termination)
> > - straight cable from port1 to port2
> >
> > Scheme:
> >
> > 
> >1--
> >|  |   <-- cat5 patch cord
> >2--
> >B410P  |
> >3
> >|
> >4
> >|
> > 
> >
> >
> >
> > /etc/dahdi/system.conf :
> > span=1,0,0,ccs,ami
> > bchan=1,2
> > dchan=3
> > span=2,1,0,ccs,ami
> > bchan=4,5
> > dchan=6
> >
> >
> > /etc/asterisk/chan_dahdi.conf:
> > context=isdntrunk
> > group=1
> > callgroup=1
> > pickupgroup=1
> > immediate=no
> >
> > signalling=bri_cpe
> > channel=>1-2
> > signalling=bri_net
> > channel=>4-5
>
> I'd seperate the NT and TE to different groups to make sure they don't
> run over each other. Put the NT in group=2, and you'd be sure you dial
> from the TE.
>
> More importantly: what do you see in:
>
>  pri debug span 1
>
> If you see nothing there: do you see anything in
>
>  pri intense debug span 1


I re-installed from scratch.

Now, I've got :

-- Executing [EMAIL PROTECTED]:1] Dial("SIP/7533-08f8c2a8",
"DAHDI/G1/0123456789") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/0123456789
[Dec  1 18:20:00] NOTICE[5770]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Dec  1 18:20:00] NOTICE[5771]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 2
[Dec  1 18:20:01] NOTICE[5770]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Dec  1 18:20:02] NOTICE[5770]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Dec  1 18:20:02] NOTICE[5771]: chan_dahdi.c:10781 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 2
  == Primary D-Channel on span 2 down
[Dec  1 18:20:03] WARNING[5771]: chan_dahdi.c:3156 pri_find_dchan: No
D-channels available!  Using Primary channel 6 as D-channel anyway!
[Dec  1 18:20:03] WARNING[5949]: app_dial.c:1067 wait_for_answer: Unable to
forward voice or dtmf
-- Hungup 'DAHDI/5-1'


While seeing this in console, I can heard a weird noise (a bit like fax
signal but not a fax signal) in calling phone earphones.


I will try to summize in a later mail.
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Re: [asterisk-users] [SPAM] - Re: CDR Desgin - Email found in subject

[Andrew Thomas]

...or something along the lines of a setting a variable (like we do for
MONITOR_EXEC)...



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Re: [asterisk-users] CDR Desgin

[Philipp Kempgen]

JD schrieb:
> Steve Murphy wrote:

>> Brian Degenhardt had some code we just gave some
>> thought
>> to, wherein we determine if the last channel involved in a linkedID set
>> has been closed. If so, then the entire set is finished. We can use this
>> facility to get you a closing attribute, that could be added to the last
>> CDR emmitted for that set; OR, we could just emit another CDR with type
>> CLOSE or FINAL or something, that signals the end of the chain.

> Just thinking out loud: how about a feature wherein, after the FINAL is 
> sent, asterisk can
>   1. create a temp text file with just those entries, and
>   2. launch a user-made script.
> 
> cdr_manager.conf
>   [general]
>   legparsecmd=/usr/local/bin/my_parser.pl
> 
> wherein the linkedID is passed as the first parameter and the text file 
> name&path as the second
> 
> Ignore this suggestion if it horribly complicates things.

I don't think it complicates things but do you really want to load
a Perl interpreter for each and every CDR?
Maybe it would be better to send a signal to a 3rd-party daemon
via dbus or a socket or somesuch.

   Philipp Kempgen

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Re: [asterisk-users] CDR Desgin

[JD]

Steve Murphy wrote:
> On Wed, 2008-11-26 at 01:13 +0100, Freddi Hansen wrote:
>   
>>> To me the obvious answer is to provide a CDR for every call leg so for
>>>   
 A calling B via Asterisk there would be two CDRs produced. It's far
 far easier to disregard the unwanted CDRs than it is to try and
 generate the missing ones and in some cases it's virtually impossible.
 If it's weighed up I think people would vote to have accurate CDRs
 ahead of anything else and if single legs are the best way to do that
 then it's the way it should be done.

 In addition with single leg CDRs it will solve the dilemna about
 acommodating every possible call scenario that I know has caused you a
 lot of consternation over the last 18 months.

 Sure it's a change from the current situation so maybe needs to use
 the standard apporach of a configuration setting to opt in initially
 before becoming the default in the subsequent major release.
 
>>>   
>>>
>>>
>>> OK, Greyman, your logic is solid. If we provide a CDR implementation
>>> that just generates the individual call legs, and ties them together via
>>> the linkedid
>>> (see team/group/newcdr), then both camps should be able to derive the
>>> info
>>> they need for billing, via hopefully not-overly-complex SQL queries to a
>>> backend db.
>>>
>>> I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of
>>> shift.
>>> And, yes, the implementation will make this new approach optional, and
>>> not
>>> default. But, pardon if I make it available via the CEL domain come
>>> implementation time.
>>>
>>>
>>> It should take me a week to rehash my document; perhaps longer (I'm in
>>> bugfix mode, and this borderline development work); in the meantime,
>>> those with decided CDR needs might make their wishes known, if they do
>>> not think this approach will work. Speak now, or forever hold your
>>> peace; or forever complain... or whatever.
>>> If this is particularly distressing to you, perhaps your fears might be
>>> slightly assuaged when you read the details...
>>>   
>>>   
>> I was part of a team that did design a multiservice billing system about 
>> 15 years ago and the solution people seems to agree on here (and me to) 
>> looks pretty much the same i.e one call may consist of several calls 
>> legs. In addition to the linkedid it would be nice to have an indication 
>> in the cdr that tells us that 'this is the lastone on this  linked id'.
>> Our experience was that  we shouldn't  for load reasons work with cdr's 
>> in the immidiate multileg format in the DB. So we did collect cdr's in a 
>> tmp DB until we got the the record with end marker set. We would then 
>> produce our final cdr for the actual service, store it in billing col. 
>> and delete it from the multileg col. When a new service is created we 
>> only have to make a the new customized cdr, we don't have to touch the 
>> generation of the multileg format.  
>>
>> Freddi
>> 
>
> Freddi--
>
> Very interesting. Brian Degenhardt had some code we just gave some
> thought
> to, wherein we determine if the last channel involved in a linkedID set
> has been closed. If so, then the entire set is finished. We can use this
> facility to get you a closing attribute, that could be added to the last
> CDR emmitted for that set; OR, we could just emit another CDR with type
> CLOSE or FINAL or something, that signals the end of the chain.
>
> murf
>   
Just thinking out loud: how about a feature wherein, after the FINAL is 
sent, asterisk can
  1. create a temp text file with just those entries, and
  2. launch a user-made script.

cdr_manager.conf
  [general]
  legparsecmd=/usr/local/bin/my_parser.pl

wherein the linkedID is passed as the first parameter and the text file 
name&path as the second

Ignore this suggestion if it horribly complicates things.

John

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Re: [asterisk-users] CDR Desgin

[Andrew Thomas]

Just seconding Freddi's idea - as it makes perfect sense.  Otherwise we
could quite easily start testing a call that hasn't actually finished
yet.



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Re: [asterisk-users] CDR Design

[JD]

Steve Murphy wrote:
> [...]
>   
> Everyone--
>
> I've just made some major changes to the CDRfix2.rfc.txt 
> file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
> to accommodate the Leg approach instead of a 
> channel-based approach.
>
> Greyman is correct. By cutting the timeline into legs, 
> we avoid all the nasty channel state problems, or so it
> appears thus far. I threw out all the text about 
> channel/peer state, fleshed in all the example 
> cases, etc.
>
> I added a section describing the linkedID field.
>
> I provide a list of CDR record types at the end,
> that will eventually be expanded to describe each
> field that set in that type, and what they mean.
>
> The types so far are:
>
> 3WAY
> AXFER 
> AXFER1
> AXFER2
> BARGE
> BXFER
> CALL
> CONF 
> HOLD 
> PARK
> RECALL
> RECONN
> USER 
> WHISPER   
>
>
> murf
>
>   

I love it! You will have it done later today, correct? (joke.)

Just a non-technical/social suggestion: don't call this CDR. Call it 
"Enhanced CEL" or something like that.

Why?: because otherwise you will forever get arguments about it. 
Traditionally, a CDR is one line per call; all inclusive. And as you 
know, that is a horrible standard for todays complex systems; but it is 
what it is.

So, perhaps Asterisk should not build native CDR at all. It should build 
your "Enhanced CEL". A separate perl/ruby/etc script could be included 
in the Asterisk distribution that build the "CDRs" after the fact. Or 
multiple CDR scripts based on the flavor-of-the-day of what CDR means.

Just my thoughts. I very much look forward to your code. This will make 
my life so much easier.

John

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Re: [asterisk-users] [SPAM] - Re: CDR Design - Email found in subject

[Andrew Thomas]

Hi murf,

Speaking as someone who designs and builds billing platforms, this is
very exciting.

One little thing I have most problems with is the good old fax
detection.  I know that NVFaxDetect et al do actually answer the call
and, therefore, get flagged as ANSWERED in the CDR.

But, if the call never gets answered after the initial detection - then,
to my customer, it is a missed (NO ANSWER) call.

Is there anyway of following that scenario?

In other words:

 Call comes in over DAHDI, ZAP or mISDN channel
 NVFaxDetect samples call
 It's not fax, so call is passed on...
 Call never gets answered and caller hangs up.

I appreciate that the NV suite are not Digium products - but it would be
nice if the new CDR information could at least flag it up as a 'type',
so that I can manually (well, via. my code) trace the call through to
check for human intervention.

Actually, the same would go for v1.6 and it's built in fax detection :).

I hope that makes sense.

Cheers
Andy


Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF 


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Re: [asterisk-users] CDR Desgin

[Steve Murphy]

On Wed, 2008-11-26 at 01:13 +0100, Freddi Hansen wrote:
> >
> > To me the obvious answer is to provide a CDR for every call leg so for
> > > A calling B via Asterisk there would be two CDRs produced. It's far
> > > far easier to disregard the unwanted CDRs than it is to try and
> > > generate the missing ones and in some cases it's virtually impossible.
> > > If it's weighed up I think people would vote to have accurate CDRs
> > > ahead of anything else and if single legs are the best way to do that
> > > then it's the way it should be done.
> > > 
> > > In addition with single leg CDRs it will solve the dilemna about
> > > acommodating every possible call scenario that I know has caused you a
> > > lot of consternation over the last 18 months.
> > > 
> > > Sure it's a change from the current situation so maybe needs to use
> > > the standard apporach of a configuration setting to opt in initially
> > > before becoming the default in the subsequent major release.
> >   
> >
> >
> > OK, Greyman, your logic is solid. If we provide a CDR implementation
> > that just generates the individual call legs, and ties them together via
> > the linkedid
> > (see team/group/newcdr), then both camps should be able to derive the
> > info
> > they need for billing, via hopefully not-overly-complex SQL queries to a
> > backend db.
> >
> > I'll modify my RFC to reflect this line of thinking. Yes, it is a bit of
> > shift.
> > And, yes, the implementation will make this new approach optional, and
> > not
> > default. But, pardon if I make it available via the CEL domain come
> > implementation time.
> >
> >
> > It should take me a week to rehash my document; perhaps longer (I'm in
> > bugfix mode, and this borderline development work); in the meantime,
> > those with decided CDR needs might make their wishes known, if they do
> > not think this approach will work. Speak now, or forever hold your
> > peace; or forever complain... or whatever.
> > If this is particularly distressing to you, perhaps your fears might be
> > slightly assuaged when you read the details...
> >   
> I was part of a team that did design a multiservice billing system about 
> 15 years ago and the solution people seems to agree on here (and me to) 
> looks pretty much the same i.e one call may consist of several calls 
> legs. In addition to the linkedid it would be nice to have an indication 
> in the cdr that tells us that 'this is the lastone on this  linked id'.
> Our experience was that  we shouldn't  for load reasons work with cdr's 
> in the immidiate multileg format in the DB. So we did collect cdr's in a 
> tmp DB until we got the the record with end marker set. We would then 
> produce our final cdr for the actual service, store it in billing col. 
> and delete it from the multileg col. When a new service is created we 
> only have to make a the new customized cdr, we don't have to touch the 
> generation of the multileg format.  
> 
> Freddi

Freddi--

Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID set
has been closed. If so, then the entire set is finished. We can use this
facility to get you a closing attribute, that could be added to the last
CDR emmitted for that set; OR, we could just emit another CDR with type
CLOSE or FINAL or something, that signals the end of the chain.

murf

-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] Anonymous callerid

[Philipp Kempgen]

Max Alex schrieb:

> Actually we are setting up the callerid in case of emergency calls when we
> got the anonymous callerid from the caller.
> But the calls are going with callerid anonymous and not set the callerid, i
> want to know how can we sent some meaning ful information to the emergency
> services so our calls will not be disconnected and recieved by them.

Sorry but I fail to understand the problem.
Enable a "sip debug" trace on the Asterisk CLI and show us the
INVITE that Asterisk sends to your SIP provider.
(Be careful not to post any passwords to the list.)

And ask your SIP provider about it. Maybe they want the caller ID
in a special format (like national or international without leading
"+" or somesuch).

> On Sun, Nov 30, 2008 at 1:07 AM, Philipp Kempgen
> <[EMAIL PROTECTED]>wrote:
> 
>> Max Alex schrieb:
>>
>> > Actully we are getting the anonymous callerid from the originated phone
>> > (blocked from phone) so we need to override the callerid and then pass to
>> > network.
>> > we need to send out caller id. That is why we tried to override it.
>> >
>> > But we are not able to override it.
>>
>> I don't quite understand the problem.
>> The phone sends "anonymous" and you want to send something meaningful
>> to emergeny services?
>> Set(CALLERID(num)=yournumber)
>> Set(CALLERID(name)=yournumber)

   Philipp Kempgen

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Re: [asterisk-users] func_odbc questions

[Tim Panton]

On 1 Dec 2008, at 13:38, Giedrius Augys wrote:




2008/12/1 Tilghman Lesher <[EMAIL PROTECTED]>
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
>   I'm working with asterisk 1.6. And I have success using  
func_odbc with
> one row query results (SELECT source,destination from cc  
WHERE ... ):

> exten => s,1,Ringing
> exten => s,n,Wait(4)
> exten => s,n,Answer
> exten =>
> s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=$ 
{ODBC_GETVARIABLES(

>${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1}
> ${REALNUMBER1}, ${STATUSAS})
>
>
> But I don't know how to retrieve data, if query returns a lot of  
rows. In

> documentation I read that need to use in config file:
> mode=multirow,
> and use function ODBC_FETCH. But how to get result-id variable and  
use

> ODBC_FETCH?

The initial result in mode=multirow is not data at all, but a  
query_id that

may be used with ODBC_FETCH to return the first row of data and every
subsequent row of data (up to the max number of rows, if any).

> And another question is, if I execute not SELECT , but stored  
procedure,
> and this procedure will return two, three tables? Is it possible  
retrieve

> these data from couple tables?

If you're talking about a JOIN, then yes.  As long as the fields  
have distinct
names, then you can retrieve each row in turn, same as any other  
query.


--
Tilghman

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Thanks for your reply, but I don't get it... Is there any  
documentation or simple examples how to use ODBC_FETCH and so on.




For your own sanity (if nothing else) I'd wrap the stored procedure in  
a view, then

get FUNC_ODBC to query that view.

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Re: [asterisk-users] CDR Design

[Steve Murphy]

On Tue, 2008-11-25 at 08:06 +, Grey Man wrote:
> >On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> > For the moment, let's not worry about the implementation. Let's
> > get consensus on the spec first. In the scenario, where A calls B,
> > B xfers A to C, C xfers A to D, or some such similar scenario,
> > half the world wants a single CDR for A, from the time it started,
> > to the time it hung up with D. The other half wants A->B's dial and
> > bridge,
> > a cdr for A & C's bridge, a CDR for A & D's bridge, and mayhaps some
> > CDRs
> > to describe the xfers, where B xfers A to C and C xfers A to D.
> >
> > My document is pointing in the former direction, and either we need to
> > spec both, or pick one.
> 
> To me the obvious answer is to provide a CDR for every call leg so for
> A calling B via Asterisk there would be two CDRs produced. It's far
> far easier to disregard the unwanted CDRs than it is to try and
> generate the missing ones and in some cases it's virtually impossible.
> If it's weighed up I think people would vote to have accurate CDRs
> ahead of anything else and if single legs are the best way to do that
> then it's the way it should be done.
> 
> In addition with single leg CDRs it will solve the dilemna about
> acommodating every possible call scenario that I know has caused you a
> lot of consternation over the last 18 months.
> 
> Sure it's a change from the current situation so maybe needs to use
> the standard apporach of a configuration setting to opt in initially
> before becoming the default in the subsequent major release.
> 
> Regards,
> 
> Greyman.
> 
> P.S. Sorry about the duplicate post. I actually sent the email to the
> list 4 times with around 8 hour spacings and I'm not sure why there
> was a problem getting it through.

Everyone--

I've just made some major changes to the CDRfix2.rfc.txt 
file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
to accommodate the Leg approach instead of a 
channel-based approach.

Greyman is correct. By cutting the timeline into legs, 
we avoid all the nasty channel state problems, or so it
appears thus far. I threw out all the text about 
channel/peer state, fleshed in all the example 
cases, etc.

I added a section describing the linkedID field.

I provide a list of CDR record types at the end,
that will eventually be expanded to describe each
field that set in that type, and what they mean.

The types so far are:

3WAY
AXFER 
AXFER1
AXFER2
BARGE
BXFER
CALL
CONF 
HOLD 
PARK
RECALL
RECONN
USER 
WHISPER   


murf


-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] func_odbc questions

[Giedrius Augys]

2008/12/1 Tilghman Lesher <[EMAIL PROTECTED]>

> On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
> >   I'm working with asterisk 1.6. And I have success using func_odbc with
> > one row query results (SELECT source,destination from cc WHERE ... ):
> > exten => s,1,Ringing
> > exten => s,n,Wait(4)
> > exten => s,n,Answer
> > exten =>
> >
> s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(
> >${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1}
> > ${REALNUMBER1}, ${STATUSAS})
> >
> >
> > But I don't know how to retrieve data, if query returns a lot of rows. In
> > documentation I read that need to use in config file:
> > mode=multirow,
> > and use function ODBC_FETCH. But how to get result-id variable and use
> > ODBC_FETCH?
>
> The initial result in mode=multirow is not data at all, but a query_id that
> may be used with ODBC_FETCH to return the first row of data and every
> subsequent row of data (up to the max number of rows, if any).
>
> > And another question is, if I execute not SELECT , but stored procedure,
> > and this procedure will return two, three tables? Is it possible retrieve
> > these data from couple tables?
>
> If you're talking about a JOIN, then yes.  As long as the fields have
> distinct
> names, then you can retrieve each row in turn, same as any other query.
>
> --
> Tilghman
>
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Thanks for your reply, but I don't get it... Is there any documentation or
simple examples how to use ODBC_FETCH and so on.

Thanks

-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] func_odbc questions

[Tilghman Lesher]

On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
>   I'm working with asterisk 1.6. And I have success using func_odbc with
> one row query results (SELECT source,destination from cc WHERE ... ):
> exten => s,1,Ringing
> exten => s,n,Wait(4)
> exten => s,n,Answer
> exten =>
> s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(
>${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1}
> ${REALNUMBER1}, ${STATUSAS})
>
>
> But I don't know how to retrieve data, if query returns a lot of rows. In
> documentation I read that need to use in config file:
> mode=multirow,
> and use function ODBC_FETCH. But how to get result-id variable and use
> ODBC_FETCH?

The initial result in mode=multirow is not data at all, but a query_id that
may be used with ODBC_FETCH to return the first row of data and every
subsequent row of data (up to the max number of rows, if any).

> And another question is, if I execute not SELECT , but stored procedure,
> and this procedure will return two, three tables? Is it possible retrieve
> these data from couple tables?

If you're talking about a JOIN, then yes.  As long as the fields have distinct
names, then you can retrieve each row in turn, same as any other query.

-- 
Tilghman

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[asterisk-users] func_odbc questions

[Giedrius Augys]

Hello,

  I'm working with asterisk 1.6. And I have success using func_odbc with one
row query results (SELECT source,destination from cc WHERE ... ):
exten => s,1,Ringing
exten => s,n,Wait(4)
exten => s,n,Answer
exten =>
s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(${NUMERIS})})
exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1} ${REALNUMBER1},
${STATUSAS})


But I don't know how to retrieve data, if query returns a lot of rows. In
documentation I read that need to use in config file:
mode=multirow,
and use function ODBC_FETCH. But how to get result-id variable and use
ODBC_FETCH?

And another question is, if I execute not SELECT , but stored procedure, and
this procedure will return two, three tables? Is it possible retrieve these
data from couple tables?

Thanks




-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Tzafrir Cohen]

On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
> Hi,
> 
> Hardware is b410p with :
> - span1 jumpers set to NT (no 100 ohm termination)
> - span2 jumpers set to TE (no 100 ohm termination)
> - straight cable from port1 to port2
> 
> Scheme:
> 
> 
>1--
>|  |   <-- cat5 patch cord
>2--
>B410P  |
>3
>|
>4
>|
> 
> 
> 
> 
> /etc/dahdi/system.conf :
> span=1,0,0,ccs,ami
> bchan=1,2
> dchan=3
> span=2,1,0,ccs,ami
> bchan=4,5
> dchan=6
> 
> 
> /etc/asterisk/chan_dahdi.conf:
> context=isdntrunk
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> 
> signalling=bri_cpe
> channel=>1-2
> signalling=bri_net
> channel=>4-5

I'd seperate the NT and TE to different groups to make sure they don't
run over each other. Put the NT in group=2, and you'd be sure you dial
from the TE.

More importantly: what do you see in:

  pri debug span 1

If you see nothing there: do you see anything in 

  pri intense debug span 1

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

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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject

[Andrew Thomas]

Apart from you were dialling out on your inbound context and
vice-versa.

 

The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.

 

Actually - have you tried putting the 100ohm termination on for
your NT port?

 

I need to do that with mISDN as it only allows ptmp for ISDN
extensions.

 

Cheers

Andy

 

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Re: [asterisk-users] [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject

[Olivier]

2008/12/1 Andrew Thomas <[EMAIL PROTECTED]>

>  It looks like you are trying to dial out on your 'NT' instead of your
> 'TE'.
>
>
>
> Try changing Dial(DAHDI/*g*1/${EXTEN:1}); to Dial(DAHDI/*G*1/${EXTEN:1});
>
Unfortunately, changing g1 to G1 didn't change end result.
Anyway, I can't really see why using NT or TE port should theorically change
end result as both allow incoming and outgoing calls.
Do you agree ?

>
>
> Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.
>
>
>
> HTH
>
>
>
>
>
>
>
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Olivier]

2008/12/1 Tzafrir Cohen <[EMAIL PROTECTED]>

> On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
> > Hi,
> >
> > Hardware is b410p with :
> > - span1 jumpers set to NT (no 100 ohm termination)
> > - span2 jumpers set to TE (no 100 ohm termination)
> > - straight cable from port1 to port2
> >
> > Scheme:
> >
> > 
> >1--
> >|  |   <-- cat5 patch cord
> >2--
> >B410P  |
> >3
> >|
> >4
> >|
> > 
> >
> >
> >
> > /etc/dahdi/system.conf :
> > span=1,0,0,ccs,ami
> > bchan=1,2
> > dchan=3
> > span=2,1,0,ccs,ami
> > bchan=4,5
> > dchan=6
>
> s/dchan/hardhdlc/
>
> This is one of the defaults that needs changing in dahdi_genconf .
> Probably will happen when we fix our BRI driver to support hardhdlc :-)


Thanks for the tip.

Now:
# cat ../dahdi/system.conf
span=1,0,0,ccs,ami
bchan=1,2
hardhdlc=3
span=2,0,0,ccs,ami
bchan=4,5
hardhdlc=6

but still :
Executing [EMAIL PROTECTED]:1] Dial("SIP/7533-08bb6508",
"DAHDI/g1/0123456789") in new stack
[Dec  1 10:46:13] WARNING[5676]: app_dial.c:1739 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)



>
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject

[Andrew Thomas]

It looks like you are trying to dial out on your 'NT' instead of your
'TE'.

 

Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1});

 

Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.

 

HTH

 

 

 

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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Tzafrir Cohen]

On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
> Hi,
> 
> Hardware is b410p with :
> - span1 jumpers set to NT (no 100 ohm termination)
> - span2 jumpers set to TE (no 100 ohm termination)
> - straight cable from port1 to port2
> 
> Scheme:
> 
> 
>1--
>|  |   <-- cat5 patch cord
>2--
>B410P  |
>3
>|
>4
>|
> 
> 
> 
> 
> /etc/dahdi/system.conf :
> span=1,0,0,ccs,ami
> bchan=1,2
> dchan=3
> span=2,1,0,ccs,ami
> bchan=4,5
> dchan=6

s/dchan/hardhdlc/

This is one of the defaults that needs changing in dahdi_genconf .
Probably will happen when we fix our BRI driver to support hardhdlc :-)

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

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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Olivier]

2008/12/1 Steve Howes <[EMAIL PROTECTED]>

> On 1 Dec 2008, at 09:09, Olivier wrote:
> > Is there something obvious in configuration keeping this "self-
> > looping trunk" to work ?
>
> You are not using a crossover cable?

No : B410P manual says using jumper is enough (but I've got no successful
experience to confirm this to be true).
Have you ever tried this with a crossover cable ?

>
>
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Re: [asterisk-users] Is using dahdi_genconf_parameters recommended to configure dahdi ?

[Olivier]

2008/12/1 Tzafrir Cohen <[EMAIL PROTECTED]>

> On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
> > Hi,
> >
> > I've just discovered that dahdi_genconf could create configuration files
> > reading a single genconf_parameter file.
> >
> > 1. Is using dahdi_genconf recommended to configure dahdi or is it a
> helpful
> > tool for specific devices or installation processes ?
>
> A helpful tool.
>
> Though my opinion is that for the common cases (analog-only systems,
> BRI, simple PRI setups) you should just use it and save yourself the
> need to configure two and more different places.
>
> I would like to know where it has bad defaults and where it could be
> improved.


IMHO, its defaults are  fine.

The only suggestion that I can think of for it would be to run in
interactive mode : asking relevant questions to users instead of reading
configuration data in a file.



>
> >
> > 2. If not, is there a custom mecanism that would prevent hand-edited
> files
> > to be written over by dahdi_genconf (you can obviously remove execution
> > rights to dahdi_genconf but is there something better) ?
>
> dahdi_genconf is not run automaitcally as part of the installation
> procedure and should not be run automatically IMHO.
>
> (Int also generates a .bak file for every file it generates. But this is
> only useful when you run it once)
>
> > If yes, is it normal that /etc/dahdi wouldn't contain any example of
> > genconf_parameter file (in my setup, I could find one in
> > /usr/src/dahdi/tools/xpp/ but none in /etc/dahdi) or did I miss a step
> when
> > I installed dahdi or asterisk ?
>
> The parameters files is optional. I'm not sure I like installing sample
> configuration files that later on get obsolete. But documentation there
> is still badly missing.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Dahdi, b410p and looping from 1 port to another

[Steve Howes]

On 1 Dec 2008, at 09:09, Olivier wrote:
> Is there something obvious in configuration keeping this "self- 
> looping trunk" to work ?

You are not using a crossover cable?

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Re: [asterisk-users] Is using dahdi_genconf_parameters recommended to configure dahdi ?

[Olivier]

2008/12/1 Tzafrir Cohen <[EMAIL PROTECTED]>

> On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
> > Hi,
> >
> > I've just discovered that dahdi_genconf could create configuration files
> > reading a single genconf_parameter file.
> >
> > 1. Is using dahdi_genconf recommended to configure dahdi or is it a
> helpful
> > tool for specific devices or installation processes ?
>
> A helpful tool.
>
> Though my opinion is that for the common cases (analog-only systems,
> BRI, simple PRI setups) you should just use it and save yourself the
> need to configure two and more different places.
>
> I would like to know where it has bad defaults and where it could be
> improved.
>
> >
> > 2. If not, is there a custom mecanism that would prevent hand-edited
> files
> > to be written over by dahdi_genconf (you can obviously remove execution
> > rights to dahdi_genconf but is there something better) ?
>
> dahdi_genconf is not run automaitcally as part of the installation
> procedure and should not be run automatically IMHO.
>
> (Int also generates a .bak file for every file it generates. But this is
> only useful when you run it once)
>
> > If yes, is it normal that /etc/dahdi wouldn't contain any example of
> > genconf_parameter file (in my setup, I could find one in
> > /usr/src/dahdi/tools/xpp/ but none in /etc/dahdi) or did I miss a step
> when
> > I installed dahdi or asterisk ?
>
> The parameters files is optional. I'm not sure I like installing sample
> configuration files that later on get obsolete. But documentation there
> is still badly missing.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Typo in dahdi_genconf man page

[Olivier]

2008/12/1 Tzafrir Cohen <[EMAIL PROTECTED]>

> On Mon, Dec 01, 2008 at 08:39:53AM +0100, Olivier wrote:
> > Hi,
> >
> > When typing "man dahdi_genconf", you can read :
> > This script generate configuration files for Dahdi hardware.  Currently
> it
> > can generate three files: dahdi, chan_dahdi, users and chan_dahdi_full
> >
> >
> > where it say :
> > This script generate configuration files for Dahdi hardware.  Currently
> it
> > can generate four files: dahdi, chan_dahdi, users and chan_dahdi_full
> >
> > I'm using latest dahdi version.
> > Should this be reported in Mantis ?
>
> Generally, yes (under dahdi-tools). However, thanks for the note. I'll
> fix that.


Fine.

While at it, "man dahdi_genconf" says :
 "Currently it can generate three files: dahdi, chan_dahdi, users and
chan_dahdi_full
   (see below).  Without arguments, it generates only dahdi and
chan_dahdi.

   dahdi - /etc/dahdi/system.conf
   Configuration for dahdi_cfg(1). It's location may be overriden by
the environment variable DAHDI_CONF_FILE.

   chan_dahdi - /etc/asterisk/dahdi-channels.conf
   Configuration for asterisk(1). It should be included in the main
/etc/asterisk/chan_dahdi.conf.  It's location may be overriden by the
   environment variable CHAN_DAHDI_CHANNELS_FILE.

   users - /etc/asterisk/users.conf
   Configuration for asterisk(1) and AsteriskGUI.  It's location may
be overriden by the environment variable USERS_FILE.

   chan_dahdi_full - /etc/asterisk/chan_dahdi.conf
   Configuration for asterisk(1) and AsteriskGUI.  It's location may
be overriden by the environment variable CHAN_DAHDI_CONF_FILE.

What if "it can generate three files: dahdi, chan_dahdi, users and
chan_dahdi_full" was replaced by "it can generate four files: system.conf,
dahdi-channels.conf, users.conf and chan_dahdi.conf" ?

Cheers


>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Is using dahdi_genconf_parameters recommended to configure dahdi ?

[Tzafrir Cohen]

On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
> Hi,
> 
> I've just discovered that dahdi_genconf could create configuration files
> reading a single genconf_parameter file.
> 
> 1. Is using dahdi_genconf recommended to configure dahdi or is it a helpful
> tool for specific devices or installation processes ?

A helpful tool. 

Though my opinion is that for the common cases (analog-only systems,
BRI, simple PRI setups) you should just use it and save yourself the
need to configure two and more different places.

I would like to know where it has bad defaults and where it could be
improved.

> 
> 2. If not, is there a custom mecanism that would prevent hand-edited files
> to be written over by dahdi_genconf (you can obviously remove execution
> rights to dahdi_genconf but is there something better) ?

dahdi_genconf is not run automaitcally as part of the installation
procedure and should not be run automatically IMHO.

(Int also generates a .bak file for every file it generates. But this is
only useful when you run it once)

> If yes, is it normal that /etc/dahdi wouldn't contain any example of
> genconf_parameter file (in my setup, I could find one in
> /usr/src/dahdi/tools/xpp/ but none in /etc/dahdi) or did I miss a step when
> I installed dahdi or asterisk ?

The parameters files is optional. I'm not sure I like installing sample
configuration files that later on get obsolete. But documentation there
is still badly missing.

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

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[asterisk-users] Dahdi, b410p and looping from 1 port to another

[Olivier]

Hi,

Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2

Scheme:


   1--
   |  |   <-- cat5 patch cord
   2--
   B410P  |
   3
   |
   4
   |




/etc/dahdi/system.conf :
span=1,0,0,ccs,ami
bchan=1,2
dchan=3
span=2,1,0,ccs,ami
bchan=4,5
dchan=6


/etc/asterisk/chan_dahdi.conf:
context=isdntrunk
group=1
callgroup=1
pickupgroup=1
immediate=no

signalling=bri_cpe
channel=>1-2
signalling=bri_net
channel=>4-5

/etc/asterisk/extensions.ael
...
context mylocal {
_10. => {
Dial(DAHDI/g1/${EXTEN:1});
Hangup();
};
_753[0356] => {
Dial(SIP/${EXTEN});
Hangup();
};
};

context isdntrunk {
_X. => {
NoOp(Call from ${CALLERID(num)});
Dial(SIP/7530);
Hangup();
};
s => {
NoOp(Call from ${CALLERID(num)});
Dial(SIP/7530);
Hangup();
};
};

# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
   wct4xxp: done   wcte12xp: done   wct1xxp: done   wcte11xp: done
wctdm24xxp: done   wcfxo: done   wctdm: done   wcb4xxp: done   xpp_usb: done
Running dahdi_cfg: done.

# dahdi_cfg -vvv
DAHDI Tools Version - 2.1.0-rc5

DAHDI Version: 2.1.0-rc5
Echo Canceller(s):
Configuration
==

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: D-channel (Default) (Slaves: 06)

6 channels to configure.

# /etc/init.d/asterisk start
# asterisk -r

after dialing 10123456789
Executing [EMAIL PROTECTED]:1] Dial("SIP/7533-095aaae0",
"DAHDI/g1/0123456789") in new stack
[Dec  1 09:59:22] WARNING[5011]: app_dial.c:1739 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)




Is there something obvious in configuration keeping this "self-looping
trunk" to work ?

Regards
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Re: [asterisk-users] Typo in dahdi_genconf man page

[Tzafrir Cohen]

On Mon, Dec 01, 2008 at 08:39:53AM +0100, Olivier wrote:
> Hi,
> 
> When typing "man dahdi_genconf", you can read :
> This script generate configuration files for Dahdi hardware.  Currently it
> can generate three files: dahdi, chan_dahdi, users and chan_dahdi_full
> 
> 
> where it say :
> This script generate configuration files for Dahdi hardware.  Currently it
> can generate four files: dahdi, chan_dahdi, users and chan_dahdi_full
> 
> I'm using latest dahdi version.
> Should this be reported in Mantis ?

Generally, yes (under dahdi-tools). However, thanks for the note. I'll
fix that.

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

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[asterisk-users] how to configure free radius server with asterisk

[krunal soni]

Hi all,

i want to configure free radius with asterisk server. my radius server and
clients are running perfectly with each other. now only thing i need to do
is to patch radius client with asterisk.

Can any one suggest how i can do that? i am using free radius 2.1.x version
and asterisk latest version. radiusclient-ng as a client.





Regards,
Krunal
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[asterisk-users] Is using dahdi_genconf_parameters recommended to configure dahdi ?

[Olivier]

Hi,

I've just discovered that dahdi_genconf could create configuration files
reading a single genconf_parameter file.

1. Is using dahdi_genconf recommended to configure dahdi or is it a helpful
tool for specific devices or installation processes ?

2. If not, is there a custom mecanism that would prevent hand-edited files
to be written over by dahdi_genconf (you can obviously remove execution
rights to dahdi_genconf but is there something better) ?
If yes, is it normal that /etc/dahdi wouldn't contain any example of
genconf_parameter file (in my setup, I could find one in
/usr/src/dahdi/tools/xpp/ but none in /etc/dahdi) or did I miss a step when
I installed dahdi or asterisk ?


Regards
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