Re: [Asterisk-Dev] Open source time card application for Asterisk

[Brian West]

Title: Re: [Asterisk-Dev] Open source time card application for Asterisk



Their is this really nice ifdef which should really be a config file option its called PRI_ANI in chan_zap.

/b


On 9/23/05 11:16 AM, "Gilmore, Gerry" <[EMAIL PROTECTED]> wrote:

Chuck,
 
Actually, Caller ID cannot – so far as I know – “easily be spoofed”. While you can usually disable sending “caller ID” by the *6x method, be aware that if you call an 800 number, that 800 number *will* get the calling party number. It’s needed for billing the 800# recipient.
 
With PRI, if you have it correctly provisioned by the carrier and they support it, etc., you can legitimately spoof a caller name and number, but I doubt a nurse or janitor would maintain a PRI line to do this. J
 
Gerry
 

There are 10 kinds of people in the world, those who understand binary and those who don't.
 
Gerry Gilmore
Field Applications Engineer
Intel Corporation
(http://www.intel.com)
 





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Friday, September 23, 2005 12:14 PM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Open source time card application for Asterisk
 
Joseph wrote: 
On Fri, 2005-09-23 at 09:34 -0600, Chuck Bunn wrote:
  
Hi,
 
I am in the process of developing a time card application that 
integrates with Asterisk and I would like to know if anyone has done 
this and if so can you recommend an open source time card application 
that might reduce the amount of work required to connect it to Asterisk. 
I do not want to reinvent the wheel and write a WEB based time card app, 
I would rather spend my time getting Asterisk to connect to it. I will 
be using LAMP so it will be written in PHP instead of C...
 
Thanks

 
Do you mean employee time-card? With mysql database this would be very
interesting project.
Do you have a short description of what it would do?
 
  
Hi,

Thanks for responding. The basic system would work as follows. An employee would call in and would transfer to a menu (either via an operator or via the phone system if no operator is available). Something like press 1 for sales and service press 2 for accounting and press 3 if you are an employee. Upon pressing three the user would be asked to enter their employee ID and password. The system would capture the employee ID , time of day and if available the caller ID from the location they are calling from. When they are finished they would repeat the process thus capturing the finish data for filling out a time card. The usage is a group of field nurses that need an easy way to enter data into there time cards since the Internet is not always available from the locations they call from. Although the caller ID can easily be spoofed this is not as important as capture of the time and employee data for filling out a time card. A second application uses the same theory but for janitors. The caller ID helps confirm they are where they are supposed to be. There are several PHP time cards out there but I am trying to find the best for interfacing with Asterisk. Phase one would be to capture the data to a flat file phase 2 would be to get it into a database, phase three would interface it to an existing LAMP based time card apt and phase 4 would allow for phone access to information stored in the time card such as how many hours worked, hours by day etc. A final phase would interface these to both Peachtree and Quickbooks.

I am working on a more definitive outline right now but I wanted some feedback from the development community before doing so, so that I did not reinvent the wheel.

Thanks

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Re: [Asterisk-Dev] I need help to make a simple iax switch/proxy

[Brian West]

Yes, I do know what he is talking about.  What he speaks of is true but I
think he failed to relay the specifics of what was going on.  I think Derek
followed up with some really good points about the implementation of IAX2 in
Asterisk suffers a bit and needs to be addressed.  If you push it as much as
we do you'll start to notice some major issues with and without the jitter
buffer on.  I personally don't think their is any problem with the IAX2
protocol itself just how its implemented in Asterisk.

/b

On 9/11/05 9:52 PM, "Brian Capouch" <[EMAIL PROTECTED]> wrote:

> Does anyone know what this guy is talking about?
> 
> Um, including this guy, of course . . .
> 
> B.


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Re: [Asterisk-Dev] ***TESTING NEEDED*** JITTER BUFFER FOR SIP

[Brian West]

Ok we tested the new jitter buffer today from a very distant network  
with about 20-55% packet loss and lots of jitter you could at least  
understand the person that the far end... granted I think it was  
better than SprintPC :P


/b

On Aug 25, 2005, at 1:10 PM, Olle E. Johansson wrote:


http://bugs.digium.com/view.php?id=3854

If you really care about a SIP jitter buffer, make sure you help us
testing this yesterday or at least as soon as possible!

Thank you very much for your help in this matter! If we can get enough
positive test reports quickly, this new jitter buffer will be a great
addition to Asterisk!

/O
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Re: [Asterisk-Dev] Henning G. Schulzrinne quote on IAX2 from vonmagazine

[Brian West]

Its a problem with the current implementation in my opinion.  /bOn Aug 13, 2005, at 2:38 PM, Michael Giagnocavo wrote:That'd be a problem in chan_iax2, not IAX2, no?  But I'll add: with the current implementation, you can't even *register* many users, let alone take calls from them...  -Michael ___
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Re: [Asterisk-Dev] Henning G. Schulzrinne quote on IAX2 from von magazine

[Brian West]

On Aug 13, 2005, at 1:45 PM, John Todd wrote:I hereby kill this thread and take it to the -users list. Actually this should be on -dev because IAX2 does have scalability issues.  It will not and CAN NOT scale to large scales.  This is due to the single thread that handles all traffic in and out of the single UDP port.  As you start to load the box with calls the call quality starts to degrade and the delay shoots thru the roof.  You have one queue for receive and one for transmit that is serviced by the same thread.  Now if you pile on 30 calls with say a 300ms of latency it starts to sound like crap while running sip in the same situation sounds perfect.  IAX2 has its place but not in a large scale deployment at this time unless we can solve the issue with the bottle neck in the tx/rx queues.  I think we have talked about splitting the tx and rx queues into two threads that only gains a little bit of speed.  The next step would be to move this operation into the kernel since it does this and does it well already.  Then again on the other hand if you had an option to use a dedicated udp port per call that would also cause the issue to go away.  We could add something to punch thru nat and cause it to work just like it does now without much if any thought on the users side.  /b ___
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Re: [Asterisk-Dev] Re: PostgreSQL support in Asterisk 1.2?

[Brian West]

On Aug 9, 2005, at 3:15 PM, Kristian Nielsen wrote:And while everybody is busy whining about the default value of the MySQL `sql_mode' variable not being set to 'strict' mode, Asterisk _still_ does not have a native PostgreSQL interface... Don't really need one, ODBC is all you really need in that case.  It works great./b___
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Re: [Asterisk-Dev] Cluecon - Who's going ?

[Brian West]

I'm going to be speaking about how to use valgrind, gdb and strace to  
help debug issues... it can be applied to more than just asterisk.


/b

On Jul 25, 2005, at 10:29 AM, Terry Moore-Read wrote:

I'm relatively new to Asterisk and I'm hoping attending Cluecon  
will be
a good way to get up to speed on the project and hear about what  
others

are doing with it.

We currently use a Cisco IP phone system at my office, although I just
added an asterisk box to provide soft phones to our travelling users.
(IAX2 is a lot easier to get through firewalls than cisco's  
protocols).


Terry Moore-Read
Lukins & Annis, P.S.
Spokane, WA

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[Asterisk-Dev] We are giving away 3 A101 single-port T1 cards during Cluecon!

[Brian West]

Cluecon's ( http://www.cluecon.com ) premier sponsor, Sangoma ( http://www.sangoma.com ), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma’s next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your chance to win.Be sure to register by this wednesday, it's the last day I can squeeze in room registrations so please register and pay by that date if possible.Thanks,Brian WestAsterlink.com___
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Re: [Asterisk-Dev] Re: res_config_mysql.c v1.7

[Brian West]

It might be most helpful if 1.0.9 HAD realtime support.  It doesn't  
exist at all in the stable branch.


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jul 6, 2005, at 11:46 AM, Leonardo F. Bauchwitz wrote:


Hi Matthew:
I have a  problem with realtime and mysql.
My Asterisk is Asterisk 1.0.9.
I  compiled the res_config_mysql.c  without problems, but when  
I start Asterisk crash.


The error message is:_



Segmentation fault
sast:/usr/src/asterisk-addons# output: fwrite: Broken pipe



If I look /var/log/messages:



ul  6 10:43:27 DEBUG[4003]: MySQL RealTime Host: 127.0.0.1
Jul  6 10:43:27 DEBUG[4003]: MySQL RealTime Port: 3306
Jul  6 10:43:27 DEBUG[4003]: MySQL RealTime User: root
Jul  6 10:43:27 DEBUG[4003]: MySQL RealTime Password: mateamargo2005
Jul  6 10:43:27 DEBUG[4003]: MySQL RealTime: Successfully  
connected to database.




The connection to database is ok

If I remove /usr/lib/asterisk/modules/res_config_mysql.so, then   
Asterisk start without a problem


Can you help me?
Leonardo
Leonardo Federico Bauchwitz

Matthew Boehm wrote:


Before hand I wasn't calling ast_category_new("") and the  
operations below

on that struct (ast_variable_append, and ast_category_append) where
operating on a NULL pointer.

-Matthew

- Original Message - From: "Kevin P. Fleming"  
<[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List" [EMAIL PROTECTED]>

Sent: Friday, January 28, 2005 12:13 PM
Subject: Re: [Asterisk-Dev] Re: res_config_mysql.c v1.7





Matthew Boehm wrote:



Seems it works.

http://bugs.digium.com/bug_view_page.php?bug_id=0003446


But that would only make a difference if the Asterisk process is  
running
out of memory. Is that likely to be the actual cause of the  
problem you

are seeing?
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Re: [Asterisk-Dev] More AEL funnies

[Brian West]

temp=(${DB(CFIM/${ext})});

I'll be that will cause it to work also.

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jul 6, 2005, at 3:32 AM, Brian Capouch wrote:

AEL doesn't like to encounter DB entries that don't exist, although  
it doesn't seem to do anything evil beyond screwing up the CLI  
display:


Jul  6 03:27:51 WARNING[4320]: ast_expr.y:486 ast_yyerror:  
ast_yyerror(): syntax error: parse error; Input:


^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^ 
^^
^^ 
^-- Executing Set("SIP/nwata2-39bb", "temp=0") in  
new stack


The line that causes this is:

temp=${DB(CFIM/${ext})};

Where in this case there is no entry in the DB for that pair.

If I put something in the DB, the ast_yyerror goes away.

B.
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Re: [Asterisk-Dev] C Code of Asterisk

[Brian West]

check out app_skel.c and www.cluecon.com  ;)

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Mar 30, 2005, at 5:56 AM, Bharat M. Sarvan wrote:


Hello Everybody,

 Has anybody gone through the C code of the  
Asterisk? Like Dial ( ) is a registered application of Asterisk, I  
wish to build an application of my own. So how does do it in  
Asterisk?  And also could anybody


Please tell me an IRC for asterisk and also the server that I  
should choose.




Thanks in Advance









Regards,

Bharat M. Sarvan



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Re: [Asterisk-Dev] How can I see what a thread is doing?

[Brian West]

You can't without pausing the process... now if you're fast enough  
you can attach snag the trace and detach and only get a small blip in  
audio.


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 27, 2005, at 3:34 PM, Matthew Boehm wrote:


Brian West wrote:


attach to it with gdb and do "thread apply all bt"
/b
---



That's destructive though isn't it? Will it kill calls? Halt the  
server? Do I attach to the main thread or the thread in question?


Thanks,
-Matthew

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Re: [Asterisk-Dev] CVS Head-- Dialplan disappeared! -- RESOLUTION

[Brian West]

Yep thats normal!

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 27, 2005, at 2:49 PM, Steve Murphy wrote:


Here I answer my question:

QUESTION_ If there's a syntax error in the extensions.conf file, won't
it tell you?

ANSWER: No!

My problem was that one of the comments at the top of the file got
munged, most likely by me, but I have no idea how I did it:

; Static extension configuration files, used by
; the pbx_config module.

Changed to:

10 extension configuration files, used by
; the pbx_config module.

I fixed it back up, and everything works fine again. Sorry for the
scare.

We really, REALLY, need to improve the error messages on the  
parsing of
config files, so the average user, or some dingbat like me, won't  
write

in, suspecting some strange change in the CVS head is at fault, when
really it's our own doing.

If the logs or messages, or ANYTHING had indicated an error in the
parsing of the config file, I'd not have wasted a couple hours  
trying to

figure things out.

Shall I roll up my sleeves, and attack the config file parsing  
code? Or

is someone else already on this?

murf



On Mbe, 2005-06-27 at 11:25 -0400, Jeremy McNamara wrote:


Steve Murphy wrote:



Just now, I CVS updated, built, and am running asterisk, and
when we pick up phones and dial, the asterisk console
reports:






Start asterisk with -vvvgc and examine for messages.


Jeremy McNamara


--
Steve Murphy <[EMAIL PROTECTED]>
Electronic Tools Company
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Re: [Asterisk-Dev] CVS Head-- Dialplan disappeared!

[Brian West]

chances are you have a char at the top before [general]  like junk chars

example:
[general] < good

ad[general] < bad


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 27, 2005, at 12:58 PM, Steve Murphy wrote:


Via the -d option in the command line to asterisk, I see this
/var/log/asterisk/messages:

(sorry in advance-- my mail composer doesn't let me determine the
line wrap... )

Jun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/manager.conf
Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/rtp.conf
Jun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/modules.conf
Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
modem.conf

Jun 27 11:45:32 VERBOSE[18902] logger.c:   == Loading modem driver
chan_modem_aopen.soJun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/modules.conf
Jun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/musiconhold.conf
Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/adsi.conf
Jun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/features.conf
Jun 27 11:45:32 DEBUG[18902] config.c:
Parsing /etc/asterisk/indications.conf
Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
skinny.conf

Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/mgcp.conf
Jun 27 11:45:32 DEBUG[18902] config.c: Parsing /etc/asterisk/sip.conf
Jun 27 11:45:47 DEBUG[18902] config.c: Parsing /etc/asterisk/iax.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
zapata.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
phone.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
agents.conf

Jun 27 11:46:00 DEBUG[18902] config.c:
Parsing /etc/asterisk/extensions.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
dundi.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf

Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/enum.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
queues.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf

Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/enum.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/ 
codecs.conf

Jun 27 11:46:00 DEBUG[18902] config.c:
Parsing /etc/asterisk/voicemail.conf
Jun 27 11:46:00 DEBUG[18902] config.c: Parsing /etc/asterisk/enum.conf


anyway, in the above, I see a "Parsing /etc/asterisk/extensions.conf",
but nothing there...

If there's any errors, wouldn't I see some sort of message?

BTW. I created a null extconf.conf  file, and that didn't make any
difference in behavior (except no message about it not being found).

murf





On Mbe, 2005-06-27 at 11:25 -0400, Jeremy McNamara wrote:


Steve Murphy wrote:



Just now, I CVS updated, built, and am running asterisk, and
when we pick up phones and dial, the asterisk console
reports:






Start asterisk with -vvvgc and examine for messages.


Jeremy McNamara


--
Steve Murphy <[EMAIL PROTECTED]>
Electronic Tools Company

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Re: [Asterisk-Dev] How can I see what a thread is doing?

[Brian West]

attach to it with gdb and do "thread apply all bt"

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 27, 2005, at 1:33 PM, Matthew Boehm wrote:


Hey guys,
 We are having some problems with asterisk and echo and dropped  
calls. I think it is because we have an old 4 proc P3 machine and  
can't put thru more than 6-7 calls before we start getting echo and  
major jitter.
 I'm using "top" and are also displaying threads. There are 7 calls  
up right now and I see about 15 threads spread amongst the CPUs.  
Most of them are using 0% cpu. The top two threads are using about  
50% each.
 How can I find out what these two threads are doing to use so much  
CPU? Can I do this without causing harm to the system? I've got 71  
peers/users and all but 20 are being "qualified". Can this be using  
up lots of CPU?


Thanks,
Matthew

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Re: [Asterisk-Dev] Queue/How to get the number of incoming calls

[Brian West]

I think this would be a user's list question./bOn Jun 17, 2005, at 8:14 PM, Gang Li wrote: Hi,all,   Now,I am working at make an realtime monitor for the call center based on asterisk. and ,I had search the archive and wiki.Through the return info from the management API, I can get the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get the number of incoming calls. The info like following :    Asterisk Call Manager/1.0 Response: Success Message: Authentication accepted Response: Success Message: Queue status will follow  Event: QueueParams Queue: test_queue Max: 18 Calls: 1 Holdtime: 127 Completed: 209 Abandoned: 18 ServiceLevel: 0 ServicelevelPerf: 0.0 Event: QueueMember Queue: test_queue  Location: Agent/101 Membership: static Penalty: 0 CallsTaken: 20 LastCall: 1118794338 Event: QueueMember Queue: test_queue  Location: Agent/102 Membership: static Penalty: 0 CallsTaken: 19 LastCall: 1118778909 Event: QueueMember Queue: test_queue  Location: Agent/103 Membership: static Penalty: 0 CallsTaken: 14 LastCall: 1118782495 Event: QueueMember Queue: test_queue  Location: Agent/104 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue  Location: Agent/105 Membership: sta tic Penalty: 0 CallsTaken: 9 LastCall: 1118779889 Event: QueueMember Queue: test_queue  Location: Agent/106 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/107 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/108 Membership: static Penalty: 0 CallsTaken: 146 LastCall: 1118795077 Event: QueueEntry Queue:  test_queue Position: 1 Channel: Zap/6-1 CallerID: 4042662907 Wait: 739  Response: Goodbye Message: Thanks for all the fish.  ***   who knows how to get it through such info ,or there are other method for getting incoming calls number?   Any advice and help will be appreciated!            Best Regards, Gary Li__赶快注册雅虎超大容量免费邮箱?http://cn.mail.yahoo.com___Asterisk-Dev mailing listAsterisk-Dev@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-devTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-dev ___
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Re: [Asterisk-Dev] Seeking a clue RE: how * does the right thing with NAT'ed SIP/SDP devices

[Brian West]

Its called "localnet"
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 17, 2005, at 3:43 PM, David Pollak wrote:


Folks,

I'm looking to understand how Asterisk determines if it's going to  
re-write an SDP contained in a SIP message from an ill-behaving  
application behind a firewall.


Specifically, the Microsoft RTC client running behind a NAT puts  
the wrong IP address (the local non-routable address, rather than  
the NAT's address) in the SDP header.  However, Asterisk does the  
right thing and send media streams to the IP address of the NAT.


I've looked through chan_sip.c:process_sdp and don't see anything  
that remotely looks like logic to re-write the host.


If anyone can lend a clue to me, I'd be grateful to the tune of a  
couple of beers.


Thanks,

David


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[Asterisk-Dev] asterisk asterisk.c, 1.155, 1.156 cdr.c, 1.38, 1.39 channel.c, 1.196, 1.197 loader.c, 1.40, 1.41 pbx.c, 1.246, 1.247

[Brian West]

cdr.c: In function `submit_unscheduled_batch':
cdr.c:939: warning: implicit declaration of function  
`use_ast_mutex_lock_instead_of_pthread_mutex_lock'
cdr.c:941: warning: implicit declaration of function  
`use_ast_mutex_unlock_instead_of_pthread_mutex_unlock'

cdr.c: In function `do_cdr':

:P

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

Begin forwarded message:


From: [EMAIL PROTECTED]
Date: June 2, 2005 9:39:29 PM CDT
To: asterisk-cvs@lists.digium.com
Subject: [Asterisk-cvs] asterisk asterisk.c, 1.155, 1.156 cdr.c,  
1.38, 1.39 channel.c, 1.196, 1.197 loader.c, 1.40, 1.41 pbx.c,  
1.246,1.247



Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv4616

Modified Files:
asterisk.c cdr.c channel.c loader.c pbx.c
Log Message:
support configurable batch posting of CDRs (off by default) (bug  
#3883)



Index: asterisk.c
===
RCS file: /usr/cvsroot/asterisk/asterisk.c,v
retrieving revision 1.155
retrieving revision 1.156
diff -u -d -r1.155 -r1.156
--- asterisk.c19 May 2005 01:57:19 -1.155
+++ asterisk.c3 Jun 2005 01:42:31 -1.156
@@ -45,6 +45,7 @@
 #include "asterisk/tdd.h"
 #include "asterisk/term.h"
 #include "asterisk/manager.h"
+#include "asterisk/cdr.h"
 #include "asterisk/pbx.h"
 #include "asterisk/enum.h"
 #include "asterisk/rtp.h"
@@ -601,6 +602,8 @@
 char filename[80] = "";
 time_t s,e;
 int x;
+/* Try to get as many CDRs as possible submitted to the  
backend engines (if in batch mode) */

+ast_cdr_engine_term();
 if (safeshutdown) {
 shuttingdown = 1;
 if (!nice) {
@@ -1952,6 +1955,10 @@
 printf(term_quit());
 exit(1);
 }
+if (ast_cdr_engine_init()) {
+printf(term_quit());
+exit(1);
+}
 ast_rtp_init();
 if (ast_image_init()) {
 printf(term_quit());

Index: cdr.c
===
RCS file: /usr/cvsroot/asterisk/cdr.c,v
retrieving revision 1.38
retrieving revision 1.39
diff -u -d -r1.38 -r1.39
--- cdr.c25 May 2005 17:18:05 -1.38
+++ cdr.c3 Jun 2005 01:42:31 -1.39
@@ -17,6 +17,8 @@
 #include 
 #include 
 #include 
+#include 
+#include 

 #include "asterisk/lock.h"
 #include "asterisk/channel.h"
@@ -27,6 +29,10 @@
 #include "asterisk/options.h"
 #include "asterisk/linkedlists.h"
 #include "asterisk/utils.h"
+#include "asterisk/sched.h"
+#include "asterisk/config.h"
+#include "asterisk/cli.h"
+#include "asterisk/module.h"

 int ast_default_amaflags = AST_CDR_DOCUMENTATION;
 char ast_default_accountcode[AST_MAX_ACCOUNT_CODE] = "";
@@ -40,6 +46,39 @@

 static AST_LIST_HEAD_STATIC(be_list, ast_cdr_beitem);

+struct ast_cdr_batch_item {
+struct ast_cdr *cdr;
+struct ast_cdr_batch_item *next;
+};
+
+static struct ast_cdr_batch {
+int size;
+struct ast_cdr_batch_item *head;
+struct ast_cdr_batch_item *tail;
+} *batch = NULL;
+
+static struct sched_context *sched;
+static int cdr_sched = -1;
+static pthread_t cdr_thread = AST_PTHREADT_NULL;
+
+#define BATCH_SIZE_DEFAULT 100
+#define BATCH_TIME_DEFAULT 300
+#define BATCH_SCHEDULER_ONLY_DEFAULT 0
+#define BATCH_SAFE_SHUTDOWN_DEFAULT 1
+
+static int enabled;
+static int batchmode;
+static int batchsize;
+static int batchtime;
+static int batchscheduleronly;
+static int batchsafeshutdown;
+
+AST_MUTEX_DEFINE_STATIC(cdr_batch_lock);
+
+/* these are used to wake up the CDR thread when there's work to  
do */

+AST_MUTEX_DEFINE_STATIC(cdr_pending_lock);
+pthread_cond_t cdr_pending_cond;
+
 /*
  * We do a lot of checking here in the CDR code to try to be sure  
we don't ever let a CDR slip
  * through our fingers somehow.  If someone allocates a CDR, it  
must be completely handled normally

@@ -370,7 +409,7 @@
 while (cdr) {
 next = cdr->next;
 chan = !ast_strlen_zero(cdr->channel) ? cdr->channel :  
"";

-if (!ast_test_flag(cdr, AST_CDR_FLAG_POSTED))
+if (!ast_test_flag(cdr, AST_CDR_FLAG_POSTED) && ! 
ast_test_flag(cdr, AST_CDR_FLAG_POST_DISABLED))
 ast_log(LOG_WARNING, "CDR on channel '%s' not posted 
\n", chan);

 if (!cdr->end.tv_sec && !cdr->end.tv_usec)
 ast_log(LOG_WARNING, "CDR on channel '%s' lacks end 
\n", chan);

@@ -724,7 +763,7 @@
 return -1;
 }

-void ast_cdr_post(struct ast_cdr *cdr)
+static void post_cdr(struct ast_cdr *cdr)
 {
 char *chan;
 struct ast_cdr_beitem *i;
@@ -755,13 +794,17 @@
 void ast_cdr_reset(struct ast_cdr *cdr, int flags)
 {
 struct ast_flags tmp = {flags};
+struct ast_cdr *dup;

 while (cdr) {
-/* Post if requested */
+/* Detach if post is requested */
 if (ast_test_flag(&tmp, AST_CDR_FLAG_LOCKED) || ! 
ast_test_flag(cdr, AST_CDR_FLAG_LOCKED)) {

 if (ast_test_flag(&tmp, AST_CDR_FLAG_POSTED)) {
 ast_cdr_end(cdr);
- 

Re: [Asterisk-Dev] CVS HEAD file version tags and header include order

[Brian West]

Shouldn't this be "version show files" to stick with the backwardness  
of the existing CLI commands? :P

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 6, 2005, at 6:49 PM, Kevin P. Fleming wrote:

Based on the discussion on this list today, I've added file version  
tags to all C source files in CVS HEAD, along with a 'show version  
files' command that can be used to display them.


Along the way, I converted all remaining files to include system  
headers before Asterisk headers and local headers; this may break  
builds on non-Linux platforms. If you find that it does, please  
don't report that "it's broken", but try to determine exactly what  
is the reason a particular system header needs to be included  
_after_ an Asterisk header. That way we can add the appropriate  
comments (or an alternative fix) when correcting the problem.

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[Asterisk-Dev] Cluecon Register Now, Pay later!

[Brian West]

Ok guys I have it setup so you can register for cluecon now and pay  
when it gets closer to the time of the conference.


/b
---
Keep Your Friends Close, But Your Enemies Even Closer...

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[Asterisk-Dev] Cluecon developer discount...

[Brian West]

We have posted the details of the discount on http:// 
www.cluecon.com,  If you wish to speak or have recommendations on  
what you would like to hear at Cluecon please feel free to let me  
know about it.   Remember this conference isn't just for Developers  
we have a large space for sponsors to setup booths to promote their  
products and services to the Developer Community.  So if you're the  
new kid on the block and wish to get up to speed faster come to  
Cluecon and interact with developers, companies and people that make  
Open Source Telephony what it is.


Developers need hardware, software, support and more... If your  
company can market to developers then this conference is perfect for  
you to show your products and services at.  Cluecon is open for any  
and all... so come join in!!!


NEXT!!!

/b
[EMAIL PROTECTED]
http://www.cluecon.com (where the fun starts)

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Re: [Asterisk-Dev] making progress; how to debug "() ??"

[Brian West]

If your module starts any threads on unload you'll need to cancel and  
join the thread back so you can do a clean up better.  Check out  
chan_zap does this on unload.  You'll do a pthread_cancel,  
pthread_kill then pthread_join.  I suspect this is what is going on.


/b
---
Keep Your Friends Close, But Your Enemies Even Closer...

On Jun 1, 2005, at 11:58 AM, Matthew Boehm wrote:


Hey guys,
 I'm making big headway in bringing an old SNMP patch up to current  
CVS.

Module loads at startup, connects with running snmpd and I can query
asterisk with snmpwalk to see all sorts of stuff.

 My problem right now is unloading the module from asterisk. If you  
tell the
module to unload, the CLI returns immediatly and asterisk core  
dumps a few

seconds later.

 When I backtrace, I get this:

#0  0x40baf957 in ?? ()
#1  0x40b982b4 in ?? ()
#2  0x400262b6 in start_thread () from /lib/tls/libpthread.so.0

My module links with about 4 other shared objects that are not part of
asterisk. I'm guessing that is why I see ?? () in the bt's?

I tried reading up on how to load more symbols into gdb but I have  
to know
the memory address to load them into. How the heck am I suposed to  
know

that? Is there a better way? A correct way?

I see that gdb loads some symbols from mysql and that isn't part of
asterisk, but res_config_mysql is a module in asterisk. If my new  
module is

part of asterisk, how come gdb doesn't load its symbols too?

Thanks,
Matthew

--
-- 
--
Matthew Boehm, IT DirectorCypress  
Telecommunications
[EMAIL PROTECTED]   3838 N. Sam Houston Parkway E  
#400
T: 832-200-8640 x3044  Houston, TX  
77032


My girlfriend was recently diagnosed with multiple personality  
disorder;

 When she called yesterday, my CallerID box exploded.
-- 
--


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Re: [Asterisk-Dev] karma on mantis?

[Brian West]

also -c is console not color just in case.. I have seen people thing  
its color.


/b

On May 30, 2005, at 7:36 PM, Andrew Kohlsmith wrote:


On Monday 30 May 2005 20:22, Brian West wrote:


your terminal.. because people in their right minds DO NOT START
asterisk like that and all reality asterisk should have died if you
did that and NOT continued to run.  But then again still not major.
NEXT!!!



I run all my asterisk servers in screen sessions with -c.  It's the  
only way I

can get colour console in a screen session.

-A.
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Re: [Asterisk-Dev] karma on mantis?

[Brian West]

I get color if I export TERM=xterm 90% of the boxes i'm on...

/b

On May 30, 2005, at 7:36 PM, Andrew Kohlsmith wrote:


On Monday 30 May 2005 20:22, Brian West wrote:


your terminal.. because people in their right minds DO NOT START
asterisk like that and all reality asterisk should have died if you
did that and NOT continued to run.  But then again still not major.
NEXT!!!



I run all my asterisk servers in screen sessions with -c.  It's the  
only way I

can get colour console in a screen session.

-A.
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[Asterisk-Dev] Fwd: Your email message was blocked

[Brian West]

For all you people that use SUCH things.. ADD the list address to your allow or face getting your email address recommended for an unsubscribed./bBegin forwarded message:From: [EMAIL PROTECTED]Date: May 30, 2005 7:27:17 PM CDTTo: [EMAIL PROTECTED]Cc: [EMAIL PROTECTED]Subject: Your email message was blocked MailMarshal (an automated content monitoring gateway) has stopped the following email it contains unacceptable language   Message: BB0249c03d.0001.mml   From:    [EMAIL PROTECTED]   To:      [EMAIL PROTECTED]   Subject: Re: [Asterisk-Dev] karma on mantis?If you believe the above email to be business related pleasecontact [EMAIL PROTECTED] to arrange for the message to be released to its intended recipients.The blocked e-mail will be automatically deleted after 5 days.MailMarshal Rule: Inbound Messages : Block Unacceptable LanguageFor more information on email virus scanning, security and contentmanagement, visit http://www.marshalsoftware.com ___
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Re: [Asterisk-Dev] karma on mantis?

[Brian West]

I recall that bug.. yes it might be a bug and I think something was  
done to correct it.. but wasn't major because you could have worked  
around it by NOT starting asterisk in that manner and NOT closing  
your terminal.. because people in their right minds DO NOT START  
asterisk like that and all reality asterisk should have died if you  
did that and NOT continued to run.  But then again still not major.   
NEXT!!!


/b

On May 30, 2005, at 6:52 PM, Roy Sigurd Karlsbakk wrote:


my point was people get their karma points by doing trivial stuff
but reporting something that seems pretty fucking major, like  
starting asterisk -cp and getting disconnected can get you a negative


which is wrong

the whole karma system is fscked up

let the code speak for itself

roy


Den 30. mai. 2005 kl. 23.48 skrev Brian West:



You usually won't get +karma till a bug is closed and resolved.   
If then they forget say something at that time.


/b

On May 30, 2005, at 12:56 PM, Roy Sigurd Karlsbakk wrote:





hi

what are the basis on which the 'karma' on the mantis is given?  
My current karma is currently -2 although I've probably done more  
good than bad in there. the latest bug I've been fighting, #4318,  
has taken quite some time, debug info posted on mantis etc, but  
this is as it seems not very important. What seems to be more  
important is fixing one's own bugs (oej gets 4 karma poits for  
fixing 3113 for "Finding and Fixing a Trivial Bug" and "markster  
says: Giving you this twice since it wasn't your bug (even if it  
was a brief fix)" and mentioning disclaimers (as in 3113  
"Remembering to mention your disclaimer when you upload a  
patch"). It all looks like a face factor where the "good guys"  
get "karma" points for whatever they do and the "not-so-good- 
guys" get nothing. As with my own first -2 for 3331 that "only  
hung the box, needing it to be restarted onsite" if you started  
asterisk -cp and disconnected from the terminal... not a major  
bug by guidelines, some say, but I don't know.


why don't you just ditch the whole karma bullshit and let the  
code speak for itself?


roy


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Re: [Asterisk-Dev] karma on mantis?

[Brian West]

You usually won't get +karma till a bug is closed and resolved.  If  
then they forget say something at that time.


/b

On May 30, 2005, at 12:56 PM, Roy Sigurd Karlsbakk wrote:


hi

what are the basis on which the 'karma' on the mantis is given? My  
current karma is currently -2 although I've probably done more good  
than bad in there. the latest bug I've been fighting, #4318, has  
taken quite some time, debug info posted on mantis etc, but this is  
as it seems not very important. What seems to be more important is  
fixing one's own bugs (oej gets 4 karma poits for fixing 3113 for  
"Finding and Fixing a Trivial Bug" and "markster says: Giving you  
this twice since it wasn't your bug (even if it was a brief fix)"  
and mentioning disclaimers (as in 3113 "Remembering to mention your  
disclaimer when you upload a patch"). It all looks like a face  
factor where the "good guys" get "karma" points for whatever they  
do and the "not-so-good-guys" get nothing. As with my own first -2  
for 3331 that "only hung the box, needing it to be restarted  
onsite" if you started asterisk -cp and disconnected from the  
terminal... not a major bug by guidelines, some say, but I don't  
know.


why don't you just ditch the whole karma bullshit and let the code  
speak for itself?


roy


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Re: [Asterisk-Dev] bug guidelines page not found

[Brian West]

Apparently it was a victim of some sort of rm accident.

http://64.233.167.104/search?q=cache:sDPFrkBS_hUJ:www.digium.com/ 
bugguidelines.html+&hl=en&client=safari


/b

On May 25, 2005, at 11:53 AM, Matthew Boehm wrote:


As linked from here: http://bugs.digium.com/main_page.php

http://www.digium.com/bugguidelines.html

403 Page Not Found

-Matthew

--
-- 
--
Matthew Boehm, IT DirectorCypress  
Telecommunications
[EMAIL PROTECTED]   3838 N. Sam Houston Parkway E  
#400
T: 832-200-8640 x3044  Houston, TX  
77032


My girlfriend was recently diagnosed with multiple personality  
disorder;

 When she called yesterday, my CallerID box exploded.
-- 
--

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Re: [Asterisk-Dev] Dialplan syntax changes.. Option: work on a radically different design..

[Brian West]

call 996 we are in here talking about it right now.
And I said the exact same thing you did moments before I received the  
email.

/b
On May 18, 2005, at 9:40 AM, Steve Kann wrote:
Here's my opinion, as an observer (personally, most of this doesn't  
matter much to me at the moment, as my dialplans are stunningly  
simple, mostly because I handle most things via FastAGI).

Taking a step back, and re-examining the problem..
[I'll do a rumsfeld-style ask myself questions and answer them  
thing here]:

What is the dialplan?  It's really a procedural language, it seems.
Do we already have well defined syntax(es) that would fit?
Sure we do -- and most of them stem from the syntax of C in one way  
or another.

So, can it make sense to make the Dialplan look more like C?
I think it does.
Here's an example (based on some stuff in configs/ 
extensions.conf.sample

 snip 
 static=yes
 writeprotect=no
 autofallthrough=yes
CONSOLE=Console/dsp ; Console interface  
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest   ; IAXtel username/ 
password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface

macro LocalDial (dialed) {
   Dial($dialed);
   switch($?) {
   }
   return something;
}
context mycontext {
   extension 2125551212 () {
 Dial(SIP/Grandstream1,DIAL_RINGBACK|DIAL_ARGUMENT)
 switch ($?) {
case NOANSWER {
   2125551333 ();
}
case CONGESTION {
   Play(busy);
}
default {}
} }
   extension _212555 () {
 Dial(Local/$EXTEN);
   }
}

I think that we could get together and define a grammar for this,  
parse it with yacc/lex, etc.
There was lots of talk about the "pluggable PBX" concept earlier,  
on dev chats, etc.  I think if you're going to break everyone's  
dialplans anyway, you might as well really break them, and do it  
right.

It seems like what's happening here is that we're making the  
language look like spreadsheet expressions..

I'm surely oversimplifying the problem here, in at least two ways:
1) The dialplan is not only looked at as calls pass through it, but  
it is also consulted to determine the existence of extensions and  
pattern matching.

2) The dialplan was originally designed to be modifiable at run- 
time, where you could, e.g. change things in the CLI, and then have  
asterisk spit out the new dialplan.  But you can't do that now anyway.

-SteveK
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Re: [Asterisk-Dev] Loss of functionality with depreciation of DBGet/Put?

[Brian West]

${ISNULL()}
${EXISTS()}
function_db_read() will never return NULL.

Thats right.. EXISTS and ISNULL are just opposites...
NEXT!!!  ;)
Back atcha.

Thanks for the other fix on the bug tracker.. I was about to get  
around to that. ;)

/b
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Re: [Asterisk-Dev] Loss of functionality with depreciation of DBGet/Put?

[Brian West]

ok this madness works:
exten => 566,1,Dial(${IF(${EXISTS(${DB(CF/${CALLERIDNUM})})}?Zap/g1/$ 
{DB(CF/${CALLERIDNUM})}:SIP/10)})

The parser needs to be more sane with spaces.. but that works.
/b
On May 17, 2005, at 10:17 AM, Brian West wrote:
exten => 100,1,Set(temp=${DB(CFIM/200)})
exten => 100,n,GotoIF(${temp} = "" ? novalue)
exten => 100,n,...
exten => 100,n(novalue),...

exten => 100,1,Set(temp=${IF(${DB(CFIM/200)} = ${DB(CFIM/200)} ?  
novalue )})

Not sure if that is 100% correct but give it a go it should work...  
then maybe we need a GOTO function to operate similar?

exten => 100,1,Dial(${IF(${DB(CFIM/200)} = ${DB(CFIM/200)} ? "SIP/ 
default" )})

The set app is not a requirement to use the return value of the  
function (or shouldn't be)

/b
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Re: [Asterisk-Dev] Re: Fax and h323

[Brian West]

On May 11, 2005, at 9:36 AM, Steve Underwood wrote:
Mark told me it will go in as one of the pure GPL addons. It will  
not be
disclaimed. I wanted to know if I could use its ASN.1 code in my  
T.38
implementation. Because of this I haven't. This seems to mean the  
new
H.323 will have problems with things like G.729
Mark paid for the channel driver.. so that is fully Digium's (just  
not the ooh323c lib)

/b
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Re: [Asterisk-Dev] Re: New h.323 channel driver

[Brian West]

Current code has nothing to do with it really.  If you do follow  
the  readme with the included chan_h323 that you wrote all you get  
is a  nice segfault when the module loads.  It works fine with  
something  like redhat 9 or older slackware.  I suspect that using  
gcc 3.4.3 has  something to do with it.  This channel driver from  
ObjSys compiles,  loads and actually makes a call.  Which is more  
than I could ever get  your driver to do on the same box.

So instead of filing a bug report you simply bitch about it?
Go back to your hole.
My my we are bitter today aren't we...  I would file a report if we  
actually used your code.

/b
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Re: [Asterisk-Dev] Re: New h.323 channel driver

[Brian West]

Let me remind all you people, Digium PAID for this newest H.323  
driver to be developed.  For whatever reason is beyond me.  I have  
always had very good luck using reasonable volumes of traffic on my  
H.323 driver when using the proper versions of Open H.323 as  
defined in the README.
If you don't actually read the documentation, you will not make it  
work.
To have something they knew would work with some sort of  
reliability.  The channel driver in CVS and ooh323 both have issues  
and do not work for everyone.  If I take your chan_h323 compile and  
install it to the letter in the README it will segfault on load.  I  
compile and install this new driver which isn't even complete yet and  
I can at least make a phone call with it.  I do not mean any  
disrespect to you or your code ... this is just one more option that  
people will have.

/b
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Re: [Asterisk-Dev] Re: New h.323 channel driver

[Brian West]

If you would run even semi-current code chan_h323 performs  
wonderfully against As5300s - both inbound and outbound.

Think before you type.
Jeremy McNamara

Current code has nothing to do with it really.  If you do follow the  
readme with the included chan_h323 that you wrote all you get is a  
nice segfault when the module loads.  It works fine with something  
like redhat 9 or older slackware.  I suspect that using gcc 3.4.3 has  
something to do with it.  This channel driver from ObjSys compiles,  
loads and actually makes a call.  Which is more than I could ever get  
your driver to do on the same box.

/b
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Re: [Asterisk-Dev] [Rant] [long] - code style and quality

[Brian West]

You mean that thing where patches get posted... and sit for weeks/months/years rotting away into nothingness?   I sure hope when Kevin starts full time that will be a thing of the past.  Because we have missed out on a lot of great patches because the people committing just get tired and give up... I know I have been there./bOn May 7, 2005, at 12:07 AM, Jeffrey C. Ollie wrote:And perhaps you could have checked the bugtracker before getting all high and mighty.  ___
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Re: [Asterisk-Dev] [Rant] [long] - code style and quality

[Brian West]

haha thats funny.
/b
I'd second that recommendation even though the book came out of the  
Microsoft Press!

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Re: [Asterisk-Dev] Manager proposal

[Brian West]

I have done a nice proof of concept in the past few hours to test the  
idea of an XMLRPC interface.  I did it using res_perl which allowed  
me to prove it was a wise idea.

I did two simple ones.. asterisk.ShowChannels and  
asterisk.ShowUptime ... its possible but alot of the code and output  
in asterisk will need to have concise or machine readable output that  
is sane.

<432>:perl channels.pl
$VAR1 = {
  'channels' => {
'SIP/16-e3e1' => [
 'SIP/16-e3e1',
 'default',
 '999',
 '4',
 'Up',
 'MusicOnHold',
 'default',
 '16',
 '',
 '3'
   ]
      }
};
<433>:perl uptime.pl
$VAR1 = {
  'uptime' => '849'
};
/b

On May 4, 2005, at 5:15 PM, Brian West wrote:
I VOTE for totally removing the manager... and replacing it with  
XML-RPC .. then each module can register methods when it loads... ;)

(and the flames start w00sh)
/b
On May 4, 2005, at 9:29 AM, Olle E. Johansson wrote:

I see a lot of managare events named "Hold" and "Unhold",  
"ParkedCall", "UnParkedCall".

I would like to have *one* event name, say "Hold"
and then an additional paramater signalling the state, like
Event: Hold
Holdstatus: On | Off
Event: ParkedCall
PCstatus: Parked | Unparked | GivingUp
Does that make any sense?
For me, it would be much easier writing a parser... We do this in the
channel with registration status - one event name, then different  
status lines marking the current status.

/O
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Re: [Asterisk-Dev] Dev call 1.2 release discussion

[Brian West]

6) unloading modules on shutdown (should help chan_h323 to
unregister from Gatekeeper on Asterisk's shutdown) -
discussed long time ago in #asterisk-dev;
Patch please??


He did in fact do a patch.  It sat on the bug tracker and rotted away  
because the powers that be never had the vision to see why it was  
actually needed.

/b
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Re: [Asterisk-Dev] Dev call 1.2 release discussion

[Brian West]

On May 6, 2005, at 12:34 AM, Paul Cadach wrote:
Hello,
I would like to see in 1.2:
1) working/stable h.323 stack - #3967 and other;
Might happen... Might not by then.
2) configurable PRI functionality (facilities, IEs per individual  
connections, not per switch type) - no ticket
available but -dev list have reports;
3) loadable language syntax modules (with russian support) - #3832  
(russian support is pending);
This is one i'm pushing for too.
4) event-based MWI - #2980;
I would like this also.  But I don't suspect it will be done soon  
unless you're going to provide a patch?

5) T.38 support;
Do pigs fly this month?  Doubt it.
6) unloading modules on shutdown (should help chan_h323 to  
unregister from Gatekeeper on Asterisk's shutdown) -
discussed long time ago in #asterisk-dev;
I see now why this is needed COME ON MARK I think paul did a  
patch for this once right?

7) internal timing for indications and moh without zaptel hardware.
NEVER EVER EVER EVER gonna happen unless someone codes it.
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Re: [Asterisk-Dev] Fedora 3 - make file issue (-lidn)?

[Brian West]

Well curl can be compiled with and without idn support.
/b
On May 2, 2005, at 6:00 PM, Rich Adamson wrote:
Seems it bailed out while compiling app_curl, and required the dev
package to proceed. So, I guess 1=1=1 and we can point at any "one"
of those; but still 'required'.
Seems a little odd that it wasn't required in RHv9, just fedora 3.
(That's why I was kind of guessing at maybe a Makefile issue.)

Why?  Its a libcurl dependancy ... Not an asterisk one.   Should be
now list the ENTIRE dependancy tree ?
Autoconf anyone?
/b
On May 2, 2005, at 4:28 PM, Rich Adamson wrote:

For archive purposes, the package required was libidn-devel
(Thanks Brian)
The asterisk.org/download page should probably be updated to
note the above 'required package'.
This test system is now running just fine with MOH, x100p, etc.
Will be swapping for a TDM04b (for test purposes) later today.


install idn-devel
/b
On May 2, 2005, at 1:03 PM, Rich Adamson wrote:

Trying to compile cvs-head from this morning on a new fedora 3
install.
Zaptel and libpri compiled fine, asterisk compile blows up with
 /usr/bin/ld: cannot find -lidn
while trying to make app_curl.so. All other asterisk/apps compiled
into
.so files except for app_curl.
idn is located in /usr/bin and I did add that to the ld config.
I'm certainly not strong on reading Makefile content, but it  
almost
looks like that is required for bsd and not for fedora 3.

Thoughts?
Rich
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Re: [Asterisk-Dev] I don't need libpri..please continue...

[Brian West]

I bet its just finding the libpri headers..
/b
On May 2, 2005, at 5:34 PM, Andrew Kohlsmith wrote:
On May 2, 2005 06:15 pm, Matthew Boehm wrote:
chan_zap.c:61:2: #error "You need newer libpri"
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
The machine I am compiling asterisk on has no PRI/Zap cards  
connected to
it. So why is libpri necessary to download? Shouldn't it say "You  
don't
have libpri installed so I am skipping the following..." and keep
compiling?

Basically it sees you have zaptel installed so it's trying to compile
chan_zap, which requires a newer libpri.
If you're sufficiently proficient, work around it.  Otherwise, just  
do what it
says; it's not like libpri's huge.  :-)

-A.
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[Asterisk-Dev] Benchmark numbers....

[Brian West]

In my testing of CVS-HEAD I can get 5551+ sip "sessions" without  
media on asterisk without a problem.  The load average is around 2-3.

On a side note... on my 3ghz P4 HT box I can get 629 ulaw sip calls  
with media (verified) without a problem.  The load average on the box  
was around 14 and it still sounded perfect... so if you had a dual  
3.4 ghz Xeon box you should have ZERO problems doing a DS3 with  
asterisk. (That is if the interrupt is 1000 per second and not 28000  
and its all ulaw)

I'll be doing more testing later this week.
/b
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Re: [Asterisk-Dev] Alphanumeric Extensions Proposal Comments

[Brian West]

I'm still trying to come up with a better solution in my mind its  
maddening to go over all this info and see what could be the best way...

Because when you stop and think about it regexp on the exten portion  
would be nice on both alpha and numeric extension types.

/b
On May 2, 2005, at 9:34 AM, Michael Giagnocavo wrote:
In response to Olle and Leif's proposal:
http://edvina.net/asterisk/alphanumericextensions.pdf
Starting with Asterisk dialplans being ASCII.. Why? Wouldn't that  
make it
relatively hard to use any other characters in Asterisk, if your  
dialplan is
so limited? I'd imagine we'd want the same format for other files  
too. Thus,
I don't understand how starting off with ASCII gets us going  
anywhere good.

The fact that some Caller ID implementations might use an old  
encoding like
ISO 8859-1 shouldn't determine how Asterisk handles things  
internally. One
of the nice things about Asterisk is that it's designed quite well  
and aims
to do things correctly, even when a lot of other surrounding  
things, well,
aren't done so well.

Quite frankly, I think that Unicode is one thing that should just  
be there,
in the background, working. In the dialplan and elsewhere, we  
shouldn't have
to manipulate those strings any differently. Asterisk should use  
Unicode all
over, and when something needs a string in a different format, then  
Asterisk
converts. That'd keep a lot more in line with what Asterisk does  
regarding
other things (say, protocols and codecs) than making us (as users of
Asterisk) worry about the details of certain protocols.

-Michael
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Re: [Asterisk-Dev] asterisk "stable"?

[Brian West]

Roy Sigurd Karlsbakk wrote:

Hi
running 1.0.6/1.0.7, I keep seeing numerous bugs such as huge  
memory leaks, rtp overflows, application bugs etc etc. Wasn't the  
1.0.x track meant to be stable? Or does "stable" only mean  
"feature freeze"? Will asterisk 1.2.x be just as badly tested as  
the 1.0.x track?
roy


What kind of response do you expect to that?
Why not just post the bug id's of the bugs you have found  
(preferably with patches).

In addition to this WHAT THE HELL are you doing to it?  I have boxes  
with MASSIVE uptimes and have NEVER seen these issues.  I do know one  
thing if you do not set the ulimit -n 10 or something similar  
before you start asterisk you'll run out of FD's around 151 calls.

/b
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Re: [Asterisk-Dev] Proposed patch.

[Brian West]

If you swap the order.. check for the IDIOT env first.. before the  
strcmp.. oh well I give up.

/b
On Apr 29, 2005, at 3:09 PM, Kevin P. Fleming wrote:
Brian West wrote:

Are you be sarcastic or truthful?  haha  Because we do this in   
frame.c already.  Plus its a one time hit on load/reload.

A one time hit? You are calling getenv() every time you see that  
pattern during a configuration load. In frame.c, it happens only if  
the user types that special command.

Granted, it's not going to affect call handling performance, but it  
would still be better to just check it once during the load (since  
it can't change while Asterisk is still running) and then reference  
an int variable with the value set appropriately.
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Re: [Asterisk-Dev] Bounty: app_chanspy

[Brian West]

Why not get a backtrace and post a bug?
/b
On Apr 28, 2005, at 8:51 PM, Dan Fernandez wrote:
I have a CVS version from a few days ago and chanspy also crashes  
asterisk.


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[Asterisk-Dev] ast_find_ourip and sip?

[Brian West]

Ok this is one that i'm sure some of you have seen before.  Why MUST  
sip resolve the hostname to an IP?  It doesn't really use it so why?   
If you set /etc/hosts with the hostname then the ip of 127.0.0.1 then  
it works.. but then again why not just fall back to that instead of  
just disabling sip totally?Also is it wise to embed the root  
nameserver's ip into the function?  I'm going for NOT

/b
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Re: [Asterisk-Dev] Queues observations

[Brian West]

; This setting controls whether callers can join a queue with no members. There are three                                                                       ; choices:                                                                                                                                                      ;                                                                                                                                                               ; yes - callers can join a queue with no members or only unavailable members                                                                                    ; no - callers cannot join a queue with no members                                                                                                              ; strict - callers cannot join a queue with no members or only unavailable members                                                                              ;                                                                                                                                                               ; joinempty = yes                                                                                                                                               ;                                                                                                                                                               ; If you wish to remove callers from the queue when new callers cannot join, set this setting                                                                   ; to one of the same choices for 'joinempty'                                                                                                                    ;                                                                                                                                                               ; leavewhenempty = yes                                       It looks like the first part of the request is done.  As for the second part I have never heard of giving 2 seconds of ring back before connecting to an agent nor have I heard of any other system I have ever called doing this./bOn Apr 26, 2005, at 9:06 PM, Stephan A. Edelman wrote: We've implemented the call Queuing using asterisk and have the following observations:   (1) The Queue() application returns zero when there is no agent available to take the call (busy on another call, for example) but also when there is no agent logged in for that queue. The latter is fairly problematic. I would prefer to have the caller dumped into voicemail rather than waiting for an agent to eventually login.   For this purpose, it looks like we need an AgentIsAvail(queuename) function, or something similar on which we can branch in the dialplan.    (2) When the caller is eventually connected to an agent, the MOH disappears and then immediately the agent is heard. The caller gets no prior indication / warning that his call is now connected. I think we need an option in the Queue() to create a 2-second ringback prior to being connected.   Any thoughts? Stephan.___Asterisk-Dev mailing listAsterisk-Dev@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-devTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-dev ___
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Re: [Asterisk-Dev] Dev Meeting this week.

[Brian West]

I had 2 concerns with ARA but when I popped into the conference  
people were
discussing bugs on mantis and I didn't feel like interrupting.

-Matthew
Na.. Dive in.. No more real bug talk unless its an issue that we are  
trying to fix that has an open bug... so don't be scared to jump in.

/b
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Re: [Asterisk-Dev] Dev Meeting this week.

[Brian West]

I just brought that up in here.  We'll see if we can get someone on  
this.. is their a bug open on this yet?

/b
On Apr 21, 2005, at 1:53 PM, Brian Capouch wrote:
Brian West wrote:
So nobody has any interest in giving input?  I hear people bitch  
and  moan all the time.. but when we have the chance to really  
give some  input.. either here on the list.. or in the conference  
call we have  WEEKLY.  So lets give some input

I have no developer's chops, as everyone knows, but pretty heavy  
user interest in the development situation.

There is a lot of latent interest in native IAX encryption out  
there. It's broken and I've done all I can do (some valgrind  
snatches) to furnish information as to where the bugs are lurking.

I'm not sure if this is the sort of input you were looking for. . .
Thx.
B.
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Re: [Asterisk-Dev] Donate some sip phones to STABLE maintiner

[Brian West]

I wouldn't trust that... :P
/b
On Apr 19, 2005, at 10:55 PM, Kevin P. Fleming wrote:
Brian West wrote:
yep it can but to fully test and make sure you don't break things you 
must make clean and start fresh.  I would expect that.
That's actually the whole point of ccache... after you have run 'make 
clean', it can tell whether some (or all) of the sources you are 
building have not changed (nor have anything they include), so it just 
gets the previously built object file from the cache.
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[Asterisk-Dev] Dev Meeting this week.

[Brian West]

Ok guys this week Mark will not be at the meeting this week.  I'm going 
to start collecting up information we want to cover.  Please reply with 
things including bugs/patches you wish to talk about.  Remember its 1PM 
CST on the 21st and usually lasts 3-5+ hours.   Also just to let you 
guys know people hang out in the conference a lot of the time all hours 
of the day.  So you can pop in and talk any time all week long.

Also join #996 on freenode (its linked to the bot in the conf)
IAX2/[EMAIL PROTECTED]/996
These meetings were not to be "lets squash bugs" meetings so we are not 
going to be doing this in the thursday meetings anymore(unless its a 
bug that we do need to talk about that might change the core).  We will 
have another day we will plan for bug squashing with mark or kevin.

Please everyone show up bring your ideas and lets hash out some 
direction... a strong push forward in development is what we are going 
for.

Thanks,
Brian
Asterlink.com
Also ask me about ClueCon
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Re: [Asterisk-Dev] Donate some sip phones to STABLE maintiner

[Brian West]

yep it can but to fully test and make sure you don't break things you 
must make clean and start fresh.  I would expect that.

/b
On Apr 19, 2005, at 2:18 PM, Scott Laird wrote:
This would be a good time to point out 'ccache'--it can make a *huge* 
difference when you're recompiling slightly-changed source trees over 
and over again.

Scott
On Apr 19, 2005, at 10:18 AM, Brian West wrote:
Also guys he compiles asterisk tons of times a day on a 700mhz box... 
  can any of us help him out a bit on that?  I don't have a good 
asterisk box myself yet (coming soon) so i know how he feels trying 
to compile asterisk 100's of times a day on a slow box.

/b
On Apr 19, 2005, at 8:33 AM, Terry Wilson wrote:
Russell, Nuvio will donate a 7960, a Polycom IP300, and a Motorola 
ATA
if you would like.  Send me your shipping info and I will get them
sent out today.

On 4/14/05, Jared Smith <[EMAIL PROTECTED]> wrote:
On Wed, 2005-04-13 at 15:20 -0500, Brian West wrote:
Does anyone have any SIP phones they could send Russell so he 
could do
more testing on stable?
I've got a BarbieTone I'll donate to the cause -- Russell, please 
send
me your address out-of-band.

-Jared
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Re: [Asterisk-Dev] Donate some sip phones to STABLE maintiner

[Brian West]

compile farm is hard to test pci cards on isn't it :P
/b
On Apr 19, 2005, at 3:05 PM, Magnus Espeland wrote:
On 4/19/05, Brian West <[EMAIL PROTECTED]> wrote:
Also guys he compiles asterisk tons of times a day on a 700mhz box...
can any of us help him out a bit on that?  I don't have a good  
asterisk
box myself yet (coming soon) so i know how he feels trying to compile
asterisk 100's of times a day on a slow box.
hmm, according to this site, asterisk has access to the OSDL, so he
should be able to compile there.. on several architectures:
http://osdl.org/lab_activities/lab_projects/active_projects/ 
display_projects.html?query=Active

Best regards,
Magnus Espeland
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Re: [Asterisk-Dev] Mini Trame destination

[Brian West]

Do you mean mini "F"rames?
/b
On Apr 18, 2005, at 9:02 AM, Tutu Lord wrote:
Hello,
I am a french student and i must understand the IAX2 protocol. This 
protocol isn't developed in France. So after few weeks of study, there 
is a point which is not clear:

How Mini Trames arrive to destination whitout Destination Call Number 
flield in their overhead.

Thanks you,
Lucas
_
MSN Hotmail : choisissez votre adresse @hotmail.fr 
http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR

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Re: [Asterisk-Dev] BOUNTY: app_hangup from exten => h

[Brian West]

And now that I think about it... having _. not match the 1 letter 
o,s,h,i,t,a extensions would be a more preferable solution to having 
the warning in CVS.

/b
On Apr 15, 2005, at 11:23 AM, Brian West wrote:
No, I meant the "i" extension doesn't work.  The _. works fine as
as catch-all no-match kind of thing for the entire dialplan.

Exactly.  Why do people doubt me?  In my example yesterday I showed an 
example of proper usage to people.  As long as you understand the 
proper usage of the _. you're fine.   I see Mark added the warning to 
CVS anyway.  That still will do no good _. has its use and then 
telling people you shouldn't use it then we might as well remove it if 
you shouldn't use it.  I also explained this at Astricon when I spoke.

Just because you don't understand the proper usage of something 
doesn't mean we should plaster a warning on it nor should we bitch and 
moan about it.  The proper course of action is to document its proper 
usage and make sure people know how to use it.

People recommend using exten => i which is totally wrong if you wish 
to capture the invalid extension the person dialed since ${EXTEN} 
would be "i" instead of the number the person dialed. (which can be 
helpful ya know).

I vote to have the warning removed from CVS it DOES NOT BELONG THERE.
/b
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Re: [Asterisk-Dev] BOUNTY: app_hangup from exten => h

[Brian West]

No, I meant the "i" extension doesn't work.  The _. works fine as
as catch-all no-match kind of thing for the entire dialplan.

Exactly.  Why do people doubt me?  In my example yesterday I showed an 
example of proper usage to people.  As long as you understand the 
proper usage of the _. you're fine.   I see Mark added the warning to 
CVS anyway.  That still will do no good _. has its use and then telling 
people you shouldn't use it then we might as well remove it if you 
shouldn't use it.  I also explained this at Astricon when I spoke.

Just because you don't understand the proper usage of something doesn't 
mean we should plaster a warning on it nor should we bitch and moan 
about it.  The proper course of action is to document its proper usage 
and make sure people know how to use it.

People recommend using exten => i which is totally wrong if you wish to 
capture the invalid extension the person dialed since ${EXTEN} would be 
"i" instead of the number the person dialed. (which can be helpful ya 
know).

I vote to have the warning removed from CVS it DOES NOT BELONG THERE.
/b
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Re: [Asterisk-Dev] BOUNTY: app_hangup from exten => h

[Brian West]

Rich,
	Your original example was correct.  Not only can it work you get to 
know what the moron dialed.

/b
On Apr 15, 2005, at 9:08 AM, Rich Adamson wrote:
On April 15, 2005 08:06 am, Rich Adamson wrote:
; This section is the Last and handles 'no valid extension'
[no-match]
exten => _.,1,Answer
exten => _.,2,Playback(invalid,skip)
exten => _.,3,Hangup

where none of the includes have a _. except for the "no-match"
catch all? That catch all is simply there to say "hey dummy,
there is no match in the dialplan. Try again."
As I suggested:
[no-match]
exten => i,1,Answer
exten => i,2,Playback(invalid,skip)
exten => i,3,Hangup
Am I missing something, or does the _. catchall catch an invalid 
extension
better than i?
Tried that, doesn't work. So far the only thing that can catch
an invalid extension is _. as shown above.
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Re: [Asterisk-Dev] Re: BOUNTY: app_hangup from exten => h

[Brian West]

On Apr 15, 2005, at 6:37 AM, Andrew Kohlsmith wrote:
On April 14, 2005 03:07 pm, Brian West wrote:
I disagree; This bounty's created some VERY GOOD discussion, and I
would love
to see some real-life useful examples of using _. to capture special
extensions.
[contexta]
include => catchall
exten => _NXXNXX,1,Dial(blah)
exten => h,1,Hangup
[catchall]
exten => _.,1,Playback(yousuck)
exten => i,1,Playback(yousuck)
Hardly "NEXT!!!", as the 'i' extension is *designed* for this.
i, doesn't let you catch the true ${EXTEN} that they tried to dial now 
does it.  ${EXTEN} would = i which is NOT WHAT YOU WANT.

NEXT!!!
/b
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Re: [Asterisk-Dev] Re: BOUNTY: app_hangup from exten => h

[Brian West]

I don't think this patch is one thats needed.  You can try all you want 
to keep stupid people from doing stupid things... Its still going to 
happen.

/b
On Apr 14, 2005, at 9:40 AM, Eric Wieling wrote:
Andrew Kohlsmith wrote:
On April 14, 2005 10:01 am, Tony Mountifield wrote:
There is no app_hangup.c. The Hangup command is implemented in pbx.c 
by
the function pbx_builtin_hangup(), which does nothing except return
non-zero. The PBX core then initiates a hangup because the app 
returned
a non-zero status, which it would for ANY app that did so.

In my opinion the correct fix would be for Asterisk, when an app 
returns
non-zero, to check if it has already called the h extension for the
channel, and if so, NOT to go to h,1 again. This should be done using
a flag rather than checking the extension, so that it still works if
the dialplan does a Goto out of the h extension to somewhere else.
Have you ever seen Asterisk loop when Hangup is called from h?  I 
certainly never have...
No, but I have seen many a time where people have exten => _.,Hangup 
which is not a good thing most of the time.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Dev] G729 Question

[Brian West]

I think we already covered this in the dev conf didn't we?
/b
On Apr 7, 2005, at 2:21 PM, Tom Dickenson wrote:
SOMEONE WANT TO EXPLAIN THIS? While finally placing an outgoing call 
using
mutualphone (requiring that I get a G.729 License to do this) the 
following
pops up on asterisk's cli when a call is made (call works wondeful) 
just
curious to the extent of the following lines...

Attempting native bridge on SIP/cisco1-5d0b and SIP/mutualphone-c59c
Apr  7 12:01:58 WARNING[726]: rtp.c:1559 ast_rtp_bridge: codec0 = 12 
is not
codec1 = 256, cannot native bridge

-Tom Dickenson
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Re: [Asterisk-Dev] Attn: Bug Marshalls

[Brian West]

Also an FYI
Bugs that are resolved (that means the ones that are in green) get 
toggled to closed when they get applied to stable when Russell gets 
around to them.  If you reopen a bug that can apply to stable please 
make sure you NOTE this and then re-resolve the bug if you can.  If not 
find a bug marshal to do this for you.

Thanks,
/b
On Apr 1, 2005, at 1:21 AM, Brian West wrote:
Russell,
	I'll help over the weekend.. we can all jump on the conf and go over 
them if you like.
/b

On Mar 31, 2005, at 8:45 PM, Russell Bryant wrote:
Hey guys,
At the time of this writing, there are about 75 bugs marked as 
"Resolved" on the bug tracker.  I plan to work hard this weekend to 
catch up, but I surely would not turn down any help if anyone is 
bored.

If you would like to help me catch up, just look through the resolved 
bugs and close the ones that obviously don't need to be considered 
for 1.0 (new features, for fixes for things only in cvs head).

As far as the ones that do need to be considered for stable, there is 
plenty of work to be done there as well.  If you want to get some 
karma, you can help port some patches!

Thanks,
Russell
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Re: [Asterisk-Dev] Attn: Bug Marshalls

[Brian West]

Russell,
	I'll help over the weekend.. we can all jump on the conf and go over 
them if you like.
/b

On Mar 31, 2005, at 8:45 PM, Russell Bryant wrote:
Hey guys,
At the time of this writing, there are about 75 bugs marked as 
"Resolved" on the bug tracker.  I plan to work hard this weekend to 
catch up, but I surely would not turn down any help if anyone is 
bored.

If you would like to help me catch up, just look through the resolved 
bugs and close the ones that obviously don't need to be considered for 
1.0 (new features, for fixes for things only in cvs head).

As far as the ones that do need to be considered for stable, there is 
plenty of work to be done there as well.  If you want to get some 
karma, you can help port some patches!

Thanks,
Russell
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Re: [Asterisk-Dev] app_meetme fix for entry beep with count users

[Brian West]

You do realize meetme can do pins ... and dynamic pins...
/b
On Mar 28, 2005, at 6:00 PM, Jared Mauch wrote:
since using the 'c' (count users in conf flag) in cvs head
there is no longer a beep when users join.
this should resolve that issue/solves it for my case.
here's the flags i usually use with my conference:
[macro-stdconf];
exten => s,1,Answer
exten => s,2,AbsoluteTimeout(86400)
exten => s,3,Wait,1
exten => s,4,Authenticate(${ARG2})
exten => s,5,Meetme(${ARG1}|cdpMs)
exten => s,6,Hangup
(i've got my disclaimer filed, but can't seem to recall
if i have a mantis login right now.. but either way, this is
fairly straightforward).
cvs server: Diffing .
Index: app_meetme.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_meetme.c,v
retrieving revision 1.91
diff -u -r1.91 app_meetme.c
--- app_meetme.c21 Mar 2005 03:23:05 -  1.91
+++ app_meetme.c28 Mar 2005 23:01:10 -
@@ -729,6 +729,7 @@
else if (res == -1)
goto outrun;
}
+   conf_play(chan, conf, ENTER);
}
}
--
Jared Mauch  | pgp key available via finger from [EMAIL PROTECTED]
clue++;  | http://puck.nether.net/~jared/  My statements are only 
mine.
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Re: [Asterisk-Dev] patches directory...

[Brian West]

Those recordings are CRAZY eh?  hahah
/b
On Mar 23, 2005, at 3:52 PM, Chris Wade wrote:
listening to the 3-17 dev conf...
the issue of the patches directory and it 'being part of the product', 
why not just create a 'asterisk-patches' just like 'asterisk-addons'?

might not resolve everyones issues with this 'make patch' idea, but 
isn't it a step in the right direction?

just a thought,
-chris
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Re: [Asterisk-Dev] G.722 and Grandstream

[Brian West]

lets see da code.. I wanna test this as soon as my get my grandstream :P
/b
On Mar 21, 2005, at 3:42 PM, Andrew Lindh wrote:

Steve Kann wrote:
You do realize that G.722 audio is 16kHz, and there's code all over
asterisk that assumes 8kHz; even if you decode G.722, you still need 
to
resample to get 8kHz, and to encode to G.722, you'll need to resample 
to
16kHz. Then there's all the code that assumes that ms = samples/8, 
etc..
Yes, and that's not important.G.722 is a 64Kbit/sec (for mode 1)
data stream. So dealing with the raw G.722 is easy because it
looks just like G.711. So with any raw G.722 it does not matter that
it's 16Khz audio. On the codec conversion side, yes it would need to be
resampled to mix and match 8Khz/16Khz. Quick and easy for testing is
to just drop or add data words to match the rate. I was not planning on
supporting raw 16Khz in asterisk at all, it would always be raw G.722 
or
converted to 8Khz slin audio. Anything that it would be converted to
(for now) is 8Khz based anyway, so there would be no issue of quality 
loss.

In my quick test I recorded a G.722 audio stream (just from the 
dialplan
record function) and I can play it back to the phone (from the 
dialplan).
When I use an external G.722 audio encoder/decoder I don't get valid 
audio
from the conversion. So my big issue is I don't have G.722 codec code 
and
information that matches the G.722 on the BT-100 phone

  Andrew
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Re: [Asterisk-Dev] Mac OS X Compile Problems and FYI

[Brian West]

bison is all I had to upgrade.. but I haven't tried in the past two 
weeks.

/b
On Mar 21, 2005, at 10:06 AM, Matt Fredrickson wrote:
On Sun, Mar 20, 2005 at 02:00:04PM -0600, Matthew Boehm wrote:
(seems my attachment was too big for the list. I've posted the 
compiler
output here: http://www.pastebin.com/260676 )

Running OS X 10.3.8, gcc version 3.3 20030304 (Apple Computer, Inc. 
build
1495)

Comes with bison v1.28 which will not work when compiling ast_expr.y
Had to upgrade to bison v2.0.
Also had to upgrade to sed 4.1.4
I think there is a page on the wiki for getting Asterisk to compile on 
MacOSX.

I believe it covers the upgrade on bison, but I can't remember 
anything about
an upgrade of sed (I don't think I had to do that at least).  You may 
check to
see if it needs updating.

Matthew Fredrickson
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Re: [Asterisk-Dev] rewriting asterisk as a state machine? - Not

[Brian West]

I would like to know too... since it was only possible to get about 185 
or so threads till very very recently.

/b
On Mar 4, 2005, at 9:14 AM, Andrew Kohlsmith wrote:
On March 4, 2005 09:52 am, Paul Mahler wrote:
We are getting over 5000 simultaneous calls with our PBX hardware 
with less
than 50% CPU utilization, so I'm not sure this is much of a problem. 
;-)
What hardware and what types (channels) of calls?  I'm just curious,
especially as to what you do for redundancy and failover.
-A.
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Re: [Asterisk-Dev] dev conference: missed again

[Brian West]

I have done this.. but what is wrong with using wget?
/b
On Mar 3, 2005, at 8:05 PM, Edwin Groothuis wrote:
On Thu, Mar 03, 2005 at 07:00:27PM -0600, Brian West wrote:
http://bkw.digiweb.com/devconf/
This is where it will show up once its over.. the others are there 
now.
Can you please tell your webserver to use the right mime-type for
it? Right now it's sent as text/plain and I suspect that's why sox
makes rubbish of it.
Edwin
--
Edwin Groothuis  |Personal website: 
http://www.mavetju.org
[EMAIL PROTECTED]|  Weblog: 
http://weblog.barnet.com.au/edwin/
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Re: [Asterisk-Dev] dev conference: missed again

[Brian West]

http://bkw.digiweb.com/devconf/
This is where it will show up once its over.. the others are there now.
/b
On Mar 3, 2005, at 6:27 PM, Brian West wrote:
Ok this conference can be purchased only in DVD or a 3 CD set for a 
Low Low Price.  At this rate we will be in here till 10,11 or 12

So by then it should fit on a DVD-R.
/b
On Mar 3, 2005, at 5:13 PM, Matthew Boehm wrote:
missed it again..geez..i should setup asterisk to call me at 1PM.
is there a "here-is-what-we-talked-about" topic listing somewhere? I 
don't
want to add to BKW's "posting-the-audio-portions-generosity" but if a 
topic
list were put somewhere, that would be great!

-Thanks,
Matthew
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Re: [Asterisk-Dev] dev conference: missed again

[Brian West]

Ok this conference can be purchased only in DVD or a 3 CD set for a Low 
Low Price.  At this rate we will be in here till 10,11 or 12

So by then it should fit on a DVD-R.
/b
On Mar 3, 2005, at 5:13 PM, Matthew Boehm wrote:
missed it again..geez..i should setup asterisk to call me at 1PM.
is there a "here-is-what-we-talked-about" topic listing somewhere? I 
don't
want to add to BKW's "posting-the-audio-portions-generosity" but if a 
topic
list were put somewhere, that would be great!

-Thanks,
Matthew
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Re: [Asterisk-Dev] MOH using native codecs in stable branch

[Brian West]

On Mar 2, 2005, at 6:59 PM, Greg Boehnlein wrote:
On Wed, 2 Mar 2005, Brian West wrote:
Just use CVS-HEAD Think just because the "stable" branch has the
word stable in it... doesn't mean its stable.  CVS-HEAD is no less
stable than the stable branch.
I just don't understand comments like this. The entire reason that a 
1.0
branch exists is to snapshot a release at a specific level, and work to

The problem is stable is old and is missing alot of the features people 
are wanting.  Honestly we run CVS-HEAD without fail.  I can get WEEKS 
of uptime and no crashing but we have done a lot of custom mods to 
asterisk that may never make it back to the public due to the fact that 
getting patches into CVS take ages and its just a pain in the ass to 
keep patches up to date when they sit on the bug tracker and rot away.

/b
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Re: [Asterisk-Dev] Digium's G.729A codec problem

[Brian West]

Ok lets fix this now... Open logger.conf and take debug,warning and 
notice away from the console line.  Then you won't have to ask WHY you 
get errors/warnings/notices.  99% of the time they are meaningless.

/b
On Mar 2, 2005, at 6:38 PM, Andrew Kohlsmith wrote:
On March 2, 2005 05:50 pm, Jeremy McNamara wrote:
There is no OPEN SOURCE implementation of G.729.Refrain from 
posting
illegal links.
He never said it was open source, and his website clearly says you 
need to
adhere to your country's patent laws...  I don't see the problem?

-A.
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Re: [Asterisk-Dev] Manager XML

[Brian West]

I recommended that already and he pretty much said NO.  He said he was 
open to new ideas... except when it comes to this one :P

/b
On Mar 2, 2005, at 4:47 PM, Jeremy McNamara wrote:
Olle E. Johansson wrote:
Remember, I am not suggesting that we remove the current format,
just add a new optional parseable format with complete data instead 
of the stripped data being presented in the CLI.

Why don't we take Tim Clark's suggestion and move the manager from the 
core to be a module?  Then anyone can create their own method of  
communication as they see fit.

Jeremy McNamara
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Re: [Asterisk-Dev] MOH using native codecs in stable branch

[Brian West]

Just use CVS-HEAD Think just because the "stable" branch has the 
word stable in it... doesn't mean its stable.  CVS-HEAD is no less 
stable than the stable branch.

/b
On Mar 2, 2005, at 2:50 PM, Juan Jose Comellas wrote:
Is anybody using MOH without mpg123 (using native codecs) in the 
current
stable (v1-0) branch. I've seen that the patch to allow this has been 
checked
into CVS (bug #2369). I tried to patch the stable branch with it but it
didn't apply cleanly, so I'm currently porting it to the stable 
branch. Is
there anything special that I have to take into account when doing 
this?

--
Juan Jose Comellas
([EMAIL PROTECTED])
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Re: [Asterisk-Dev] Digium's G.729A codec problem

[Brian West]

G729A vs G729B ... they are stream compatible.  B on the other hand has 
VAD and a few other things.  B is more complex than A but they are 100% 
compatible.

/b
On Mar 2, 2005, at 5:14 AM, Jacky wrote:
Hi, all,
I have buy 5 Digium's G.729A codec(it just support G.729A license)
When I  calll with 2 SIP UA that support G.729A and G.729B, its rtp 
frame
have some problem when softswitch with Asterisk.

The voice frame have been drop, so sometime I can't hear voice.
If I want to fix the problem when softswitch G.729A and G.729B codec.
What source code I must to modify ?
Or some people have finished the issue, Could you show me how to do?
--
Jacky
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Re: [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)

[Brian West]

Well in this case his patch would break it.
/b
On Feb 26, 2005, at 11:10 AM, Jared Smith wrote:
On Sat, 2005-02-26 at 18:45 +0100, Frank van Dijk wrote:
I ran into an issue with the way asterisk sends rfc2833 DTMF events. 
As
my days of experience with asterisk can be counted on one hand I would
like to hear your expert opinion on the attached patch that solves the
problem for me, or maybe your opinion on other ways to solve the 
problem.

On the non-technical side of things, standard operating procedure is to
add the bug to the bug tracking system (bugs.digium.com), and make sure
you have a disclaimer on file with Digium.  After that's done, then 
it's
a good idea to announce your patch in this mailing list so that it can
be discussed.

On the technical side, it looks pretty straightforward to me.  Does
anybody with more Asterisk coding experience have anything to say about
this patch?
-Jared Smith
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[Asterisk-Dev] Re: rfc2833 DTMFs sent with bad timestamps (patch)

[Brian West]

Frank,
	The RFC states that the timestamp you send on RFC2833 SHOULD NOT 
increment from the first one sent and each following dtmf frame sent 
should use the exact same timestamp as the first one.  Only sequence 
number should increment.  If you use RTP debug in asterisk and compare  
how say a Cisco 7960 or a Sipura send DTMF they act in the same manner. 
 If you increase the timestamp on the DTMP packet you'll break it and 
the gateways such as a Lucent TNT or cisco will totally ignore the 
digits being dialed and all you'll hear is a clipy missing dtmf frame.  
I know this because i'm the one that spent 3 hours reading the mind 
warping RFC and talking to mark.  I think your gateway is what is 
broken.

http://bugs.digium.com/bug_view_page.php?bug_id=0002928
/b
On Feb 26, 2005, at 11:45 AM, Frank van Dijk wrote:
Hi
I ran into an issue with the way asterisk sends rfc2833 DTMF events. 
As my days of experience with asterisk can be counted on one hand I 
would like to hear your expert opinion on the attached patch that 
solves the problem for me, or maybe your opinion on other ways to 
solve the problem.

I ran into the issue using 1.0.3 on BSD, but looking at the latest 
rtp.c the issue is still there. I have an asterisk forwarding RTP 
streams between a cisco ISDN/VOIP gateway and an intel HMP. When a 
caller (in the PSTN) enters some DTMFs in quick succession, asterisk 
sends the digits without voice packets in between. That is ok, except 
that it re-uses the timestamp of the first digit for the following 
digits. The HMP box receives digits that claim to have occurred 
simultaneously and it ignores all but the first one. I've seen mention 
of this issue in some bug reports, but it seems the problem persists 
(CMIIW).

The cause of the issue is the fact that ast_rtp_senddigit() in rtp.c 
uses rtp->lastts as the timestamp. lastts is the timestamp of the 
latest voice packet. If no voice packets are sent, lastts does not 
change. The patch below uses (lastts + number of ms elapsed since 
then) as timestamp for sending DTMF events. The diff is against rtp.c 
from cvs-head.

thanks for your time.
--
mvrgr Frank van Dijk
--- rtp.c.orig  2005-02-26 04:01:11.0 +0100
+++ rtp.c   2005-02-26 16:47:17.854001472 +0100
@@ -1036,6 +1036,14 @@
free(rtp);
 }
+static inline unsigned int timeofdaydiff_ms(const struct timeval 
*nw,const struct timeval *old)
+{
+	unsigned int ms;
+	ms = (nw->tv_sec - old->tv_sec) * 1000;
+	ms += (100 + nw->tv_usec - old->tv_usec) / 1000 - 1000;
+	return ms;
+}
+
 static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval 
*delivery)
 {
 	struct timeval now;
@@ -1047,14 +1055,12 @@
 	}
 	if (delivery && (delivery->tv_sec || delivery->tv_usec)) {
 		/* Use previous txcore */
-		ms = (delivery->tv_sec - rtp->txcore.tv_sec) * 1000;
-		ms += (100 + delivery->tv_usec - rtp->txcore.tv_usec) / 1000 - 
1000;
+		ms = timeofdaydiff_ms(delivery,&rtp->txcore);
 		rtp->txcore.tv_sec = delivery->tv_sec;
 		rtp->txcore.tv_usec = delivery->tv_usec;
 	} else {
 		gettimeofday(&now, NULL);
-		ms = (now.tv_sec - rtp->txcore.tv_sec) * 1000;
-		ms += (100 + now.tv_usec - rtp->txcore.tv_usec) / 1000 - 1000;
+		ms = timeofdaydiff_ms(&now,&rtp->txcore);
 		/* Use what we just got for next time */
 		rtp->txcore.tv_sec = now.tv_sec;
 		rtp->txcore.tv_usec = now.tv_usec;
@@ -1069,6 +1075,7 @@
 	int res;
 	int x;
 	int payload;
+	unsigned int timestamp;
 	char data[256];
 	char iabuf[INET_ADDRSTRLEN];

@@ -1093,6 +1100,11 @@
return 0;
 	gettimeofday(&rtp->dtmfmute, NULL);
+
+	/* make distinct digits have distinct timestamps */
+	timestamp = timeofdaydiff_ms(&rtp->dtmfmute,&rtp->txcore) * 8
+	  + rtp->lastts;
+
 	rtp->dtmfmute.tv_usec += (500 * 1000);
 	if (rtp->dtmfmute.tv_usec > 100) {
 		rtp->dtmfmute.tv_usec -= 100;
@@ -1102,7 +1114,7 @@
 	/* Get a pointer to the header */
 	rtpheader = (unsigned int *)data;
 	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | 
(rtp->seqno++));
-	rtpheader[1] = htonl(rtp->lastts);
+	rtpheader[1] = htonl(timestamp);
 	rtpheader[2] = htonl(rtp->ssrc);
 	rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
 	for (x=0;x<6;x++) {
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Re: [Asterisk-Dev] Problem with too many files open

[Brian West]

you sure about that?  That is a very old way of doing it...
bkw
On Feb 14, 2005, at 4:54 AM, roberto wrote:
El dom, 30-01-2005 a las 21:01 +0200, George Konstantoulakis escribió:
The problem is that you reach the max number of open files.
Do as root :
ulimit -n 8192
before running asterisk.
To see  that  you changed the number of open files :
ulimit -a
George.
At least in Debian distribution you can't doo this. I have change max
number file descriptor patching include/limits.h and include/fs.h in 
the
kernel tree.

Roberto.
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Re: [Asterisk-Dev] Dev Conf 2pm CST

[Brian West]

i'll put it up later tonight... or tommorow ;)
bkw
On Feb 10, 2005, at 7:19 PM, Nick Bachmann wrote:
Brian West wrote:
Yes it was recorded
Where, oh where could it be? :)
Nick
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RE: [Asterisk-Dev] is this a bug?

[Brian West]

0.59r-p1 has the fixes too.  And works fine.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Kenneth Long
> Sent: Wednesday, January 26, 2005 1:10 AM
> To: Asterisk Developers Mailing List
> Subject: RE: [Asterisk-Dev] is this a bug?
> 
> LOL!
> 
> I think 's' is the next letter in the alphabet after
> 'r'.
> Its the latest release, fixing some security issue...
> 
> http://www.gentoo.org/security/en/glsa/glsa-200501-14.xm
> http://bugs.gentoo.org/show_bug.cgi?id=76862
> 
> 
> > Do not use 0.59s use 0.59r and do not install form
> > portage.  0.59s (the s
> > must stand for SHIT)  because it doesn't work right.
> >
> 
> 
> 
> __
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> Meet the all-new My Yahoo! - Try it today!
> http://my.yahoo.com
> 
> 
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RE: [Asterisk-Dev] Re: is this a bug?

[Brian West]

> I beg to differ.  An interactive system like this, with a command
> line, and command line editing using control characters, should NEVER
> crash on a ^C.  It would be polite for it to respond to ^C with a

You're doing ctrl-c to exit it IS NOT A CRASH.  -c causes asterisk to not
detach from the controlling term just like EVERY OTHER APPLICATION that is
still attached where ctrl-c causes it to exit.

bkw  

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RE: [Asterisk-Dev] is this a bug?

[Brian West]

I start asterisk with asterisk -vgc

I press ctrl-c 

I get

*CLI> Beginning asterisk shutdown
Beginning asterisk shutdown
Killed

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Kenneth Long
> Sent: Tuesday, January 25, 2005 12:53 PM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] is this a bug?
> 
> thanks... but this is not the point I'm really
> asking... copying is not the issue...
> 
> I'm worried about the invalid pointer message.
> 
> The ctrl-c crashes Asterisk in xterm.
> 
> I wish such commands would not so easily crash it.
> 
> Is the crash "normal" behavior?
> 
> 
> regards,
> Ken
> 
> 
> --- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> 
> > Hello -user question from a windows user.
> >
> > On Tue, 2005-01-25 at 09:14 -0800, Kenneth Long
> > wrote:
> > > I'm launching Asterisk on a xterm session.
> > > I wanted to highlight some text and out of
> > > habit I did a ctrl-C. That reliably and
> > consistantly
> > > shuts down Asterisk with a invalid pointer.
> > > Same error message everytime.
> >
> > Ctrl-c is actually an interupt. Windows users needed
> > easy to remember
> > shortcuts, thats why you are used to ctrl-c.
> >
> > > Some other Ctrl- key sequences lock the session,
> > too.
> > > like ctrl-S.
> >
> > >From terminal days, ctrl-s is flow control stop.
> > ctrl-q is what you use
> > to restart from a ctrl-s.
> >
> > Most terminals should automatically copy on select.
> > You will find unix
> > traditions are full of as short as can be short
> > cuts. Select implies
> > copy and middle click pastes. Click select click and
> > your done.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
> >
> > ___
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> >
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> >
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> 
> 
> 
> 
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RE: [Asterisk-Dev] is this a bug?

[Brian West]

The same reason when you do "more somefile" and ctrl C exits that too.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Tuesday, January 25, 2005 12:39 PM
> To: 'Asterisk Developers Mailing List'
> Subject: Re: [Asterisk-Dev] is this a bug?
> 
> On January 25, 2005 12:57 pm, Brian West wrote:
> > No its not a bug.. it will always do that if you start it with -c
> 
> It's not a bug that enabling colour will "allow" asterisk to die with ^C,
> ^S??
> If it's not a bug then why does ^C, ^S, etc. not cause * to die?
> 
> i.e. why does enabling colour enable these particular crashes?  This
> doesn't
> seem to be the kind of "undocumented feature" that is conducive to a good
> environment.  :-)
> 
> -A.
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RE: [Asterisk-Dev] is this a bug?

[Brian West]

-c  Provide console CLI

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Tuesday, January 25, 2005 12:42 PM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] is this a bug?
> 
> On January 25, 2005 12:59 pm, Steven Critchfield wrote:
> > Ctrl-c is actually an interupt. Windows users needed easy to remember
> > shortcuts, thats why you are used to ctrl-c.
> 
> Yes but why does ^C only interrupt asterisk when -c (colour) is given?  To
> me
> this is a bug.  Asterisk should either always trap these key sequences or
> always pass them, irrespective of your desire to see coloured console
> messages.
> 
> -A.
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RE: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72

[Brian West]

I just spoke with mark and we are all gonna get together and come up with
MeetMe2 (it will be in addons for a while)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Sunday, January 23, 2005 2:18 PM
> To: 'Asterisk Developers Mailing List'
> Subject: RE: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0003402
> 
> Yes we posted a new bug with ours.. but since deleted the attachments. I
> did
> voice my opinion to mark and everyone else involved but nobody would
> listen
> to me.  So I gave up.
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> > [EMAIL PROTECTED] On Behalf Of Tony Mountifield
> > Sent: Sunday, January 23, 2005 2:04 PM
> > To: asterisk-dev@lists.digium.com
> > Subject: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72
> >
> > In article <[EMAIL PROTECTED]>,
> > Brian West <[EMAIL PROTECTED]> wrote:
> > > I think this should have been talked about a bit longer before it was
> > > commited.  I don't feel its 100% correct.  I think the way anthm
> > > implemented it was better... it included outcalling... let you start
> > > and stop recording from the admin menu(reusing existing routines)...
> > > converted it to use flags.
> >
> > Just looked at your comments on that bug report #3393. You didn't say
> > anything like the above in there - just a couple of helpful comments
> > about implementation details, but nothing to suggest to wait because you
> > thought there was a better overall approach.
> >
> > Cheers
> > Tony
> > --
> > Tony Mountifield
> > Work: [EMAIL PROTECTED] - http://www.softins.co.uk
> > Play: [EMAIL PROTECTED] - http://tony.mountifield.org
> > ___
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RE: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0003402

Yes we posted a new bug with ours.. but since deleted the attachments. I did
voice my opinion to mark and everyone else involved but nobody would listen
to me.  So I gave up.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Tony Mountifield
> Sent: Sunday, January 23, 2005 2:04 PM
> To: asterisk-dev@lists.digium.com
> Subject: [Asterisk-Dev] Re: FW: asterisk/apps app_meetme.c,1.71,1.72
> 
> In article <[EMAIL PROTECTED]>,
> Brian West <[EMAIL PROTECTED]> wrote:
> > I think this should have been talked about a bit longer before it was
> > commited.  I don't feel its 100% correct.  I think the way anthm
> > implemented it was better... it included outcalling... let you start
> > and stop recording from the admin menu(reusing existing routines)...
> > converted it to use flags.
> 
> Just looked at your comments on that bug report #3393. You didn't say
> anything like the above in there - just a couple of helpful comments
> about implementation details, but nothing to suggest to wait because you
> thought there was a better overall approach.
> 
> Cheers
> Tony
> --
> Tony Mountifield
> Work: [EMAIL PROTECTED] - http://www.softins.co.uk
> Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Dev] Asterisk Source Code Intensive - Proposal forAstricon

[Brian West]

I have the meetme box up and ready for many many ulaw connections... 

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Leif Madsen
> Sent: Monday, January 03, 2005 1:12 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] Asterisk Source Code Intensive - Proposal
> forAstricon
> 
> Brian West said the following in another thread:
> > Leif and I have done one on one code/doc writing line by line in some
> cases.
> 
> And Jim Van Meggelen said in another thread:
> > You know what, if we could pick an app and pick the brains of the dev
> > team as to what it does, we could possibly have a weekly conference call
> > of, say and hour or two, go through it line by line, and document the
> > code as the minutes of the meeting. Nothing heavy, just a few lines per
> > night - no more than an hour. Betcha we'd end up with an awesome amount
> > of source code comments. Once started, it'll become much easier to
> > maintain.
> 
> I think this is a fantastic idea.  Have someone "in the know" lead the
> conference call, have someone dedicated to the actual writing of the
> documentation (someone with fast typing skills), and the others would
> participate in the conference either as silent observers or actively
> speaking.  Certain people will obviously choose how to participate.
> 
> As Brian said... start at A and document to Zed.
> 
> Thanks,
> Leif Madsen.
> http://www.leifmadsen.com.
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RE: [Asterisk-Dev] Asterisk Source Code Intensive - Proposal forAstricon

[Brian West]

> Funny enough bkw mentioned the same thing tonight on the conference.
> I think going through and documenting the code would be a great way to
> learn it, and help to lower the learning curve for those who come
> after.

Leif and I have done one on one code/doc writing line by line in some cases.

bkw

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RE: [Asterisk-Dev] Features requests on bugs.digium.com

[Brian West]

> I think a lot of people will agree that -dev is underutilized.  My own
> theory is that people stay away because of too many accidental posts
> here (i.e. -users problems).  I'm not sure what the solution to that is,
> but that's my opinion.

I can agree with this we need to use -dev more ;)

HAPPY NEW YEAR!

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RE: [Asterisk-Dev] Features requests on bugs.digium.com

[Brian West]

> I know this makes a lot of sense to you Brian, but it's not as 'open' of a
> process as you might think. Personally, I'm glad you can invest a lot of
> time 'in the clubhouse'. I'm sure you've formed a lot of very meaningful
> personal and professional relationships that will enrich your life with
> moose penis stories for many years to come, and the motivation you derive
> from that has led you to do great things with Asterisk for which we are
> all
> grateful. But the current model doesn't scale well ... there's a lot of
> very
> talented people out in the world who can't spent 4 hours of their day in
> an
> IAX teleconference, and forgive me but I've sat on the IRC channels you're
> talking about and the s/n isn't wonderful there either. I think the
> largest
> barrier to getting involved with what you're talking about is that it's
> somewhat of an all-or-nothing approach. If these discussions instead
> appeared in Mantis or on the -devel list, people could contribute as/when
> they are able, instead of devoting hours a day to a teleconference on top
> of
> their day job.

Well asterisk is my job so I have to devote time to it.  As for an all or
nothing not everyone including myself are on here every night... I skip days
and sometimes weeks when I'm too busy to work with it... but we do touch
base often about ideas... We just all called Allison and wished her a happy
new year ;)  

bkw

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RE: [Asterisk-Dev] Features requests on bugs.digium.com

[Brian West]

Ok guys we work with mark daily and we know what will fly or not fly.  We
talk on conf calls NIGHTLY and we communicate VERY fast on bugs and we don't
just close things out without talking about it.  If you all want to get
involved join #asterisk-bugs and get the conf iax url we are usually hanging
out in there 4-8 hours a day between all of us we cover bugs and lots of
random banter but we do discuss all the options and mark even comes in from
time to time.  (Hell we even had Allison in there one night)  And we'll be
in there tonight over new years... maybe if you join you'll hear the story
of where "moose penis" came from :P

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Darren Nickerson
> Sent: Friday, December 31, 2004 4:56 PM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] Features requests on bugs.digium.com
> 
> "Josh Roberson" <[EMAIL PROTECTED]> wrote:
> 
> > This discussion was actually already here, we just jumped into it.  If
> my
> > reactions to your viewpoints don't prove that I'm open-minded about
> this,
> > I don't know what will.
> 
> Aw, I knew you were open minded ;-) You're one of the most level-headed of
> the core group, and I appreciate your patience and willingness to engage
> in
> constructive debate. I also appreciate all the housekeeping you do in
> Mantis!
> 
> > Sorry for the misunderstanding - dealing with the project and other
> > people's dumps on what you're trying to accomplish really can increase
> the
> > stress level on one side of the fence.
> 
> I know exactly how you feel. Open source development can be like that.
> 
> -Darren
> 
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[Asterisk-Dev] Features requests on bugs.digium.com

[Brian West]

Ok I think all feature requests need to be gone from bugs.digium.com... if
it has a patch it should be ok.  But feature requests I don't think should
be on the "bug" tracker.  It is a bug tracker right?

So we can focus... what do you think... all raise your hands if the bug
tracker should only be for bugs and bugs with patches, or new features with
patches...?

bkw

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RE: [Asterisk-Dev] RE: Latest CVS: Segmentation fault (core dumped)

[Brian West]

Make sure you update.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of VoIP
> Sent: Monday, December 27, 2004 12:45 PM
> To: 'Asterisk Developers Mailing List'
> Subject: [Asterisk-Dev] RE: Latest CVS: Segmentation fault (core dumped)
> 
> I've installed latest CVS on 2 servers, one with digium card and the other
> w/o digium card. Both got segmentation fault. Thru gdb, I got the below
> results which is not the app_voicemail problem.
> 
> (gdb) bt full
> #0  0x4003c89e in pthread_mutex_lock () from /lib/tls/libpthread.so.0
> No symbol table info available.
> #1  0x41b3ddbd in register_verify (p=0x8158cf0, sin=0x431313cc,
> req=0x431313dc, uri=0x431315f9 "sip:mycom.com", ignore=0) at
> chan_sip.c:5017
> destroyme = 0
> res = -1
> peer = (struct sip_peer *) 0x0
> tmp = "
> iabuf = "\024\n\023BÓ\026\023C¬ý\022C\fý\022C"
> name = 0x4312fd11 "admin"
> c = 0x4312fd16 ""
> t = 0x43131605 ""
> #2  0x41b4984f in handle_request (p=0x8158cf0, req=0x431313dc,
> sin=0x431313cc,
> recount=0x431313b8, nounlock=0x431313bc) at chan_sip.c:7843
> resp = {rlPart1 = 0x0, rlPart2 = 0x0, len = 0, headers = 0, header
> =
> {
> 0x0 }, lines = 0, line = {0x0 },
>   data = '\0' }
> cmd = 0x431315f0 "REGISTER"
> cseq = 0x431316ca " REGISTER"
> from = 0x0
> e = 0x431315f9 "sip:mycom.com"
> useragent = 0x41b50df9 ""
> c = (struct ast_channel *) 0x0
> ---Type  to continue, or q  to quit---
> transfer_to = (struct ast_channel *) 0x0
> seqno = 1
> len = 1
> ignore = 0
> respid = 0
> res = -1
> gotdest = 0
> iabuf = '\0' 
> af = {frametype = 5, subclass = 0, datalen = 0, samples = 0,
>   mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec = 0,
> tv_usec = 0}, prev = 0x0, next = 0x0}
> debug = 0
> #3  0x41b49f9d in sipsock_read (id=0x8128b18, fd=9, events=1, ignore=0x0)
> at chan_sip.c:7941
> req = {rlPart1 = 0x431315f0 "REGISTER",
>   rlPart2 = 0x431315f9 "sip:mycom.com", len = 349, headers = 9, header = {
> 0x431315f0 "REGISTER", 0x4313160f "Content-Length: 0",
> 0x43131622 "Contact: ",
> 0x4313164b "Call-ID:
> [EMAIL PROTECTED]",
> 0x43131688 "Max-Forwards: 70",
> 0x4313169a "From: ;tag=14422979",
> 0x431316c3 "CSeq: 1 REGISTER", 0x431316d5 "To: ",
> 0x431316ef "Via: SIP/2.0/UDP
> 192.168.1.101:5060;branch=z9hG4bKc0a801650131c9---Type  to
> continue,
> or q  to quit---
> b141d0551c10a10001", 0x4313174b "", 0x0 },
>   lines = 0, line = {0x4313174d "", 0x0 },
>   data = "REGISTER\000sip:mycom.com\000SIP/2.0\000\000Content-Length:
> 0\000\000Contact: \000\000Call-ID:
> [EMAIL PROTECTED]:
> 70\000\000From: ;tag"...}
> sin = {sin_family = 2, sin_port = 50743, sin_addr = {
> s_addr = 616355034}, sin_zero = "\000\000\000\000\000\000\000"}
> p = (struct sip_pvt *) 0x8158cf0
> res = 349
> len = 16
> nounlock = 0
> recount = 0
> debug = 0
> #4  0x08053345 in ast_io_wait (ioc=0x811ed80, howlong=1000) at io.c:267
> res = 1
> x = 0
> origcnt = 1
> #5  0x41b4a6bc in do_monitor (data=0x0) at chan_sip.c:8088
> res = 1000
> sip = (struct sip_pvt *) 0x0
> peer = (struct sip_peer *) 0x0
> t = 1104172719
> ---Type  to continue, or q  to quit---
> fastrestart = 0
> lastpeernum = -1
> curpeernum = 0
> reloading = 0
> #6  0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
> No symbol table info available.
> (gdb)
> 
> 
> ___
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RE: [Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining

[Brian West]

> Why are you worried? Anyone who knows jack or shit, should easily figure
> out you know what you are talking about. I don't think Mark would need a
> cert to compete on an asterisk consulting gig. If your or Mark did, the
> company would assuredly have it's head way to far up it's HR departments
> ass to be worth working for.

You do have a point... I was just a wee bit pissed off over the post at
first... but after talking to Mark he calmed me down... and said he would
take care of it... :P

bkw

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[Asterisk-Dev] RE: [Asterisk-biz] Asterisk training and certification :: AstriconTraining

[Brian West]

I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified...  I think me and MANY
others are about to walk out of the project over this.  I have already
spoken with many people that are close to the project.  You're hurting US
and our ability to make money.  I still know the code better than most of
the people that will be paying to be certified.  You're pushing it here. 

I REFUSE TO PAY!!!  I know you guys mean well but you didn't take any of us
into account that know this software and know it well.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-biz-
> [EMAIL PROTECTED] On Behalf Of Olle E. Johansson
> Sent: Sunday, December 19, 2004 1:32 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-biz] Asterisk training and certification ::
> AstriconTraining
> 
> *** AsteriskT Open Source Linux PBX Training and Certification
> 
> Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc.,
> Edvina AB and Sokol & Associates today released a new program for
> training and certification of Asterisk professionals. Asterisk is the
> leading Open Source PBX for Linux, with support for both PSTN
> connectivity and many VoIP protocols.
> 
> The first class in the Astricon Training product line is the five-day
> bootcamp "Introduction to Asterisk". This class will be held in the US
> and Europe six times during 2005. The organizers and teachers is the
> same team that set up the Astricon 2004 conference and expo in September
> this year, an event that gathered over 450 Asterisk users and developers
> in Atlanta, GA.
> 
> The new Asterisk certification is named dCAP, Digium Certified Asterisk
> Professional. To get the certification, one has to go through a 150
> question exam as well as a practical exam, where the student builds and
> configures a PBX. The certification will be given by the Astricon team
> under license from Digium.
> 
> "This is an important step towards greater acceptance of Asterisk in the
> enterprise", says Olle E. Johansson of Edvina in Sweden. "With a
> professional training and certification, you can ensure that your staff
> or your consultants has the required skills to setup and manage a
> mission-critical PBX platform based on Asterisk."
> 
> "The Asterisk Open Source project is building a professional business
> ecosystem", says Mark Spencer, the founder of Digium and creator of
> Asterisk. "Many companies are now selling Asterisk-based solutions. With
> the 1.0 stable release in September, the Digium hardware that ranges
> from the IAXy end-user device to carrier-class quad-T1 cards and the
> Digium commercial support we have a professional platform for partnering
> with major enterprises. The Astricon training and dCAP certification
> enables us to build a network of consultants that we know will and are
> able to assist us working on the continued success of Asterisk."
> 
> The first training class will be held in Kansas City, MO, January 17-21
> 2005. The cost for a five-day bootcamp with certification is $3,275 USD.
> Details can be found on http://www.astricon.net
> 
> AsteriskT is the leading open source PBX, used all over the world. Since
> it is Linux-based, it inherits all of the power and stability of the
> operating system. Linux provides open source alternatives to proprietary
> applications. Asterisk is the first package to fit all telecommunication
> needs in a broad variety of environments.
> 
> DigiumT is the creator and primary developer of Asterisk, the industry's
> first open source PBX. Used in combination with Digium's PCI telephony
> interface cards, Asterisk offers a strategic, highly cost-effective
> approach to voice and data transport over TDM, switched, IP, and
> Ethernet architectures.
> Digium provides a highly refined selection of quality hardware and
> software products, developed and implemented using innovative
> engineering techniques (primarily open source development). A full range
> of professional services complement these product lines, including
> consulting, technical support, and custom software development services.
> The open source communications revolution is here, and Digium is leading
> the way.
> 
> Contacts:
> . Olle E. Johansson, Edvina AB, Phone +46 8 594 788 10,
>   http://www.astricon.net
> . Steven M. Sokol, Sokol & Associates,
>   Phone: +1.816.822.1807, IaxTel: 700.613.9004
> . Digium, press contact Rick Segrest,
>   Phone: +1 (256) 428-6000
>   http://www.digium.com
> 
> 
> 
> ___
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RE: [Asterisk-Dev] Choice of astdb backend -- questions...

[Brian West]

Ya we hope to get it in CVS-HEAD sometime soon.. it's a very nice addition.


bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of TC
> Sent: Thursday, December 09, 2004 1:07 PM
> To: Rob Fugina; Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] Choice of astdb backend -- questions...
> 
> No licensing issues at ALLL
>  just cvs backlog issues & integration
> http://sqlite.org/
> Sources are in the public domain. Use for any purpose.
> http://sqlite.org/copyright.html
> 
> - Original Message -
> From: "Rob Fugina" <[EMAIL PROTECTED]>
> To: "Asterisk Developers Mailing List" <[EMAIL PROTECTED]>
> Sent: Thursday, December 09, 2004 11:01 AM
> Subject: Re: [Asterisk-Dev] Choice of astdb backend -- questions...
> 
> 
> > On Thu, 9 Dec 2004 11:33:42 -0600, Brian West <[EMAIL PROTECTED]> wrote:
> > > Go get res_sqlite from asterisk-addons
> > >
> > > It blows away the db1 support.
> > >
> > > bkw
> >
> > By its location in asterisk-addons, I infer that there are similar
> > licensing incompatabilities...  Is there no interest in
> > upgrading/updating what's distributed with the normal asterisk
> > distribution?
> >
> > Rob
> > ___
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RE: [Asterisk-Dev] chan->priority = 0 ?

[Brian West]

Have you rebuilt your app since the channel.h changes.  You'll also need to
make sure you have the latest everything in CVS-HEAD it is not broken.  If
channel.h changes you really need to make clean and rebuild EVERY SINGLE app
you have otherwise you'll have strange issues such as this.

Also:

memset(cd.cardid,0,sizeof(cd.cardid));
memset(cd.credit,0,sizeof(cd.credit));
memset(cd.destination,0,sizeof(cd.destination));
memset(cd.tariffid,0,sizeof(cd.tariffid));
memset(cd.active,0,sizeof(cd.active));
memset(cd.rate,0,sizeof(cd.rate));
memset(cd.dialstr,0,sizeof(cd.dialstr));
memset(cd.prefixid,0,sizeof(cd.prefixid));
memset(cd.providerid,0,sizeof(cd.providerid));
memset(cd.newdestination,0,sizeof(cd.newdestination));
memset(cd.ccprefix,0,sizeof(cd.ccprefix));


All those lines can be reduced to exactly one:

memset(cd,0,sizeof(struct calldata));

Thanks,
Brian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-dev-
> [EMAIL PROTECTED] On Behalf Of Wolfgang Pichler
> Sent: Thursday, December 09, 2004 5:41 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Dev] chan->priority = 0 ?
> 
> hi all,
> 
> i do have a problem with the chan->priority variable - my application
> (app_prepaid_auth_cid) gets called from an extension at priority 3 - so
> the variable chan->priority should contain the value 3. But it always
> contains 0 as priority.
> 
> Here is the part from extension.conf
> --
> exten => _.,3,PrepaidAuthCID(${CALLERID})
> --
> 
> and attached is the c source file - the error happens on line 342
> 
> can someone help me with this strange error ?
> 
> regards,
> Wolfgang


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[Asterisk-Dev] blah

[Brian West]

knock knock
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