Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote:


Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a  
solution, but
unfortunately the system I'm working with needs the separate  
macros. I'll

update the list if anything gets worked out.


pbx-1*CLI> show application gosub
pbx-1*CLI>
  -= Info about application 'Gosub' =-

[Synopsis]
Jump to label, saving return address

[Description]
Gosub([[context|]exten|]priority)
  Jumps to the label specified, saving the return address.

pbx-1*CLI>


Already considered this option, but I want to give it some arguments.  
And that isn't possible with gosub.



regards,
Attilla
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Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

Well I solved the problem, by just making it one macro, not a macro  
inside another one.


[macro-record]
exten => s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$ 
{TIMESTAMP})

exten => s,2,Monitor(wav,${CALLFILENAME},m})
exten => s,3,setcallerid(${ARG2})
exten => s,4,dial(${ARG3})
exten => h,1,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $ 
{CALLFILENAME} &)



Regards,
Attilla



On Aug 13, 2006, at 11:21 PM, Attilla De Groot wrote:


On Aug 13, 2006, at 7:53 PM, Gonzalo Servat wrote:


I think what you probably want is:

exten => _*22*X.,1,Macro(record,conference,${EXTEN:4})

If you have _*23*., it means it will match *23 as well as
*23*, but not *23*123456 which is probably what you
want. Try:

exten => _*23*X.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4} 
@voipbuster)


I made both changes, but that wasn't a problem. If I understand it  
correctly, this only changes that the extension got more specific,  
right ?





Also, from memory, the "h" extension gets executed from the main
context. After making the above changes, try adding this:

exten => _*23*X.,h,System(/etc/asterisk/mail.sh [EMAIL PROTECTED]
${CALLFILENAME} &)

.. and remove the "h" extension from macro-record.

Let me know if the above helps.


I did make the changes, but it didn't solve the problem. It should  
be  h,1,System by  the way ;)

I have some log here:

 -- Executing Macro("SIP/attilla-0dce", "record|dialout|31455200025| 
SIP/[EMAIL PROTECTED]") in new stack
-- Executing SetVar("SIP/attilla-0dce", "CALLFILENAME=CALL- 
dialout-08001234-20060813-231921") in new stack
-- Executing Monitor("SIP/attilla-0dce", "wav|CALL- 
dialout-08001234-20060813-231921|m}") in new stack
-- Executing GotoIf("SIP/attilla-0dce", "0?macro-record|s| 
4:macro-record|s|5") in new stack

-- Goto (macro-record,s,5)
-- Executing Macro("SIP/attilla-0dce", "dialout|31455200025|SIP/ 
[EMAIL PROTECTED]") in new stack
-- Executing SetCallerID("SIP/attilla-0dce", "31455200025") in  
new stack
-- Executing Dial("SIP/attilla-0dce", "SIP/ 
[EMAIL PROTECTED]") in new stack

-- Called [EMAIL PROTECTED]
-- SIP/sip1.voipbuster.com-f715 is making progress passing it  
to SIP/attilla-0dce

-- SIP/sip1.voipbuster.com-f715 answered SIP/attilla-0dce
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce' in macro 'record'
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce'


As you can see it executes everything perfectly and I was expecting  
that after this the script would be executed.



Regards,
Attilla


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Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

On Aug 13, 2006, at 7:53 PM, Gonzalo Servat wrote:


I think what you probably want is:

exten => _*22*X.,1,Macro(record,conference,${EXTEN:4})

If you have _*23*., it means it will match *23 as well as
*23*, but not *23*123456 which is probably what you
want. Try:

exten => _*23*X.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4} 
@voipbuster)


I made both changes, but that wasn't a problem. If I understand it  
correctly, this only changes that the extension got more specific,  
right ?





Also, from memory, the "h" extension gets executed from the main
context. After making the above changes, try adding this:

exten => _*23*X.,h,System(/etc/asterisk/mail.sh [EMAIL PROTECTED]
${CALLFILENAME} &)

.. and remove the "h" extension from macro-record.

Let me know if the above helps.


I did make the changes, but it didn't solve the problem. It should  
be  h,1,System by  the way ;)

I have some log here:

 -- Executing Macro("SIP/attilla-0dce", "record|dialout|31455200025| 
SIP/[EMAIL PROTECTED]") in new stack
-- Executing SetVar("SIP/attilla-0dce", "CALLFILENAME=CALL- 
dialout-08001234-20060813-231921") in new stack
-- Executing Monitor("SIP/attilla-0dce", "wav|CALL- 
dialout-08001234-20060813-231921|m}") in new stack
-- Executing GotoIf("SIP/attilla-0dce", "0?macro-record|s| 
4:macro-record|s|5") in new stack

-- Goto (macro-record,s,5)
-- Executing Macro("SIP/attilla-0dce", "dialout|31455200025|SIP/ 
[EMAIL PROTECTED]") in new stack
-- Executing SetCallerID("SIP/attilla-0dce", "31455200025") in  
new stack
-- Executing Dial("SIP/attilla-0dce", "SIP/ 
[EMAIL PROTECTED]") in new stack

-- Called [EMAIL PROTECTED]
-- SIP/sip1.voipbuster.com-f715 is making progress passing it to  
SIP/attilla-0dce

-- SIP/sip1.voipbuster.com-f715 answered SIP/attilla-0dce
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce' in macro 'record'
  == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
attilla-0dce'


As you can see it executes everything perfectly and I was expecting  
that after this the script would be executed.



Regards,
Attilla


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Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

On Aug 13, 2006, at 5:08 PM, Aaron Daniel wrote:

Just a thought, but shouldn't the exten => h be priority 1, not  
priority
6?  The h extension is totally separate from the s extension, the  
server

doesn't think it exists because there's no priority 1 for it.  Try
that :)



As far as I know that isn't a recruirement, but ofcourse I tried, but  
it didn't solve the problem.



Regards,
Attilla
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Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

On Aug 13, 2006, at 11:32 AM, Michael Strelnikov wrote:


I have the same problem even with AEL.
When call is finished from macro "dialout" asterisk just stops all  
futher processing.

I haven't found any solution yet.


Well, in my case I can solve it by adjusting the script with a loop  
to check if the file exists (it will only be created on hangup) and  
the send the mail. But I'm sure you have to agree that that isn't the  
best solution.



Regards,
Attilla

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Re: [asterisk-users] Macro inside macro

[Attilla De Groot]

On Aug 13, 2006, at 11:22 AM, Dovid Bender wrote:

Please include what you send to the macro from your extensions.conf  
so we can see what you are sending down to the macro.



- Original Message - From: "Attilla De Groot"  
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"  


Sent: Sunday, August 13, 2006 4:56 AM
Subject: [asterisk-users] Macro inside macro



Hi all,


I'm making a little macro te record conversations if a user want  
so  and if it's recorded the recording should be e-mailed.

This is what I have come up with right now:

[macro-record]
exten => s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$  
{TIMESTAMP})

exten => s,2,Monitor(wav,${CALLFILENAME},m})
exten => s,3,GotoIf($["${ARG1}" = "conference"]?macro-record|s|  
4:macro-record|s|5)

exten => s,4,Macro(conference|${ARG2})
exten => s,5,Macro(dialout|${ARG2}|${ARG3})
exten => h,6,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $  
{CALLFILENAME} &)



The conversation gets recorded perfectly and it's also possible  
to  dial, but somehow the script doesn't get executed and I don't  
know  why (the script works manually, I already tested this).


Can anyone helpyp me with the solution ?


Best regards,
Attilla
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Sorry, didn't thought it was relevant, since the entire macro gets  
executed, but here it is.


;recording
exten => _*22*XXX,1,Macro(record,conference,${EXTEN:4))
exten => _*23*.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4} 
@voipbuster)


[macro-record]
exten => s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$ 
{TIMESTAMP})

exten => s,2,Monitor(wav,${CALLFILENAME},m})
exten => s,3,GotoIf($["${ARG1}" = "conference"]?macro-record|s| 
4:macro-record|s|5)

exten => s,4,Macro(conference|${ARG2})
exten => s,5,Macro(dialout|${ARG2}|${ARG3})
exten => h,6,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $ 
{CALLFILENAME} &)


[macro-dialout]
exten => s,1,setcallerid(${ARG1})
exten => s,2,dial(${ARG2})
exten => s,3,Hangup()

[macro-conference]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,MeetMe(${ARG1}|p)
exten => s,4,Hangup


Regards,
Attilla
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[asterisk-users] Macro inside macro

[Attilla De Groot]

Hi all,


I'm making a little macro te record conversations if a user want so  
and if it's recorded the recording should be e-mailed.

This is what I have come up with right now:

[macro-record]
exten => s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$ 
{TIMESTAMP})

exten => s,2,Monitor(wav,${CALLFILENAME},m})
exten => s,3,GotoIf($["${ARG1}" = "conference"]?macro-record|s| 
4:macro-record|s|5)

exten => s,4,Macro(conference|${ARG2})
exten => s,5,Macro(dialout|${ARG2}|${ARG3})
exten => h,6,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $ 
{CALLFILENAME} &)



The conversation gets recorded perfectly and it's also possible to  
dial, but somehow the script doesn't get executed and I don't know  
why (the script works manually, I already tested this).


Can anyone helpyp me with the solution ?


Best regards,
Attilla
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Re: [asterisk-users] Queue menu

[Attilla De Groot]

On Jul 12, 2006, at 1:01 PM, Tristan wrote:


Hi,

In the queue.conf when you define the queue:

[myqueue]
context = thecontextforusers
...

and in your extensions.conf

[thecontextforusers]
exten => 1,1,NoOp(1 WAS PRESSED)
exten => 2,1,NoOp(2 WAS PRESSED)


If you need more help, just ask ! ;)


Hi Tristan,


Perfectly clear now and it works, thank you.


regards,
Attilla
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[asterisk-users] Queue menu

[Attilla De Groot]

Hi all,


I'm trying to setup a menu in a queue, something like "All our agents  
are busy, press 1 to leave a voicemail, press 2 for another  
department" etc.


Anyway, the only thing I found is this on the wiki:

Menu for the user
You can define a menu for the user, while waiting. For this menu, you  
can only use one-digit extensions. Define the context for the menu in  
the configuration for the queue to enable this option.



But I don't know what this meanse or atleast how to implement it.
Could someone give me a hint in the right direction ?


Regards,
Attilla
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[asterisk-users] Freeware sip/iax client windows mobile

[Attilla De Groot]

Hi all,


I have two pda's and I want to be able to make calls, but I need a  
client for this. The only problem is Windows Mobile 5.0, I can't find  
a freeware client for this, the only one is Sjphone. But this one is  
still beta for windows mobile and it just doesn't work good.


Does anyone have an alternative ?



regards,
Attilla
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Re: [Asterisk-Users] Asterisk & Eyebeam chat function

[Attilla De Groot]

Hi Doug,


I didn't knew this.
Thank you.


Regards,
Attilla

On Jun 13, 2006, at 4:52 PM, Douglas Garstang wrote:

Unless it's changed recently, Asterik doesn't support the SIP  
'MESSAGE' command.


Doug.


-Original Message-----
From: Attilla De Groot [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk & Eyebeam chat function


Hi all,


Eyebeam has a sip-chat function and it would be nice if I would be
able to use it. But the problem is that I can't really find
information about it.

I can just try to send a message and on the Asterisk console a
message like this appears:

Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message:
Received message to  from "Bla
Sheep";tag=1d072048, dropped it...
   Content-Type:text/plain
   Message: ?

Can anyone tell me more about this or give me a link with some
information about it ?


Regards,
Attilla de GrootÎ

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[Asterisk-Users] Asterisk & Eyebeam chat function

[Attilla De Groot]

Hi all,


Eyebeam has a sip-chat function and it would be nice if I would be  
able to use it. But the problem is that I can't really find  
information about it.


I can just try to send a message and on the Asterisk console a  
message like this appears:


Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message:  
Received message to  from "Bla  
Sheep";tag=1d072048, dropped it...

  Content-Type:text/plain
  Message: ?

Can anyone tell me more about this or give me a link with some  
information about it ?



Regards,
Attilla de GrootÎ

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Re: [Asterisk-Users] Call-pickup function in Queue application

[Attilla De Groot]

On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote:


Hi Attilla,

I'm not sure if there is something like that available or not, but  
I know there are some alternatives. You can set the time out limit  
to say 15 seconds, which for me is about 3-4 rings on the phone  
before it goes looking for the next agent. The other option you can  
manually remove the interface from the queue via the CLI by the  
following:


remove queue member  from 

However, I'm not sure if that will have an effect on the  
call...hopefully it will just send the caller looking for the next  
number. I haven't personally tried it.


I know some phones like the Polycom 601 have a buddy watch option.  
As far as I know, and someone can step in and correct me if I am  
wrong, that will just show if the person is on the phone or not. I  
don't think you can pick up on the line.


Kevin


Hi Kevin,


Well I thought about those alternatives and I suggested them, but the  
person who wants them said that such a feature was avalible on  
another pbx where he used to work. And well, he would like the same  
thing on the Asterisk PBX.


I already have the time at 15 seconds, and well removing a member  
from the queue might send it to the next agent. But if there are more  
then two agents in the queue there is not really a point.



Regards,
Attilla
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[Asterisk-Users] Call-pickup function in Queue application

[Attilla De Groot]

Hi All,


I need a function that I believe isn't available in Asterisk, but I  
don't know if I'm correct about this.


I have a queue and I want agents that are in that queue to have the  
ability to answer a call in the queue with calling an extention. For  
example, if I'm an agent and my colleague forgot to logout I could  
take the call when his phone is still ringing without walking to his  
desk or waiting for round robin.


Can anyone tell me if this already is avalible ?



Regards,
Attilla
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Re: [Asterisk-Users] Monitor application and e-mailing attachment

[Attilla De Groot]

On Jun 4, 2006, at 8:11 PM, Doug Lytle wrote:

Okay,

I decided that I would give it a try, I ended up having the same  
issues as you.  I got it figured out though.


When using the monitor application, it splits the wave files into  
incoming and outgoing legs.  Once the call has been completed, it  
mixes them into the final file.  The problem is, it doesn't  
consider the h extension as the call completed.  It waits until the  
h extension has terminated before merging.  I proved this point by  
adding a sleep 20 in my script and the incoming/outgoing waves sat  
there for 20 seconds.  My fix was to ad an ampersand at the end of  
the System script entry.  As so:


exten => h,1,System(/usr/local/bin/monitor.sh ${CALLFILENAME} &)

This will cause the script to detach from the current shell.

Doug




Your a real life saver. Thank you.


Attilla
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Re: [Asterisk-Users] Monitor application and e-mailing attachment

[Attilla De Groot]

On Jun 4, 2006, at 5:29 PM, Doug Lytle wrote:

I pass variables to my scripts all the time.

When I first started using scripts with variables, I was having  
issues with the scripts running, but producing no results. Echoing  
the variables to a log, I was able to track down why they weren't  
working.  You may want to do the same.


Doug


Can you show me an example from you ?
Because if I echo everything works fine, but it just won't work.

This doesn't work, while the echo is ok:
exten => h,1,System(/bin/sh /etc/asterisk/mail.sh ${CALLFILENAME})

This does work:
exten => h,1,System(/bin/sh /etc/asterisk/mail.sh  
CALL-008000200570-20060604-181756)



Call me stupid, but thats exactly the same.


Attilla
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Re: [Asterisk-Users] Monitor application and e-mailing attachment

[Attilla De Groot]

On Jun 4, 2006, at 2:16 PM, Doug Lytle wrote:


Check the execute permissions on the script.

If that doesn't work, then echo the output of the script to a log  
and review the log to find the error.


Doug


Hi Doug,


Well the permissions are set ok, but I just found out that the  
problem is that I give an argument in the system application. Because  
I say System(script.sh ${var}), but if I do that variable static in  
my script and then run it without argument, it's working without any  
problems.



Regards,
Attilla
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Re: [Asterisk-Users] How to make this into a Macro?

[Attilla De Groot]

On Jun 4, 2006, at 11:50 AM, Ronald Wiplinger wrote:


I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?

exten => 8863959,1,Dial(SIP/8863959,60,r)
exten => 8863959,2,NoOp(${DIALSTATUS})
exten => 8863959,3,Voicemail,[EMAIL PROTECTED]
exten => 8863959,104,Voicemail,[EMAIL PROTECTED]
exten => 8863959,105,hangup



Hi Ronald,


Just make something like this:

exten => 8863959,1,Macro(dial,8863959)

[macro-dial]
exten => s,1,Dial(SIP/${ARG1},60,r)
exten => s,2,NoOp(${DIALSTATUS})
exten => s,3,Voicemail,[EMAIL PROTECTED]
exten => s,104,Voicemail,b$([EMAIL PROTECTED]
exten => s,105,hangup


Regards,
Attilla
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[Asterisk-Users] Monitor application and e-mailing attachment

[Attilla De Groot]

Hi all,


I'm trying to make a context that will monitor a call and when it's  
completed it would e-mail the wav to a specified mail adres.


So I made a standard context that records a call, like this:

exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ 
{TIMESTAMP})

exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m})
exten => _*31*00[1-9].,3,Dial(SIP/0031${EXTEN:[EMAIL PROTECTED])

exten => h,1,System(/etc/asterisk/mail.sh ${CALLFILENAME})


Ok, this is my mail.sh:

/usr/bin/mime-construct --to [EMAIL PROTECTED] --subject "Asterisk  
Recording" --attachment $1.wav --file /var/spool/asterisk/monitor/$1.wav



-- Executing System("SIP/attilla-8407", "/etc/asterisk/mail.sh  
CALL-008000200570-20060604-115659") in new stack



But this just doesn't work.

Asterisk says it's executing the script, but the mail doesn't get  
send. If I check my log-files I don't see that the mail gets send. I  
tried to run the script manually and that worked without problems. I  
checked if it was a file rights issue, but even when I run Asterisk  
as root, it doesn't work.


Then I just mad a little scipt that said "touch /etc/blaat" and if I  
executed that with the system command, it worked perfectly.


Can anyone give a hint about what I'm doing wrong ?



Regards,
Attilla
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Re: [Asterisk-Users] Change g729 payload

[Attilla De Groot]

On Jun 1, 2006, at 10:57 PM, Dan Austin wrote:

It was put in a development branch, but has not seen any action in
over two months.

I have had it running against chan_ooh323 for six months and chan_sip
for three weeks in production for a moderately loaded conferencing
server.  Absolutely no issues with either channel.  It would not be
too hard to add support to any of the RTP-based channels, but IAX
will not work with the code as it stands.


Dan
__


Hi Dan,


That's very good news.

But to be honest, I'm not really a programmer and I'm not sure how to  
apply this patch on the source code. Could you give any instuctions  
about how to apply the patch ?



Attilla
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Re: [Asterisk-Users] Change g729 payload

[Attilla De Groot]

On Jun 1, 2006, at 5:38 PM, Jean-Michel Hiver wrote:



You're wrong :)


Nobody is perfect. ;)

I had the same problem. Unfortunately this value is hard coded in  
Asterisk's code. I don't know if recent versions of Asterisk  
support this.


Well I just found this:
http://bugs.digium.com/view.php?id=5162

So it seems that there is a patch and that it's ready for 2 months,  
but I just checked the rtc.c code and it doesn't include this patch.  
And I don't like to use "beta" patches on a production machine.



Attilla
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[Asterisk-Users] Change g729 payload

[Attilla De Groot]

Hi All,


I have a SIP provider that tells me that my RTP stream uses a  
"20bytes payload in the g729 coded data". And they would like that we  
change this to 30bytes (3 frames).


But maybe I'm wrong but isn't a certain payload just a standard for a  
codec ?


And if I'm wrong, how can I change the payload for my g729 calls in   
Asterisk.



Greetings,
Attilla
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Re: [Asterisk-Users] Recent debian packages?

[Attilla de Groot]

Massimo Nuvoli wrote:

>Also you can use the "unstable" branch of debian, all things are near
>ok, from the asterisk core to the kernel.
>
>Bye
>
>  
>
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a bad advice to run unstable on production systems.


Gr,
Attilla
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Re: [Asterisk-Users] Memory-leak 1.2.7.1

[Attilla de Groot]

Anthony Rodgers wrote:

> Is there any chance you're connecting to a remote share using CIFS?
>
> What does slabtop look like on your machines?


I would like to answer both question, but I don't know what CIFS of
slabtop is. But I'm sure you can tell me.


Greetings,
Attilla

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Re: [Asterisk-Users] Memory-leak 1.2.7.1

[Attilla de Groot]

Vij wrote:

> May be "updatedb" or some other such heavy application, which runs at
> night is causing heavy load on the system and spoils the working of
> asterisk.
>
> See if this phenomenon happens at the same time of the day everyday.
> Also, see what processes run at *that time*.
>
> Cheers,
> Vij
>
Hi Vij,


Well since the problem occurs on diffrent machines, I'm not so sure
about this. I'm going to try if I can see what processes run at *that
time*, but like I said it often occurs at night when I'm at sleep.

So I'm first going to downgrade 1.2.3, someone told me, that he was 100%
sure there are no memory leaks in that version.


Greetings,
Attilla
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Re: [Asterisk-Users] registration at Voipbuster times out

[Attilla de Groot]

Remko Muis wrote:

> Hi,
>
> I am new here on this list, and have a problem of which I hope that
> somebody here can help me with it.
> I have a Voipbuster account, with which I would like to make phone
> calls via my Asterisk PBX. If I let X-Lite register directly at
> voipbuster.com, everything is OK, but if I let Asterisk register
> there, it says "registration for [EMAIL PROTECTED]
>  timed out, trying again", even
> though all settings are precisely as in X-Lite (username, password,
> and sip-proxy settings). Also I am sure the right ports are forwarded
> or open, both in my router and in iptables (firewall of Asterisk
> server). The log files of X-Lite and the output of "sip debug" show no
> differences, except this one:
> 

Hi Remko,


If the account is working properly most of the time registration
timeouts are because of wrong time settings. So check the time and date
on your server. :)


Greetings,
Attilla
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Re: [Asterisk-Users] Memory-leak 1.2.7.1

[Attilla de Groot]

Steve Totaro wrote:

> Hardware platform and specs?  Call volume?  Any messages in your
> logs?  I had this problem on an Itanium2 box, went away when I
> "downgraded" to a Xeon.
>
Hi Steve,


At this moment I don't have real acurate statistics. But think of 40
calls a day. So nothing fancy I think. And the hardware configuration
are all just simple x86 machines, 2 Pentium4 machines, 1 Amd and a
Celeron I believe.


Greetings,
Attilla
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Re: [Asterisk-Users] Memory-leak 1.2.7.1

[Attilla de Groot]

Marco Mouta wrote:

> I'm also not an expert, but could it as any relationship with your
> Telephony card drivers??
>
> Which Telephony boards do u use?
>
>
None. :)

I only use Asterisk as an VoIP pbx. Only the zaptel drivers installed
for ztdummy as a timer interface.


Greetings,
Attilla
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[Asterisk-Users] Memory-leak 1.2.7.1

[Attilla de Groot]

Hi All,


First off all, this is my first mail to this mailing-list, so if I am
doing something wrong please tell me. And apologies for my english in
advance, it's not my native language.

Anyway, I have few machines running Asterisk 1.2.7.1. All machines but
one are Gentoo (other one is Debian). The problem is that Asterisk keeps
eating my memory.

Just random (mostly at night) all my free memory is gone and of course
Asterisk doesn't work anymore (no moh, queues doesn't work etc.). At
first I didn't knew for sure it was Asterisk causing this problem, so I
made sure Asterisk was the only extensive application running (no
apache, mysql etc.) and the problem kept returning. And of course if I
didn't happen when I won't start Asterisk.

So, this left me only one conclusion. The application with the memory
leak is Asterisk. So I thought of these things:

- Problem with linux distribution ?
Nope, got 2 different distributions and happens on both ?
- Bug with mpg123 ?
Nope, used native Asterisk moh thing, still happens.
- Memory leak due to a loop in my configuration ?
Nope, 3 diffrent configurations, 2 of them are _really_ simple (like 3
context and some dial strings :P).


Since I'm no programmer and not really a bug-hunter I'm out of idea's
right now. The only solution for me nog is to reboot the machines if and
when it happens. I was unable to find a similar problem in the
mailinglist and nobody had the same problem in the irc channel.

So I'm hoping somebody has a solution of a tip for me.


Greetings,
Attilla de Groot


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