astlinux-users — AstLinux Users Mailing List

Re: [Astlinux-users] Call Forwarding in my Astlinux I have no success.

[Gary G. H. <ghe...@co...>]

Nedi:

I prefer a simple dialplan approach ...  In my EXTENSIONS.CONF I have a few
"Alias" extensions ... If you dial "5000" it rings a sip phone on my desk
... If you dial "5001" the call get routed out my VoIP provider and rings my
cell phone ... So ext 5001 is just an alias that calls a "DIAL <My Cell
Phone>" command ... I also use similar alias extensions along with a
"dbput/dbget" to set internal extensions to forward to another internal
extension when someone is out of office ...

I have also setup a pretty determined "find me" scenario for my paying
clients ... They have a special extension that they have been told to dial
when trying to reach me ... In this scenario, the system tries me at my
desk, then at my home number and finally at my cell phone ... Asterisk stays
in control all the way to the cell phone handoff ... At the cell phone
handoff, Asterisk just lets it ring until voicemail on the cell phone takes
the call as that was the last resort ...

In any case, the summary is that there are many ways to handle call
forwarding depending on what you are trying to get done ... Most involve
using either a "queue" or some form of "manual set forwarding number" at the
phone ... There are many examples of various methods on voip-info.org ...
The way I am doing it now was taken directly from one of the examples there
and tweaked a bit for my needs ...

G.Hendershot

-----Original Message-----
From: ast...@li...
[mailto:ast...@li...] On Behalf Of Nedi
Sent: Wednesday, May 30, 2007 6:25 PM
To: Astlinux list
Subject: [Astlinux-users] Call Forwarding in my Astlinux I have no success.

Hello,

I come back,  has anyone idea how can i make call forwarding it would be
great  anyone cann help me with.

nedi 


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Re: [Astlinux-users] is dyndns supported in the version 0.4.5

[Gary G. H. <ghe...@co...>]

Nedi:

The answer if I understand properly is, "YES" ... 

I currently use this function in v0.4.5 to register my Astlinux box with
DynDNS.Org ... I understand the function supports a couple other popular
services that are similar ... The settings you need to make are made in
RC.CONF as with earlier versions ... I do not think this feature has changed
much in quite a while ...

G.Hendershot


-----Original Message-----
From: ast...@li...
[mailto:ast...@li...] On Behalf Of Nedi
Sent: Wednesday, May 30, 2007 6:19 PM
To: Astlinux list
Subject: [Astlinux-users] is dyndns supported in the version 0.4.5

is dyndns functions supported in the version 0.4.5 regards nedi

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[Astlinux-users] Call Forwarding in my Astlinux I have no success.

[Nedi <ne...@gm...>]

Hello,

I come back,  has anyone idea how can i make call forwarding it would be 
great  anyone cann help me with.

nedi 

[Astlinux-users] is dyndns supported in the version 0.4.5

[Nedi <ne...@gm...>]

is dyndns functions supported in the version 0.4.5
regards nedi

Re: [Astlinux-users] Astlinux 0.4.5on ViaEN12000EG - asteriskSIGSEGV

[Kristian K. <kri...@gm...>]

On 5/30/07, Ioan Indreias <ioa...@mo...> wrote:
> We experience crashes with both images - the generic i586 (this is the one
> used when I start this thread) and via (which is the one we have currently
> on our machine).
>
> Thanks,
> Ioan.
>

Ioan,

  I have compiled gdb for AstLinux 0.4.x here:

http://admin.star2star.com/gdb.gz

sha1:
cbedf5d68460a8a43a690559a49eebcd7f436e99  gdb.gz

  Follow the instructions here:

http://www.voip-info.org/wiki/view/Asterisk+debugging#Backtracingacoredumpfileintmp

  Let me know how it goes!


-- 
Kristian Kielhofner

Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Kristian K. <kri...@gm...>]

On 5/30/07, Ioan Indreias <ioa...@mo...> wrote:
> Hello Mathias,
>
> 1. I use no transcoding.
> In Sipp tests I use the standard uac_pcap (using alaw g711a.pcap file)  and
> uas with rtp_echo scenarios.
> Also, as I have mentioned I run uac+uas scenario (with no RTP traffic) and
> asterisk crashed with segmentation fault.
>
> I have core files but I do not know how to debug them.
>

Ioan,

  You will have to load that corefile into gdb.  I will try to compile
an AstLinux gdb for you today.

-- 
Kristian Kielhofner

Re: [Astlinux-users] Astlinux 0.4.5on ViaEN12000EG - asteriskSIGSEGV

[Ioan I. <ioa...@mo...>]

We experience crashes with both images - the generic i586 (this is the one 
used when I start this thread) and via (which is the one we have currently 
on our machine).

Thanks,
Ioan.

----- Original Message ----- 
From: "Darrick Hartman" <dha...@dj...>
To: "AstLinux Users Mailing List" <ast...@li...>
Sent: Wednesday, May 30, 2007 5:41 PM
Subject: Re: [Astlinux-users] Astlinux 0.4.5on ViaEN12000EG - 
asteriskSIGSEGV


> Ioan Indreias wrote:
>> Hello Darrick,
>>
>> Our machine have 1GB of memory - so I think it is more than enough. And 
>> as
>> you seen from my "free" reports, only 35-45MB are used.
>>
>
> Whoops!  My bad.  I was up til 4:30am last night so I'll blame it on that.
>
>> /var and /tmp was never full (I monitor them carrefully).
>>
>> Maybe I miss something?
>>
>
> I was only making the comment because I wouldn't have wanted you to run
> out of system memory (when I thought you only had 128MB).  With 1GB of
> ram there would be no problem in making those partitions larger (using
> the variables in rc.conf), but as you say, you're not filling those up.
>
> Are you running the VIA or generic i586 image?
>
>> Ioan.

Re: [Astlinux-users] Astlinux 0.4.5 on ViaEN12000EG - asteriskSIGSEGV

[Darrick H. <dha...@dj...>]

Ioan Indreias wrote:
> Hello Darrick,
>
> Our machine have 1GB of memory - so I think it is more than enough. And as 
> you seen from my "free" reports, only 35-45MB are used.
>   

Whoops!  My bad.  I was up til 4:30am last night so I'll blame it on that.

> /var and /tmp was never full (I monitor them carrefully).
>
> Maybe I miss something?
>   

I was only making the comment because I wouldn't have wanted you to run 
out of system memory (when I thought you only had 128MB).  With 1GB of 
ram there would be no problem in making those partitions larger (using 
the variables in rc.conf), but as you say, you're not filling those up.

Are you running the VIA or generic i586 image?

> Ioan.
>
> ----- Original Message ----- 
> From: "Darrick Hartman" <dha...@dj...>
> To: "AstLinux Users Mailing List" <ast...@li...>
> Sent: Wednesday, May 30, 2007 5:23 PM
> Subject: Re: [Astlinux-users] Astlinux 0.4.5 on ViaEN12000EG - 
> asteriskSIGSEGV
>
>
> Mathias WOLFF wrote:
>   
>> Hello,
>>
>> What is the codec used : alaw to gsm ; ulaw to G729 ... ?
>> Have you modify tmpfs sizes in rc.conf ?
>>
>>     
>
> You'll need to be very careful there.  From prior emails, Ioan only has
> 128MB of ram on this machine.  If you increase /tmp or /var which are
> both tmpfs file systems, your available ram will be reduced as the
> partitions fill up.  It would be interesting to repeat the same test
> with more memory (say 512MB) and increased /tmp and /var partitions.
>
>
>   
>> regards
>>
>> Mathias
>>
>>     
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>   


-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

Re: [Astlinux-users] Astlinux 0.4.5 on ViaEN12000EG - asteriskSIGSEGV

[Ioan I. <ioa...@mo...>]

Hello Darrick,

Our machine have 1GB of memory - so I think it is more than enough. And as 
you seen from my "free" reports, only 35-45MB are used.

/var and /tmp was never full (I monitor them carrefully).

Maybe I miss something?

Ioan.

----- Original Message ----- 
From: "Darrick Hartman" <dha...@dj...>
To: "AstLinux Users Mailing List" <ast...@li...>
Sent: Wednesday, May 30, 2007 5:23 PM
Subject: Re: [Astlinux-users] Astlinux 0.4.5 on ViaEN12000EG - 
asteriskSIGSEGV


Mathias WOLFF wrote:
> Hello,
>
> What is the codec used : alaw to gsm ; ulaw to G729 ... ?
> Have you modify tmpfs sizes in rc.conf ?
>

You'll need to be very careful there.  From prior emails, Ioan only has
128MB of ram on this machine.  If you increase /tmp or /var which are
both tmpfs file systems, your available ram will be reduced as the
partitions fill up.  It would be interesting to repeat the same test
with more memory (say 512MB) and increased /tmp and /var partitions.


> regards
>
> Mathias
>

Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Darrick H. <dha...@dj...>]

Mathias WOLFF wrote:
> Hello,
>
> What is the codec used : alaw to gsm ; ulaw to G729 ... ?
> Have you modify tmpfs sizes in rc.conf ?
>   

You'll need to be very careful there.  From prior emails, Ioan only has 
128MB of ram on this machine.  If you increase /tmp or /var which are 
both tmpfs file systems, your available ram will be reduced as the 
partitions fill up.  It would be interesting to repeat the same test 
with more memory (say 512MB) and increased /tmp and /var partitions.


> regards
>
> Mathias
>
> Ioan Indreias a écrit :
>   
>> Hello,
>>
>> I come back with the following report:
>>
>> 1. I add some swap but I experience the same crash (asterisk segmentation 
>> fault at 46 simultaneous SIP calls - with or without RTP traffic).
>>
>> 2. In order to avoid any discussions regarding Sipp, I create a simple test 
>> scenario:
>>     a. extensions.conf
>>         [test]
>>         exten => 000,1,Answer
>>         exten => 000,2,Echo
>>         exten => 000,3,Hangup
>>
>>         exten => 001,1,Answer
>>         exten => 001,2,Playback(demo-congrats)
>>         exten => 001,3,Hangup
>>
>>     b. create a call file (/tmp/test.call)
>>         Channel: Local/000@test
>>         MaxRetries: 0
>>         Context: test
>>         Extension: 001
>>         Priority: 1
>>
>>     c. Start the test:
>>         in one terminal run:
>>          # watch -n 1 asterisk -rx "show channels"
>>         in the seccond terminal run several times:
>>          # cp /tmp/test.call /var/spool/asterisk/outgoing/`date +%s`.call; 
>> free
>>         this will add 2 active calls.
>>
>>
>> Result of the test: On our PBX (astlinux 0.4.5 on Via EN12000EG), asterisk 
>> crash at the transition from 46 to 48 channels:
>> ....
>> pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
>> +%s`.call; free
>>               total         used         free       shared      buffers
>>   Mem:       971136        37144       933992            0         1396
>>  Swap:      2097136            0      2097136
>> Total:      3068272        37144      3031128
>>
>> pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
>> +%s`.call; free
>>               total         used         free       shared      buffers
>>   Mem:       971136        32704       938432            0         1396
>>  Swap:      2097136            0      2097136
>> Total:      3068272        32704      3035568
>>
>>
>> Could anybody confirmed that using astlinux 0.4.5 on their system, they 
>> could have more than 48 active calls using this simple test?
>>
>> Best regards,
>> Ioan.
>> www.modulo.ro
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by DB2 Express
>> Download DB2 Express C - the FREE version of DB2 express and take
>> control of your XML. No limits. Just data. Click to get it now.
>> http://sourceforge.net/powerbar/db2/
>> _______________________________________________
>> Astlinux-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>>   
>>     
>
>   


-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Ioan I. <ioa...@mo...>]

Hello Mathias,

1. I use no transcoding.
In Sipp tests I use the standard uac_pcap (using alaw g711a.pcap file)  and 
uas with rtp_echo scenarios.
Also, as I have mentioned I run uac+uas scenario (with no RTP traffic) and 
asterisk crashed with segmentation fault.

I have core files but I do not know how to debug them.


2. Regarding tmpfs sizes - I do not modified them and here are the current 
settings:

pbx asterisk # df -k
Filesystem           1k-blocks      Used Available Use% Mounted on
/dev/sda1                63413     57044      6369  90% /
none                       200         0       200   0% /dev
none                      5000       192      4808   4% /var
none                     10000       184      9816   2% /tmp
/dev/sda2             24730020       456  23473320   0% /mnt/kd

Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Mathias W. <mat...@gr...>]

Hello,

What is the codec used : alaw to gsm ; ulaw to G729 ... ?
Have you modify tmpfs sizes in rc.conf ?

regards

Mathias

Ioan Indreias a écrit :
> Hello,
>
> I come back with the following report:
>
> 1. I add some swap but I experience the same crash (asterisk segmentation 
> fault at 46 simultaneous SIP calls - with or without RTP traffic).
>
> 2. In order to avoid any discussions regarding Sipp, I create a simple test 
> scenario:
>     a. extensions.conf
>         [test]
>         exten => 000,1,Answer
>         exten => 000,2,Echo
>         exten => 000,3,Hangup
>
>         exten => 001,1,Answer
>         exten => 001,2,Playback(demo-congrats)
>         exten => 001,3,Hangup
>
>     b. create a call file (/tmp/test.call)
>         Channel: Local/000@test
>         MaxRetries: 0
>         Context: test
>         Extension: 001
>         Priority: 1
>
>     c. Start the test:
>         in one terminal run:
>          # watch -n 1 asterisk -rx "show channels"
>         in the seccond terminal run several times:
>          # cp /tmp/test.call /var/spool/asterisk/outgoing/`date +%s`.call; 
> free
>         this will add 2 active calls.
>
>
> Result of the test: On our PBX (astlinux 0.4.5 on Via EN12000EG), asterisk 
> crash at the transition from 46 to 48 channels:
> ....
> pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
> +%s`.call; free
>               total         used         free       shared      buffers
>   Mem:       971136        37144       933992            0         1396
>  Swap:      2097136            0      2097136
> Total:      3068272        37144      3031128
>
> pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
> +%s`.call; free
>               total         used         free       shared      buffers
>   Mem:       971136        32704       938432            0         1396
>  Swap:      2097136            0      2097136
> Total:      3068272        32704      3035568
>
>
> Could anybody confirmed that using astlinux 0.4.5 on their system, they 
> could have more than 48 active calls using this simple test?
>
> Best regards,
> Ioan.
> www.modulo.ro
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>   

-- 
Mathias WOLFF
Groupe COMTEP
Siège : 1, rue Meunier 44880 SAUTRON
Tél. : 0 826 103 100
Fax. : 02 40 63 41 33
Site : http://www.groupe-comtep.com

OPERATEUR & INTEGRATEUR TELECOM

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Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Ioan I. <ioa...@mo...>]

Hello,

I come back with the following report:

1. I add some swap but I experience the same crash (asterisk segmentation 
fault at 46 simultaneous SIP calls - with or without RTP traffic).

2. In order to avoid any discussions regarding Sipp, I create a simple test 
scenario:
    a. extensions.conf
        [test]
        exten => 000,1,Answer
        exten => 000,2,Echo
        exten => 000,3,Hangup

        exten => 001,1,Answer
        exten => 001,2,Playback(demo-congrats)
        exten => 001,3,Hangup

    b. create a call file (/tmp/test.call)
        Channel: Local/000@test
        MaxRetries: 0
        Context: test
        Extension: 001
        Priority: 1

    c. Start the test:
        in one terminal run:
         # watch -n 1 asterisk -rx "show channels"
        in the seccond terminal run several times:
         # cp /tmp/test.call /var/spool/asterisk/outgoing/`date +%s`.call; 
free
        this will add 2 active calls.


Result of the test: On our PBX (astlinux 0.4.5 on Via EN12000EG), asterisk 
crash at the transition from 46 to 48 channels:
....
pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
+%s`.call; free
              total         used         free       shared      buffers
  Mem:       971136        37144       933992            0         1396
 Swap:      2097136            0      2097136
Total:      3068272        37144      3031128

pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date 
+%s`.call; free
              total         used         free       shared      buffers
  Mem:       971136        32704       938432            0         1396
 Swap:      2097136            0      2097136
Total:      3068272        32704      3035568


Could anybody confirmed that using astlinux 0.4.5 on their system, they 
could have more than 48 active calls using this simple test?

Best regards,
Ioan.
www.modulo.ro

Re: [Astlinux-users] [Astlinux-announce] Available for testing: Digium G729 registration work-around for AstLinux

[Leonardo K. (Gmail) <lka...@gm...>]

Hi everyone;

News about that issue?




On 5/23/07, Kristian Kielhofner <kri...@gm...> wrote:
> On 5/23/07, Darrick Hartman <dha...@dj...> wrote:
> > > Darrick,
> > >
> > >   Try doing this:
> > >
> > > echo "216.207.245.3 register.digium.com" >> /tmp/etc/hosts
> > >
> > >   And then re-run it.
> > >
> > >
> > >
> > Ok.  Now the register utility runs properly.  Is that some sort of dns
> > issue with uclibc again?
> >
> > I do get the following error when going through the process.
> >
> > Could not write license file /var/lib/asterisk/licenses/G729-12193349.lic!
> >
> > Looks like we need to add a symbolic link from
> > /var/lib/asterisk/licenses to /mnt/kd/licenses
> >
> > Darrick
> >
>
> Darrick,
>
>   The problem is that DNS resolution in glibc requires another library
> - libresolv and I didn't copy that into the tarball.  I will now.
>
>  I agree on the symlink.
>
>
> --
> Kristian Kielhofner
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>

Re: [Astlinux-users] Call Forwarding in my Astlinux I have no success.

[Mark P. <g7...@g7...>]

They do say that 13 is an unlucky number ;-}


On Sun, 2007-05-27 at 21:16 +0200, Nedi wrote:
> I try to forward all incoming calls of my internal 13 to my mobile phone if 
> I use followed code in my astlinux extensions.conf   I have no success.
> 
> i dial my internal 13 and 13 rings 13 the call was not forwarded to my 
> mobile.
> 
> I typed on my softphone *21*and my mobile number
> 
> if I call normal from  13 to mobile it works
> 
> -------------------------------------------------------
> 
> ; Unconditional Call Forward
> 
> exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
> 
> exten => _*21*X.,2,Playback(call-fwd-unconditional)
> 
> exten => _*21*X.,3,Hangup
> 
> exten => _*22*,1,DBdel(CFIM/${CALLERIDNUM})
> 
> exten => _*22*,2,Playback(call-fwd-cancelled)
> 
> exten => _*22*,3,Hangup
> 
> ------------------------------
> 
> can be i use something wrong or old in my extensions.conf  or sip.conf   or 
> I must something activate to work with dbput and dgdel
> 
> or is there another way to realise that with calls forwarding
> 
> I have in my extensions.conf
> 
> 
> 
> exten =>13,1,Dial(SIP/13,25,r)
> exten =>13,n,Answer
> exten =>13,n,Playback(vm-nobodyavail)
> exten =>13,n,Voicemail(13)
> exten =>13,n,Hangup
> 
> 
> 
> and in my sip
> 
> 
> 
> [13]
> 
> type=friend
> username=13
> secret=13
> callerid="13" <13>
> host=dynamic
> mailbox=13@default
> dtmfmode=rfc2833
> canreinvite=no
> context=13
> 
> 
> 
> regards nedi
> 
> 
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

Re: [Astlinux-users] NETDEV WATCHDOG: ethx: transmit timed out (net4801)

[anuar m. <an...@ro...>]

--- Kristian Kielhofner
<kri...@gm...> wrote:

Thanks Kristian,
See comments below.

> 
> Anuar,
> 
>   The last time I saw something like this on a
> Soekris it was because
> of the NS DP83815 short cable issues.  Evidently the
> DP83815 had some
> problems figuring out the proper attenuation on
> various cable lengths.
>  It caused all sorts of problems.
> 
>   I could swear that I also saw it with some
> DP83816s a couple of
> years ago.  Driver updates have been able to work
> around the issue
> (for the most part) and I have not heard of this in
> years.
> 
>   Unfortunately, the natsemi driver does not
> distinguish between the
> 83815 and the 83816 (which fixed the issue -
> supposedly).  Can you
> open your case and read the DP8381x part number off
> of your ethernet
> chips?

Part number: DP83816AVNG

> 
>   Also, try this:
> 
> 1)  Use longer or shorter ethernet cables (trial and
> error) on both
> ethernet interfaces (although eth1 seems to be the
> culprit).
> 
I have try numbers of cables from 0.5 meter to 3
meters   but the problem is still there. I am pretty
sure the cables are ok.

> 2)  Is your switch managed?  See if there are an
> excessive number of
> link state changes and/or input errors on the
> connected ethernet port.
>  Also check and make sure that it is negotiating the
> same speed
> (100/full).
> 
I am not using any switch in between the PC1 <-->
Net4801 <--> PC2; all are directly connected.

> 3)  Have you tried a different switch?  Try
> inserting another switch
> in between the systems or use a different switch
> completely.
> 
I cannot test this configuration since I don't have a
switch. I will try to find somebody to borrow from.

>   Of course it could always be something simple
> (like a bad cable),
> but step #1 should cross that off the list too.
> 
> -- 

I found some interesting, since I using vsftp as a
server and by default local authenticated user get
unlimited transfer rate.

By setting "local_max_rate=1000000" bytes/second in
/etc/vsftpd/vsftpd.conf for example to limit the
transfer rate. Restart vsftpd and re-run the ftp test
again, the no issue. I got about 5464 Kbytes/sec which
is quite good. 

If I set to unlimited transfer rate (local_max_rate=0)
and restart vsftpd and re-run the test I got time out
again. I seem that the driver has some problem
handling high traffic rate. 

Anyway, I try to switch different ports for example
using eth0 and eth2, eth0 and eth1, etc but using
unlimitted transfer rate the problem persist.

Any pointers appreciated becuase in real world one
cannot control other people ftp server or other server
for that matters.

Thanks,
Anuar

> Kristian Kielhofner
> 
>
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[Astlinux-users] Call Forwarding in my Astlinux I have no success.

[Nedi <ne...@gm...>]

I try to forward all incoming calls of my internal 13 to my mobile phone if 
I use followed code in my astlinux extensions.conf   I have no success.

i dial my internal 13 and 13 rings 13 the call was not forwarded to my 
mobile.

I typed on my softphone *21*and my mobile number

if I call normal from  13 to mobile it works

-------------------------------------------------------

; Unconditional Call Forward

exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})

exten => _*21*X.,2,Playback(call-fwd-unconditional)

exten => _*21*X.,3,Hangup

exten => _*22*,1,DBdel(CFIM/${CALLERIDNUM})

exten => _*22*,2,Playback(call-fwd-cancelled)

exten => _*22*,3,Hangup

------------------------------

can be i use something wrong or old in my extensions.conf  or sip.conf   or 
I must something activate to work with dbput and dgdel

or is there another way to realise that with calls forwarding

I have in my extensions.conf



exten =>13,1,Dial(SIP/13,25,r)
exten =>13,n,Answer
exten =>13,n,Playback(vm-nobodyavail)
exten =>13,n,Voicemail(13)
exten =>13,n,Hangup



and in my sip



[13]

type=friend
username=13
secret=13
callerid="13" <13>
host=dynamic
mailbox=13@default
dtmfmode=rfc2833
canreinvite=no
context=13



regards nedi

[Astlinux-users] Off topioc but interesting

[Mark P. <g7...@g7...>]

You know when you see a gizmo doing some pretty neat task and you think
to yourself "I could use that for my xyz project"?

I was at the NJQRP meeting today and happened to have my demo AstLinux
NET4801 with me in my laptop bag.

Not half an hour later it was stripped down and "re-purposed" into a
remote satellite telemetry receiver. 

We had a visit from one of the NASA dudes and he told us about how they
are trying to make a cheap satellite receiver that would listen to the
telemetry data and then upload it to NASA in real time for later display
in a free application.

By dumb luck we had most everything one would need (computers, radio's,
soundcards etc) within the group and so we built a proof of concept rig
right there on the table.

NASA dude reckons he could probably mass produce these things for about
$250 and drop them into schools, research stations, ham shacks etc all
over the world thus creating a receiver network for almost nothing
(comparatively speaking).

And it all started with an AstLinux box!

Mark 

Re: [Astlinux-users] app_addon_sql_mysql.so on astlinux

[Sebastian A. <sp...@sy...>]

Hello list,

For what it's worth, I would also like to see app_addon_sql_mysql.so in
astlinux. I wouldn't mind if it were optional or not.

Kind regards,

Sebastian


> -----Original Message-----
> From: ast...@li... 
> [mailto:ast...@li...] On 
> Behalf Of gramels
> Sent: 23 May 2007 19:16
> To: ast...@li...
> Subject: [Astlinux-users] app_addon_sql_mysql.so on astlinux
> 
> Hi,
> 
> while browsing through the list I recognized some discussions but no
> solution to get a mysql client running on a astlinux to enable remote
> logging on a mysql server in the net.
> 
> I am in the process to move asterisk from my main server to on a
> dedicated device, but would like to keep the features and the data.
> 
> For this I need app_addon_sql_mysql.so running. I played around a
> little with, trying to copy over the modules from my server, but it
> failed so far.
> 
> Any hints how to get the right files/ binaries would be great.
> 
> Even more appreciated would be if, a future version might include
> those mysql client modules in the distro (may be as optional?)
> 
> -gramels
> --
> http://blog.gramels.info/blog
> 
> --------------------------------------------------------------
> -----------
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> Download DB2 Express C - the FREE version of DB2 express and take
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> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via 
> PayPal to pa...@kr....
> 

Re: [Astlinux-users] Astlinux 0.4.5 on Via EN12000EG - asteriskSIGSEGV

[Ioan I. <ioa...@mo...>]

Hello,

Any idea on how to debug the asterisk crash?

Last test was running 2 cps with calls of 10 sec (resulting 20
simultaneous G711 calls) and after aprox. 3 hours asterisk crash. Same
behaviour - no "visible" traces.

If no other ideas I will configure some swap space and try again.

Best regards,
Ioan.

> Hello Kris,
>
> Thank you for your quick answer and tips. Still I do not think it is a
> problem on ulimit.
> I verify fds for asterisk pid and was less that 1024. Also, for generating
> the core I run unlimit -c unlimited....but this is just to get the core.
> How
> to debug it?
>
> Regarding swap - I see (with free) that there is plenty of free memory. So
> no need for a swap. Or am I wrong?
>
> Bellow you have some data from such a crash:
>
> ------------------------------------------------------------------------
> pbx ~ # asterisk -rx "show channels"; free
> Channel              Location             State   Application(Data)
> 0 active channels
> 0 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        57964       913172            0         8304
>  Swap:            0            0            0
> Total:       971136        57964       913172
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 10 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        58808       912328            0         8304
>  Swap:            0            0            0
> Total:       971136        58808       912328
>  22:16:56 up  7:42, load average: 0.09, 0.02, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 20 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        59656       911480            0         8304
>  Swap:            0            0            0
> Total:       971136        59656       911480
>  22:17:22 up  7:42, load average: 0.11, 0.03, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 31 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        60596       910540            0         8304
>  Swap:            0            0            0
> Total:       971136        60596       910540
>  22:17:46 up  7:43, load average: 0.14, 0.05, 0.01
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 35 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        60940       910196            0         8304
>  Swap:            0            0            0
> Total:       971136        60940       910196
>  22:17:58 up  7:43, load average: 0.11, 0.04, 0.01
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 41 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        61580       909556            0         8304
>  Swap:            0            0            0
> Total:       971136        61580       909556
>  22:18:14 up  7:43, load average: 0.08, 0.04, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 42 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        61660       909476            0         8304
>  Swap:            0            0            0
> Total:       971136        61660       909476
>  22:18:29 up  7:44, load average: 0.06, 0.04, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 43 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        61752       909384            0         8304
>  Swap:            0            0            0
> Total:       971136        61752       909384
>  22:18:37 up  7:44, load average: 0.05, 0.04, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 44 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        61832       909304            0         8304
>  Swap:            0            0            0
> Total:       971136        61832       909304
>  22:18:44 up  7:44, load average: 0.05, 0.04, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 45 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        61924       909212            0         8304
>  Swap:            0            0            0
> Total:       971136        61924       909212
>  22:18:53 up  7:44, load average: 0.04, 0.03, 0.00
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
> 46 active calls
>               total         used         free       shared      buffers
>   Mem:       971136        62004       909132            0         8304
>  Swap:            0            0            0
> Total:       971136        62004       909132
>  22:18:59 up  7:44, load average: 0.03, 0.03, 0.00
>
> --- When I tried 47th call, asterisk crashed ---
>
> pbx ~ # asterisk -rx "show channels" | grep "active call"; free; uptime
>               total         used         free       shared      buffers
>   Mem:       971136        55480       915656            0         8304
>  Swap:            0            0            0
> Total:       971136        55480       915656
>  22:19:05 up  7:44, load average: 0.51, 0.13, 0.03
> pbx ~ #
>
> ------------------------------------------------------------------------
>
>
> What do you think?
>
> BR,
> Ioan.

Re: [Astlinux-users] Using FreePBX conf files

[<ipf...@gm...>]

"Thank you very much Kristian Kielhofner."

He is really a very nice person...

I thank you to.

Miklos 

-----Mensagem original-----
De: ast...@li...
[mailto:ast...@li...] Em nome de cfh
Enviada: quarta-feira, 23 de maio de 2007 18:04
Para: AstLinux Users Mailing List
Assunto: Re: [Astlinux-users] Using FreePBX conf files

hi all,


>   Trixbox certainly has it's time and place.  I don't think that I 
> have ever publicly addressed this but I might as well...
> 
>   Trixbox and AstLinux have very different design goals.  AstLinux is 
> not really designed to be a complete out of the box beginner's PBX 
> system.  It is meant to be a rock solid, bare bones toolkit for making 
> an appliance, PBX, etc.
> 
In my old company my project manager want only trixbox solutions on epia on
4GB Microdrive with many problem of performance, misnd channel, backup and
restore. But  the new asterisk@home have a intuitive GUI to manage it.

Now I am free :0 (no project manager) and I can use astlinux :) and other my
derivations from buildroot toolchain like IPS, firewall, openvpn server,
access point ...


With astlinux all the txt config  stay in a usb disk and with

ssh root@myastlinuxpbx 'cat /dev/sda' > backup.img cat backup.img > /dev/sda
(whre sda is my usbdisk deposit in the vault)

is possibile make fantastic backup without problem :)

In this moment i use my build script to create SIP/IAX internals and i edit
my extension because i cant find other solutions.
The problem is for the noshellperson :( .



>   For instance - Matt Riddell has done a great job with the 
> VentureVoip appliance.  They took the functionality that AstLinux had, 
> added provisioning, hardware, and a web gui (among many other things 
> I'm sure) and ended up with a VERY nice finished product.  Well done 
> guys!
I can't find the web gui of VentureVoip appliance . Isnt it GPL ? Why?



Thank you very much Kristian Kielhofner.

With you fantastic ASTLINUX you have take me the way of buildroot :)

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Re: [Astlinux-users] LcdProc

[Manuel D. <man...@te...>]

Thanks Darrick

-----Mensaje original-----
De: ast...@li...
[mailto:ast...@li...] En nombre de Darrick
Hartman
Enviado el: jueves, 24 de mayo de 2007 17:52
Para: AstLinux Users Mailing List
Asunto: Re: [Astlinux-users] LcdProc

Manuel Dominguez wrote:
> Hi again,
>
> I am again with the lcd. I have found a patch for lcdproc to support
> picoLCD. I have tried to include in package but some errors appears whe
try
> to apply. My experience in linux is very limited. Can somebody take a look
> to the patch and oriented me where is the problem? 
>   
Manuel,

Just wanted to let you know that lcdproc has been updated in trunk and 
the 0.4 branch to the latest version (0.5.2).  This version now has 
support for the picoLCD device.

Darrick

-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com


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Re: [Astlinux-users] LcdProc

[Darrick H. <dha...@dj...>]

Manuel Dominguez wrote:
> Hi again,
>
> I am again with the lcd. I have found a patch for lcdproc to support
> picoLCD. I have tried to include in package but some errors appears whe try
> to apply. My experience in linux is very limited. Can somebody take a look
> to the patch and oriented me where is the problem? 
>   
Manuel,

Just wanted to let you know that lcdproc has been updated in trunk and 
the 0.4 branch to the latest version (0.5.2).  This version now has 
support for the picoLCD device.

Darrick

-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

Re: [Astlinux-users] NETDEV WATCHDOG: ethx: transmit timed out (net4801)

[Kristian K. <kri...@gm...>]

On 5/22/07, anuar musa <an...@ro...> wrote:
> Hi,
> I have download Astlinux-0.4.5.img.gz but it's still
> have the same problem as I mentioned before.
>
> I did make some change to configuration by enabling
> packet forwarding to enable routing in net4801 besides
> assigning IP address to eth1 and eth2. e.g.
> pbx ~ # ifconfig eth1 192.168.3.1 netmask
> 255.255.255.0
> pbx ~ # ifconfig eth2 192.168.2.1 netmask
> 255.255.255.0
> pbx ~ # echo 1 >
> /proc/sys/net/ipv4/conf/all/forwarding
>
> Some more detail info:
>
> Routing table:
> pbx ~ # netstat -nr
> Kernel IP routing table
> Destination     Gateway         Genmask         Flags
>  MSS Window  irtt Iface
> 192.168.3.0     0.0.0.0         255.255.255.0   U
>    0 0          0 eth1
> 192.168.2.0     0.0.0.0         255.255.255.0   U
>    0 0          0 eth2
>
> pbx ~ # cat /proc/interrupts
>            CPU0
>   0:     793413          XT-PIC  timer
>   2:          0          XT-PIC  cascade
>   4:       1022          XT-PIC  serial
>   5:          0          XT-PIC  ohci_hcd:usb1
>   8:     796142          XT-PIC  rtc
>  10:       2371          XT-PIC  eth0, eth1, eth2
>  14:      23244          XT-PIC  ide0
> NMI:          0
> ERR:          1
>
> Attached is the dmesg file.
>
> Regards,
> Anuar
>

Anuar,

  The last time I saw something like this on a Soekris it was because
of the NS DP83815 short cable issues.  Evidently the DP83815 had some
problems figuring out the proper attenuation on various cable lengths.
 It caused all sorts of problems.

  I could swear that I also saw it with some DP83816s a couple of
years ago.  Driver updates have been able to work around the issue
(for the most part) and I have not heard of this in years.

  Unfortunately, the natsemi driver does not distinguish between the
83815 and the 83816 (which fixed the issue - supposedly).  Can you
open your case and read the DP8381x part number off of your ethernet
chips?

  Also, try this:

1)  Use longer or shorter ethernet cables (trial and error) on both
ethernet interfaces (although eth1 seems to be the culprit).

2)  Is your switch managed?  See if there are an excessive number of
link state changes and/or input errors on the connected ethernet port.
 Also check and make sure that it is negotiating the same speed
(100/full).

3)  Have you tried a different switch?  Try inserting another switch
in between the systems or use a different switch completely.

  Of course it could always be something simple (like a bad cable),
but step #1 should cross that off the list too.

-- 
Kristian Kielhofner

Re: [Astlinux-users] Using FreePBX conf files

[cfh <llc...@gm...>]

hi all,


>   Trixbox certainly has it's time and place.  I don't think that I
> have ever publicly addressed this but I might as well...
> 
>   Trixbox and AstLinux have very different design goals.  AstLinux is
> not really designed to be a complete out of the box beginner's PBX
> system.  It is meant to be a rock solid, bare bones toolkit for making
> an appliance, PBX, etc.
> 
In my old company my project manager want only trixbox solutions on epia
on 4GB Microdrive with many problem of performance, misnd channel,
backup and restore. But  the new asterisk@home have a intuitive GUI to
manage it.

Now I am free :0 (no project manager) and I can use astlinux :) and
other my derivations from buildroot toolchain like IPS, firewall,
openvpn server, access point ...


With astlinux all the txt config  stay in a usb disk and with

ssh root@myastlinuxpbx 'cat /dev/sda' > backup.img
cat backup.img > /dev/sda (whre sda is my usbdisk deposit in the vault)

is possibile make fantastic backup without problem :)

In this moment i use my build script to create SIP/IAX internals and i
edit my extension because i cant find other solutions.
The problem is for the noshellperson :( .



>   For instance - Matt Riddell has done a great job with the
> VentureVoip appliance.  They took the functionality that AstLinux had,
> added provisioning, hardware, and a web gui (among many other things
> I'm sure) and ended up with a VERY nice finished product.  Well done
> guys!
I can't find the web gui of VentureVoip appliance . Isnt it GPL ? Why?



Thank you very much Kristian Kielhofner.

With you fantastic ASTLINUX you have take me the way of buildroot :)