[Astlinux-users] curl enabled in trunk

[Darrick Hartman]

Per someone's request in irc, I added the curl binary to default builds 
of trunk.  This does add a total of about 300K to the system, but does 
add quite a bit of functionality.  I won't be adding this to Astlinux 
0.6.x.  I don't think that 0.7.0 will be too far down the road though.

Darrick


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

[Darrick Hartman]

John Novack wrote:
> You are most welcome.
> 
> What these lists are supposed to be all about.
> 
> If only the asterisk-users list were populated with members that are as 
> nice as this group.
> 

You mean like the posting I made this afternoon with regards to the 
change in parked calls?  If you haven't already noticed call parking has 
changed significantly in 1.4.23.1.  Most of the changes are good.  (like 
parking the same call twice actually works now).  However, if a parked 
call reaches it's timeout, the parked call is no longer returned to the 
device that parked the call.  It's returned to the 's' extension in the 
context of the device that parked the call.  Totally whacked.  I posted 
this to the Asterisk-users mailing list this afternoon and have yet to 
have anyone reply to it.

About the only way to 'fix' this behavior is to have each phone have 
it's own context (which is just lame).

I didn't really see a detailed explanation in the Changelog either.  I 
knew there were some significant changes to res_features.so (and 
features.conf) only because of a bug in the parked call logic.  By 
default in prior versions, someone who was parked could initiate a 
transfer if they pressed the blind transfer feature key (# by default).

If you ARE moving from Asterisk 1.2 to Asterisk 1.4 there is a very good 
Upgrade.txt file that's available in the source code and on Digium's 
website (use google).

Darrick


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Minimal modules.conf ( was Re: Astlinux 0.6.3 Warning & Error's)

[Cleveland Electronic Services]

Hi John,

 

Thanks once again for the useful information, I used some of your examples
below to get rid of some more untoward stuff showing up on the logs, now my
logs are nice and clean. I am writing up a document as I go through as it is
a bit of a learning curve since 0.4.8 would you mind if I add this
information to it with a link to this post?

 

Thanks

 

Cheers

 

Cleve 

 

From: John Novack [mailto:jnov...@stromberg-carlson.org] 
Sent: Friday, 6 March 2009 1:34 PM
To: AstLinux Users Mailing List; thely...@gmail.com
Subject: [Astlinux-users] Minimal modules.conf ( was Re: Astlinux 0.6.3
Warning & Error's)

 

This is for Astlinux 0.5  a late version of Asterisk 1.2
Your results may vary
You may have to add to the list for 1.4

Tom Lynn wrote:;
[modules]
autoload=yes
;
; If you want, load the GTK console right away.  
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss.  Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
noload => chan_modem.so
noload => res_musiconhold.so
;
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_modem_i4l.so
noload => chan_modem_bestdata.so
noload => chan_modem_aopen.so
noload => chan_mgcp.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_skinny.so
noload => chan_misdn.so
noload => res_odbc.so
noload => cdr_odbc.so
noload => cdr_pgsql.so
noload => chan_sccp.so
noload => app_alarmreceiver.so
noload => app_realtime.so
noload => cdr_addon_mysql.so
noload => codec_g729c.so
noload => format_g729c.so
noload => pbx_dundi.so
noload => pbx_ael.so
load => app_macro.so
;
Not sure why I needed to add the last line.



 

On Thu, Mar 5, 2009 at 8:12 AM, John Novack 
wrote:



Darrick Hartman wrote: 




Message-ID: <3a248f9d4dd820e398a51733736c4...@localhost>
X-Sender: dhart...@djhsolutions.com
Received: from rrcs-74-87-125-110.west.biz.rr.com [74.87.125.110] with
HTTP/1.1
(POST); Thu, 05 Mar 2009 09:49:36 -0600
User-Agent: RoundCube Webmail/0.1
Content-Type: text/plain; charset="UTF-8"
Content-Transfer-Encoding: 8bit
 
Cleve,
 
You can either ignore the warnings or disable the modules that are trying to
load (that you're not using).  
 
If you look at /etc/asterisk/modules.conf, you'll see several modules
listed.
 
In your case, you would want to noload res_smdi.so and chan_misdn.so since
it doesn't look like you're using those.
 
  

I no load a whole host of unused modules.
Somewhere, many moons ago, I found a posting on one of the lists that had
suggestions. If modules.conf is set to "autoload" then one can no load
these, as well as skinny, mgcp, musiconhold if you aren't using it and a
raft of others.
Anyone interested I can post my "minimal" load.
Also on some versions, I found that the application macro wasn't loading,
and had to specify that to load. Never figured out why.
 





Not sure why you're seeing RED ALARM for dahdi (which is really zaptel, but
Digium made the change to call it dahdi even if we're still using zaptel...)
 
  

In my experience, if FXO ports on a TDM400 aren't connected to a CO to see
battery, they will give a RED alarm.

John Novack 






Darrick
 
On Fri, 6 Mar 2009 01:47:36 +1100, "Cleveland Electronic Services"
 
 wrote:
  

Hi All.
 
I am seeing a lot of this in my logs, any ideas how to fix it.
 
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
res_smdi.c:1335 in load_module: No SMDI interfaces are available to


listen
  

on, not starting SMDI listener.
Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
codec_dahdi.c:419 in find_transcoders: Failed to open /dev/zap/transcode:
No
such file or directory
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_sip.c:16731 in set_insecure_flags: insecure=very at line 170 is
deprecated; use insecure=port,invite instead
Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
chan_misdn.c:5053 in load_module: Unable to initialize mISDN
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 1: Red


Alarm
  

Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 2: Red


Alarm
 
 

--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation

Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

[John Novack]

You are most welcome.

What these lists are supposed to be all about.

If only the asterisk-users list were populated with members that are as 
nice as this group.


John Novack



Cleveland Electronic Services wrote:


Hi John,

 


Thanks for your reply on the following.

 

[In my experience, if FXO ports on a TDM400 aren't connected to a CO 
to see battery, they will give a RED alarm.]


 

Indeed you are perfectly correct, Once I plugged in 2 x CO, no red 
alarm in the logs.


 


Thanks for the pointer's

 


Cheers

 


*Cleve *

 

 


*From:* John Novack [mailto:jnov...@stromberg-carlson.org]
*Sent:* Friday, 6 March 2009 3:13 AM
*To:* AstLinux Users Mailing List
*Subject:* Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

 




Darrick Hartman wrote:

mailto:rkba...@clevelandelectronicservices.com>> 

 

Message-ID: <3a248f9d4dd820e398a51733736c4...@localhost>
X-Sender: dhart...@djhsolutions.com 
Received: from rrcs-74-87-125-110.west.biz.rr.com [74.87.125.110] with HTTP/1.1
(POST); Thu, 05 Mar 2009 09:49:36 -0600
User-Agent: RoundCube Webmail/0.1
Content-Type: text/plain; charset="UTF-8"
Content-Transfer-Encoding: 8bit
 
Cleve,
 
You can either ignore the warnings or disable the modules that are trying to load (that you're not using).  
 
If you look at /etc/asterisk/modules.conf, you'll see several modules listed.
 
In your case, you would want to noload res_smdi.so and chan_misdn.so since it doesn't look like you're using those.
 
  


I no load a whole host of unused modules.
Somewhere, many moons ago, I found a posting on one of the lists that 
had suggestions. If modules.conf is set to "autoload" then one can no 
load these, as well as skinny, mgcp, musiconhold if you aren't using 
it and a raft of others.

Anyone interested I can post my "minimal" load.
Also on some versions, I found that the application macro wasn't 
loading, and had to specify that to load. Never figured out why.



Not sure why you're seeing RED ALARM for dahdi (which is really zaptel, but 
Digium made the change to call it dahdi even if we're still using zaptel...)
 
  

In my experience, if FXO ports on a TDM400 aren't connected to a CO to 
see battery, they will give a RED alarm.


John Novack


Darrick
 
On Fri, 6 Mar 2009 01:47:36 +1100, "Cleveland Electronic Services"

 
 wrote:
  


Hi All.

 


I am seeing a lot of this in my logs, any ideas how to fix it.

 


Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:

res_smdi.c:1335 in load_module: No SMDI interfaces are available to




listen
  


on, not starting SMDI listener.

Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:

codec_dahdi.c:419 in find_transcoders: Failed to open /dev/zap/transcode:

No

such file or directory

Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:

chan_sip.c:16731 in set_insecure_flags: insecure=very at line 170 is

deprecated; use insecure=port,invite instead

Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:

chan_misdn.c:5053 in load_module: Unable to initialize mISDN

Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:

chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 1: Red




Alarm
  


Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:

chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 2: Red




Alarm
 
 
--

Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net 

https://lists.sourceforge.net/lists/listinfo/astlinux-users
 
Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org .
 
  




--
Dog is my co-pilot


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
--

Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

[Cleveland Electronic Services]

Hi John,

 

Thanks for your reply on the following.

 

[In my experience, if FXO ports on a TDM400 aren't connected to a CO to see
battery, they will give a RED alarm.]

 

Indeed you are perfectly correct, Once I plugged in 2 x CO, no red alarm in
the logs.

 

Thanks for the pointer's

 

Cheers

 

Cleve 

 

 

From: John Novack [mailto:jnov...@stromberg-carlson.org] 
Sent: Friday, 6 March 2009 3:13 AM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

 



Darrick Hartman wrote: 




Message-ID: <3a248f9d4dd820e398a51733736c4...@localhost>
X-Sender: dhart...@djhsolutions.com
Received: from rrcs-74-87-125-110.west.biz.rr.com [74.87.125.110] with
HTTP/1.1
(POST); Thu, 05 Mar 2009 09:49:36 -0600
User-Agent: RoundCube Webmail/0.1
Content-Type: text/plain; charset="UTF-8"
Content-Transfer-Encoding: 8bit
 
Cleve,
 
You can either ignore the warnings or disable the modules that are trying to
load (that you're not using).  
 
If you look at /etc/asterisk/modules.conf, you'll see several modules
listed.
 
In your case, you would want to noload res_smdi.so and chan_misdn.so since
it doesn't look like you're using those.
 
  

I no load a whole host of unused modules.
Somewhere, many moons ago, I found a posting on one of the lists that had
suggestions. If modules.conf is set to "autoload" then one can no load
these, as well as skinny, mgcp, musiconhold if you aren't using it and a
raft of others.
Anyone interested I can post my "minimal" load.
Also on some versions, I found that the application macro wasn't loading,
and had to specify that to load. Never figured out why.




Not sure why you're seeing RED ALARM for dahdi (which is really zaptel, but
Digium made the change to call it dahdi even if we're still using zaptel...)
 
  

In my experience, if FXO ports on a TDM400 aren't connected to a CO to see
battery, they will give a RED alarm.

John Novack




Darrick
 
On Fri, 6 Mar 2009 01:47:36 +1100, "Cleveland Electronic Services"
 
 wrote:
  

Hi All.
 
I am seeing a lot of this in my logs, any ideas how to fix it.
 
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
res_smdi.c:1335 in load_module: No SMDI interfaces are available to


listen
  

on, not starting SMDI listener.
Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
codec_dahdi.c:419 in find_transcoders: Failed to open /dev/zap/transcode:
No
such file or directory
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_sip.c:16731 in set_insecure_flags: insecure=very at line 170 is
deprecated; use insecure=port,invite instead
Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
chan_misdn.c:5053 in load_module: Unable to initialize mISDN
Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 1: Red


Alarm
  

Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 2: Red


Alarm
 
 

--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
 
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
 
  





-- 
Dog is my co-pilot
--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Can't hear phone ring in my cell headset when calling my Astlinux box

[Tod Fitch]

On Mar 6, 2009, at 3:53 PM, Cleveland Electronic Services wrote:


[If it were a bug in 0.6.3, I would think that other people would be
having problems too.  I would look at the Asterisk configuration  
files.

 Asterisk has changed between 1.4.21 and 1.4.23.1.]

Darrick is perfectly correct, It is not an issue with Astlinux but  
in the
different versions of Asterisk, I have a fair few Asterisk installs  
running

1.4.21.1 some of the settings/configs that work well, will not work on
1.4.23.1 unfortunately you have to make some changes. I am sure this  
is
Asterisk's way of building up or you could say preparing for  
migration to
Asterisk 1.6, we went through a similar phase with Asterisk 1.2 to  
1.4.


Cheers

Cleve


Does anyone have a link to a migration guide for these Asterisk  
configuration changes?


--Tod

smime.p7s
Description: S/MIME cryptographic signature
--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Can't hear phone ring in my cell headset when calling my Astlinux box

[Cleveland Electronic Services]

[If it were a bug in 0.6.3, I would think that other people would be 
having problems too.  I would look at the Asterisk configuration files. 
  Asterisk has changed between 1.4.21 and 1.4.23.1.]

Darrick is perfectly correct, It is not an issue with Astlinux but in the
different versions of Asterisk, I have a fair few Asterisk installs running
1.4.21.1 some of the settings/configs that work well, will not work on
1.4.23.1 unfortunately you have to make some changes. I am sure this is
Asterisk's way of building up or you could say preparing for migration to
Asterisk 1.6, we went through a similar phase with Asterisk 1.2 to 1.4.

Cheers  

Cleve 


-Original Message-
From: Darrick Hartman [mailto:dhart...@djhsolutions.com] 
Sent: Saturday, 7 March 2009 4:36 AM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Can't hear phone ring in my cell headset when
calling my Astlinux box

If it were a bug in 0.6.3, I would think that other people would be 
having problems too.  I would look at the Asterisk configuration files. 
  Asterisk has changed between 1.4.21 and 1.4.23.1.

If you want to send me your config files, I can put them on a test box 
and see if I can duplicate the problem.

Darrick

Ionel Chila wrote:
> Interesting. Downgraded back to 0.6.2 fresh install and everything works
fine. Is gotta be some kind of bug in 0.6.3
> 
> 
> 
> 
> 
> - Original Message 
> From: Darrick Hartman 
> To: AstLinux Users Mailing List 
> Sent: Thursday, March 5, 2009 2:57:18 PM
> Subject: Re: [Astlinux-users] Can't hear phone ring in my cell headset
when calling my Astlinux box
> 
> Ionel Chila wrote:
>> So just upgraded my Soekris 4801 box to the latest version of
>> Astlinux 0.6.3 and it looks like it broke some functions.
> 
> The changes that were made between Astlinux 0.6.2 and 0.6.3 should not
> have any negative affect on a working Asterisk 1.4.x configuration.  If
> you're upgrading from an Asterisk 1.2.x version, then several things may
> need to change.
> 
>> If I call my home phone running on this Soekris Astlinux box, I can't
>> hear the ringing in my cell phone. It just makes a click and then
>> after 4 rings my answering machine will pick up.
> 
> Hold the phone!  Why would you have your answering machine and the 
> Astlinux box on the same line?  That may be confusing things.  Unplug 
> the answering machine and try again.  Also what Zaptel hardware are you 
> using?  You're really leaving out some key pieces of information.
> 
>> Phisicaly the phone in my house does ring but I just can't hear the
>> ring in my headset when calling
>>
>> I did test this with the previous Astlinux and works fine and I did
>> test it with ISO (astlinux-0.6.3-geni586.iso) image running on a
>> laptop using the same very configuration files from the USB
>> tocken I even re-installed Astlinux from scratch
> 
> I would look closely at the Asterisk configuration files on the net4801 
> device.
> 
> Darrick
> 
>

--
> Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco,
CA
> -OSBC tackles the biggest issue in open source: Open Sourcing the
Enterprise
> -Strategies to boost innovation and cut costs with open source
participation
> -Receive a $600 discount off the registration fee with the source code:
SFAD
> http://p.sf.net/sfu/XcvMzF8H
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
> 
> 
>

--
> Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco,
CA
> -OSBC tackles the biggest issue in open source: Open Sourcing the
Enterprise
> -Strategies to boost innovation and cut costs with open source
participation
> -Receive a $600 discount off the registration fee with the source code:
SFAD
> http://p.sf.net/sfu/XcvMzF8H
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.



--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support 

Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's

[Cleveland Electronic Services]

Hi Darrick,

Thankyou for the information below, I have now made the changes and all is
working very well. Still have to try the fix for the red alarm.

Cheers

Cleve 

-Original Message-
From: Darrick Hartman [mailto:dhart...@djhsolutions.com] 
Sent: Friday, 6 March 2009 2:50 AM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Astlinux 0.6.3 Warning & Error's



Message-ID: <3a248f9d4dd820e398a51733736c4...@localhost>
X-Sender: dhart...@djhsolutions.com
Received: from rrcs-74-87-125-110.west.biz.rr.com [74.87.125.110] with
HTTP/1.1
(POST); Thu, 05 Mar 2009 09:49:36 -0600
User-Agent: RoundCube Webmail/0.1
Content-Type: text/plain; charset="UTF-8"
Content-Transfer-Encoding: 8bit

Cleve,

You can either ignore the warnings or disable the modules that are trying
to load (that you're not using).  

If you look at /etc/asterisk/modules.conf, you'll see several modules
listed.

In your case, you would want to noload res_smdi.so and chan_misdn.so since
it doesn't look like you're using those.

Also change your sip.conf settings to use the new format for 'insecure'. 
The warning below tells you exactly what to put in.

Not sure why you're seeing RED ALARM for dahdi (which is really zaptel, but
Digium made the change to call it dahdi even if we're still using
zaptel...)

Darrick

On Fri, 6 Mar 2009 01:47:36 +1100, "Cleveland Electronic Services"
 wrote:
> Hi All.
> 
> I am seeing a lot of this in my logs, any ideas how to fix it.
> 
> Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
> res_smdi.c:1335 in load_module: No SMDI interfaces are available to
listen
> on, not starting SMDI listener.
> Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
> codec_dahdi.c:419 in find_transcoders: Failed to open /dev/zap/transcode:
> No
> such file or directory
> Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
> chan_sip.c:16731 in set_insecure_flags: insecure=very at line 170 is
> deprecated; use insecure=port,invite instead
> Mar  6 00:39:18 pbx local0.err asterisk[1462]: ERROR[1462]:
> chan_misdn.c:5053 in load_module: Unable to initialize mISDN
> Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
> chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 1: Red
Alarm
> Mar  6 00:39:18 pbx local0.warn asterisk[1462]: WARNING[1462]:
> chan_dahdi.c:3787 in handle_alarms: Detected alarm on channel 2: Red
Alarm



--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Can't hear phone ring in my cell headset when calling my Astlinux box

[Darrick Hartman]

If it were a bug in 0.6.3, I would think that other people would be 
having problems too.  I would look at the Asterisk configuration files. 
  Asterisk has changed between 1.4.21 and 1.4.23.1.

If you want to send me your config files, I can put them on a test box 
and see if I can duplicate the problem.

Darrick

Ionel Chila wrote:
> Interesting. Downgraded back to 0.6.2 fresh install and everything works 
> fine. Is gotta be some kind of bug in 0.6.3
> 
> 
> 
> 
> 
> - Original Message 
> From: Darrick Hartman 
> To: AstLinux Users Mailing List 
> Sent: Thursday, March 5, 2009 2:57:18 PM
> Subject: Re: [Astlinux-users] Can't hear phone ring in my cell headset when 
> calling my Astlinux box
> 
> Ionel Chila wrote:
>> So just upgraded my Soekris 4801 box to the latest version of
>> Astlinux 0.6.3 and it looks like it broke some functions.
> 
> The changes that were made between Astlinux 0.6.2 and 0.6.3 should not
> have any negative affect on a working Asterisk 1.4.x configuration.  If
> you're upgrading from an Asterisk 1.2.x version, then several things may
> need to change.
> 
>> If I call my home phone running on this Soekris Astlinux box, I can't
>> hear the ringing in my cell phone. It just makes a click and then
>> after 4 rings my answering machine will pick up.
> 
> Hold the phone!  Why would you have your answering machine and the 
> Astlinux box on the same line?  That may be confusing things.  Unplug 
> the answering machine and try again.  Also what Zaptel hardware are you 
> using?  You're really leaving out some key pieces of information.
> 
>> Phisicaly the phone in my house does ring but I just can't hear the
>> ring in my headset when calling
>>
>> I did test this with the previous Astlinux and works fine and I did
>> test it with ISO (astlinux-0.6.3-geni586.iso) image running on a
>> laptop using the same very configuration files from the USB
>> tocken I even re-installed Astlinux from scratch
> 
> I would look closely at the Asterisk configuration files on the net4801 
> device.
> 
> Darrick
> 
> --
> Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
> -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
> -Strategies to boost innovation and cut costs with open source participation
> -Receive a $600 discount off the registration fee with the source code: SFAD
> http://p.sf.net/sfu/XcvMzF8H
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.
> 
> 
> --
> Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
> -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
> -Strategies to boost innovation and cut costs with open source participation
> -Receive a $600 discount off the registration fee with the source code: SFAD
> http://p.sf.net/sfu/XcvMzF8H
> ___
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org.


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] Can't hear phone ring in my cell headset when calling my Astlinux box

[Ionel Chila]

Interesting. Downgraded back to 0.6.2 fresh install and everything works fine. 
Is gotta be some kind of bug in 0.6.3





- Original Message 
From: Darrick Hartman 
To: AstLinux Users Mailing List 
Sent: Thursday, March 5, 2009 2:57:18 PM
Subject: Re: [Astlinux-users] Can't hear phone ring in my cell headset when 
calling my Astlinux box

Ionel Chila wrote:
> So just upgraded my Soekris 4801 box to the latest version of
> Astlinux 0.6.3 and it looks like it broke some functions.

The changes that were made between Astlinux 0.6.2 and 0.6.3 should not
have any negative affect on a working Asterisk 1.4.x configuration.  If
you're upgrading from an Asterisk 1.2.x version, then several things may
need to change.

> If I call my home phone running on this Soekris Astlinux box, I can't
> hear the ringing in my cell phone. It just makes a click and then
> after 4 rings my answering machine will pick up.

Hold the phone!  Why would you have your answering machine and the 
Astlinux box on the same line?  That may be confusing things.  Unplug 
the answering machine and try again.  Also what Zaptel hardware are you 
using?  You're really leaving out some key pieces of information.

> Phisicaly the phone in my house does ring but I just can't hear the
> ring in my headset when calling
> 
> I did test this with the previous Astlinux and works fine and I did
> test it with ISO (astlinux-0.6.3-geni586.iso) image running on a
> laptop using the same very configuration files from the USB
> tocken I even re-installed Astlinux from scratch

I would look closely at the Asterisk configuration files on the net4801 
device.

Darrick

--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.


--
Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
___
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Re: [Astlinux-users] callback with 0.6.3

[noel . 4sip]

Hi every body,

to David: my 'top monitoring' seems normal(!) with only a lot of asterisk
processes taking near of 0% of cpu.

I'm just wondering why a 'simple' working extension.conf configuration
working with Asterisk 1.4.21.2 in astlinux version 0.6.2 isn't working
anymore after the upgrade to Asterisk 1.4.23.1
I don't see any relevant warnings in the change logs.
I don't have any error messages in the CLI with verbosity and debug set to
10.
The only thing is that my system freezes after displaying the congestion
as last command with the only power off/on hardware reset solution.

[code]
-- Executing [...@phonext-in:1] Answer("SIP/voxbone.com-081c9100", "")
in new stack
-- Executing [...@phonext-in:2] NoOp("SIP/voxbone.com-081c9100", "appel
entrant 32 pour callback dans phonext-in") in new stack
-- Executing [...@phonext-in:3] Goto("SIP/voxbone.com-081c9100",
"moncontexte-disa||1") in new stack
-- Goto (moncontexte-disa,,1)
-- Executing [9...@moncontexte-disa:1]
NoOp("SIP/voxbone.com-081c9100", "appel entrant 32
moncontexte-disa ") in new stack
-- Executing [9...@moncontexte-disa:2]
PlayTones("SIP/voxbone.com-081c9100", "congestion") in new stack
-- Executing [9...@moncontexte-disa:3]
Congestion("SIP/voxbone.com-081c9100", "5") in new stack
[/code]

here is the corresponding code of my extension.conf (sorry for
comments/noop messages in french)

[code]

[globals]
CALLBACKEXT=32

[general]
autofallthrough=yes


[phonext-in]
;en provenance de l'extension autorisée ${CALLBACKEXT}?
exten => s/${CALLBACKEXT},1,Answer()
exten => s/${CALLBACKEXT},n,NoOp(appel entrant ${CALLERID(num)} pour
callback dans phonext-in)
exten => s/${CALLBACKEXT},n,Goto(moncontexte-disa,,1)
;sinon appel générique
exten => s,1,Answer()
exten => s,n,NoOp(appel entrant ${CALLERID(num)} generic phonext-in)
exten => s,n,Goto(moncontexte-generic,1234,1)


[moncontexte-generic]
exten => 1234,1,NoOp(appel entrant ${CALLERID(num)} moncontexte 1234)
exten => 1234,n,Set(CHANNEL(language)=fr)
exten => 1234,n,Answer()
exten => 1234,n,Wait(1)
exten => 1234,n,Playback(hello-world)
exten => 1234,n,Playtones(congestion)
exten => 1234,n,Hangup()


[moncontexte-disa]
exten => ,1,NoOp(appel entrant ${CALLERID(num)} moncontexte-disa )
;exten => ,n,Set(CHANNEL(language)=fr)
exten => ,n,Playtones(congestion)
exten => ,n,Congestion()
exten => ,n,Hangup(5)
;exten => ,n,Hangup()
;passage de l'appel au canal h
exten => h,1,NoOp(appel entrant ${CALLERID(num)} vers canal h)
;ajouter le 00 dans le numéro de l'appelant
exten => h,n,Set(CALLERID(num) = 00${CALLERID(num)})
exten => h,n,NoOp(ajout du 00)

 rest of the file

[/code]

and the corresponding reload debug message are

[code]

  == Parsing '/etc/asterisk/extensions.conf': Found
  == Setting global variable 'CALLBACKEXT' to '32'
-- Registered extension context 'phonext-in'
-- Added extension 's' priority 1 (CID match '32')to phonext-in
  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2727
  == Using TOS bits 0
-- Added extension 's' priority 2 (CID match '32')to phonext-in
-- Added extension 's' priority 3 (CID match '32')to phonext-in
-- Added extension 's' priority 1 to phonext-in
-- Added extension 's' priority 2 to phonext-in
-- Added extension 's' priority 3 to phonext-in
-- Registered extension context 'moncontexte-generic'
-- Added extension '1234' priority 1 to moncontexte-generic
-- Added extension '1234' priority 2 to moncontexte-generic
-- Added extension '1234' priority 3 to moncontexte-generic
-- Added extension '1234' priority 4 to moncontexte-generic
-- Added extension '1234' priority 5 to moncontexte-generic
-- Added extension '1234' priority 6 to moncontexte-generic
-- Added extension '1234' priority 7 to moncontexte-generic
-- Registered extension context 'moncontexte-disa'
-- Added extension '' priority 1 to moncontexte-disa
-- Added extension '' priority 2 to moncontexte-disa
-- Added extension '' priority 3 to moncontexte-disa
-- Added extension '' priority 4 to moncontexte-disa
-- Added extension 'h' priority 1 to moncontexte-disa
-- Added extension 'h' priority 2 to moncontexte-disa
-- Added extension 'h' priority 3 to moncontexte-disa
-- Added extension 'h' priority 4 to moncontexte-disa
-- Added extension 'h' priority 5 to moncontexte-disa

... rest of the messages

[/code]

the only difference I see in the debug message are those

  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2727
  == Using TOS bits 0

lines present only in the 0.6.3 reload messages, just after
Added extension 's' priority 1
and before processing the rest of my extension.conf file.

Could that be a path to the understanding of this situation?

(btw I'm still working with the previous working version)