[asterisk-users] Error compiling Asterisk on Centos 8

2021-10-18 Thread cio-alves

I am compiling asterisk11 (eleven) on Centos 8 (eight)
with
./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64 
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-asteriskssl 
-enable-xmldoc NOISY_BUILD=no


Since I did the configure with "--disable-asteriskssl", why is even 
trying to compile libasteriskssl?


make[1]: Entering directory '/usr/src/asterisk/main'
gcc -o libasteriskssl.o -c libasteriskssl.c -MD -MT libasteriskssl.o -MF 
.libasteriskssl.o.d -MP -pthread -I/usr/src/asterisk/include   
-I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes 
-Wmissing-prototypes -Wmissing-declarations   -g3  -O3  
-U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -march=native 
-DAST_MODULE=\"core\" -DAST_IN_CORE
libasteriskssl.c:77:26: error: macro "SSL_library_init" passed 1 
arguments, but takes just 0

 int SSL_library_init(void)
  ^
libasteriskssl.c:78:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or 
‘__attribute__’ before ‘{’ token

 {
 ^
libasteriskssl.c:87:33: error: macro "SSL_load_error_strings" passed 1 
arguments, but takes just 0

 void SSL_load_error_strings(void)
 ^
libasteriskssl.c:88:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or 
‘__attribute__’ before ‘{’ token

 {
 ^
libasteriskssl.c:97:1: error: expected identifier or ‘(’ before ‘{’ 
token

 {
 ^
libasteriskssl.c:106:1: error: expected identifier or ‘(’ before ‘{’ 
token

 {
 ^

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Re: [asterisk-users] error compiling current git

2020-02-28 Thread hw
On Thursday, February 27, 2020 4:29:01 PM CET Kevin Harwell wrote:
> On Thu, Feb 27, 2020 at 8:51 AM hw  wrote:
> > Hi,
> > 
> > compiling the current git version on Centos 7 gives me:
> >[CC] res_statsd.c -> res_statsd.o
> > 
> > res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified
> > in initializer
> > 
> >   .on_valid_pair = ast_rtp_on_valid_pair,
> >   ^
> > 
> > res_rtp_asterisk.c:2669:2: warning: initialization from incompatible
> > pointer type [enabled by default]
> > res_rtp_asterisk.c:2669:2: warning: (near initialization for
> > ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default]
> > 
> >[CC] res_format_attr_g729.c -> res_format_attr_g729.o
> > 
> > Is this to be expected or should I make a bug report?
> 
> When you pulled the lasted code this change would have forced a
> re-configure. If you haven't already try doing a full clean and rebuild,
> and see if you still have the error:
> 
> $ make distclean
> $ ./configure [your options]
> $ make

Thanks, that worked :)





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Re: [asterisk-users] error compiling current git

2020-02-27 Thread Kevin Harwell
On Thu, Feb 27, 2020 at 8:51 AM hw  wrote:

> Hi,
>
> compiling the current git version on Centos 7 gives me:
>
>
>[CC] res_statsd.c -> res_statsd.o
> res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified
> in initializer
>   .on_valid_pair = ast_rtp_on_valid_pair,
>   ^
> res_rtp_asterisk.c:2669:2: warning: initialization from incompatible
> pointer type [enabled by default]
> res_rtp_asterisk.c:2669:2: warning: (near initialization for
> ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default]
>[CC] res_format_attr_g729.c -> res_format_attr_g729.o
>
>
> Is this to be expected or should I make a bug report?
>
>
When you pulled the lasted code this change would have forced a
re-configure. If you haven't already try doing a full clean and rebuild,
and see if you still have the error:

$ make distclean
$ ./configure [your options]
$ make

-- 
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Software Developer
Sangoma Technologies
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Re: [asterisk-users] error compiling current git

2020-02-27 Thread Joshua C. Colp
On Thu, Feb 27, 2020 at 10:51 AM hw  wrote:

> Hi,
>
> compiling the current git version on Centos 7 gives me:
>
>
>[CC] res_statsd.c -> res_statsd.o
> res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified
> in initializer
>   .on_valid_pair = ast_rtp_on_valid_pair,
>   ^
> res_rtp_asterisk.c:2669:2: warning: initialization from incompatible
> pointer type [enabled by default]
> res_rtp_asterisk.c:2669:2: warning: (near initialization for
> ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default]
>[CC] res_format_attr_g729.c -> res_format_attr_g729.o
>
>
> Is this to be expected or should I make a bug report?
>

You should make a bug report[1], including which version of Asterisk (every
version is in git) and whether you are used bundled or unbundled PJSIP.

[1] https://issues.asterisk.org/jira

-- 
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Sangoma Technologies
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[asterisk-users] error compiling current git

2020-02-27 Thread hw
Hi,

compiling the current git version on Centos 7 gives me:


   [CC] res_statsd.c -> res_statsd.o
res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in 
initializer
  .on_valid_pair = ast_rtp_on_valid_pair,
  ^
res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer 
type [enabled by default]
res_rtp_asterisk.c:2669:2: warning: (near initialization for 
‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default]
   [CC] res_format_attr_g729.c -> res_format_attr_g729.o


Is this to be expected or should I make a bug report?




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Re: [asterisk-users] Error compile dahdi 3 on CentOS 7.7

2019-10-03 Thread Shaun Ruffell
On Thu, Oct 03, 2019 at 09:53:40AM -0400, Jerry Geis wrote:
> >although it looks like the dahdi-linux-complete tarball has not been
> >updated since the final 3.1.0 was released)
> 
> This works - when will the tar be updated ?

Unfortunately, I do not have any specific insight that would allow me to
answer this question.

Cheers,
Shaun

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Re: [asterisk-users] Error compile dahdi 3 on CentOS 7.7

2019-10-03 Thread Jerry Geis
>although it looks like the dahdi-linux-complete tarball has not been
>updated since the final 3.1.0 was released)

This works - when will the tar be updated ?

Thanks,


Jerry
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Re: [asterisk-users] Error compile dahdi 3 on CentOS 7.7

2019-10-03 Thread Shaun Ruffell
On Thu, Oct 03, 2019 at 08:59:29AM -0400, Jerry Geis wrote:
> What should I do about the below error?  Compiling from source. CentOS 7.7
> is fully updated.
> 
> Thanks
> 
> Jerry
> 
> In file included from
> digium/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/dahdi-base.c:68:0:
> digium/dahdi-linux-complete-3.0.0+3.0.0/linux/include/dahdi/kernel.h:1422:1:
> error: conflicting types for ‘timer_setup’
>  timer_setup(struct timer_list *timer,
>  ^
> In file included from include/linux/workqueue.h:8:0,
>  from include/linux/srcu.h:34,
>  from include/linux/notifier.h:15,
>  from include/linux/memory_hotplug.h:6,
>  from include/linux/mmzone.h:881,
>  from include/linux/gfp.h:5,
>  from include/linux/kmod.h:22,
>  from include/linux/module.h:13,
>  from
> digium/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/dahdi-base.c:40:
> include/linux/timer.h:164:20: note: previous definition of ‘timer_setup’
> was here
>  static inline void timer_setup(struct timer_list *timer,

You should be able to install
dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz [1] which contains a
commit [2] that handles the fact that RHEL backported the new
timer_setup interface.

[1] 
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz
[2] 
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=8c8b9b6df0d5757cad25b40efeb15ba1a7c95bdf

(although it looks like the dahdi-linux-complete tarball has not been
updated since the final 3.1.0 was released)

Cheers,
Shaun

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[asterisk-users] Error compile dahdi 3 on CentOS 7.7

2019-10-03 Thread Jerry Geis
What should I do about the below error?  Compiling from source. CentOS 7.7
is fully updated.

Thanks

Jerry

In file included from
digium/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/dahdi-base.c:68:0:
digium/dahdi-linux-complete-3.0.0+3.0.0/linux/include/dahdi/kernel.h:1422:1:
error: conflicting types for ‘timer_setup’
 timer_setup(struct timer_list *timer,
 ^
In file included from include/linux/workqueue.h:8:0,
 from include/linux/srcu.h:34,
 from include/linux/notifier.h:15,
 from include/linux/memory_hotplug.h:6,
 from include/linux/mmzone.h:881,
 from include/linux/gfp.h:5,
 from include/linux/kmod.h:22,
 from include/linux/module.h:13,
 from
digium/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/dahdi-base.c:40:
include/linux/timer.h:164:20: note: previous definition of ‘timer_setup’
was here
 static inline void timer_setup(struct timer_list *timer,
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Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread John Novack

Too bad.

LOTS of users will still want to continue to use these cards, example the OP!


Good news it probably suppresses prices on used cards!


John Novack



Malcolm Davenport wrote:

Howdy,

That is correct.

The list of supported cards is in the README file (not the -complete package 
README, but the dahdi-linux README)

Cheers

On Thu, Jun 6, 2019 at 2:52 PM John Novack SCII_U mailto:jnov...@comcast.net>> wrote:

Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the 
TDM400 and 410?


John Novack



Greg Woods wrote:



On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:

Howdy,

There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.

Try that.


I noticed that was there, but I didn't try it originally because it's 
obviously a beta version. However, I did download it and try it. It does 
compile, but doesn't work correctly. For one thing, it thinks my Digium card is 
an Ethernet controller:

# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog 
card (rev 11)

Attempting to start the dahdi service results in:

Short version:Jun 06 13:11:38 worldsys.gregandeva.net 
 sh[1026]: using 
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: Selected signaling not supported
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         FXO signaling is being used on a FXO interface (use a FXS signaling 
variant)
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         RBS signaling is being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         Signaling is being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net  
systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net  
systemd[1]: dahdi.service: Failed with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net  
systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to 
accept incoming data and voice interfaces.

(The assigned-spans.conf file has nothing in it but comments)

Long version:
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: 
Unable to register channel '1'
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: 
Channel '1' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: 
Unable to register channel '2'
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: 
Channel '2' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: 
Unable to register channel '3'
Jun 06 13:11:42 

Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread Malcolm Davenport
Howdy,

That is correct.

The list of supported cards is in the README file (not the -complete
package README, but the dahdi-linux README)

Cheers

On Thu, Jun 6, 2019 at 2:52 PM John Novack SCII_U 
wrote:

> Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the
> TDM400 and 410?
>
>
> John Novack
>
>
>
> Greg Woods wrote:
>
>
>
> On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport 
> wrote:
>
>> Howdy,
>>
>> There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.
>>
>> Try that.
>>
>
> I noticed that was there, but I didn't try it originally because it's
> obviously a beta version. However, I did download it and try it. It does
> compile, but doesn't work correctly. For one thing, it thinks my Digium
> card is an Ethernet controller:
>
> # lspci | grep Digium
> 07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog
> card (rev 11)
>
> Attempting to start the dahdi service results in:
>
> Short version:Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: using
> '/etc/dahdi/assigned-spans.conf'
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: DAHDI_CHANCONFIG failed
> on channel 1: Invalid argument (22)
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Selected signaling not
> supported
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Possible causes:
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: FXO signaling
> is being used on a FXO interface (use a FXS signaling variant)
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: RBS signaling
> is being used on a E1 CCS span
> Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Signaling is
> being assigned to channel 16 of an E1 CAS span
> Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: dahdi.service: Main
> process exited, code=exited, status=1/FAILURE
> Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: dahdi.service: Failed
> with result 'exit-code'.
> Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: Failed to start The
> DAHDI drivers allow you to use your linux computer to accept incoming data
> and voice interfaces.
>
> (The assigned-spans.conf file has nothing in it but comments)
>
> Long version:
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 1:
> Invalid argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 1: Invalid
> argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
> '1'
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '1' failure
> ignored: ignore_failed_channels.
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 2:
> Invalid argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 2: Invalid
> argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
> '2'
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '2' failure
> ignored: ignore_failed_channels.
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 3:
> Invalid argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 3: Invalid
> argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
> '3'
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '3' failure
> ignored: ignore_failed_channels.
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 4:
> Invalid argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 4: Invalid
> argument
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
> '4'
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '4' failure
> ignored: ignore_failed_channels.
> Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
> WARNING[1333]: chan_dahdi.c:18883 process_dahdi: 

Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread Greg Woods
On Thu, Jun 6, 2019 at 1:51 PM John Novack SCII_U 
wrote:

> Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the
> TDM400 and 410?
>

If it does, they have a funny way of implementing it, since the proper
modules are still being loaded.

This is a $600+ card, which is about 2/3 of the price of an entire PC built
from parts, and I really don't want to have to buy a new telephony card. If
this is true, then I'll have to go back to finding a way to make 2.11.1
work.

--Greg
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Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread John Novack SCII_U

Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the 
TDM400 and 410?


John Novack



Greg Woods wrote:



On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:

Howdy,

There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.

Try that.


I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work 
correctly. For one thing, it thinks my Digium card is an Ethernet controller:


# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card 
(rev 11)

Attempting to start the dahdi service results in:

Short version:Jun 06 13:11:38 worldsys.gregandeva.net 
 sh[1026]: using 
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: Selected signaling not supported
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         FXO signaling is being used on a FXO interface (use a FXS signaling 
variant)
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         RBS signaling is being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net  
sh[1026]:         Signaling is being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net  
systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net  
systemd[1]: dahdi.service: Failed with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net  systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to accept incoming 
data and voice interfaces.


(The assigned-spans.conf file has nothing in it but comments)

Long version:
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 1: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '1'
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '1' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 2: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '2'
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '2' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 3: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '3'
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '3' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
4: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net  asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 4: 
Invalid argument

Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread Greg Woods
On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport 
wrote:

> Howdy,
>
> There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.
>
> Try that.
>

I noticed that was there, but I didn't try it originally because it's
obviously a beta version. However, I did download it and try it. It does
compile, but doesn't work correctly. For one thing, it thinks my Digium
card is an Ethernet controller:

# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog
card (rev 11)

Attempting to start the dahdi service results in:

Short version:Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: using
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: DAHDI_CHANCONFIG failed
on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Selected signaling not
supported
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: FXO signaling is
being used on a FXO interface (use a FXS signaling variant)
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: RBS signaling is
being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net sh[1026]: Signaling is
being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: dahdi.service: Main
process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: dahdi.service: Failed
with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net systemd[1]: Failed to start The
DAHDI drivers allow you to use your linux computer to accept incoming data
and voice interfaces.

(The assigned-spans.conf file has nothing in it but comments)

Long version:
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 1:
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 1: Invalid
argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
'1'
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '1' failure
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 2:
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 2: Invalid
argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
'2'
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '2' failure
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 3:
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 3: Invalid
argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
'3'
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '3' failure
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 4:
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 4: Invalid
argument
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel
'4'
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '4' failure
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:18883 process_dahdi: Ignoring any changes to
'userbase' (on reload) at line 23.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:18883 process_dahdi: Ignoring any changes to
'vmsecret' (on reload) at line 31.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:18883 process_dahdi: Ignoring any changes to
'hassip' (on reload) at line 35.
Jun 06 13:11:42 worldsys.gregandeva.net asterisk[1333]: [Jun  6 13:11:42]
WARNING[1333]: chan_dahdi.c:18883 

Re: [asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread Malcolm Davenport
Howdy,

There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.

Try that.

On Thu, Jun 6, 2019 at 1:11 PM Greg Woods  wrote:

> Seems like I post about this about once a year, when it's time to upgrade
> Fedora.
>
> I first got this error trying to compile a patched version of
> dahdi-linux-2.11.1; I noticed that there is now a
> dahdi-linux-complete-3.0.0+3.0.0, so I tried that one with the same result.
>
> If I compile it while running kernel-4.16.8-300.fc28.x86_64, it compiles
> fine, but when I try to compile it while
> running kernel-5.0.16-100.fc28.x86_64, I get this error:
>
> CC [M]
>  
> /local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.o
> /local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.c:
> In function ‘Oct6100UserGetTime’:
> /local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.c:38:2:
> error: implicit declaration of function ‘do_gettimeofday’; did you mean
> ‘do_settimeofday64’? [-Werror=implicit-function-declaration]
>   do_gettimeofday();
>   ^~~
>   do_settimeofday64
> cc1: some warnings being treated as errors
> make[4]: *** [scripts/Makefile.build:277:
> /local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.o]
> Error 1
> make[3]: *** [scripts/Makefile.build:492:
> /local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x]
> Error 2
> make[2]: *** [Makefile:1581:
> _module_/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi]
> Error 2
> make[2]: Leaving directory '/usr/src/kernels/5.0.16-100.fc28.x86_64'
> make[1]: *** [Makefile:74: modules] Error 2
> make[1]: Leaving directory
> '/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux'
> make: *** [Makefile:9: all] Error 2
>
> At first I thought this might be due to using a newer version of gcc that
> was being stricter about turning warnings into errors, but it compiles fine
> with the older kernel. So I tried some stupid stuff like calling
> do_gettimeofday64() instead; same error. I also tried the suggestion of
> using settimeofday64() instead; different error (argument type mismatch,
> but even if it had compiled, I wouldn't have expected that code to actually
> work).  Also tried explicitly declaring "void do_gettimeofday()" (since the
> return value isn't being used); different error.
>
> I expect this isn't a hard thing to fix, but it has been many years since
> I've done any C programming and I am quite rusty.
>
> Anyone else run into this, and have a fix?
>
> Thanks,
> --Greg
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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Digium - a Sangoma company | Senior Product Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Tel/Fax: +1 256 428 6252
malco...@sangoma.com
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[asterisk-users] error compiling dahdi for recent kernels

2019-06-06 Thread Greg Woods
Seems like I post about this about once a year, when it's time to upgrade
Fedora.

I first got this error trying to compile a patched version of
dahdi-linux-2.11.1; I noticed that there is now a
dahdi-linux-complete-3.0.0+3.0.0, so I tried that one with the same result.

If I compile it while running kernel-4.16.8-300.fc28.x86_64, it compiles
fine, but when I try to compile it while
running kernel-5.0.16-100.fc28.x86_64, I get this error:

CC [M]
 
/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.o
/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.c:
In function ‘Oct6100UserGetTime’:
/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.c:38:2:
error: implicit declaration of function ‘do_gettimeofday’; did you mean
‘do_settimeofday64’? [-Werror=implicit-function-declaration]
  do_gettimeofday();
  ^~~
  do_settimeofday64
cc1: some warnings being treated as errors
make[4]: *** [scripts/Makefile.build:277:
/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x/oct612x-user.o]
Error 1
make[3]: *** [scripts/Makefile.build:492:
/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/oct612x]
Error 2
make[2]: *** [Makefile:1581:
_module_/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi]
Error 2
make[2]: Leaving directory '/usr/src/kernels/5.0.16-100.fc28.x86_64'
make[1]: *** [Makefile:74: modules] Error 2
make[1]: Leaving directory
'/local/src/dahdi/dahdi-linux-complete-3.0.0+3.0.0/linux'
make: *** [Makefile:9: all] Error 2

At first I thought this might be due to using a newer version of gcc that
was being stricter about turning warnings into errors, but it compiles fine
with the older kernel. So I tried some stupid stuff like calling
do_gettimeofday64() instead; same error. I also tried the suggestion of
using settimeofday64() instead; different error (argument type mismatch,
but even if it had compiled, I wouldn't have expected that code to actually
work).  Also tried explicitly declaring "void do_gettimeofday()" (since the
return value isn't being used); different error.

I expect this isn't a hard thing to fix, but it has been many years since
I've done any C programming and I am quite rusty.

Anyone else run into this, and have a fix?

Thanks,
--Greg
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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-05 Thread Joseph Smith
Thank you for the response Mike,

I did run into a CDR bottleneck as well and have already disabled it,

> module show like cdr
Module Description  Use 
Count  Status  Support Level
0 modules loaded

# grep enable= /etc/asterisk/cdr.conf
enable=no

At this point I'm really just not sure what the current bottleneck is and how 
to prevent the tasks for pooling.  I expected that the CPU would cap out before 
this occurred.  I do feel like there must be something I'm missing but just 
can't to it.

Any further suggestions are very welcome.

Thanks
Joseph

From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Mike 
<mich...@virtutel.ca>
Sent: Friday, September 1, 2017 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

I had that problem before – I believe “task processor queue reached 500 
scheduled tasks” crashing means your CDR records (queue) are being written as 
the call ends, and if you had many thousands of entries being written to disk 
it crashes asterisk (each ring to one phone is an entry, so it goes up fast – 
for example 10 busy phones, with a between-ring delay of 1 second means every 
second there are 10 entries being put in memory)

I was using a MySQL CDR, but I had left the “CSV” type of CDR on. I 
removed/disabled the CSV CDR module, kept on the SQL CDR only and things have 
been working fine ever since.

Mike

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan


Thanks for the suggestion Tony,


I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(


From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield 
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.outlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
--
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Work: t...@softins.co.uk - http://www.softins.co.uk
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Mike
I had that problem before - I believe "task processor queue reached 500
scheduled tasks" crashing means your CDR records (queue) are being written
as the call ends, and if you had many thousands of entries being written
to disk it crashes asterisk (each ring to one phone is an entry, so it
goes up fast - for example 10 busy phones, with a between-ring delay of 1
second means every second there are 10 entries being put in memory)

 

I was using a MySQL CDR, but I had left the "CSV" type of CDR on. I
removed/disabled the CSV CDR module, kept on the SQL CDR only and things
have been working fine ever since.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

 

Thanks for the suggestion Tony,

 

I installed each codec for MoH, core sounds, and extra sound packages.
Unfortunately the tests produce the same results. 

 

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0
(

 

continuously for a while followed by a 

 

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue
reached 500 scheduled tasks.

 

Then this time Asterisk actually crashed. :(

 

  _  

From: asterisk-users-boun...@lists.digium.com
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan 

 

In article
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.ou
tlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
> 
> Thanks for the feedback.
> 
> I do agree with having multiple smaller servers.  When I was first
approached with this task I mentioned as much. 
> However, the current desire is to work with already existing hardware.
That is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it
as well.
> 
> 
> I am struggling to find what the bottle neck is in this scenario.  Does
anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up
because of transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio
files I am using are gsm.

You will find it less taxing on the server if you have MoH files and
sounds files
available in all the possible native formats. Then Asterisk can use the
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and
g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk 


 <http://www.softins.co.uk/> Software Insight - Welcome

www.softins.co.uk

Welcome. Software Insight Ltd is a small but expert company specialising
in software and systems development and systems administration. We pride
ourselves in ...



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Tony Mountifield's Home Page. This page is still under construction
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Joseph Smith
Thanks for the suggestion Tony,


I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(


From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield 
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.outlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Software Insight - Welcome<http://www.softins.co.uk/>
www.softins.co.uk
Welcome. Software Insight Ltd is a small but expert company specialising in 
software and systems development and systems administration. We pride ourselves 
in ...



Play: t...@mountifield.org - http://tony.mountifield.org
[http://tony.mountifield.org/images/tony2.jpg]<http://tony.mountifield.org/>

Tony Mountifield's Home Page<http://tony.mountifield.org/>
tony.mountifield.org
Tony Mountifield's Home Page. This page is still under construction (despite 
having been started a long time ago!) It will grow as I think of more things to 
put in ...




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The Asterisk Community's home for Discussion ... Is there any way to share same 
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...




New to Asterisk? Start here:
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Tony Mountifield
In article 
,
Joseph Smith  wrote:
> 
> Thanks for the feedback.
> 
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much. 
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
> 
> 
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Joseph Smith
Thanks for the feedback.

I do agree with having multiple smaller servers.  When I was first approached 
with this task I mentioned as much.  However, the current desire is to work 
with already existing hardware.  That is out of my hands at the moment unless 
it just can't be done.  I will explore Freeswitch a bit soon to compare it as 
well.


I am struggling to find what the bottle neck is in this scenario.  Does anyone 
have any advice on what that could be or on steps to discover it?   Do you 
think that tasks are pooling up because of transcoding?  If so would it help to 
change the codec that is being used?  I am not sure about the MoH but the audio 
files I am using are gsm.


Thanks

Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Pete Mundy 
<p...@fiberphone.co.nz>
Sent: Thursday, August 31, 2017 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

>> On Thu, 31 Aug 2017, Joseph Smith wrote:
>>
>> So I am looking for a better way to allow several thousand callers to listen 
>> to this IVR menu at the same time.


> On 1/09/2017, at 7:10 AM, Steve Edwards <asterisk@sedwards.com> wrote:
>
> I'm thinking multiple hosts.
>
> I'm not a fan of 4,000 eggs in one basket.


+1 for horizontal scaling as the best solution in this situation.

Pete

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Pete Mundy
>> On Thu, 31 Aug 2017, Joseph Smith wrote:
>> 
>> So I am looking for a better way to allow several thousand callers to listen 
>> to this IVR menu at the same time.


> On 1/09/2017, at 7:10 AM, Steve Edwards  wrote:
> 
> I'm thinking multiple hosts.
> 
> I'm not a fan of 4,000 eggs in one basket.


+1 for horizontal scaling as the best solution in this situation.

Pete



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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Steve Edwards

On Thu, 31 Aug 2017, Joseph Smith wrote:

So I am looking for a better way to allow several thousand callers to 
listen to this IVR menu at the same time.


I'm thinking multiple hosts.

I'm not a fan of 4,000 eggs in one basket.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Daniel Tryba
On Thu, Aug 31, 2017 at 05:54:43PM +, Joseph Smith wrote:
> 
> So I am looking for a better way to allow several thousand callers to listen 
> to this IVR menu at the same time.
> 

An alternative that comes to mind is to have 1 conference with 1 channel
playing MoH in it and then add callers in a muted state to it. Never
tried this, don't know if it fits your case.


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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Joseph Smith
It is meant to simulate simultaneous calls on an IVR.  I have also tested with 
a separate set of audio files closer to what the actual IVR menu.  This 
produced the same result.


I apologize for not clearly stating the use case up front.  I will try to give 
a bit more detail on that now.


I have an IVR menu and submenu that users may dial into. I initially tested 
with the IVR audio files.  When I began experiencing this issue I used MoH as 
an attempt to narrow down the problem to the simplest dialplan possible.


If I continue my test at this volume or a higher volume, I begin to get errors 
about reaching the maximum queue size for that particular taskprocessor.  
Since, these error proceeded that I thought that they may be the key to 
preventing the queue from maxing out.


It sounds like Richard is saying that these refcount logs may not actually be 
errors and can be ignored in this scenario.  If that is the case then is there 
anything that can be done about the task processor queue size?  Is that simply 
a side effect of having so many callers listening to the IVR at the same time?

pjsip.conf is currently setup with a trunk allowing incoming calls from a 
specific IP.  This is the task processor that is maxing out.

So I am looking for a better way to allow several thousand callers to listen to 
this IVR menu at the same time.

Thank you for the feedback thus far.

Any info and advice is helpful.

Thanks
Joseph




From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Antony Stone 
<antony.st...@asterisk.open.source.it>
Sent: Thursday, August 31, 2017 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:

> I was hoping Asterisk would handle more than 4k simultaneous calls.

I know from experience that Asterisk can handle more than 4k simultaneous
calls, however it's an extreme case to have all of them playing music on hold.

I think that if you tested 4k simultaneous calls with standard media streams
on the majority of them, you would not experience the problem.

Is this a real problem for you - that Asterisk can't manage 4k MoH sessions
simultaneously, even though it can manage 4k standard phone calls?


Antony.

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...




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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Antony Stone
On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:

> I was hoping Asterisk would handle more than 4k simultaneous calls.

I know from experience that Asterisk can handle more than 4k simultaneous 
calls, however it's an extreme case to have all of them playing music on hold.

I think that if you tested 4k simultaneous calls with standard media streams 
on the majority of them, you would not experience the problem.

Is this a real problem for you - that Asterisk can't manage 4k MoH sessions 
simultaneously, even though it can manage 4k standard phone calls?


Antony.

-- 
Someone has stolen all the toilets from New Scotland Yard.  Police say they 
have absolutely nothing to go on.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Richard Mudgett
On Thu, Aug 31, 2017 at 11:15 AM, Joseph Smith 
wrote:

> Is there any more information I can provide to give insight to these
> errors?
>
> Any further advice on avoiding these during high call volume?
>
>
> I was hoping Asterisk would handle more than 4k simultaneous calls.
>

* There is no user configurable option to change the excessive ref count
trigger value.  However, you could change the EXCESSIVE_REF_COUNT define
value in the main/astobj2.c file and recompile.

Richard
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Joseph Smith
Is there any more information I can provide to give insight to these errors?

Any further advice on avoiding these during high call volume?


I was hoping Asterisk would handle more than 4k simultaneous calls.

Thanks

Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Joseph Smith 
<warlock1...@hotmail.com>
Sent: Monday, August 28, 2017 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan


Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Richard Mudgett 
<rmudg...@digium.com>
Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
<warlock1...@hotmail.com<mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can any

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Doug Lytle

On 08/28/2017 06:00 PM, Joseph Smith wrote:



I set no optimize and better backtrace through "make menuselect" and 
the output is now




Please ignore the noise, I need to slow down when I read.

Doug

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Doug Lytle

On 08/28/2017 06:00 PM, Joseph Smith wrote:



I set no optimize and better backtrace through "make menuselect" and 
the output is now




menuselect => Compiler Flags => Better Backtraces

Doug

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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Joseph Smith
Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Richard Mudgett 
<rmudg...@digium.com>
Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
<warlock1...@hotmail.com<mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

This particular FRACK is meant to help find ao2 object reference leaks.  It 
acts as an early warning for excessive references to any particular ao2 object 
used in the code.  The FRACK itself is benign.  Based upon the inline backtrace 
the ao2 object is likely to be a codec format.

* What codecs are you using in this setup?

* With 500 calls/sec and the calls lasting 8 seconds that comes to 4000 active 
channels.  Hitting the FRACK would result in

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Richard Mudgett
On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
wrote:

> Hello,
>
> I've recently setup a small load test against an instance of Asterisks.
> I've tested on asterisk 13.5 and 14.6 with the same results.
>
I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

> I am using PJSIP.  My dial plan is,
>
> [test]
>
> exten => 1001,1,Answer
>
> exten => 1001,n,MusicOnHold(15)
>
> exten => 1001,n,Hangup
>
>
> I am using SIPP to test.  I can share XML if desired but it simply waits
> on the line while music plays for 8 seconds.  I used sippycup to generate
> it with the following steps in the yaml file.
>
>
> steps:
>
>   - invite
>
>   - wait_for_answer
>
>   - ack_answer
>
>   - sleep 8
>
>   - send_bye
>
>
> At around 500 calls per second I begin to see the following ERRORs,
>
>
> [Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup:
> Excessive refcount 10 reached on ao2 object 0x26bffc0
>
> [Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!,
> Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0
> (0)
>
> Got 19 backtrace records
>
> #0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]
>
> #1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]
>
> #2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]
>
> #3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]
>
> #4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b)
> [0x7efeb578230b]
>
> #5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]
>
> #6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]
>
> #7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]
>
> #8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d)
> [0x7efeb578478d]
>
> #9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]
>
> #10: [0x582e84] /usr/sbin/asterisk() [0x582e84]
>
> #11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]
>
> #12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]
>
> #13: [0x60002a] /usr/sbin/asterisk() [0x60002a]
>
>
This inline backtrace would be more useful if you had BETTER_BACKTRACES
enabled.


>
> I've also seen similar behavior when using playback instead of
> MusicOnHold.  CPU usage gets around 50%.  Can anyone enlighten me on the
> meaning and cause of the error?  Is there some steps (config etc) that can
> be taken to alleviate the issue?
>

This particular FRACK is meant to help find ao2 object reference leaks.  It
acts as an early warning for excessive references to any particular ao2
object used in the code.  The FRACK itself is benign.  Based upon the
inline backtrace the ao2 object is likely to be a codec format.

* What codecs are you using in this setup?

* With 500 calls/sec and the calls lasting 8 seconds that comes to 4000
active channels.  Hitting the FRACK would result in an average of 25
references to the format per channel.  This is a simplistic calculation as
there are going to be some references that have nothing to do with a call.
The number of base references would depend upon which codec is involved.

* There is no user configurable option to change the excessive ref count
trigger value.  However, you could change the EXCESSIVE_REF_COUNT define
value in the main/astobj2.c file and recompile.

Richard
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[asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Joseph Smith
Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]


I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

Thanks
Joseph




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Re: [asterisk-users] Error when installing asterisk on Ubuntu 16.03LTS

2016-10-08 Thread Антон Сацкий
try to uncheck chan_unistim.so module

2016-10-08 14:57 GMT+03:00 christopher kamutumwa :

> Hello,
> I am trying to install asterisk 14 on ubuntu 16 but i am getting below
> error message please assist with how to resolve that after i run make
> && make install && make config && make samples
>
>[LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
>[CPP] chan_unistim.c -> chan_unistim.i
>[CCi] chan_unistim.i -> chan_unistim.o
>[LD] chan_unistim.o -> chan_unistim.so
> /usr/bin/ld: /usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: relocation
> R_X86_64_32 against `__TMC_END__' can not be used when making a shared
> object; recompile with -fPIC
> /usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: error adding symbols: Bad
> value
> collect2: error: ld returned 1 exit status
> /usr/src/asterisk-14.0.2/Makefile.rules:176: recipe for target
> 'chan_unistim.so' failed
> make[1]: *** [chan_unistim.so] Error 1
> Makefile:397: recipe for target 'channels' failed
> make: *** [channels] Error 2
>
> thanks
>
> chri
>
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[asterisk-users] Error when installing asterisk on Ubuntu 16.03LTS

2016-10-08 Thread christopher kamutumwa
Hello,
I am trying to install asterisk 14 on ubuntu 16 but i am getting below
error message please assist with how to resolve that after i run make
&& make install && make config && make samples

   [LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
   [CPP] chan_unistim.c -> chan_unistim.i
   [CCi] chan_unistim.i -> chan_unistim.o
   [LD] chan_unistim.o -> chan_unistim.so
/usr/bin/ld: /usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: relocation
R_X86_64_32 against `__TMC_END__' can not be used when making a shared
object; recompile with -fPIC
/usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: error adding symbols: Bad value
collect2: error: ld returned 1 exit status
/usr/src/asterisk-14.0.2/Makefile.rules:176: recipe for target
'chan_unistim.so' failed
make[1]: *** [chan_unistim.so] Error 1
Makefile:397: recipe for target 'channels' failed
make: *** [channels] Error 2

thanks

chri

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Re: [asterisk-users] Error compiling dahdi on CentOS 7

2016-02-28 Thread Tzafrir Cohen
On Wed, Feb 24, 2016 at 03:55:09PM -0600, Carlos Chavez wrote:
> I am having a problem trying to compile
> dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server.  Version 2.10.2
> compiles fine.  Is there a new dependency for 2.11.0 that was not required
> for previous versions?  Here are some of the errors I get:
> 
>   INSTALL
> /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi.ko
> Can't read private key
>   INSTALL 
> /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic.ko
> Can't read private key
>   INSTALL 
> /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic_eth.ko
> Can't read private key

I'm not sure what this is. However,

> 
> /usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or
> directory
> /usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory
> /usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or
> directory
> /usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory

This one shouldl be fixed in dahdi-tools 2.11.1-rc1 .

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[asterisk-users] Error compiling dahdi on CentOS 7

2016-02-24 Thread Carlos Chavez
I am having a problem trying to compile 
dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server.  Version 
2.10.2 compiles fine.  Is there a new dependency for 2.11.0 that was not 
required for previous versions?  Here are some of the errors I get:


  INSTALL 
/usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi.ko

Can't read private key
  INSTALL 
/usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic.ko

Can't read private key
  INSTALL 
/usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic_eth.ko

Can't read private key

/usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or 
directory

/usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory
/usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory
/usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory
/usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory
/usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or 
directory

/usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory
make[4]: *** [install-man8] Error 1
make[4]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp'

make[3]: *** [install-am] Error 2
make[3]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp'

make[2]: *** [install-recursive] Error 1
make[2]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp'

make[1]: *** [install-recursive] Error 1
make[1]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools'

make: *** [install] Error 2

Compiling version 2.10.2 on the same machine does not give any 
errors and works as expected.


--
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Re: [asterisk-users] Error making dahdi linux compete 2.11.0

2016-02-17 Thread Tzafrir Cohen
On Mon, Feb 15, 2016 at 05:28:14PM +0200, Tzafrir Cohen wrote:
> On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote:
> > Getting the some errors making dahdi 2.11.0.
> > 
> > Seems same as listed here 
> > http://forums.asterisk.org/viewtopic.php?f=1=96455
> > 
> > In that link they say to use 2.10.2 but that's from December. Is there a 
> > fix yet for this?
> 
> My bad, I forgot to push the fix for that.
> 
> http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=e1da7b528467a8f8f82058993b2e01333677ee39
> 
> You may need to run autoreconf after applying this patch to rebuild the
> Makefile (though IIRC the makefile will be re-generated by running
> 'make').

And while I'm at it:

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=6057ef25e984a2c7f8327b872233ba610b9aabe6

Another thing to note: you should install pkg-config (though practically
you only need it to detect libusb, that is: for building
Astribank-related utilities). If you installed libusb(1)-dev but it is
still not detected, maybe it is because pkg-config is not installed.

-- 
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http://www.xorcom.com

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Re: [asterisk-users] Error making dahdi linux compete 2.11.0

2016-02-15 Thread Tzafrir Cohen
On Mon, Feb 15, 2016 at 02:15:58PM +, Ryan, Travis wrote:
> Getting the some errors making dahdi 2.11.0.
> 
> Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1=96455
> 
> In that link they say to use 2.10.2 but that's from December. Is there a fix 
> yet for this?

My bad, I forgot to push the fix for that.

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=e1da7b528467a8f8f82058993b2e01333677ee39

You may need to run autoreconf after applying this patch to rebuild the
Makefile (though IIRC the makefile will be re-generated by running
'make').

-- 
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http://www.xorcom.com

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[asterisk-users] Error making dahdi linux compete 2.11.0

2016-02-15 Thread Ryan, Travis
Getting the some errors making dahdi 2.11.0.

Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1=96455

In that link they say to use 2.10.2 but that's from December. Is there a fix 
yet for this?

Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102

We're not the IT departmentWe're the I-TEAM department!

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Re: [asterisk-users] Error while compiling asterisk asterisk-1.8.32.3

2015-11-25 Thread Rusty Newton
On Sat, Nov 21, 2015 at 1:51 AM, Jayesh Labade  wrote:
> Hi,
>
> I encountered following error while compiling asterisk-1.8.32.3. I am
> using Debian 8(Jessie) 64 bit version.
>
> make[1]: *** [chan_dahdi.so] Error 1
> Makefile:351: recipe for target 'channels' failed
> make: *** [channels] Error 2
>
> Detailed error attached in log file.

I'm not sure what is going on there but I wanted to mention that
Asterisk 1.8 is completely EOL, there will be no further fixes, even
security fixes. For new installations you should use Asterisk 13 which
is the most recent LTS.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

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direct: +1 256 428 6200

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[asterisk-users] Error while compiling asterisk asterisk-1.8.32.3

2015-11-20 Thread Jayesh Labade
Hi,

I encountered following error while compiling asterisk-1.8.32.3. I am
using Debian 8(Jessie) 64 bit version.

make[1]: *** [chan_dahdi.so] Error 1
Makefile:351: recipe for target 'channels' failed
make: *** [channels] Error 2

Detailed error attached in log file.

Best Regards,
Jayesh Labade
   [CC] sig_ss7.c -> sig_ss7.o
   [LD] chan_dahdi.o sig_analog.o sig_pri.o sig_ss7.o -> chan_dahdi.so
sig_analog.o: In function `ast_atomic_fetchadd_int':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:600: multiple 
definition of `ast_atomic_fetchadd_int'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:600: 
first defined here
sig_analog.o: In function `ast_atomic_dec_and_test':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:646: multiple 
definition of `ast_atomic_dec_and_test'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:646: 
first defined here
sig_analog.o: In function `ast_tvdiff_sec':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:45: multiple 
definition of `ast_tvdiff_sec'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:45: first 
defined here
sig_analog.o: In function `ast_tvdiff_us':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:64: multiple 
definition of `ast_tvdiff_us'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:64: first 
defined here
sig_analog.o: In function `ast_tvdiff_ms':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:78: multiple 
definition of `ast_tvdiff_ms'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:78: first 
defined here
sig_analog.o: In function `ast_tvzero':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:95: multiple 
definition of `ast_tvzero'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:95: first 
defined here
sig_analog.o: In function `ast_tvcmp':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:106: multiple 
definition of `ast_tvcmp'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:106: 
first defined here
sig_analog.o: In function `ast_tveq':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:125: multiple 
definition of `ast_tveq'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:125: 
first defined here
sig_analog.o: In function `ast_tvnow':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:135: multiple 
definition of `ast_tvnow'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:135: 
first defined here
sig_analog.o: In function `ast_tv':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:171: multiple 
definition of `ast_tv'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:171: 
first defined here
sig_analog.o: In function `ast_samp2tv':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:186: multiple 
definition of `ast_samp2tv'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:186: 
first defined here
sig_analog.o: In function `_ast_malloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:477: multiple 
definition of `_ast_malloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:477: 
first defined here
sig_analog.o: In function `_ast_calloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:500: multiple 
definition of `_ast_calloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:500: 
first defined here
sig_analog.o: In function `_ast_realloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:536: multiple 
definition of `_ast_realloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:536: 
first defined here
sig_analog.o: In function `_ast_strdup':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:563: multiple 
definition of `_ast_strdup'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:563: 
first defined here
sig_analog.o: In function `_ast_strndup':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:592: multiple 
definition of `_ast_strndup'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:592: 
first defined here
sig_analog.o: In function `_ast_vasprintf':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:631: multiple 
definition of `_ast_vasprintf'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:631: 
first defined here
sig_analog.o: In function `ast_threadstorage_get':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/threadstorage.h:173: multiple 
definition of `ast_threadstorage_get'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/threadstorage.h:173:
 first defined here
sig_analog.o: In function `ast_skip_blanks':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/strings.h:90: multiple 
definition of `ast_skip_blanks'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/strings.h:90: 
first defined here
sig_analog.o: 

Re: [asterisk-users] error trying to get PJSIP working

2015-06-19 Thread Matthew Jordan
On Thu, Jun 18, 2015 at 1:52 PM, Ryan, Travis ry...@oscarwinski.com wrote:
 I’m doing an upgrade from Asterisk 11 to 13. I’m following the guide at
 https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to
 setup realtime, as I use realtime on Asterisk 11 too.



 I’m getting the following error when trying to connect the peer to the
 server.



 Help? J



 Thanks,



 Travis



 [Jun 15 16:20:03] NOTICE[5116] res_odbc.c: res_odbc: Connected to laf [laf]

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] ERROR[5116] res_pjsip_registrar.c: Unable to bind contact
 'sip:812@10.1.80.112:5062' to AOR '812'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM
 ps_contacts WHERE id LIKE ? ORDER BY id

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') =
 '812;@%'

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM
 ps_contacts WHERE id LIKE ? ORDER BY id

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') =
 '812;@%'

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: INSERT INTO
 ps_contacts (id, outbound_proxy, expiration_time, path, qualify_frequency,
 user_agent, uri) VALUES (?, ?, ?, ?, ?, ?, ?)

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id') =
 '812;@sip:812@10.1.80.112:5062'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 2
 ('outbound_proxy') = ''

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 3
 ('expiration_time') = '1434399723'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 4 ('path') = ''

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 5
 ('qualify_frequency') = '0'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 6 ('user_agent')
 = 'Media5-fone/4.1.3.3034 iOS/8.3'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 7 ('uri') =
 'sip:812@10.1.80.112:5062'

 [Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute returned an error
 -1: 42S22: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1]Unknown
 column 'outbound_proxy' in 'field list' (103)

 [Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute error -1! Verifying
 connection to laf [laf]...


It looks like you are missing the outbound_proxy column on the
ps_contacts table. If you're missing that column, you are probably
missing some other columns as well.

Note that the schema for the realtime tables for PJSIP has been
updated many times, as new features have been added. The alembic
scripts bundled with Asterisk can manage your DB schemas for you, or
can be used to generate the schema used by your specific version of
Asterisk.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] error trying to get PJSIP working

2015-06-18 Thread Ryan, Travis
I'm doing an upgrade from Asterisk 11 to 13. I'm following the guide at 
https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup 
realtime, as I use realtime on Asterisk 11 too.

I'm getting the following error when trying to connect the peer to the server.

Help? :)

Thanks,

Travis

[Jun 15 16:20:03] NOTICE[5116] res_odbc.c: res_odbc: Connected to laf [laf]
[Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) 
called (obj-txf = (nil))
[Jun 15 16:20:03] ERROR[5116] res_pjsip_registrar.c: Unable to bind contact 
'sip:812@10.1.80.112:5062' to AOR '812'
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM 
ps_contacts WHERE id LIKE ? ORDER BY id
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') = 
'812;@%'
[Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) 
called (obj-txf = (nil))
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM 
ps_contacts WHERE id LIKE ? ORDER BY id
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') = 
'812;@%'
[Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) 
called (obj-txf = (nil))
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: INSERT INTO 
ps_contacts (id, outbound_proxy, expiration_time, path, qualify_frequency, 
user_agent, uri) VALUES (?, ?, ?, ?, ?, ?, ?)
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id') = 
'812;@sip:812@10.1.80.112:5062'
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 2 ('outbound_proxy') 
= ''
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 3 
('expiration_time') = '1434399723'
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 4 ('path') = ''
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 5 
('qualify_frequency') = '0'
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 6 ('user_agent') = 
'Media5-fone/4.1.3.3034 iOS/8.3'
[Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 7 ('uri') = 
'sip:812@10.1.80.112:5062'
[Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute returned an error -1: 
42S22: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1]Unknown column 
'outbound_proxy' in 'field list' (103)
[Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute error -1! Verifying 
connection to laf [laf]...
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Re: [asterisk-users] Error writing CDR

2015-04-26 Thread jg

Hi All

I have dozens of these messages on CLI complaining about database
connection and error writing CDR to disk.

The curious thing is I can find them all inside the database. I
selected them using uniqueid and manually compared each column
with the cdr_adaptive_odbc.c error line.

mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both
returned OK for all tables.

Environment is: in production Asterisk 11.7.0~dfsg-1ubuntu1 Ubuntu
14.04.1 LTS

Any thoughts?

Thanx

Ethy

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
23000: [MySQL][ODBC 5.1
Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
'-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)

Hi Ethy,

why date and time are empty?

At least date is used as a unique key and a unique key has to be
unique. In other words, the same key can not exist twice like in your
case.

Check why there is no date and time anymore ...



Or define your table with and independent primary key that gets added 
automatically:

mysql describe cdr;
+--+--+--+-+-++
| Field| Type | Null | Key | Default | Extra
  |
+--+--+--+-+-++
*| id   | int(11) | NO   | PRI | NULL| 
auto_increment |*
| clid | varchar(80)  | NO   | | |  
  |
| src  | varchar(80)  | NO   | MUL | |  
  |
| dst  | varchar(80)  | NO   | | |  
  |
...
| lastapp  | varchar(80)  | NO   | | |  
  |
| lastdata | varchar(80)  | NO   | | |  
  |
| duration | int(11)  | NO   | | 0   |  
  |
| billsec  | int(11)  | NO   | | 0   |  
  |
| disposition  | varchar(45)  | NO   | | |  
  |
| start| datetime | NO   | MUL | -00-00 00:00:00 |  
  |
| answer   | datetime | NO   | | -00-00 00:00:00 |  
  |
| end  | datetime | NO   | | -00-00 00:00:00 |  
  |
| uniqueid | varchar(45)  | NO   | | |  
  |
...

Just in case you get bogus records with offending primary keys due to some other problem, you 
would still have valid data base entries and you would be able to look at the pattern.


jg


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Re: [asterisk-users] Error writing CDR

2015-04-26 Thread Ethy H. Brito
On Sun, 26 Apr 2015 11:11:10 +0200
jg webaccounts...@jgoettgens.de wrote:

  Hi All
 
  I have dozens of these messages on CLI complaining about database
  connection and error writing CDR to disk.
 
  The curious thing is I can find them all inside the database. I
  selected them using uniqueid and manually compared each column
  with the cdr_adaptive_odbc.c error line.
 
  mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both
  returned OK for all tables.
 
  Environment is: in production Asterisk 11.7.0~dfsg-1ubuntu1 Ubuntu
  14.04.1 LTS
 
  Any thoughts?
 
  Thanx
 
  Ethy
 
  [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
  ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
  23000: [MySQL][ODBC 5.1
  Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
  '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)
  Hi Ethy,
 
  why date and time are empty?
 
  At least date is used as a unique key and a unique key has to be
  unique. In other words, the same key can not exist twice like in your
  case.
 
  Check why there is no date and time anymore ...
 
 
 Or define your table with and independent primary key that gets added
 automatically:
 
 mysql describe cdr;
 +--+--+--+-+-++
 | Field| Type | Null | Key | Default |
 Extra  |
 +--+--+--+-+-++
 *| id   | int(11) | NO   | PRI | NULL|
 auto_increment |* | clid | varchar(80)  | NO   |
 | || | src  |
 varchar(80)  | NO   | MUL | || |
 dst  | varchar(80)  | NO   | |
 || ... | lastapp  | varchar(80)  | NO   |
 | || | lastdata |
 varchar(80)  | NO   | | || |
 duration | int(11)  | NO   | | 0
 || | billsec  | int(11)  | NO   | |
 0   || | disposition  | varchar(45)
 | NO   | | || | start|
 datetime | NO   | MUL | -00-00 00:00:00 || |
 answer   | datetime | NO   | | -00-00 00:00:00
 || | end  | datetime | NO   | |
 -00-00 00:00:00 || | uniqueid | varchar(45)
 | NO   | | || ...
 
 Just in case you get bogus records with offending primary keys due to
 some other problem, you would still have valid data base entries and you
 would be able to look at the pattern.
 
 jg

Hi guys

I maybe have encountered a bug or I am doing something very stupid.

That is what happened: I disabled a connection I am playing around on
res_odbc.conf and the problem stopped. Just did this, nothing else.

By the beginning of this month I was trying to do a CDR extension to
accommodate calls statistics. I did not touched the CDR as you can see by
the describe above.

I just follow some stepping stones at 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html
and http://.../3rd_Edition/asterisk-book-html-chunk/database_storing-cdr.html to
activate ODBC and then I created a h extension at the extensions.conf file to
harvest some stats. 
No attributions at all (like Set(CDR(foo)=bar)), since I did not tampered the 
CDR.

What is happening to ODBC that it is trying to insert data into the CDR
when I did not told it to do so?

What I did not do to prevent this?

Thanx for your answers

Ethy

 
 

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Re: [asterisk-users] Error writing CDR

2015-04-25 Thread Ethy H. Brito
On Sat, 25 Apr 2015 17:05:44 -0300
Ethy H. Brito ethy.br...@inexo.com.br wrote:

 On Sat, 25 Apr 2015 17:11:34 +0200
 jg webaccounts...@jgoettgens.de wrote:
 
  
   Hi All
  
   I have dozens of these messages on CLI complaining about database
   connection and error writing CDR to disk.
  
   The curious thing is I can find them all inside the database.
   I selected them using uniqueid and manually compared each column
   with the cdr_adaptive_odbc.c error line.
  
   mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both
   returned OK for all tables.
  
   Environment is:
 in production Asterisk 11.7.0~dfsg-1ubuntu1
 Ubuntu 14.04.1 LTS
  
   Any thoughts?
  
   Thanx
  
   Ethy
  
   [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
   ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
   23000: [MySQL][ODBC 5.1
   Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
   '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)
  
   [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:657
   ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
   connection to MyAsterisk-asterisk [MyAsterisk-asterisk]...
  
   [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:761
   ast_odbc_sanity_check: Connection is down attempting to reconnect...
  
   [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1527
   odbc_obj_connect: Connecting MyAsterisk-asterisk
  
   [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1559
   odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk
   [MyAsterisk-asterisk]
  
   [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:645
   ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
   23000: [MySQL][ODBC 5.1
   Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
   '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)
  
   [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:657
   ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
   connection to MyAsterisk-asterisk [MyAsterisk-asterisk]...
  
   [Apr 25 10:57:01]
   WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check:
   Connection is down attempting to reconnect...
  
   [Apr 25 10:57:02]
   WARNING[7666]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on
   dialog '34f3f3481b8d1e4772dc111f42d13...@ip.ip.ip.ip:5060' with owner
   SIP/CLIENT_ID_1-0178 in place (Method: BYE). Rescheduling
   destruction for 1 ms
  
   [Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1527
   odbc_obj_connect: Connecting MyAsterisk-asterisk
  
   [Apr 25 10:57:06]
   NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect:
   res_odbc: Connected to MyAsterisk-asterisk [MyAsterisk-asterisk]
  
   [Apr 25 10:57:06]
   WARNING[19013][C-02cb]: cdr_adaptive_odbc.c:739 odbc_log:
   cdr_adaptive_odbc: Insert failed on 'MyAsterisk-asterisk:cdr'.  CDR
   failed: INSERT INTO cdr
   (dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid)
   VALUES (blahblahblah, ... ,'1429970147.612')
  
  Can you post the output of describe schema;? At least the first
  error message is probably related to a not so optimal primary key
  definition.
 
 Thanx for the reply.
 
 request follows...
 
 mysql describe cdr ;
 +-+--+--+-+-+---+
 | Field   | Type | Null | Key | Default | Extra |
 +-+--+--+-+-+---+
 | calldate| datetime | NO   | PRI | -00-00 00:00:00 |   |
 | dst | varchar(80)  | NO   | PRI | NULL|   |
 | accountcode | varchar(20)  | NO   | PRI | NULL|   |
 | clid| varchar(80)  | NO   | | NULL|   |
 | src | varchar(80)  | NO   | MUL | NULL|   |
 | dcontext| varchar(80)  | NO   | | NULL|   |
 | channel | varchar(80)  | NO   | | NULL|   |
 | dstchannel  | varchar(80)  | NO   | | NULL|   |
 | lastapp | varchar(80)  | NO   | | NULL|   |
 | duration| int(11)  | NO   | | 0   |   |
 | billsec | int(11)  | NO   | | 0   |   |
 | disposition | varchar(45)  | NO   | MUL | NULL|   |
 | amaflags| int(11)  | NO   | | 0   |   |
 | userfield   | varchar(255) | NO   | | NULL|   |
 | lastdata| varchar(80)  | NO   | | NULL|   |
 | uniqueid| varchar(32)  | YES  | MUL | NULL|   |
 +-+--+--+-+-+---+
 16 rows in set (0.00 sec)
 
 
 FYI this has been running smooth for years.
 
 This problem started a few days ago.
 
 Ethy

Further informations.

For 

Re: [asterisk-users] Error writing CDR

2015-04-25 Thread Ethy H. Brito
On Sat, 25 Apr 2015 17:11:34 +0200
jg webaccounts...@jgoettgens.de wrote:

 
  Hi All
 
  I have dozens of these messages on CLI complaining about database
  connection and error writing CDR to disk.
 
  The curious thing is I can find them all inside the database.
  I selected them using uniqueid and manually compared each column
  with the cdr_adaptive_odbc.c error line.
 
  mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both
  returned OK for all tables.
 
  Environment is:
  in production Asterisk 11.7.0~dfsg-1ubuntu1
  Ubuntu 14.04.1 LTS
 
  Any thoughts?
 
  Thanx
 
  Ethy
 
  [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
  ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
  [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate
  entry '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)
 
  [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:657
  ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
  connection to MyAsterisk-asterisk [MyAsterisk-asterisk]...
 
  [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:761
  ast_odbc_sanity_check: Connection is down attempting to reconnect...
 
  [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1527
  odbc_obj_connect: Connecting MyAsterisk-asterisk
 
  [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1559
  odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk
  [MyAsterisk-asterisk]
 
  [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:645
  ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
  [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate
  entry '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)
 
  [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:657
  ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
  connection to MyAsterisk-asterisk [MyAsterisk-asterisk]...
 
  [Apr 25 10:57:01]
  WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check:
  Connection is down attempting to reconnect...
 
  [Apr 25 10:57:02]
  WARNING[7666]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on
  dialog '34f3f3481b8d1e4772dc111f42d13...@ip.ip.ip.ip:5060' with owner
  SIP/CLIENT_ID_1-0178 in place (Method: BYE). Rescheduling
  destruction for 1 ms
 
  [Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1527
  odbc_obj_connect: Connecting MyAsterisk-asterisk
 
  [Apr 25 10:57:06]
  NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect: res_odbc:
  Connected to MyAsterisk-asterisk [MyAsterisk-asterisk]
 
  [Apr 25 10:57:06]
  WARNING[19013][C-02cb]: cdr_adaptive_odbc.c:739 odbc_log:
  cdr_adaptive_odbc: Insert failed on 'MyAsterisk-asterisk:cdr'.  CDR
  failed: INSERT INTO cdr
  (dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid)
  VALUES (blahblahblah, ... ,'1429970147.612')
 
 Can you post the output of describe schema;? At least the first
 error message is probably related to a not so optimal primary key
 definition.

Thanx for the reply.

request follows...

mysql describe cdr ;
+-+--+--+-+-+---+
| Field   | Type | Null | Key | Default | Extra |
+-+--+--+-+-+---+
| calldate| datetime | NO   | PRI | -00-00 00:00:00 |   |
| dst | varchar(80)  | NO   | PRI | NULL|   |
| accountcode | varchar(20)  | NO   | PRI | NULL|   |
| clid| varchar(80)  | NO   | | NULL|   |
| src | varchar(80)  | NO   | MUL | NULL|   |
| dcontext| varchar(80)  | NO   | | NULL|   |
| channel | varchar(80)  | NO   | | NULL|   |
| dstchannel  | varchar(80)  | NO   | | NULL|   |
| lastapp | varchar(80)  | NO   | | NULL|   |
| duration| int(11)  | NO   | | 0   |   |
| billsec | int(11)  | NO   | | 0   |   |
| disposition | varchar(45)  | NO   | MUL | NULL|   |
| amaflags| int(11)  | NO   | | 0   |   |
| userfield   | varchar(255) | NO   | | NULL|   |
| lastdata| varchar(80)  | NO   | | NULL|   |
| uniqueid| varchar(32)  | YES  | MUL | NULL|   |
+-+--+--+-+-+---+
16 rows in set (0.00 sec)


FYI this has been running smooth for years.

This problem started a few days ago.

Ethy


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Re: [asterisk-users] Error writing CDR

2015-04-25 Thread Guenther Boelter
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 04/25/2015 10:51 PM, Ethy H. Brito wrote:
 
 Hi All
 
 I have dozens of these messages on CLI complaining about database
 connection and error writing CDR to disk.
 
 The curious thing is I can find them all inside the database. I
 selected them using uniqueid and manually compared each column
 with the cdr_adaptive_odbc.c error line.
 
 mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both
 returned OK for all tables.
 
 Environment is: in production Asterisk 11.7.0~dfsg-1ubuntu1 Ubuntu
 14.04.1 LTS
 
 Any thoughts?
 
 Thanx
 
 Ethy
 
 [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645 
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1:
 23000: [MySQL][ODBC 5.1
 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry 
 '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)

Hi Ethy,

why date and time are empty?

At least date is used as a unique key and a unique key has to be
unique. In other words, the same key can not exist twice like in your
case.

Check why there is no date and time anymore ...

Regards
Guenther

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[asterisk-users] Error writing CDR

2015-04-25 Thread Ethy H. Brito

Hi All

I have dozens of these messages on CLI complaining about database connection 
and error writing CDR to disk.

The curious thing is I can find them all inside the database.
I selected them using uniqueid and manually compared each column with the 
cdr_adaptive_odbc.c error line.

mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both returned OK 
for all tables.

Environment is:
in production Asterisk 11.7.0~dfsg-1ubuntu1
Ubuntu 14.04.1 LTS

Any thoughts?

Thanx

Ethy

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
[MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
'-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) 

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:657
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
MyAsterisk-asterisk [MyAsterisk-asterisk]... 

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:761 
ast_odbc_sanity_check: Connection is down attempting to reconnect... 

[Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1527 odbc_obj_connect: 
Connecting MyAsterisk-asterisk 

[Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1559
odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk 
[MyAsterisk-asterisk]

[Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
[MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
'-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) 

[Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:657
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
MyAsterisk-asterisk [MyAsterisk-asterisk]... 

[Apr 25 10:57:01]
WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is
down attempting to reconnect... 

[Apr 25 10:57:02]
WARNING[7666]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog
'34f3f3481b8d1e4772dc111f42d13...@ip.ip.ip.ip:5060' with owner
SIP/CLIENT_ID_1-0178 in place (Method: BYE). Rescheduling destruction
for 1 ms 

[Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1527
odbc_obj_connect: Connecting MyAsterisk-asterisk 

[Apr 25 10:57:06]
NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect: res_odbc:
Connected to MyAsterisk-asterisk [MyAsterisk-asterisk] 

[Apr 25 10:57:06]
WARNING[19013][C-02cb]: cdr_adaptive_odbc.c:739 odbc_log:
cdr_adaptive_odbc: Insert failed on 'MyAsterisk-asterisk:cdr'.  CDR failed: 
INSERT
INTO cdr
(dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid)
VALUES (blahblahblah, ... ,'1429970147.612')

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Re: [asterisk-users] Error writing CDR

2015-04-25 Thread jg



Hi All

I have dozens of these messages on CLI complaining about database connection 
and error writing CDR to disk.

The curious thing is I can find them all inside the database.
I selected them using uniqueid and manually compared each column with the 
cdr_adaptive_odbc.c error line.

mysqlcheck -a -e -v DBase  and mysqlcheck -c -e -v DBase both returned OK 
for all tables.

Environment is:
in production Asterisk 11.7.0~dfsg-1ubuntu1
Ubuntu 14.04.1 LTS

Any thoughts?

Thanx

Ethy

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
[MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
'-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:657
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
MyAsterisk-asterisk [MyAsterisk-asterisk]...

[Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:761 
ast_odbc_sanity_check: Connection is down attempting to reconnect...

[Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1527 odbc_obj_connect: 
Connecting MyAsterisk-asterisk

[Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1559
odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk 
[MyAsterisk-asterisk]

[Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000:
[MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry
'-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133)

[Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:657
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
MyAsterisk-asterisk [MyAsterisk-asterisk]...

[Apr 25 10:57:01]
WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is
down attempting to reconnect...

[Apr 25 10:57:02]
WARNING[7666]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog
'34f3f3481b8d1e4772dc111f42d13...@ip.ip.ip.ip:5060' with owner
SIP/CLIENT_ID_1-0178 in place (Method: BYE). Rescheduling destruction
for 1 ms

[Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1527
odbc_obj_connect: Connecting MyAsterisk-asterisk

[Apr 25 10:57:06]
NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect: res_odbc:
Connected to MyAsterisk-asterisk [MyAsterisk-asterisk]

[Apr 25 10:57:06]
WARNING[19013][C-02cb]: cdr_adaptive_odbc.c:739 odbc_log:
cdr_adaptive_odbc: Insert failed on 'MyAsterisk-asterisk:cdr'.  CDR failed: 
INSERT
INTO cdr
(dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid)
VALUES (blahblahblah, ... ,'1429970147.612')

Can you post the output of describe schema;? At least the first error message is probably 
related to a not so optimal primary key definition.


jg

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[asterisk-users] error retrieving a video voicemail in asterisk 11

2015-04-13 Thread Steve Dolloff
Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video 
attachment while using any video phone.  This does work in my 1.8.23.1 
installation.  The file is skipped with the ast_streamfile error (and moved to 
OLD), and the prompts following that message display the ast_best_codec error.

[Apr  7 16:05:50] WARNING[17497][C-6fdd]: file.c:1017 ast_streamfile: 
Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg (format 
(ulaw|h264)): No such file or directory [Apr  7 16:05:50] 
WARNING[17497][C-6fdd]: app_voicemail.c:8609 play_message: Playback of 
message /var/spool/asterisk/voicemail/default/2036/INBOX/msg failed
[Apr  7 16:05:50] -- SIP/2036-00ee Playing 'vm-advopts.gsm' (language 
'en')
[Apr  7 16:05:50] WARNING[17497][C-6fdd]: channel.c:940 ast_best_codec: 
Don't know any of (h264) formats

The file does exist in h264 format

-rw-r--r-- 1 root root 298102 Apr  7 16:05 msg.h264
-rw-r--r-- 1 root root301 Apr  7 16:05 msg.txt
-rw-r--r-- 1 root root 124524 Apr  7 16:05 msg.wav

Passthrough h264 video does work.  I do have h264 and ulaw codecs on the peer 
and videosupport=yes in sip.conf.  I also tried enabling h264 in the general 
section of sip.conf and gsm in voicemail.conf with the same results.

If I disable the h264 codec for the peer, I can listen to the audio portion of 
the message:

[Apr  7 16:41:05] -- SIP/2036-00f3 Playing 
'/var/spool/asterisk/voicemail/default/2036/Old/msg.slin' (language 'en')

Any guesses what I might be doing wrong?  Did something related change in 
asterisk 11?

-- Stephen


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Re: [asterisk-users] Error saving cdr at h exten in Asterisk13

2014-11-21 Thread Alonso Genis


 
 If you are setting the userfield in the 'h' extension, then this is
 what I would expect. CDRs are finalized when the path of communication
 between channels is finished; altering the data after that point
 updates the next CDR for that channel. It isn't retroactive.
 
 The 'h' extension is special in that 'endbeforehexten' explicitly
 ignores updates in 'h' extension. If you disable 'endbeforehexten',
 then you will get a CDR for the channel while it updates the hangup
 logic - but again, modifications occur on that CDR, not on previous
 ones.
 
 If you want the CDR for the channel prior to the 'h' extension to have
 a userfield entry, you have to apply it before the channel hangs up.

I see. Thank you for your answer.
Alonso.

 

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[asterisk-users] Error saving cdr at h exten in Asterisk13

2014-11-20 Thread Alonso Genis
Dears,

I need to save some information on userfield when calls end in Asterisk13, but 
I have two error cases:

1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but userfield 
is not set.
2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src e 
dst correct, and a second line with userfield setting and dst h.

I am using cdr_odbc.conf, with Asterisk11.14.0 it works fine. May be this is a 
bug in asterisk13's cdr?




pabx=# select calldate,src,dst,duration,billsec,uniqueid,userfield from cdr 
order by calldate desc limit 2;
calldate| src  | dst  | duration | billsec |   uniqueid   | 
userfield 
+--+--+--+-+--+---
 2014-11-20 14:37:03-02 | 1901 | h|0 |   0 | 1416501411.0 | 
teste
 2014-11-20 14:36:51-02 | 1901 | 1234 |   11 |   9 | 1416501411.0 | 
(2 registros)






Atenciosamente, 


Alonso Genis 
Analista de Desenvolvimento 
alo...@planetfone.com.br 







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Re: [asterisk-users] Error saving cdr at h exten in Asterisk13

2014-11-20 Thread Matthew Jordan
On Thu, Nov 20, 2014 at 12:10 PM, Alonso Genis alo...@planetfone.com.br wrote:
 Dears,

 I need to save some information on userfield when calls end in Asterisk13, 
 but I have two error cases:

 1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but 
 userfield is not set.
 2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src 
 e dst correct, and a second line with userfield setting and dst h.

 I am using cdr_odbc.conf, with Asterisk11.14.0 it works fine. May be this is 
 a bug in asterisk13's cdr?




 pabx=# select calldate,src,dst,duration,billsec,uniqueid,userfield from cdr 
 order by calldate desc limit 2;
 calldate| src  | dst  | duration | billsec |   uniqueid   | 
 userfield
 +--+--+--+-+--+---
  2014-11-20 14:37:03-02 | 1901 | h|0 |   0 | 1416501411.0 | 
 teste
  2014-11-20 14:36:51-02 | 1901 | 1234 |   11 |   9 | 1416501411.0 |
 (2 registros)


If you are setting the userfield in the 'h' extension, then this is
what I would expect. CDRs are finalized when the path of communication
between channels is finished; altering the data after that point
updates the next CDR for that channel. It isn't retroactive.

The 'h' extension is special in that 'endbeforehexten' explicitly
ignores updates in 'h' extension. If you disable 'endbeforehexten',
then you will get a CDR for the channel while it updates the hangup
logic - but again, modifications occur on that CDR, not on previous
ones.

If you want the CDR for the channel prior to the 'h' extension to have
a userfield entry, you have to apply it before the channel hangs up.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems

2014-09-22 Thread Larry Moore


On 21/09/2014 9:11 PM, Bart Remmerie wrote:

Dear Joshua,

I don't think this is it:
first, this has been working in the past
second, why would I get a message like the one below exited non-zero
if everything is normal.



Joshua is correct, to reliably process the received fax you will need to 
process it in the 'h' extension.


You may want to refer to this thread 
http://lists.digium.com/pipermail/asterisk-users/2013-June/279625.html, 
note the afax2email script will e-mail the received tiff fax image if 
the PDF conversion fails.



Larry.

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[asterisk-users] error receiving a fax ... but with a fax that was received without problems

2014-09-21 Thread Bart Remmerie
Dear all,

When receiving a fax, the extension is spawned, despite nothing but
positive messages (see below)

The sending fax considers it a success  the verbose output of asterisk
gives a FAX_SUCCESS and a NO_ERROR error in the ReceiveFax command.

The problem is that all the next steps (conversion of the fax to pdf 
sending it to a mailbox) are never executed.  When I do this manually,
there seems nothing wrong with the received file 

Any hints ?

Asterisk output:

*-- *Executing [502@LocalSets:5] *ReceiveFAX*(*DAHDI/4-1*, 
*/tmp/201409211416.tif*) in new stack

*-- *Channel 'DAHDI/4-1' receiving FAX '/tmp/201409211416.tif'

*-- *Channel 'DAHDI/4-1' FAX session '0' started

*-- *FAX handle 0: [ 037.139057 ], entering CLOSING state

*-- *FAX handle 0: [ 037.139097 ], entering CLOSING state

*-- *Channel 'DAHDI/4-1' FAX session '0' is complete, result: 'SUCCESS'
(FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x196', transfer
rate: '14400', remoteSID: '059323634'

*  == *Spawn extension (LocalSets, 502, 5) exited non-zero on 'DAHDI/4-1'



Extensions-snippet

exten = 502,1,Verbose(3,Incoming Fax)

same = n,Set(FAXDEST=/tmp)

same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})

same = n,Verbose(3,- destination: ${FAXDEST}/${tempfax}.tif)

same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)

same = n,Verbose(3,- Fax receipt completed with status:
${FAXSTATUS})

same = n,Verbose(3,converting fax)

same = n,System(/usr/bin/tiff2pdf ${FAXDEST}/${tempfax}.tif -o
${FAXDEST}/${tempfax}.pdf)

same = n,Verbose(3,fax converted ... sending fax)

same = n,System(echo [TC] new fax | mutt -s TC:fax from
${CALLERID} myn...@mymail.com -F /root/.muttrc -a ${FAXDEST}/${tempfax}
.pdf)

same = n,Hangup()
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Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems

2014-09-21 Thread Joshua Colp

Bart Remmerie wrote:

Dear all,


Kia ora,


When receiving a fax, the extension is spawned, despite nothing but
positive messages (see below)

The sending fax considers it a success  the verbose output of asterisk
gives a FAX_SUCCESS and a NO_ERROR error in the ReceiveFax command.

The problem is that all the next steps (conversion of the fax to pdf 
sending it to a mailbox) are never executed.  When I do this manually,
there seems nothing wrong with the received file 

Any hints ?


After the fax is completed the call is hung up so subsequent dialplan 
logic does not execute. You need to place the rest in the 'h' extension 
which is executed upon hangup.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems

2014-09-21 Thread Bart Remmerie
Dear Joshua,

I don't think this is it:
first, this has been working in the past
second, why would I get a message like the one below exited non-zero if
everything is normal.

In the past, all following lines were executed just fine (converting the
tif to a pdf and sending it a gmail mailbox)
I'm not convinced that everything is dropped after the ReceiveFax line,
especially since the Spawn extension message applies to the ReceiveFax
line itself (as if something goes wrong there) = any hints on where/how I
can get a more detailed logging of this ?

* == *Spawn extension (LocalSets, 502, 5) exited non-zero on 'DAHDI/4-1'

On Sun, Sep 21, 2014 at 2:45 PM, Joshua Colp jc...@digium.com wrote:

 Bart Remmerie wrote:

 Dear all,


 Kia ora,

  When receiving a fax, the extension is spawned, despite nothing but
 positive messages (see below)

 The sending fax considers it a success  the verbose output of asterisk
 gives a FAX_SUCCESS and a NO_ERROR error in the ReceiveFax command.

 The problem is that all the next steps (conversion of the fax to pdf 
 sending it to a mailbox) are never executed.  When I do this manually,
 there seems nothing wrong with the received file 

 Any hints ?


 After the fax is completed the call is hung up so subsequent dialplan
 logic does not execute. You need to place the rest in the 'h' extension
 which is executed upon hangup.

 Cheers,

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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-- 
Bart Remmerie
+32 (0477) 78.88.76
remme...@gmail.com
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Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems

2014-09-21 Thread Joshua Colp

Bart Remmerie wrote:

Dear Joshua,

I don't think this is it:
first, this has been working in the past


If the either side didn't hang up (or it took awhile for the hangup to 
be recognized) then the dialplan would continue.



second, why would I get a message like the one below exited non-zero
if everything is normal.


Because the message is there to inform you of what happened. It's not 
indicative of a problem in all cases.




In the past, all following lines were executed just fine (converting the
tif to a pdf and sending it a gmail mailbox)
I'm not convinced that everything is dropped after the ReceiveFax line,
especially since the Spawn extension message applies to the ReceiveFax
line itself (as if something goes wrong there) = any hints on where/how
I can get a more detailed logging of this ?

* == *Spawn extension (LocalSets, 502, 5) exited non-zero on 'DAHDI/4-1'


This means that whatever was executing either encountered an error or 
the channel was hung up, which is when dialplan execution stops. This is 
normal as I previously mentioned.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-19 Thread Mitch Claborn

No, that's not it.  The wording is different.

Mitch




On 08/18/2014 02:28 PM, Paul Greenberg wrote:

Mitch,

Is it the below error?

 if ((fd = open(filename, O_RDONLY))  0) {
 ast_log(LOG_WARNING, Cannot open file '%s' for reading: 
%s\n, filename, strerror(errno));
 return NULL;
 }

Regards,
Paul

From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Mitch Claborn 
mitch...@claborn.net
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Error opening file for reading: Permission denied

Asterisk 12.4

I am seeing message Error opening file for reading: Permission denied
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file.  Is there a way to find out which file is having
trouble?

--

Mitch


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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-19 Thread Matthew Jordan
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net wrote:
 No, that's not it.  The wording is different.


grep doesn't turn up your phrase:

~/projects/12$ grep --include=*.c --include=*.h -r Error opening file .
~/projects/12$

Are you using any 3rd party modules that aren't delivered with Asterisk?

-- 
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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-19 Thread Mitch Claborn

I tried grep too.

No 3rd party modules - this is an out-of-the box download and build.  
I'm guessing that some library function is being called to read a file 
and the error is happening there?


Mitch


On 08/19/2014 02:33 PM, Matthew Jordan wrote:

On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net wrote:

No, that's not it.  The wording is different.


grep doesn't turn up your phrase:

~/projects/12$ grep --include=*.c --include=*.h -r Error opening file .
~/projects/12$

Are you using any 3rd party modules that aren't delivered with Asterisk?




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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-19 Thread Steve Edwards

On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net



No, that's not it.  The wording is different.


Can you run Asterisk via strace? Something like:

sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-19 Thread Mitch Claborn

Grepping the output of the strace revealed this:

stat(/root/.terminfo, 0x7fff8622ed50) = -1 EACCES (Permission denied)
open(/root/.asterisk_history, O_RDONLY) = -1 EACCES (Permission denied)
open(/root/.odbcinst.ini, O_RDONLY)   = -1 EACCES (Permission denied) 
[this one many times]


That must be because I'm starting asterisk as root.   When I su to 
asterisk first, then start it, those above disappear. Problem solved!


Thanks Steve!

Mitch

On 08/19/2014 03:39 PM, Steve Edwards wrote:

On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net



No, that's not it.  The wording is different.


Can you run Asterisk via strace? Something like:

sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk




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[asterisk-users] Error opening file for reading: Permission denied

2014-08-18 Thread Mitch Claborn

Asterisk 12.4

I am seeing message Error opening file for reading: Permission denied 
several times during the asterisk startup (asterisk -cv) but it 
doesn't say which file.  Is there a way to find out which file is having 
trouble?


--

Mitch


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Re: [asterisk-users] Error opening file for reading: Permission denied

2014-08-18 Thread Paul Greenberg
Mitch,

Is it the below error?

if ((fd = open(filename, O_RDONLY))  0) {
ast_log(LOG_WARNING, Cannot open file '%s' for reading: %s\n, 
filename, strerror(errno));
return NULL;
}

Regards,
Paul

From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Mitch Claborn 
mitch...@claborn.net
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Error opening file for reading: Permission denied

Asterisk 12.4

I am seeing message Error opening file for reading: Permission denied
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file.  Is there a way to find out which file is having
trouble?

--

Mitch


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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

2014-08-10 Thread Deepak Rawat
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:

 Hi,

 I modified the query in res/res_config_odbc.c.
 Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
 Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'

 I rebuilt the code and installed Asterisk again. Now static realtime is
 working. Should I file a bug report?


 On Mon, Aug 11, 2014 at 2:37 AM, Deepak Rawat deepaksingh.ra...@gmail.com
  wrote:

 Hi,

 I am using Asterisk 12.4 static realtime. I am storing agents.conf
 related data in MS SQL Server 2012 database. But I keep getting *'LENGTH'
 is not a recognized built-in function name* error message. I debugged
 and found that it's due to the changes done in this ticket:
 https://issues.asterisk.org/jira/browse/ASTERISK-23582. As there is no
 LENGTH function in MS SQL Server, therefore, I am getting this error. Is
 this error due to some config issue at my end?

 Thank you,
 Deepak



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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

2014-08-10 Thread Matthew Jordan
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra...@gmail.com wrote:



 On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
 wrote:

 Hi,

 I modified the query in res/res_config_odbc.c.
 Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
 Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'

 I rebuilt the code and installed Asterisk again. Now static realtime is
 working. Should I file a bug report?


You're more than welcome to; at the same time, the number of people
using MS SQL Server with Asterisk is not tremendously high - at least
when compared with the alternatives. Unfortunately, this is one place
where making things compatible is problematic: MySQL and PostgreSQL
(which are far more likely to be used with Asterisk) along with Oracle
use LENGTH, not LEN.

Your solution, as it is currently, wouldn't be acceptable, as it would
cause far more problems than it would solve. About the only way I
could see solving this would be to make it configurable some place.

Given the relatively few number of people who use MS SQL Server, I
wouldn't expect this issue to receive a lot of attention without a
patch.

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

2014-08-10 Thread Deepak Rawat
Thank you for the response Matthew. It makes perfect sense to cater to the
majority of databases. The changes I did are custom to our setup so we can
use Asterisk 12.4 with MS SQL Server 2012. I wanted to document this error
so anyone else facing it could resolve it. Like we have backslash_is_escape
setting in res_odbc.conf maybe we can provide a database_name setting and
drive the query using this setting. If I get time, I will submit a patch
for review.

On Mon, Aug 11, 2014 at 8:01 AM, Matthew Jordan mjor...@digium.com wrote:

 On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
 deepaksingh.ra...@gmail.com wrote:
 
 
 
  On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat 
 deepaksingh.ra...@gmail.com
  wrote:
 
  Hi,
 
  I modified the query in res/res_config_odbc.c.
  Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
  Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'
 
  I rebuilt the code and installed Asterisk again. Now static realtime is
  working. Should I file a bug report?
 

 You're more than welcome to; at the same time, the number of people
 using MS SQL Server with Asterisk is not tremendously high - at least
 when compared with the alternatives. Unfortunately, this is one place
 where making things compatible is problematic: MySQL and PostgreSQL
 (which are far more likely to be used with Asterisk) along with Oracle
 use LENGTH, not LEN.

 Your solution, as it is currently, wouldn't be acceptable, as it would
 cause far more problems than it would solve. About the only way I
 could see solving this would be to make it configurable some place.

 Given the relatively few number of people who use MS SQL Server, I
 wouldn't expect this issue to receive a lot of attention without a
 patch.

 --
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 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

2014-08-10 Thread Brian LaVallee


On 8/11/14, 11:31, Matthew Jordan wrote:

On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra...@gmail.com wrote:



On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:

Hi,

I modified the query in res/res_config_odbc.c.
Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'

I rebuilt the code and installed Asterisk again. Now static realtime is
working. Should I file a bug report?


You're more than welcome to; at the same time, the number of people
using MS SQL Server with Asterisk is not tremendously high - at least
when compared with the alternatives. Unfortunately, this is one place
where making things compatible is problematic: MySQL and PostgreSQL
(which are far more likely to be used with Asterisk) along with Oracle
use LENGTH, not LEN.


Another option, just add LENGTH as a user-defined function to MS SQL.

||-- Something like this...|
CREATE FUNCTION LENGTH (@input)
||BEGIN
   RETURN ( LEN(@input) )
END|




Your solution, as it is currently, wouldn't be acceptable, as it would
cause far more problems than it would solve. About the only way I
could see solving this would be to make it configurable some place.

Given the relatively few number of people who use MS SQL Server, I
wouldn't expect this issue to receive a lot of attention without a
patch.






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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

2014-08-10 Thread Deepak Rawat
I tried that but for some reason couldn't get it working. My custom
function was getting created in dbo and the asterisk kept throwing error.
Maybe some mistake on my part. I will try it again.


On Mon, Aug 11, 2014 at 10:44 AM, Brian LaVallee b.laval...@globaltank.jp
wrote:


 On 8/11/14, 11:31, Matthew Jordan wrote:

 On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
 deepaksingh.ra...@gmail.com wrote:



 On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat 
 deepaksingh.ra...@gmail.com
 wrote:

 Hi,

 I modified the query in res/res_config_odbc.c.
 Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
 Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'

 I rebuilt the code and installed Asterisk again. Now static realtime is
 working. Should I file a bug report?

  You're more than welcome to; at the same time, the number of people
 using MS SQL Server with Asterisk is not tremendously high - at least
 when compared with the alternatives. Unfortunately, this is one place
 where making things compatible is problematic: MySQL and PostgreSQL
 (which are far more likely to be used with Asterisk) along with Oracle
 use LENGTH, not LEN.


 Another option, just add LENGTH as a user-defined function to MS SQL.

 ||-- Something like this...|
 CREATE FUNCTION LENGTH (@input)
 ||BEGIN
RETURN ( LEN(@input) )
 END|




 Your solution, as it is currently, wouldn't be acceptable, as it would
 cause far more problems than it would solve. About the only way I
 could see solving this would be to make it configurable some place.

 Given the relatively few number of people who use MS SQL Server, I
 wouldn't expect this issue to receive a lot of attention without a
 patch.





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Re: [asterisk-users] error cant write to function ODBC_DEVICES

2013-10-23 Thread Rusty Newton
On Sun, Oct 20, 2013 at 6:12 AM, Israel Gottlieb isr...@gmail.com wrote:

snip

 but from the dialplan gives me a error  cant write to function
 ODBC_DEVICES

 happy to hear any ideas

I don't use func_odbc on a regular basis, but from looking at the
sample file and looking at the functions provided within Asterisk. The
ODBC_DEVICES function does not exist.

The three functions available for configuration within func_odbc.conf
appear to be ODBC_SQL,ODBC_ANTIGF,ODBC_PRESENCE

Also the only examples of the string ODBC_DEVICES out on the web
according to Google show up at the various forums you have asked about
it. :)

So.. can't write to function is definitely expected behavior.

Hope that helps!

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Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] error cant write to function ODBC_DEVICES

2013-10-23 Thread isrlgb
Thanks for replying (I only asked on this list)

Whatever function you add to that file becomes a function and that was a odbc 
function I added

Anyhow after a restart of asterisk it started working ok

It worked like a charm (I had more than 5 inserts to a database within a 
few hours) 
-Original Message-
From: Rusty Newton rnew...@digium.com
Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 23 Oct 2013 12:15:50 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] error cant write to function ODBC_DEVICES

On Sun, Oct 20, 2013 at 6:12 AM, Israel Gottlieb isr...@gmail.com wrote:

snip

 but from the dialplan gives me a error  cant write to function
 ODBC_DEVICES

 happy to hear any ideas

I don't use func_odbc on a regular basis, but from looking at the
sample file and looking at the functions provided within Asterisk. The
ODBC_DEVICES function does not exist.

The three functions available for configuration within func_odbc.conf
appear to be ODBC_SQL,ODBC_ANTIGF,ODBC_PRESENCE

Also the only examples of the string ODBC_DEVICES out on the web
according to Google show up at the various forums you have asked about
it. :)

So.. can't write to function is definitely expected behavior.

Hope that helps!

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] error cant write to function ODBC_DEVICES

2013-10-23 Thread Rusty Newton
On Wed, Oct 23, 2013 at 12:27 PM,  isr...@gmail.com wrote:
 Thanks for replying (I only asked on this list)

 Whatever function you add to that file becomes a function and that was a odbc 
 function I added

 Anyhow after a restart of asterisk it started working ok

 It worked like a charm (I had more than 5 inserts to a database within a 
 few hours)

You can tell I haven't used func_odbc in ages. :) Glad you figured it
out despite me misleading you unwittingly. An Asterisk restart! It is
usually the simple things.

-- 
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Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] error cant write to function ODBC_DEVICES

2013-10-20 Thread Israel Gottlieb
Hi all

asterisk 1.8.23

I have odbc all setup to mysql but cant figure out why the dialplan wont
write to the odbc function

fubc_odbc.conf

[DEVICES]
dsn=device-conn;dsn in res_odbc not odbc.ini
readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM
call INNER JOIN devices ON call.id = devices.callId WHERE deviceNumber = '${
SQL_ESC(${ARG1})}'

writesql=insert into voted (callId,callNum,city,deviceId,SerialNum,
serverResponse) values (${VAL1},${VAL2},${VAL3},${VAL4},${VAL5},${VAL6}


extension.conf

the relevant line


same = n,set(ODBC_DEVICES()=${callid},${call},1,${deviceid},${num},${
serverupdate})


when sending the values from the cli using odbc write it works ok
reading from the dialplan  works ok
i tried sending plain values without variables

but from the dialplan gives me a error  cant write to function ODBC
_DEVICES

happy to hear any ideas
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Re: [asterisk-users] Error 488 Not Acceptable Here

2013-05-24 Thread Karsten Wemheuer
Hi,

Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker:
 Am 22.05.2013 16:39, schrieb Andrew Colin:
  Hi guys,
  
  Any idea why I am getting this error when someone tries to send me a T38
  Fax?
 Hi,
 
 Maybe you have not allowed T.38 as acceptable codec ;-)
 You can try with allow=all in your sip.conf.

No, T.38 is not a codec and so allow=all will not help.

To use T.38 You have to enable T.38 with t38pt_udptl = yes in
sip.conf.

The reason, why You get a 488 Not Acceptable Here488 Not Acceptable
Here, is only detectable with a SIP Trace. There are many reasons e.g.
- Your asterisk version does not support T.38
- T.38 is not enabled (see above)
- T.38 is enabled, but not at the relevant peers (in most versions of
asterisk there is only support for T.38 passthrough, so both peers have
to support T.38)
- There are problems in the transmission and some peers wants to switch
back to audio level and the other or asterisk is not willing to support
this.
The last reason may occur, if You have NAT and do not correctly forward
the data ports of T.38 (UDPTL Ports).

Best way is to get a SIP Trace to analyse. If You provide one, You
should also tell, which version of asterisk.

HTH,

Karsten



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Re: [asterisk-users] Error 488 Not Acceptable Here

2013-05-23 Thread Maximilian Grobecker
Hi,

Maybe you have not allowed T.38 as acceptable codec ;-)
You can try with allow=all in your sip.conf.


Am 22.05.2013 16:39, schrieb Andrew Colin:
 Hi guys,
 
 Any idea why I am getting this error when someone tries to send me a T38
 Fax?
 
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-- 


--
- Portunity GmbH - Werner-Seelenbinder-Str. 23
-- 42477 Radevormwald - Germany
-
- Portal:  http://www.portunity.de
-
- General: Phone: +49 (0)202 - 69555 - 0
-  eMail/SIP: i...@portunity.de
-  Fax:   +49 (0)202 - 69555 - 190
-
- Support: Phone: +49 (0)202 - 69555 - 300
-  eMail/SIP: supp...@portunity.de
-
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Re: [asterisk-users] Error 488 Not Acceptable Here

2013-05-23 Thread Gopalakrishnan N
488 not acceptable is due to codec error. Make sure you have right codec in
place between the end points.


On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker 
m.grobec...@portunity.de wrote:

 Hi,

 Maybe you have not allowed T.38 as acceptable codec ;-)
 You can try with allow=all in your sip.conf.


 Am 22.05.2013 16:39, schrieb Andrew Colin:
  Hi guys,
 
  Any idea why I am getting this error when someone tries to send me a T38
  Fax?
 
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 --


 --
 - Portunity GmbH - Werner-Seelenbinder-Str. 23
 -- 42477 Radevormwald - Germany
 -
 - Portal:  http://www.portunity.de
 -
 - General: Phone: +49 (0)202 - 69555 - 0
 -  eMail/SIP: i...@portunity.de
 -  Fax:   +49 (0)202 - 69555 - 190
 -
 - Support: Phone: +49 (0)202 - 69555 - 300
 -  eMail/SIP: supp...@portunity.de
 -
 - Amtsgericht Koeln HRB 38162
 - USt-Identnummer DE206277867
 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell
 --

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[asterisk-users] Error 488 Not Acceptable Here

2013-05-22 Thread Andrew Colin

Hi guys,

Any idea why I am getting this error when someone tries to send me a T38 
Fax?


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Re: [asterisk-users] ERROR: Unknown signalling method ss7

2013-03-17 Thread mohsen feyzzadeh


 libss7-trunk cannot be used with any released version of Asterisk.

Thank you Richard.
I could install libss7-branche 1.0 successfully.
But i faced another problem. When i`m dialling over my SS7 enabled E1, 
caller-id did not presentaded correctly on the destination phone.
I`m using these two method to change screeningPresentation parameters of IAM 
message:

1. 
exten = _X.,1,Dial(DAHDI/g1/${EXTEN}, 10, u(allowed)f(75462541))

2.
exten = _X.,1,Set(CALLERID(num)=75462541)
exten = _X.,n,Set(CALLERID(pres)=allowed)
;exten = _X.,n,Set(CALLERID(num-pres)=allowed) 
exten = _X.,n,Dial(DAHDI/g1/${EXTEN})

With both method, in the outgoing IAM message from asterisk to Telecom, 
PresentationScreening are as follow but caller-id on the destination phone is 
''.

[1]         Calling Party Number:
[1]             Nature of address: 3
[1]             NI: 0
[1]             Numbering plan: 1
[1]             Presentation: 0
[1]             Screening: 3
[1]             Address signals: 75462541

Can you please guide me?
Best Regards.


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[asterisk-users] ERROR: Unknown signalling method ss7

2013-03-14 Thread mohsen feyzzadeh
Hi all
I installed 
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.

Now i`m unable to load chan_dahdi and libss7:

myserver*CLI module load chan_dahdi.so
 ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 
'ss7' at line 37.

myserver*CLI module load libss7.so
Unable to load module libss7.so
Command 'module load libss7.so' failed.
[Mar 14 22:30:05] WARNING[10124]: loader.c:423 load_dynamic_module: Module 
'libss7.so' did not register itself during load
[Mar 14 22:30:05] WARNING[10124]: loader.c:878 load_resource: Module 
'libss7.so' could not be loaded.

what is the problem? Can you please help me to solve this problem?

Here is my config
 files:

system.conf:
=
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
mtp2=16
#dchan=16 

loadzone    = us
defaultzone = us
==


chan_dahdi.conf:
===

[trunkgroups]

[channels]

callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes

;General options
usecallerid = yes
hidecallerid = no
callwaiting = yes
threewaycalling = yes
transfer = yes
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0

switchtype = national
group = 1
signalling = ss7
ss7type = itu
linkset = 1
ss7type = itu
linkset = 1
pointcode = 
adjpointcode = 
defaultdpc = 
cicbeginswith = 1
channel = 1-15
cicbeginswith = 17
channel =
 17-31
sigchan = 16
==

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Re: [asterisk-users] ERROR: Unknown signalling method ss7

2013-03-14 Thread Richard Mudgett
 I installed
 DAHDI Version - 2.6.1
 DAHDI Tools Version - 2.6.1
 libss7-trunk
 Asterisk 11.0.1
 from source on Fedora 12 x86_64.
 
 Now i`m unable to load chan_dahdi and libss7:
 
 myserver*CLI module load chan_dahdi.so
 ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling
 method 'ss7' at line 37.
 
 myserver*CLI module load libss7.so
 Unable to load module libss7.so
 Command 'module load libss7.so' failed.
 [Mar 14 22:30:05] WARNING[10124]: loader.c:423 load_dynamic_module:
 Module 'libss7.so' did not register itself during load
 [Mar 14 22:30:05] WARNING[10124]: loader.c:878 load_resource: Module
 'libss7.so' could not be loaded.
 
 what is the problem? Can you please help me to solve this problem?

libss7-trunk cannot be used with any released version of Asterisk.

Richard

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Re: [asterisk-users] ERROR: Unknown signalling method ss7

2013-03-14 Thread Patrick Lists

On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote:

Hi all
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.


In case the 12 in Fedora 12 was not a typo, you do realize that Fedora 
12 has been end-of-line for years and has more security holes than Swiss 
cheese? It makes sense to upgrade to the latest version of Fedora (which 
is 18) or switch to CentOS 6.4 which is more suited for server 
applications. You may also want to look at the latest versions of DAHDI 
(2.6.2/2.6.3rc) and Asterisk (11.2.1) assuming both work with an 
appropriate version of libss7.


Regards,
Patrick


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Re: [asterisk-users] Error to install Asterisk‏

2013-03-07 Thread Thorsten Göllner

That should be ok.

Try the following: open 2 shells. In the first one type watch df -h. 
In the second one you start the compilation. While compilation is 
running watch the first shell. The given command refreshes all 2 seconds 
the display and shows the used/free disk space. _Perhaps_ it will give 
you a hint, what mount point is running out of space.


Am 06.03.2013 15:41, schrieb termo termosel:

I have executed make in the same console where I had written

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Is this way ok?


Date: Wed, 6 Mar 2013 14:25:50 +0100
From: t...@ovm-group.com
To: fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Did you execute the make command in the same environment so that 
make really uses the TMPDIR directory? (no su or other shell)


Am 06.03.2013 13:37, schrieb termo termosel:

Hi,

the same error, I write your commands:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

but the same error happens

/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
make[1]: se sale del directorio
«/home/ubuntu/Downloads/asterisk-11.2.1»
make: *** [_cleantest_all] Error 2

Jordi


Date: Wed, 6 Mar 2013 13:29:24 +0100
From: t...@ovm-group.com mailto:t...@ovm-group.com
To: fermit...@hotmail.com mailto:fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Try to set the tmp variable. In your case:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Am 06.03.2013 13:20, schrieb termo termosel:

Hi,

I read it but I don't find the solution. How Can I alocate
more free space in tmp?

Thanks,
Jordi


Date: Wed, 6 Mar 2013 13:12:34 +0100
From: t...@ovm-group.com mailto:t...@ovm-group.com
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
CC: fermit...@hotmail.com mailto:fermit...@hotmail.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Take a look here:

http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G 8,7G  34% /
udev  999M  4,0K 999M   1% /dev
tmpfs 403M  860K 402M   1% /run
/dev/sdb1 799M  693M 106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0 5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi



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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread termo termosel
Hi,

this is the outpu to df -h command:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi

From: fermit...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Mar 2013 17:40:32 +
Subject: Re: [asterisk-users] Error to install Asterisk




Hi,
 
Ok, tomorrow I will send the output when I will be in the office!
 
Thanks!
 
 From: asterisk_l...@earthshod.co.uk
 To: asterisk-users@lists.digium.com
 Date: Tue, 5 Mar 2013 16:11:01 +
 Subject: Re: [asterisk-users] Error to install Asterisk
 
 On Tuesday 05 March 2013, termo termosel wrote:
  Hi,
  when I try to install Asterisk 11.2.1 the console return error which it
  tells: /usr/bin/ld: final link failed: No space left on device
  and the process exits installation.
  How can I solve this problem? Tmp folder is empty.
  Thanks,Jordi
 
 Try entering this command:
 # df -h
 and paste the complete output in a message.
 
 This will show the amount of space used and remaining on all filesystems, in 
 human-readable notation  (i.e. it will automatically select the units: bytes, 
 kilo, mega, giga or terabytes, so as to get a sensible figure).
 
 You'll almost certainly have to move some files out of the way.  Have you 
 got, 
 or can you get, a USB external HDD; which either already has a Linux ext4 
 file 
 system on it, or contains only sacrificial data?  
 
 -- 
 AJS
 
 Answers come *after* questions.
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
  

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   http://lists.digium.com/mailman/listinfo/asterisk-users  
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Re: [asterisk-users] Error to install Asterisk‏

2013-03-06 Thread Thorsten Göllner

Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi




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Re: [asterisk-users] Error to install Asterisk‏

2013-03-06 Thread Thorsten Göllner

Try to set the tmp variable. In your case:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Am 06.03.2013 13:20, schrieb termo termosel:

Hi,

I read it but I don't find the solution. How Can I alocate more free 
space in tmp?


Thanks,
Jordi


Date: Wed, 6 Mar 2013 13:12:34 +0100
From: t...@ovm-group.com
To: asterisk-users@lists.digium.com
CC: fermit...@hotmail.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Take a look here:
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi



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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Error to install Asterisk‏

2013-03-06 Thread termo termosel
Hi,

the same error, I write your commands:

mkdir /var/ext_tmp

export TMPDIR=/var/ext_tmp

make

but the same error happens

/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1»
make: *** [_cleantest_all] Error 2

Jordi

Date: Wed, 6 Mar 2013 13:29:24 +0100
From: t...@ovm-group.com
To: fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]  Error to install Asterisk‏


  

  
  
Try to set the tmp variable. In your case:



mkdir /var/ext_tmp

export TMPDIR=/var/ext_tmp

make



Am 06.03.2013 13:20, schrieb termo termosel:


  
  Hi,



I read it but I don't find the solution. How Can I alocate more
free space in tmp?



Thanks,

Jordi




  Date: Wed, 6 Mar 2013 13:12:34 +0100

  From: t...@ovm-group.com

  To: asterisk-users@lists.digium.com

  CC: fermit...@hotmail.com

  Subject: Re: [asterisk-users] Error to install Asterisk‏

  

  Take a look here:

  
http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

  

  Am 06.03.2013 13:00, schrieb
termo termosel:

  
  

  
  Hi,



df -h output:




root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
df -h

S.ficherosTam.  Usado Disp. % Uso
Montado en

/cow   14G  4,5G  8,7G  34% /

udev  999M  4,0K  999M   1% /dev

tmpfs 403M  860K  402M   1% /run

/dev/sdb1 799M  693M  106M  87%
/cdrom

/dev/loop0668M  668M 0 100%
/rofs

tmpfs1006M   44K 1006M   1% /tmp

none  5,0M 0  5,0M   0%
/run/lock

none 1006M  100K 1006M   1%
/run/shm



Jordi

  
  

  

  
  

  
  


  

  
  


  
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_
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Re: [asterisk-users] Error to install Asterisk‏

2013-03-06 Thread Thorsten Göllner
Did you execute the make command in the same environment so that make 
really uses the TMPDIR directory? (no su or other shell)


Am 06.03.2013 13:37, schrieb termo termosel:

Hi,

the same error, I write your commands:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

but the same error happens

/usr/bin/ld: final link failed: No space left on device
collect2: ld devolvió el estado de salida 1
make[2]: *** [asterisk] Error 1
make[1]: *** [main] Error 2
make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1»
make: *** [_cleantest_all] Error 2

Jordi


Date: Wed, 6 Mar 2013 13:29:24 +0100
From: t...@ovm-group.com
To: fermit...@hotmail.com
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Try to set the tmp variable. In your case:

mkdir /var/ext_tmp
export TMPDIR=/var/ext_tmp
make

Am 06.03.2013 13:20, schrieb termo termosel:

Hi,

I read it but I don't find the solution. How Can I alocate more
free space in tmp?

Thanks,
Jordi


Date: Wed, 6 Mar 2013 13:12:34 +0100
From: t...@ovm-group.com mailto:t...@ovm-group.com
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
CC: fermit...@hotmail.com mailto:fermit...@hotmail.com
Subject: Re: [asterisk-users] Error to install Asterisk‏

Take a look here:

http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device

Am 06.03.2013 13:00, schrieb termo termosel:

Hi,

df -h output:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1#
mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G 34% /
udev  999M  4,0K  999M 1% /dev
tmpfs 403M  860K  402M 1% /run
/dev/sdb1 799M  693M  106M 87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M 1% /tmp
none  5,0M 0  5,0M 0% /run/lock
none 1006M  100K 1006M 1% /run/shm

Jordi



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_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Carlos Alvarez
I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact.  If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers.  You really need to know how to use the
OS before you can learn to manage services running on it.  I strongly
suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is
simply no way to run a production server without having to do systems
management regularly.


On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.comwrote:

 Hi,

 this is the outpu to df -h command:

 root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
 S.ficherosTam.  Usado Disp. % Uso Montado en
 /cow   14G  4,5G  8,7G  34% /
 udev  999M  4,0K  999M   1% /dev
 tmpfs 403M  860K  402M   1% /run
 /dev/sdb1 799M  693M  106M  87% /cdrom
 /dev/loop0668M  668M 0 100% /rofs
 tmpfs1006M   44K 1006M   1% /tmp
 none  5,0M 0  5,0M   0% /run/lock
 none 1006M  100K 1006M   1% /run/shm

 Jordi

 --
 From: fermit...@hotmail.com
 To: asterisk-users@lists.digium.com
 Date: Tue, 5 Mar 2013 17:40:32 +

 Subject: Re: [asterisk-users] Error to install Asterisk

 Hi,

 Ok, tomorrow I will send the output when I will be in the office!

 Thanks!

  From: asterisk_l...@earthshod.co.uk
  To: asterisk-users@lists.digium.com
  Date: Tue, 5 Mar 2013 16:11:01 +
  Subject: Re: [asterisk-users] Error to install Asterisk
 
  On Tuesday 05 March 2013, termo termosel wrote:
   Hi,
   when I try to install Asterisk 11.2.1 the console return error which it
   tells: /usr/bin/ld: final link failed: No space left on device
   and the process exits installation.
   How can I solve this problem? Tmp folder is empty.
   Thanks,Jordi
 
  Try entering this command:
  # df -h
  and paste the complete output in a message.
 
  This will show the amount of space used and remaining on all
 filesystems, in
  human-readable notation (i.e. it will automatically select the units:
 bytes,
  kilo, mega, giga or terabytes, so as to get a sensible figure).
 
  You'll almost certainly have to move some files out of the way. Have you
 got,
  or can you get, a USB external HDD; which either already has a Linux
 ext4 file
  system on it, or contains only sacrificial data?
 
  --
  AJS
 
  Answers come *after* questions.
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 -- _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
 to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
 or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Gertjan Baarda
Couldn't agree more, Carlos. But then again, haven't we all started this
way? ;-) The best way to understand Linux is learning the hard way. After
all, it takes a genius to understand the simplicity of Linux.

Sent from my iPhone

On 6 mrt. 2013, at 17:53, Carlos Alvarez car...@televolve.com wrote:

I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact.  If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers.  You really need to know how to use the
OS before you can learn to manage services running on it.  I strongly
suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is
simply no way to run a production server without having to do systems
management regularly.


On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.comwrote:

 Hi,

 this is the outpu to df -h command:

 root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
 S.ficherosTam.  Usado Disp. % Uso Montado en
 /cow   14G  4,5G  8,7G  34% /
 udev  999M  4,0K  999M   1% /dev
 tmpfs 403M  860K  402M   1% /run
 /dev/sdb1 799M  693M  106M  87% /cdrom
 /dev/loop0668M  668M 0 100% /rofs
 tmpfs1006M   44K 1006M   1% /tmp
 none  5,0M 0  5,0M   0% /run/lock
 none 1006M  100K 1006M   1% /run/shm

 Jordi

 --
 From: fermit...@hotmail.com
 To: asterisk-users@lists.digium.com
 Date: Tue, 5 Mar 2013 17:40:32 +

 Subject: Re: [asterisk-users] Error to install Asterisk

 Hi,

 Ok, tomorrow I will send the output when I will be in the office!

 Thanks!

  From: asterisk_l...@earthshod.co.uk
  To: asterisk-users@lists.digium.com
  Date: Tue, 5 Mar 2013 16:11:01 +
  Subject: Re: [asterisk-users] Error to install Asterisk
 
  On Tuesday 05 March 2013, termo termosel wrote:
   Hi,
   when I try to install Asterisk 11.2.1 the console return error which it
   tells: /usr/bin/ld: final link failed: No space left on device
   and the process exits installation.
   How can I solve this problem? Tmp folder is empty.
   Thanks,Jordi
 
  Try entering this command:
  # df -h
  and paste the complete output in a message.
 
  This will show the amount of space used and remaining on all
 filesystems, in
  human-readable notation (i.e. it will automatically select the units:
 bytes,
  kilo, mega, giga or terabytes, so as to get a sensible figure).
 
  You'll almost certainly have to move some files out of the way. Have you
 got,
  or can you get, a USB external HDD; which either already has a Linux
 ext4 file
  system on it, or contains only sacrificial data?
 
  --
  AJS
 
  Answers come *after* questions.
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 -- _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
 to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
 or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Carlos Alvarez
TelEvolve
602-889-3003

 --
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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Carlos Alvarez
On Wed, Mar 6, 2013 at 10:02 AM, Gertjan Baarda gertjan.baa...@gmail.comwrote:

 Couldn't agree more, Carlos. But then again, haven't we all started this
 way? ;-) The best way to understand Linux is learning the hard way. After
 all, it takes a genius to understand the simplicity of Linux.


If you're going to learn Linux, then learn it, not via some service running
on it.  It's clear in context that the original poster believes that he can
install and run Asterisk without knowing the OS.  This is obviously not
true.  If it's going to be someone's production server, that is scary.  It
also has led to many ASTERISK SUCKS! discussions I've had because there
were problems at the OS level that made the Asterisk server unreliable.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Alex Villací­s Lasso

El 06/03/13 11:52, Carlos Alvarez escribió:
I'm going to make an observation here that may upset you, and I don't mean it to, but it's fact.  If you are so unfamiliar with Linux, you will have a bad time managing Asterisk servers.  You really need to know how to use the OS before you can learn to 
manage services running on it.  I strongly suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is simply no way to run a production server without having to do systems management regularly.



On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.com 
mailto:fermit...@hotmail.com wrote:

Hi,

this is the outpu to df -h command:

root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
S.ficherosTam.  Usado Disp. % Uso Montado en
/cow   14G  4,5G  8,7G  34% /
udev  999M  4,0K  999M   1% /dev
tmpfs 403M  860K  402M   1% /run
/dev/sdb1 799M  693M  106M  87% /cdrom
/dev/loop0668M  668M 0 100% /rofs
tmpfs1006M   44K 1006M   1% /tmp
none  5,0M 0  5,0M   0% /run/lock
none 1006M  100K 1006M   1% /run/shm

Jordi


Look carefully at the df -h output. It seems that the OP is trying to install 
and run asterisk from inside an Ubuntu livecd session. Whatever the result of 
the installation, it will be wiped out on the next restart.
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[asterisk-users] Error to install Asterisk

2013-03-05 Thread termo termosel
Hi,
when I try to install Asterisk 11.2.1 the console return error which it tells:
/usr/bin/ld: final link failed: No space left on device
and the process exits installation.
How can I solve this problem? Tmp folder is empty.
Thanks,Jordi  --
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Re: [asterisk-users] Error to install Asterisk

2013-03-05 Thread Gertjan Baarda
Is there enough space on the device left?
Check this with: du -sh
--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel fermit...@hotmail.comwrote:

 Hi,

 when I try to install Asterisk 11.2.1 the console return error which it
 tells:

 /usr/bin/ld: final link failed: No space left on device

 and the process exits installation.

 How can I solve this problem? Tmp folder is empty.

 Thanks,
 Jordi

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Re: [asterisk-users] Error to install Asterisk

2013-03-05 Thread termo termosel
Hi,
if I write du -sh the response is 271M. I don't know that it means.
Thanks,Jordi

From: gertjan.baa...@gmail.com
Date: Tue, 5 Mar 2013 14:29:37 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk

Is there enough space on the device left? Check this with: du -sh--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel fermit...@hotmail.com wrote:





Hi,
when I try to install Asterisk 11.2.1 the console return error which it tells:
/usr/bin/ld: final link failed: No space left on device


and the process exits installation.
How can I solve this problem? Tmp folder is empty.
Thanks,Jordi  

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Re: [asterisk-users] Error to install Asterisk

2013-03-05 Thread Gertjan Baarda
It means 271 megabyte. Can you post the complete output?

Sent from my iPhone

On 5 mrt. 2013, at 15:31, termo termosel fermit...@hotmail.com wrote:

 Hi,

if I write du -sh the response is 271M. I don't know that it means.

Thanks,
Jordi

--
From: gertjan.baa...@gmail.com
Date: Tue, 5 Mar 2013 14:29:37 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk

Is there enough space on the device left?
Check this with: du -sh
--Gertjan

On Tue, Mar 5, 2013 at 2:20 PM, termo termosel fermit...@hotmail.comwrote:

Hi,

when I try to install Asterisk 11.2.1 the console return error which it
tells:

/usr/bin/ld: final link failed: No space left on device

and the process exits installation.

How can I solve this problem? Tmp folder is empty.

Thanks,
Jordi

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Re: [asterisk-users] Error to install Asterisk

2013-03-05 Thread Hans Witvliet
-Original Message-
From: termo termosel fermit...@hotmail.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Error to install Asterisk
Date: Tue, 5 Mar 2013 14:30:05 +

Hi,


if I write du -sh the response is 271M. I don't know that it means.


Thanks,
Jordi



-Original Message-

Hi Jordi,

The du utility will show you the Disk Utilisation (hence the
abbriviation du)

What might be more relevant, is how much space is free.
That you can examine with: df -h


hw


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Re: [asterisk-users] Error to install Asterisk

2013-03-05 Thread Gertjan Baarda
I stand corrected..

On Tue, Mar 5, 2013 at 4:58 PM, Hans Witvliet aster...@a-domani.nl wrote:

 -Original Message-
 From: termo termosel fermit...@hotmail.com
 Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Error to install Asterisk
 Date: Tue, 5 Mar 2013 14:30:05 +

 Hi,


 if I write du -sh the response is 271M. I don't know that it means.


 Thanks,
 Jordi



 -Original Message-

 Hi Jordi,

 The du utility will show you the Disk Utilisation (hence the
 abbriviation du)

 What might be more relevant, is how much space is free.
 That you can examine with: df -h


 hw


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