Re: [asterisk-users] Cisco 7941G & Auth

[Sasa]

"Jonathan Thurman" wrote:
> What does your SEP.cnf.xml file look like?  In my experience,
> the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I 
> had
> to specify the firmware version in each SEP file.  I am using 8-4-4S, but
> for you this would be something like this:
>
> 
> 
> SIP41.8-0-2SR1S
> 
> 

Hi, I have already writed also in SEP.cnf.xml file (other at 
XMLDefault.cnf.xml file) the parameter:

SIP41.8-0-2SR1S

..but the problem isn't resolved !.
Can I try to change some parameters ?..are desperate ! I think I have tried 
everything !
Thanks.

--

   Salvatore.



- Original Message - 
From: "Jonathan Thurman" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, June 19, 2009 6:04 PM
Subject: Re: [asterisk-users] Cisco 7941G & Auth


> What does your SEP.cnf.xml file look like?  In my experience,
> the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I 
> had
> to specify the firmware version in each SEP file.  I am using 8-4-4S, but
> for you this would be something like this:
>
> 
> 
> SIP41.8-0-2SR1S
> 
> 
>
>
> And you shouldn't need the tlv file.
>
> -Jonathan
>
>
>
> On Fri, Jun 19, 2009 at 8:25 AM, Sasa  wrote:
>
>> "David Gibbons" wrote:
>> > I've found that different types of TFTP servers return differing errors
>> > when a file doesn't exist. You don't need the TLV file, but >you do 
>> > need
>> a
>> > distro that tells the phone it's not there correctly. I have not had 
>> > ANY
>> > luck with windows tftp servers, only linux.
>>
>> I have tried with tftp on linux machine but the result isn't changed.
>> Thanks.
>>
>> --
>>
>>   Salvatore.
>>
>>
>>
>> - Original Message -
>> From: "David Gibbons" 
>> To: ; "'Asterisk Users MailingList - Non-Commercial
>> Discussion'" 
>> Sent: Friday, June 19, 2009 4:50 PM
>> Subject: Re: [asterisk-users] Cisco 7941G & Auth
>>
>>
>> > I've found that different types of TFTP servers return differing errors
>> > when a file doesn't exist. You don't need the TLV file, but you do need 
>> > a
>> > distro that tells the phone it's not there correctly. I have not had 
>> > ANY
>> > luck with windows tftp servers, only linux.
>> >
>> > -Dave
>> >
>> > -Original Message-
>> > From: asterisk-users-boun...@lists.digium.com
>> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John
>> Novack
>> > Sent: Friday, June 19, 2009 10:38 AM
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > Subject: Re: [asterisk-users] Cisco 7941G & Auth
>> >
>> >
>> >
>> > Sasa wrote:
>> >> Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with 
>> >> Cisco
>> >> 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my
>> >> problem
>> >> is that Cisco phone isn't authenticated on Asterisk.
>> >> In tftp directory I have:
>> >>
>> >> apps41.1-1-1-15.sbn
>> >> cnu41.3-1-1-15.sbn
>> >> copstart.sh
>> >> cvm41sip.8-0-1-18.sbn
>> >> dialplan.xml
>> >> dsp41.1-1-1-15.sbn
>> >> jar41sip.8-0-1-18.sbn
>> >> load115
>> >> load308
>> >> load309
>> >> load30018
>> >> SIP41.8-0-2SR1S.loads
>> >> term41.default.loads
>> >> term61.default.loads
>> >> XMLDefault.cnf
>> >> SEPmac_address.cnf.xml
>> >>
>> >> ..and in tftp log I have:
>> >>
>> >> Connection received from 192.168.1.61 on port 49153 [19/06 
>> >> 10:16:35.968]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:35.968]
>> >> File  : error 2 in system call CreateFile
>> >> Impossibile
>> >> trovare il file specificato. [19/06 10:16:35.968]
>> >> Connection received from 192.168.1.61 on port 49154 [19/06 
>> >> 10:16:36.109]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:36.109]
>> >> Using local port 3995 [19/06 10:16:36.109]
>> >> : sent 15 blks, 7239 bytes in 0 s. 0 blk 
>> >> resent
>> >> [19/06 10:16:36.171]
>> >> Connection received from 192.168.1.61 on port 49155 [19/06 
>> >> 10:16:40.046]
>> >> Read request for file . Mode octet [19/06 10:16:40.046]
>> >> File <\mk-sip.jar> : error 2 in system call CreateFile Impossibile
>> >> trovare
>> >> il file specificato. [19/06 10:16:40.046]
>> >> Connection received from 192.168.1.61 on port 49156 [19/06 
>> >> 10:16:40.984]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:40.999]
>> >> File  : error 3 in system call CreateFile
>> Impossibile
>> >> trovare il percorso specificato. [19/06 10:16:40.999]
>> >> Connection received from 192.168.1.61 on port 49164 [19/06 
>> >> 10:16:42.843]
>> >> Read request for file . Mode octet [19/06 10:16:42.859]
>> >> Using local port 3998 [19/06 10:16:42.859]
>> >> : sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06
>> >> 10:16:42.906]
>> >>
>> >> In XMLDefault.cnf I have:
>> >>
>> >> SIP41.8-0-2SR1S
>> >>
>> >> ..and on 7941G I have:
>> >>
>> >> App Load IDjar41sip.8-0-1-18.sbn
>> >> Boot Load ID7941G_64-02070631Amd64megRel.bin
>> >> VersionSIP41.8-0-2SR1S
>> >>
>> >> Thanks.
>> >>
>> >> --
>> >>
>> >>Salvatore.
>> >>
>> >>
>> > I have had sucess wit

Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi Moy,
many thanks for clarifying. I'll do some further investigations about it 
and I'll post the result here.

Regards
Jose


Moises Silva escribió:

On Fri, Jun 19, 2009 at 5:32 AM, Jose Arias wrote:
  

Hi Moy,

I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI
patch, which fixes a bug about stopping AsyncAGI applications, as may be you
can recall from the thread [asterisk-users] async agi question in
http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html.

This patched asterisk works fine and it stops the async agi applications
launched from the AsyncAGI loop before the Redirect as it's expected. It's
for that I don't think stopping the mixmonitor application launched from the
AsyncAGI loop would be a bug if I redirect the call. I would be only getting
the same behavior than I got with the stream file application as you
explained it should be at
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365

I'm only asking if there's any way to prevent stopping applications launched
on a channel from the AsyncAGI loop if this channel is redirected afterward,
with something like a continue_running_in_background flag in the previous
AGI invocation from AMI. Of course, it bring us the problem we'll need some
kind of identifier and some stop action to be able to stop those
applications running in background launched from the AsyncAGI loop

Anyway, as you asked me some days ago, I have published at
http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple configuration
and a simple scenario in order you can try to reproduce what I'm saying.

I don't need anyone to do anything for me. I'm willing to do the work, I
like programming and trying new things as well, but I'll need some
guidelines to go straight ahead.




Jose, the thing is that MixMonitor IS a background application in
nature, that's why I say is unexpected that after a redirect the
recording no longer works. In fact, that's why StopMixMonitor
application is needed, because all MixMonitor does is to launch a
background thread that hooks into the channel audio, then the channel
continues to execute other applications in the dial plan while this
background thread monitors its audio, on a redirect StopMixMonitor
thread should continue saving audio until StopMixMonitor is called.

  


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Re: [asterisk-users] Cisco 7941G & Auth

[David Gibbons]

Hey Sasa,

I have templates of all the files you need here (SEP file, extension file):
http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip

If you need further assistance, let me know.

Thanks
Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sasa
Sent: Monday, June 22, 2009 4:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7941G & Auth

"Jonathan Thurman" wrote:
> What does your SEP.cnf.xml file look like?  In my experience,
> the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I
> had
> to specify the firmware version in each SEP file.  I am using 8-4-4S, but
> for you this would be something like this:
>
> 
> 
> SIP41.8-0-2SR1S
> 
> 

Hi, I have already writed also in SEP.cnf.xml file (other at
XMLDefault.cnf.xml file) the parameter:

SIP41.8-0-2SR1S

..but the problem isn't resolved !.
Can I try to change some parameters ?..are desperate ! I think I have tried
everything !
Thanks.

--

   Salvatore.



- Original Message -
From: "Jonathan Thurman" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, June 19, 2009 6:04 PM
Subject: Re: [asterisk-users] Cisco 7941G & Auth


> What does your SEP.cnf.xml file look like?  In my experience,
> the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I
> had
> to specify the firmware version in each SEP file.  I am using 8-4-4S, but
> for you this would be something like this:
>
> 
> 
> SIP41.8-0-2SR1S
> 
> 
>
>
> And you shouldn't need the tlv file.
>
> -Jonathan
>
>
>
> On Fri, Jun 19, 2009 at 8:25 AM, Sasa  wrote:
>
>> "David Gibbons" wrote:
>> > I've found that different types of TFTP servers return differing errors
>> > when a file doesn't exist. You don't need the TLV file, but >you do
>> > need
>> a
>> > distro that tells the phone it's not there correctly. I have not had
>> > ANY
>> > luck with windows tftp servers, only linux.
>>
>> I have tried with tftp on linux machine but the result isn't changed.
>> Thanks.
>>
>> --
>>
>>   Salvatore.
>>
>>
>>
>> - Original Message -
>> From: "David Gibbons" 
>> To: ; "'Asterisk Users MailingList - Non-Commercial
>> Discussion'" 
>> Sent: Friday, June 19, 2009 4:50 PM
>> Subject: Re: [asterisk-users] Cisco 7941G & Auth
>>
>>
>> > I've found that different types of TFTP servers return differing errors
>> > when a file doesn't exist. You don't need the TLV file, but you do need
>> > a
>> > distro that tells the phone it's not there correctly. I have not had
>> > ANY
>> > luck with windows tftp servers, only linux.
>> >
>> > -Dave
>> >
>> > -Original Message-
>> > From: asterisk-users-boun...@lists.digium.com
>> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John
>> Novack
>> > Sent: Friday, June 19, 2009 10:38 AM
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > Subject: Re: [asterisk-users] Cisco 7941G & Auth
>> >
>> >
>> >
>> > Sasa wrote:
>> >> Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with
>> >> Cisco
>> >> 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my
>> >> problem
>> >> is that Cisco phone isn't authenticated on Asterisk.
>> >> In tftp directory I have:
>> >>
>> >> apps41.1-1-1-15.sbn
>> >> cnu41.3-1-1-15.sbn
>> >> copstart.sh
>> >> cvm41sip.8-0-1-18.sbn
>> >> dialplan.xml
>> >> dsp41.1-1-1-15.sbn
>> >> jar41sip.8-0-1-18.sbn
>> >> load115
>> >> load308
>> >> load309
>> >> load30018
>> >> SIP41.8-0-2SR1S.loads
>> >> term41.default.loads
>> >> term61.default.loads
>> >> XMLDefault.cnf
>> >> SEPmac_address.cnf.xml
>> >>
>> >> ..and in tftp log I have:
>> >>
>> >> Connection received from 192.168.1.61 on port 49153 [19/06
>> >> 10:16:35.968]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:35.968]
>> >> File  : error 2 in system call CreateFile
>> >> Impossibile
>> >> trovare il file specificato. [19/06 10:16:35.968]
>> >> Connection received from 192.168.1.61 on port 49154 [19/06
>> >> 10:16:36.109]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:36.109]
>> >> Using local port 3995 [19/06 10:16:36.109]
>> >> : sent 15 blks, 7239 bytes in 0 s. 0 blk
>> >> resent
>> >> [19/06 10:16:36.171]
>> >> Connection received from 192.168.1.61 on port 49155 [19/06
>> >> 10:16:40.046]
>> >> Read request for file . Mode octet [19/06 10:16:40.046]
>> >> File <\mk-sip.jar> : error 2 in system call CreateFile Impossibile
>> >> trovare
>> >> il file specificato. [19/06 10:16:40.046]
>> >> Connection received from 192.168.1.61 on port 49156 [19/06
>> >> 10:16:40.984]
>> >> Read request for file . Mode octet [19/06
>> >> 10:16:40.999]
>> >> File  : error 3 in system call CreateFile
>> Impossibile
>> >> trovare il percorso specificato. [19/06 10:16:40.999]
>> >> Connection received from 192.168.1.61 on port 49164 [19/06
>> >> 10:16:42.843]
>> >> Read request for file . Mode o

[asterisk-users] Crash process Asterisk

[Adrien Lemoine]

Hi all,

 

I posted here some days ago about a crash of my Asterisk process.

 

To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4.

 

The process crashed again but this time I have previously put the option -g.

 

By following this : http://www.voip-info.org/wiki/view/Asterisk+debugging
and the section "Backtracing a core dump file in /tmp" I have obtained the
output in attached.

 

But I don't really understand what that's means. If someone is a coredump
reader, it will be fine to help me understanding the reason(s) of the crash.

 

Regards,

 

Adrien

gdb asterisk core.24156 
GNU gdb Red Hat Linux (6.3.0.0-1.63rh)
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "i386-redhat-linux-gnu"...Using host libthread_db 
library "/lib/tls/libthread_db.so.1".

Core was generated by `asterisk -g'.
Program terminated with signal 11, Segmentation fault.
Reading symbols from /lib/libdl.so.2...done.
Loaded symbols for /lib/libdl.so.2
Reading symbols from /lib/tls/libpthread.so.0...done.
Loaded symbols for /lib/tls/libpthread.so.0
Reading symbols from /usr/lib/libncurses.so.5...done.

...

Reading symbols from /lib/libgcc_s.so.1...done.
Loaded symbols for /lib/libgcc_s.so.1
#0  0x00a3fc94 in pthread_mutex_lock () from /lib/tls/libpthread.so.0
(gdb) bt
#0  0x00a3fc94 in pthread_mutex_lock () from /lib/tls/libpthread.so.0
#1  0x00295220 in expire_register (data=0xb677b180) at 
../include/asterisk/lock.h:592
#2  0x08056688 in ast_sched_runq (con=0x982c808) at sched.c:373
#3  0x002bf777 in do_monitor (data=0x0) at chan_sip.c:11324
#4  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#5  0x008be6fe in clone () from /lib/tls/libc.so.6
(gdb) 
(gdb) bt full
#0  0x00a3fc94 in pthread_mutex_lock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1  0x00295220 in expire_register (data=0xb677b180) at 
../include/asterisk/lock.h:592
newcount = 24172
peer = Variable "peer" is not available.

(gdb) thread apply all bt

Thread 16 (process 24156):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008b49e4 in poll () from /lib/tls/libc.so.6
#2  0x080bdfce in main (argc=2, argv=0xbff818c4) at asterisk.c:2449

Thread 15 (process 24158):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008b49e4 in poll () from /lib/tls/libc.so.6
#2  0x080b9646 in listener (unused=0x0) at asterisk.c:592
#3  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#4  0x008be6fe in clone () from /lib/tls/libc.so.6

Thread 14 (process 24159):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x00a43608 in accept () from /lib/tls/libpthread.so.0
#2  0x080b7067 in accept_thread (ignore=0x0) at manager.c:1442
#3  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#4  0x008be6fe in clone () from /lib/tls/libc.so.6

Thread 13 (process 24160):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x00a433fb in __read_nocancel () from /lib/tls/libpthread.so.0
#2  0x001524db in vio_read () from /usr/lib/mysql/libmysqlclient.so.14
#3  0x00153f42 in net_write_command () from /usr/lib/mysql/libmysqlclient.so.14
#4  0x00154279 in my_net_read () from /usr/lib/mysql/libmysqlclient.so.14
#5  0x0014d934 in net_safe_read () from /usr/lib/mysql/libmysqlclient.so.14
#6  0x001508cc in cli_advanced_command () from 
/usr/lib/mysql/libmysqlclient.so.14
#7  0x0014e9c0 in mysql_select_db () from /usr/lib/mysql/libmysqlclient.so.14
#8  0x0002 in ?? ()
#9  0x in ?? ()

Thread 12 (process 24161):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008b7151 in ___newselect_nocancel () from /lib/tls/libc.so.6
#2  0x00113683 in do_monitor (data=0x0) at ../include/asterisk/channel.h:1027
#3  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#4  0x008be6fe in clone () from /lib/tls/libc.so.6

Thread 11 (process 24162):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008b7151 in ___newselect_nocancel () from /lib/tls/libc.so.6
#2  0x0023f41c in do_parking_thread (ignore=0x0) at 
../include/asterisk/channel.h:1148
#3  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#4  0x008be6fe in clone () from /lib/tls/libc.so.6

Thread 10 (process 24163):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008815b6 in __nanosleep_nocancel () from /lib/tls/libc.so.6
#2  0x008813bc in sleep () from /lib/tls/libc.so.6
#3  0x006ac7d4 in scan_thread (unused=0x0) at pbx_spool.c:364
#4  0x00a3e341 in start_thread () from /lib/tls/libpthread.so.0
#5  0x008be6fe in clone () from /lib/tls/libc.so.6

Thread 9 (process 24164):
#0  0x007de7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
#1  0x008b49

Re: [asterisk-users] Cisco 7941G & Auth

[Murray Blakeman]

I had similar problems with the tftp service within Solaris.

I installed the tftp-hpa server and used that instead of the Solaris one 
and now it works fine.

Even in Linux/Unix some tftp servers return the wrong error to the phone 
if the file doesn't exist.

You definitely don't need the tlv file.

tftp-hpa works fine.

I'm using a Cisco 7941G with Asterisk via SIP perfectly.

Sasa wrote:
> Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 
> 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem 
> is that Cisco phone isn't authenticated on Asterisk.
> In tftp directory I have:
>
> apps41.1-1-1-15.sbn
> cnu41.3-1-1-15.sbn
> copstart.sh
> cvm41sip.8-0-1-18.sbn
> dialplan.xml
> dsp41.1-1-1-15.sbn
> jar41sip.8-0-1-18.sbn
> load115
> load308
> load309
> load30018
> SIP41.8-0-2SR1S.loads
> term41.default.loads
> term61.default.loads
> XMLDefault.cnf
> SEPmac_address.cnf.xml
>
> ..and in tftp log I have:
>
> Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968]
> Read request for file . Mode octet [19/06 
> 10:16:35.968]
> File  : error 2 in system call CreateFile Impossibile 
> trovare il file specificato. [19/06 10:16:35.968]
> Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109]
> Read request for file . Mode octet [19/06 
> 10:16:36.109]
> Using local port 3995 [19/06 10:16:36.109]
> : sent 15 blks, 7239 bytes in 0 s. 0 blk resent 
> [19/06 10:16:36.171]
> Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046]
> Read request for file . Mode octet [19/06 10:16:40.046]
> File <\mk-sip.jar> : error 2 in system call CreateFile Impossibile trovare 
> il file specificato. [19/06 10:16:40.046]
> Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984]
> Read request for file . Mode octet [19/06 10:16:40.999]
> File  : error 3 in system call CreateFile Impossibile 
> trovare il percorso specificato. [19/06 10:16:40.999]
> Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843]
> Read request for file . Mode octet [19/06 10:16:42.859]
> Using local port 3998 [19/06 10:16:42.859]
> : sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 
> 10:16:42.906]
>
> In XMLDefault.cnf I have:
>
> SIP41.8-0-2SR1S
>
> ..and on 7941G I have:
>
> App Load IDjar41sip.8-0-1-18.sbn
> Boot Load ID7941G_64-02070631Amd64megRel.bin
> VersionSIP41.8-0-2SR1S
>
> Thanks.
>
> --
>
>Salvatore.
>
>  
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   

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Re: [asterisk-users] Strange res_config_odbc error messages in 1.6.1.1

[Benny Amorsen]

Tilghman Lesher  writes:

> Looks like you're using a widevarchar column, which is something I didn't plan
> for.  It should be in SVN shortly, however.  On the regseconds, it's a typo in
> the code (UINTEGER4 should be INTEGER4).

I have:
column type (-9) unrecognized for column 'name'

but:
`name` varchar(80) collate utf8_danish_ci NOT NULL,

Now, this being MySQL, it is likely to do things I cannot possibly
imagine, but it looks to me like 'name' is a regular varchar.

Is it possible the the problem is related to the use of the MySQL ODBC
connector version 5.1.x instead of 5.0.x?


/Benny

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Re: [asterisk-users] Realtime extensions

[Jared Smith]

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote:
> For example when I have these two extensions:
> 
> -- _0699[134]X
> -- _06[069]XXX
> 
> that are in the database and number 0699123123 comes in asterisk will
> always choose exten _06[069]XXX
> and when they are in the extensions.conf file asterisk always choose
> exten _0699[134]X.
> 
> My question is why? Is it my misconfiguration or that's how it works.

I'm no expert on Asterisk realtime, but this definitely sounds like a
bug to me.  Mind opening a bug on the issue tracker
(issues.asterisk.org) so that the developers can investigate further?


-- 
Jared Smith
Training Manager
Digium, Inc.


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[asterisk-users] Different inbound routes for each interface on a TDM800P card.

[Jeremy Winder]

I'm new to Asterisk and inherited this project so I apologize if this
question has been asked a hundred time before. I did start with Google
but I may not be asking the right questions, because I wasn't finding
any answers.

I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium
TDM800P to interface with our six analog phone lines from the telco.
Currently I have a single trunk setup ZAP/G0 that will allow all
incoming call go to your IVR and allow outgoing calls.

My problem is I need two of those lines to be routed to one IVR for our
tech support people and the other four to be routed to our main business
IVR. For the life of me, I can't seem to find anything online about how
to do this. I thought about setting up six separate trunks; but I didn't
see anywhere you specify the channel for the trunk and right now my
single trunk is working with all of the analog lines.

Any help will be greatly appreciated,

Jeremy


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Re: [asterisk-users] Crash process Asterisk

[Stefan Schmidt]

Adrien Lemoine schrieb:
> Hi all,
> 
>  
> 
> To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4.

Hello,

i am not sure which bug this may be, but i am sure that it has been
fixed since the last 6 years since 1.2.7.1 was up2date.

update to 1.2.31 or newer and you wount have the bug again.

lg

steve

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Re: [asterisk-users] Different inbound routes for each interface on a TDM800P card.

[Kerem Erciyes]

Start by creating two ZAP Groups, i.e. ZAP/G0 and ZAP/G1 then you can route
the calls according to ZAP groups inbound and outbound.

   - /etc/zaptel.conf: Configuration of your hardware interfaces
   - 
/etc/asterisk/zapata.conf:
   Asterisk configuration to use your hardware interfaces

I have done the same setup on a Sangoma A101 connected to a channelbank.

http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf
http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf



On Mon, Jun 22, 2009 at 5:25 PM, Jeremy Winder wrote:

> I'm new to Asterisk and inherited this project so I apologize if this
> question has been asked a hundred time before. I did start with Google
> but I may not be asking the right questions, because I wasn't finding
> any answers.
>
> I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium
> TDM800P to interface with our six analog phone lines from the telco.
> Currently I have a single trunk setup ZAP/G0 that will allow all
> incoming call go to your IVR and allow outgoing calls.
>
> My problem is I need two of those lines to be routed to one IVR for our
> tech support people and the other four to be routed to our main business
> IVR. For the life of me, I can't seem to find anything online about how
> to do this. I thought about setting up six separate trunks; but I didn't
> see anywhere you specify the channel for the trunk and right now my
> single trunk is working with all of the analog lines.
>
> Any help will be greatly appreciated,
>
> Jeremy
>
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
Kerem Erciyes
Sistem Danismani
http://proje.keremerciyes.com

kerem.erci...@gmail.com
+90 532 737 05 83
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Re: [asterisk-users] Different inbound routes for each interface on aTDM800P card.

[Danny Nicholas]

Here is one way:
- exten => s,1,Answer
- exten => s,2,Gotoif($["${CHANNEL}" = "Zap-1"]?techsupp|s|1)
- exten => s,3,Gotoif($["${CHANNEL}" = "Zap-2"]?techsupp|s|1)
- exten => s,4,Goto(regivr|s|1)
- exten => s,5,hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Winder
Sent: Monday, June 22, 2009 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Different inbound routes for each interface on
aTDM800P card.

I'm new to Asterisk and inherited this project so I apologize if this
question has been asked a hundred time before. I did start with Google
but I may not be asking the right questions, because I wasn't finding
any answers.

I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium
TDM800P to interface with our six analog phone lines from the telco.
Currently I have a single trunk setup ZAP/G0 that will allow all
incoming call go to your IVR and allow outgoing calls.

My problem is I need two of those lines to be routed to one IVR for our
tech support people and the other four to be routed to our main business
IVR. For the life of me, I can't seem to find anything online about how
to do this. I thought about setting up six separate trunks; but I didn't
see anywhere you specify the channel for the trunk and right now my
single trunk is working with all of the analog lines.

Any help will be greatly appreciated,

Jeremy


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Re: [asterisk-users] Meetme Talker Optimization

[John A. Sullivan III]

On Mon, 2009-06-22 at 10:58 -0400, David Backeberg wrote:
> On Sat, Jun 20, 2009 at 8:15 PM, John A. Sullivan
> III wrote:
> > Hello, all.  I've been playing with MeetMe and talker optimization
> > seemed like a great idea.  I activated it as follows:
> > exten => 201,1,MeetMe(100201,cTo)
> > No one can hear the talker.  It doesn't matter how loudly the talker
> > shouts! Is there something else I'm supposed to do to enable this? I am
> 
> Assuming that you don't have any other problems, as in a call between
> these parties outside of meetme, like with a Bridge() works just fine,
> you could have some problems specific to optimization. Specifically,
> look at dsp.conf and try tuning your silence threshold. If this is SIP
> I also recommend disabling vad / silence suppression.
> 
> In my experience, I had a lot of trouble with talker optimization and
> I turned it off completely. If you read the archives from the last few
> weeks you'll see further pointers I made on this topic, as well as
> links to a lengthy bugs.digium.com discussion on talker optimization.

These are SIP connections so silence suppression may be the problem.  I
did also notice the threads and issues about talker optimization.  I was
hoping those were resolved in 1.6.1.1 - perhaps a bit naive.  Thanks -
John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Meetme Talker Optimization

[David Backeberg]

On Sat, Jun 20, 2009 at 8:15 PM, John A. Sullivan
III wrote:
> Hello, all.  I've been playing with MeetMe and talker optimization
> seemed like a great idea.  I activated it as follows:
> exten => 201,1,MeetMe(100201,cTo)
> No one can hear the talker.  It doesn't matter how loudly the talker
> shouts! Is there something else I'm supposed to do to enable this? I am

Assuming that you don't have any other problems, as in a call between
these parties outside of meetme, like with a Bridge() works just fine,
you could have some problems specific to optimization. Specifically,
look at dsp.conf and try tuning your silence threshold. If this is SIP
I also recommend disabling vad / silence suppression.

In my experience, I had a lot of trouble with talker optimization and
I turned it off completely. If you read the archives from the last few
weeks you'll see further pointers I made on this topic, as well as
links to a lengthy bugs.digium.com discussion on talker optimization.

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Re: [asterisk-users] Different inbound routes for each interface on aTDM800P card.

[Tzafrir Cohen]

On Mon, Jun 22, 2009 at 09:56:39AM -0500, Danny Nicholas wrote:
> Here is one way:
> - exten => s,1,Answer
> - exten => s,2,Gotoif($["${CHANNEL}" = "Zap-1"]?techsupp|s|1)

Zap/1-1 ?

Elsewhere I have:

exten => s,1,Set(DAHDI_CHAN=${CUT(CHANNEL,-,1)})
exten => s,n,Set(DAHDI_CHAN=${CUT(DAHDI_CHAN,/,2)})

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Crash process Asterisk

[Adrien Lemoine]

Hi Stefan,

Thanks for your answer.

You mean that you're not sure if it's a bug, or you don't know the bug
reference ?

I'm interested to find the bug report but I don't know how to formulate my
search.

Regards,

Adrien

-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
Schmidt
Envoyé : lundi 22 juin 2009 16:32
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Crash process Asterisk



Adrien Lemoine schrieb:
> Hi all,
> 
>  
> 
> To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4.

Hello,

i am not sure which bug this may be, but i am sure that it has been
fixed since the last 6 years since 1.2.7.1 was up2date.

update to 1.2.31 or newer and you wount have the bug again.

lg

steve

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Re: [asterisk-users] Different inbound routes for each interface onaTDM800P card.

[Danny Nicholas]

Presumably this would work on Zap as well since OP was on an earlier 1.4
rel.  Taking that ball and running with, a cleaner piece of logic;

exten => s,1,Set(IVR_CHAN=${CUT(CHANNEL,-,1)})
exten => s,n,Set(IVR_CHAN=${CUT(IVR_CHAN,/,2)})
exten => s,n,Gotoif($["${IVR_CHAN}" < "3"]?techsupp|s|1)
exten => s,n,Goto(regivr,s,1)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Monday, June 22, 2009 10:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Different inbound routes for each interface
onaTDM800P card.

On Mon, Jun 22, 2009 at 09:56:39AM -0500, Danny Nicholas wrote:
> Here is one way:
> - exten => s,1,Answer
> - exten => s,2,Gotoif($["${CHANNEL}" = "Zap-1"]?techsupp|s|1)

Zap/1-1 ?

Elsewhere I have:

exten => s,1,Set(DAHDI_CHAN=${CUT(CHANNEL,-,1)})
exten => s,n,Set(DAHDI_CHAN=${CUT(DAHDI_CHAN,/,2)})

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] RTP/SIP traffic prioritization and Linux issues

[John A. Sullivan III]

Hello, all.  I've stumbled across what seems to be a traffic
prioritization issue in a Linux environment and wonder if anyone else
has encountered or addressed this issue.

We had planned to use expedited forwarding for our RTP and perhaps our
SIP packets.  Our plan was to set DSCP to 101110 (by the way, I think
document http://www.voip-info.org/wiki/view/snom+360 is in error as I'm
almost certain the expedited forwarding bits are 101110 and not 100010).
However, we realized that when these passed through Linux based routers
or firewalls using the default pfifo_fast packet scheduler, it would
look at bits 3-7 for placement in band 0, 1, or 2.  Using the standard
expedited forwarding DSCP means pfifo_fast will see 1100 and place the
packets in band 1 - the default band for all traffic.  Thus, they will
receive no prioritization.

We are planning to thus change the DSCP to 101100 (b0 instead of b8 for
Asterisk, 176 instead of 184 for our Snom phones) and map 101100 to
802.1p priority 7 on our switches.

I am imagining this or is it a real issue when using Linux based
firewalls and routers with default packet schedulers and expedited
forwarding? Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation

Street Preacher: Are you SAVED?!!
Educated Skeptic: Saved from WHAT?!!
Educated Believer: From our selfishness that hurts the ones we love
   and condemns us to an eternity of hurting each other.
http://www.spiritualoutreach.com
Christianity that makes sense


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Re: [asterisk-users] Realtime extensions

[Danny Nicholas]

You can prove this by switching the order of the two filters in
extensions.conf.  IMO these evaluate the same given the number you are
testing with.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, June 22, 2009 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote:
> Hi
> 
> I am having a problem with extension matching in asterisk (I am using
> asterisk 1.6.0.6). Is there a difference between extensions matching
> in realtime architecture and extensions matching in extensions.conf
> file.
> 
> For example when I have these two extensions:
> 
> -- _0699[134]X
> -- _06[069]XXX
> 
> that are in the database and number 0699123123 comes in asterisk will
> always choose exten _06[069]XXX
> and when they are in the extensions.conf file asterisk always choose
> exten _0699[134]X.
> 
> My question is why? Is it my misconfiguration or that's how it works.
> 
It probably has to do with the way your database is sorting the
dialplan.  In extensions.conf you have complete control over the order
of the dialplan, in realtime the database sorts the dialplan depending
on the query so one rule may appear before another.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Realtime extensions

[Carlos Chavez]

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote:
> Hi
> 
> I am having a problem with extension matching in asterisk (I am using
> asterisk 1.6.0.6). Is there a difference between extensions matching
> in realtime architecture and extensions matching in extensions.conf
> file.
> 
> For example when I have these two extensions:
> 
> -- _0699[134]X
> -- _06[069]XXX
> 
> that are in the database and number 0699123123 comes in asterisk will
> always choose exten _06[069]XXX
> and when they are in the extensions.conf file asterisk always choose
> exten _0699[134]X.
> 
> My question is why? Is it my misconfiguration or that's how it works.
> 
It probably has to do with the way your database is sorting the
dialplan.  In extensions.conf you have complete control over the order
of the dialplan, in realtime the database sorts the dialplan depending
on the query so one rule may appear before another.


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] IMAP voice mail storage

[Leif Madsen]

Philipp Kempgen wrote:
> John A. Sullivan III schrieb:
> 
>> By the way, how does one disable IMAP storage?
> 
>> I did not see a module for IMAP storage.  It would seem strange that I
>> would have to recompile.
> 
> Sadly you have to recompile.
> Disable voicemail IMAP storage in `make menuselect`.

Additionally, bugs should be reported to http://issues.asterisk.org if you 
would 
like any action to be taken on the issue :)

Thanks!
Leif Madsen.

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Re: [asterisk-users] Crash process Asterisk

[Stefan Schmidt]

Hello,

i'found this here:

https://issues.asterisk.org/view.php?id=7176

normally you will find the best information from a gdb trace at the
bottom, where the last function has been called. in your case this was
#1  0x00295220 in expire_register (data=0xb677b180) at
../include/asterisk/lock.h:592

in the changelog of version 1.2 there is a comment about an error with
expire_register and also the link to the issue above.

i hope this helps you.


best regards.

steve

Adrien Lemoine schrieb:
> Hi Stefan,
> 
> Thanks for your answer.
> 
> You mean that you're not sure if it's a bug, or you don't know the bug
> reference ?
> 
> I'm interested to find the bug report but I don't know how to formulate my
> search.
> 
> Regards,
> 
> Adrien
> 
> -Message d'origine-
> De : asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
> Schmidt
> Envoyé : lundi 22 juin 2009 16:32
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [asterisk-users] Crash process Asterisk
> 
> 
> 
> Adrien Lemoine schrieb:
>> Hi all,
>>
>>  
>>
>> To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4.
> 
> Hello,
> 
> i am not sure which bug this may be, but i am sure that it has been
> fixed since the last 6 years since 1.2.7.1 was up2date.
> 
> update to 1.2.31 or newer and you wount have the bug again.
> 
> lg
> 
> steve
> 
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] Learn Asterisk

[David @ULC]

What the best website and book to start learning asterisk ?
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Re: [asterisk-users] Realtime extensions

[Jared Smith]

On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas wrote:
> You can prove this by switching the order of the two filters in
> extensions.conf.  

The order that the extensions in extensions.conf appear has no bearing
on the sort order.  I've explained the sorting mechanism at length in
this list before, but I'd be happy to go over it again if anyone wants
me to.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] RTP/SIP traffic prioritization and Linux issues

[Dave Fullerton]

John A. Sullivan III wrote:
> Hello, all.  I've stumbled across what seems to be a traffic
> prioritization issue in a Linux environment and wonder if anyone else
> has encountered or addressed this issue.
> 
> We had planned to use expedited forwarding for our RTP and perhaps our
> SIP packets.  Our plan was to set DSCP to 101110 (by the way, I think
> document http://www.voip-info.org/wiki/view/snom+360 is in error as I'm
> almost certain the expedited forwarding bits are 101110 and not 100010).
> However, we realized that when these passed through Linux based routers
> or firewalls using the default pfifo_fast packet scheduler, it would
> look at bits 3-7 for placement in band 0, 1, or 2.  Using the standard
> expedited forwarding DSCP means pfifo_fast will see 1100 and place the
> packets in band 1 - the default band for all traffic.  Thus, they will
> receive no prioritization.
> 
> We are planning to thus change the DSCP to 101100 (b0 instead of b8 for
> Asterisk, 176 instead of 184 for our Snom phones) and map 101100 to
> 802.1p priority 7 on our switches.
> 
> I am imagining this or is it a real issue when using Linux based
> firewalls and routers with default packet schedulers and expedited
> forwarding? Thanks - John

You are correct, EF is 101110.

I recently started using dscp on my network and ran into similar issues 
as you. I have cisco routers (not on smartnet) in my environment and 
some (v 12.x) understood dscp and some (<=v 11.x) did not. For those 
that did not I had to match on the precedence bits instead and 
everything thus far is working like it is supposed to.

As for linux, I couldn't find anything online that actually implemented 
diffserv-style traffic management. I ended up writing a script that 
would generate a set of queues and used the dscp to drop packets into 
the appropriate queues and another script to set the dscp for programs 
that could not on their own.

It's still a bit of a work in process and I'm sure there are 
improvements to be made, but if you'd like to look at it I can send it 
to you off-list.

-Dave

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Re: [asterisk-users] Learn Asterisk

[Jared Smith]

On Mon, 2009-06-22 at 22:50 +0530, David @ULC wrote:

> What the best website and book to start learning asterisk ?

I'm obviously biased (as I'm co-author of the book), but I recommend
O'Reilly Media's "Asterisk: The Future of Telephony", Second Edition.
You can download a free PDF of the book at http://www.asteriskdocs.org/
or you can obviously buy a dead-tree version of the book from you
favorite bookseller.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Realtime extensions

[Danny Nicholas]

Hey Jared,  Do you have a FAQ reference for some commonly asked questions
like this?  I've got an email archive that goes back almost a year, but
finding a reference in 4K+ emails is sometimes difficult, even if I can
remember some keyword to help out.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Monday, June 22, 2009 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas wrote:
> You can prove this by switching the order of the two filters in
> extensions.conf.  

The order that the extensions in extensions.conf appear has no bearing
on the sort order.  I've explained the sorting mechanism at length in
this list before, but I'd be happy to go over it again if anyone wants
me to.


-- 
Jared Smith
Training Manager
Digium, Inc.


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[asterisk-users] "internal_timing" not working (re: SIP silence suppression)

[Bryan Field-Elliot]

We are trying to get Asterisk to behave correctly when our SIP clients  
have "Silence Suppression" turn on, but are not having any luck.  
Basically, there are several apps in Asterisk which won't send any  
audio to the SIP client, unless the SIP client itself sends audio to  
Asterisk (which it won't do if Silence Suppression is enabled and the  
caller is quiet).

We think we've done everything right in terms of setup, but obviously  
we're missing something. Can anyone please advise? Here are the  
relevant data points:

- Asterisk version is 1.6.1 (r200516M)
- Dahdi (version 2.2.0 rc2) is installed and running. Linux module is  
loaded.
- Command-line tool "dahdi_test" returns +99% accuracy
- res_timing_dahdi.so is loaded in Asterisk "modules.conf" file
- "internal_timing" is set to "yes" in Asterisk "asterisk.conf" file
- Asterisk CLI command "timing test" reports:

Attempting to test a timer with 50 ticks per second.
Using the 'DAHDI' timing module for this test.
It has been 1019 milliseconds, and we got 51 timer ticks

Based upon my understanding of things, all of the above points to "go"  
with respect to proper support of SIP Silence Suppression in the  
client, but we're just not seeing it.

Any help would be appreciated,

Thank you,

Bryan










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Re: [asterisk-users] Learn Asterisk

[Danny Nicholas]

www.voip-info.org   is a good resource IMO; the
O'Reilly online book would be considered the standard, although it is
oriented toward a 1.2 or 1.4 user.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, June 22, 2009 12:21 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Learn Asterisk

 

 

What the best website and book to start learning asterisk ?

 

 

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Re: [asterisk-users] Learn Asterisk

[Alan Lord (News)]

On 22/06/09 18:20, David @ULC wrote:
>
> What the best website and book to start learning asterisk ?

Website: Google, http://www.voip-info.org

Book: TFOT (The future of Telephony) Google for it , it is 
freely/legally downloadable.




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Re: [asterisk-users] Learn Asterisk

[David @ULC]

I am from the eastern part of India and there is No institute to have a
formal training for Asterisk.
I would like to open One Institute dedicated for Training. ( Though I am
also learning )

Any advice is highly appreciated.

On Mon, Jun 22, 2009 at 10:50 PM, David @ULC  wrote:

>
> What the best website and book to start learning asterisk ?
>
>
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Re: [asterisk-users] Realtime extensions

[Jared Smith]

On Mon, 2009-06-22 at 12:30 -0500, Danny Nicholas wrote:
> Hey Jared,  Do you have a FAQ reference for some commonly asked questions
> like this?  I've got an email archive that goes back almost a year, but
> finding a reference in 4K+ emails is sometimes difficult, even if I can
> remember some keyword to help out.

I wish I did... the only thing I really have is my brain.  When
searching mailing list archives, I tend to use www.markmail.org, but I
did a quick search and didn't find the post I was looking for, so I'll
re-iterate my understanding of the pattern matching here for posterity's
sake:

Pattern Matching: Asterisk searches the extensions and patterns digit by
digit, from left to right, and applies the following three rules:

Rule 1) For the current digit, order the possible matching extensions
based on the most constrained match.  For example, let's say Asterisk
was looking at the first second digit in this context, and the caller
dialed extension 123:

[pattern-test]
exten => _1XX,1,NoOp(Option 1)
exten => _1[2-4]X,1,NoOp(Option 2)
exten => _1NX,1,NoOp(Option 3)

In this example, the second option would be given priority over options
one or three, as there are only three possible matches for this digit
(2, 3, or 4), while the other options have more possibilities (8 in the
case of the N, or 10 in the case of the X).

Rule 2) In the case of a tie (when the number of possibilities for this
digit) is the same between two extensions, the extensions are then
sorted into ASCII sort order.  Consider this context for a moment:

[pattern-test-two]
exten => _1[1-8]X,1,NoOp(Option 4)
exten => _1NX,1,NoOp(Option 5)

In this example, the second digit has eight possibilities in both
extensions... so the extensions will be sorted in ASCII sort order.  The
1 comes before the N in the ASCII table, so option 4 will be selected
before option 5.

Rule 3) If the dialed digit can't match a particular pattern, exclude
the pattern from the list of matches.  This means that if a pattern was
more constrained in earlier matches and therefore at the top of the list
of matching extensions, later digits can disqualify it.  To illustrate
this point, let's look at the following example:

[pattern-test-three]
exten => 1[2-4]N,1,NoOp(Option 6)
exten => 1NX,1,NoOp(Option 7)

In this case, let's assume the caller dialed extension 130.  After the
first digit, the two patterns are tied.  After the second digit, option
6 gets sorted above option 7 because it is more constrained.  After the
third digit, however, option 6 is eliminated because the last digit
can't be a zero.  That means that Option 7 will match.

Clear as mud?


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Learn Asterisk

[Jared Smith]

On Mon, 2009-06-22 at 23:22 +0530, David @ULC wrote:
> I am from the eastern part of India and there is No institute to have
> a formal training for Asterisk.

Digium does have an authorized training partner in India, but since this
is a non-commercial list, I kindly ask that you contact me directly for
more information.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Asterisk on AVR32

[Doug Bailey]

When you run configure, you need to spec the "host" parameter for the 
architecture 
and environment you will be running under.  

For example, a 1.4 distro being built for a blackfin running uclinux would run 
./configure host=bfin-uclinux

This will imply that the compiler being used to build the code will be named 
--gcc and is accessible in the PATH that you provide to the make 
system.  

In the example above, the compiler is named: 
bfin-uclinux-gcc 

Doug 

- "Kyle Kienapfel"  wrote:

> why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing,
> or something you did? it should be something line "avr" or "avr32"
> 
> 
> 
> 
> 
> 
> On Thu, Jun 18, 2009 at 3:08 AM, Paulo Santos < paulo.r.san...@sapo.pt
> > wrote:
> 
> 
> Greetings everyone,
> 
> I'm trying to compile asterisk for an AVR32 (Atmel NGW100).
> Buildroot for AVR32 already has the asterisk package, though it has
> bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting
> the contents of the patch file did the trick.
> 
> Now, the problem is making asterisk. The first error is because
> asterisk
> needed to be ./configure:ed.
> 
> Trying to just do ./configure, make gives an error [1].
> 
> Trying to do ./configure with the same args as make plus --host it
> can't
> even configure [2]
> 
> I don't know much about cross-compiling, or even regular compiling for
> that matter. Does any one have any idea on how to do this?
> 
> Thanks in advance,
> Best regards,
> Paulo Santos
> 
> 
> [1]
> menuselect/menuselect --check-deps menuselect.makeopts
> /bin/bash: menuselect/menuselect: cannot execute binary file
> make[1]: *** [menuselect.makeopts] Error 126
> make[1]: Leaving directory
> `/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6'
> make: ***
> [/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6/asterisk]
> Error 2
> 
> [2]
> configure: WARNING: If you wanted to set the --build type, don't use
> --host.
> If a cross compiler is detected then cross compile mode will be used.
> checking build system type... i686-pc-linux-gnu
> checking host system type... Invalid configuration `CROSS_ARCH=Linux':
> machine `CROSS_ARCH=Linux' not recognized
> configure: error: /bin/bash ./config.sub CROSS_ARCH=Linux failed
> 
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Re: [asterisk-users] 400 calls at g711 how much cpu power

[ContactTel Business]

Lol , simply lol, don't forget the super duper, top secret patch ,everyone
is hiding from you  that makes asterisk able to do 4000 calls on a p3, 

 

PS. don't tell anyone i said this .

 

But yeah , since you need to blast 500 calls+, you should be aware that
normal blasting even 4 seconds audio will run you quite a bit of money,

 

20 seconds * 400 channels = 8000 seconds every 20 seconds, +- prep times...

Or 

133 minutes every 20 secs...

399 minutes every minute.. @ 0.015 let's say 

 

400 * 0.015 is 6$ a minute, $360 an hour, 3600$ a day, and ill let you do
the weekly fees

 

Now, that's starting to be expensive for a pet project ;) if not and gov
related, then ill just pass the remarks..

 

I always knew there's money in fear, but broadcasting it could be worth it
too ;) 

Can't wait for the day when we get voice calls about buying water in bulk
and storing crackers.

 

Anyhow let me know how you manage to do 400 calls on asterisk with or
without transcoding 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erick Perez
Sent: June-20-09 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power

 

I am fairly certain he was simply reporting the results (for posterity) of
the event having already happened.  Good to know (I guess?) that such
small hardware can acheive the performance that was squeezed out of it.
Impressive.

All THAT said, I am unconvinced that there was no sales effort involved in
sending out millions of unsolicited calls.  Claim if you like that this
was some public information event (which you fail to expand much upon) and
convict me of mistrust, but who would have paid for such a thing.  TV ads,
radio spots, billboards, etc., are much more effective for public
information.  Unsolicited calls on that order mean only one thing to me -
SPAM.  So what wonderful product were you "informing" the public about
with regard to the looming threat of illness?

 

Jeff, indeed i was posting for posterity. Maybe someone will benefit in an
outbound-only scenario that he/she will not need a supercomputer to pump a
20sec audio clip.

Again, this was a public service. And indeed TV and radio was used. Unless
you live in a bubble, you may have heard about AH1N1 virus. Which
unfortunately hit us (Panama, Republic of Panama, Central America) very
hard. I foud very repetitive to tell in my posts that i am from panama,
central america, blah,blah blah.

 

Anyways, a quick google search of this forum will also revealed that i am
kind of a regular poster and even my cellphone is listed here (Jon Pounder,
my cellphone is +507 6675 5083 in case YOU want to sell me a car loan, i
dont mind getting a call. Im a IT consultant and i have a chargeback line.
Please call me as many times as you want...please do so between 10pm and 6am
where my chargeback is the most expensive).

 

Guys, Grow up!

 

Next time someone needs to learn mouth-to-mouth and CPR lessons, please DONT
teach him. Because, following your inmature way of thinking, the person who
wants to learn CPR may as well be looking for information to learn how to
suffocate people.

Next time your son wants to know how gasoline works or how is being
produced. Please keep your familiy in ignorance. You may be training the
next crazy person who will burn things all around the world.

 

But, you wont do that, do you?

 

Again, I always tell my familiy that keeping others in ignorance is bad. but
sometimes it must be done for the sake of a greater good, and my comment is
always followed with good and sound examples (atomic technology, viruses,
etc).

 

But I forgot that Asterisk, the phone lines and a calling system is the way
the world is going to be dominated by the martians. So the secret about
phone system calculations must be keept in Area 51.

 

Now I understand Kevin Mitnick.

 

Cheers to all. Bye.

 

 

 

 


Erick Perez
Cel +(507) 6675-5083


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Re: [asterisk-users] 400 calls at g711 how much cpu power

[Jason Aarons (US)]

Is this Project Eagle Eye ?  Call every phone at once to tell them about
H1N1 in their neighborhood

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel
Business
Sent: Monday, June 22, 2009 3:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power

 

Lol , simply lol, don't forget the super duper, top secret patch
,everyone is hiding from you  that makes asterisk able to do 4000 calls
on a p3, 

 

PS. don't tell anyone i said this .

 

But yeah , since you need to blast 500 calls+, you should be aware that
normal blasting even 4 seconds audio will run you quite a bit of money,

 

20 seconds * 400 channels = 8000 seconds every 20 seconds, +- prep
times...

Or 

133 minutes every 20 secs...

399 minutes every minute.. @ 0.015 let's say 

 

400 * 0.015 is 6$ a minute, $360 an hour, 3600$ a day, and ill let you
do the weekly fees

 

Now, that's starting to be expensive for a pet project ;) if not and gov
related, then ill just pass the remarks..

 

I always knew there's money in fear, but broadcasting it could be worth
it too ;) 

Can't wait for the day when we get voice calls about buying water in
bulk and storing crackers.

 

Anyhow let me know how you manage to do 400 calls on asterisk with or
without transcoding 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erick
Perez
Sent: June-20-09 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power

 

I am fairly certain he was simply reporting the results (for
posterity) of
the event having already happened.  Good to know (I guess?) that
such
small hardware can acheive the performance that was squeezed out
of it.
Impressive.

All THAT said, I am unconvinced that there was no sales effort
involved in
sending out millions of unsolicited calls.  Claim if you like
that this
was some public information event (which you fail to expand much
upon) and
convict me of mistrust, but who would have paid for such a
thing.  TV ads,
radio spots, billboards, etc., are much more effective for
public
information.  Unsolicited calls on that order mean only one
thing to me -
SPAM.  So what wonderful product were you "informing" the public
about
with regard to the looming threat of illness?

 

Jeff, indeed i was posting for posterity. Maybe someone will benefit in
an outbound-only scenario that he/she will not need a supercomputer to
pump a 20sec audio clip.

Again, this was a public service. And indeed TV and radio was used.
Unless you live in a bubble, you may have heard about AH1N1 virus. Which
unfortunately hit us (Panama, Republic of Panama, Central America)
very hard. I foud very repetitive to tell in my posts that i am from
panama, central america, blah,blah blah.

 

Anyways, a quick google search of this forum will also revealed that i
am kind of a regular poster and even my cellphone is listed here (Jon
Pounder, my cellphone is +507 6675 5083 in case YOU want to sell me a
car loan, i dont mind getting a call. Im a IT consultant and i have a
chargeback line. Please call me as many times as you want...please do so
between 10pm and 6am where my chargeback is the most expensive).

 

Guys, Grow up!

 

Next time someone needs to learn mouth-to-mouth and CPR lessons, please
DONT teach him. Because, following your inmature way of thinking, the
person who wants to learn CPR may as well be looking for information to
learn how to suffocate people.

Next time your son wants to know how gasoline works or how is being
produced. Please keep your familiy in ignorance. You may be training the
next crazy person who will burn things all around the world.

 

But, you wont do that, do you?

 

Again, I always tell my familiy that keeping others in ignorance is bad.
but sometimes it must be done for the sake of a greater good, and my
comment is always followed with good and sound examples (atomic
technology, viruses, etc).

 

But I forgot that Asterisk, the phone lines and a calling system is the
way the world is going to be dominated by the martians. So the secret
about phone system calculations must be keept in Area 51.

 

Now I understand Kevin Mitnick.

 

Cheers to all. Bye.

 

 

 

 


Erick Perez
Cel +(507) 6675-5083





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Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have receive

Re: [asterisk-users] 400 calls at g711 how much cpu power

[hh174]

Let me also know, I just have a business with Tamiflu :)
I need to contact 6.000.000.000 people to help them.
No spam, I promise.

Olivier


ContactTel Business a écrit :

  
  
  

  
  Lol
, simply lol, don’t forget the super duper, top secret
patch ,everyone is hiding from you  that makes asterisk able to do 4000
calls on a p3, 
   
  PS.
don’t tell anyone i said this .
   
  But
yeah , since you need to blast 500 calls+, you should be
aware that normal blasting even 4 seconds audio will run you quite a
bit of
money,
   
  20
seconds * 400 channels = 8000 seconds every 20 seconds, +-
prep times...
  Or
  
  133
minutes every 20 secs...
  399
minutes every minute.. @ 0.015 let’s say 
   
  400
* 0.015 is 6$ a minute, $360 an hour, 3600$ a day, and ill
let you do the weekly fees
   
  Now,
that’s starting to be expensive for a pet project ;)
if not and gov related, then ill just pass the remarks..
   
  I
always knew there’s money in fear, but broadcasting it
could be worth it too ;) 
  Can’t
wait for the day when we get voice calls about buying
water in bulk and storing crackers.
   
  Anyhow
let me know how you manage to do 400 calls on asterisk
with or without transcoding 
   
   
   
  
  
  
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erick
Perez
  Sent: June-20-09 9:34 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] 400 calls at g711 how much cpu
power
  
  
   
  
  
I am fairly certain he was simply reporting
the results (for
posterity) of
the event having already happened.  Good to know (I guess?) that such
small hardware can acheive the performance that was squeezed out of it.
Impressive.

All THAT said, I am unconvinced that there was no sales effort involved
in
sending out millions of unsolicited calls.  Claim if you like that this
was some public information event (which you fail to expand much upon)
and
convict me of mistrust, but who would have paid for such a thing.  TV
ads,
radio spots, billboards, etc., are much more effective for public
information.  Unsolicited calls on that order mean only one thing to me
-
SPAM.  So what wonderful product were you "informing" the public
about
with regard to the looming threat of illness?
  
  
   
  
  
  Jeff, indeed i was posting for posterity. Maybe
someone will
benefit in an outbound-only scenario that he/she will not need a
supercomputer
to pump a 20sec audio clip.
  
  
  Again, this was a public service. And indeed TV
and radio
was used. Unless you live in a bubble, you may have heard about AH1N1
virus.
Which unfortunately hit us (Panama, Republic of Panama, Central
America)
very hard. I foud very repetitive to tell in my posts that i am from
panama,
central america, blah,blah blah.
  
  
   
  
  
  
  Anyways, a quick google search of this forum
will also
revealed that i am kind of a regular poster and even my cellphone is
listed
here (Jon Pounder, my cellphone is +507 6675 5083 in case YOU want to
sell me a car loan, i dont mind getting a call. Im a IT consultant and
i have a
chargeback line. Please call me as many times as you want...please do
so
between 10pm and 6am where my chargeback is the most expensive).
  
  
   
  
  
  Guys, Grow up!
  
  
   
  
  
  Next time someone needs to learn mouth-to-mouth
and CPR lessons,
please DONT teach him. Because, following your inmature way of
thinking, the
person who wants to learn CPR may as well be looking for information to
learn
how to suffocate people.
  
  
  Next time your son wants to know how gasoline
works or how
is being produced. Please keep your familiy in ignorance. You may be
training
the next crazy person who will burn things all around the world.
  
  
   
  
  
  But, you wont do that, do you?
  
  
   
  
  
  Again, I always tell my familiy that keeping
others in
ignorance is bad. but sometimes it must be done for the sake of a
greater good,
and my comment is always followed with good and sound examples (atomic
technology, viruses, etc).
  
  
   
  
  
  But I forgot that Asterisk, the phone lines and
a calling
system is the way the world is going to be dominated by the martians.
So the
secret about phone system calculations must be keept in Area 51.
  
  
   
  
  
  Now I understand Kevin Mitnick.
  
  
   
  
  
  Cheers to all. Bye.
  
  
   
  
  
   
  
  
   
  
  
   
  
  



Erick Perez
Cel +(507) 6675-5083



  
  
  
  

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Re: [asterisk-users] 400 calls at g711 how much cpu power

[Christopher Stamper]

So what happened to the OP? Seems he would be eager to help us fight the
swine flu...

-- 
Christopher Stamper

Email: christopherstam...@gmail.com
Web: http://tinyurl.com/2ooncg
gTalk: http://tinyurl.com/6e359r
Skype: cdstamper
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[asterisk-users] Documenting configuration with Real Time

[John A. Sullivan III]

Hello, all.  As we work through our design issues, we are very
interested in moving immediately to real time database since we
anticipate expanding our system in to a clustered system within a year
or two.

One of the biggest disadvantages we anticipate is the lack of
configuration documentation.  In other words, the .conf files are
self-documenting.  If I want to understand my dialplan, I read the
extensions.conf file.  If I want an overview of my SIP users, I read
sip.conf.  It seems we lose this in the database (yes, I could run sql
queries but it's not nearly as convenient).  I would imagine this is
particularly important when making a change to the configuration.  I
like being able to see how the part fits into the whole.

I suppose we could create static sql files and organize them similarly
to the conf files, i.e., have a comment such as #[context-client1]
followed by all the sql statements to create the extensions or mailboxes
under [context-client1].

How are others dealing with the loss of self-documenting .conf files
when moving to real time? Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Asterisk + mySQL

[jonas kellens]

[Jun 22 17:51:48] ERROR[13726]: cdr_addon_mysql.c:249 mysql_log:
mysql_cdr: Failed to insert into database: (1054) Unknown column
'calldate' in 'field list'  == Spawn extension 

I have the following columns (for billing) :

accountcode src dst dcontextclidchannel
dstchannel  lastapp lastdatastart   answer  end duration
billsec disposition amaflags

Why does it want to write to a column calldate ?? Where is this
defined ??

Thanks for the help !
Jonas.


On Fri, 2009-06-19 at 14:13 -0500, Miguel Molina wrote:

> jonas kellens escribió: 
> 
> > On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote: 
> > 
> > > In modules.conf:  noload => cdr_csv.so
> > > 
> > > 
> > 
> > 
> > Are there other modules I need to load or unload ??
> > 
> > asterisk*CLI> module show like cdr
> > Module Description
> > Use Count 
> > cdr_addon_mysql.so MySQL CDR Backend
> > 0 
> > app_setcdruserfield.so CDR user field apps
> > 0 
> > func_cdr.soCDR dialplan function
> > 0 
> > app_cdr.so Tell Asterisk to not maintain a CDR
> > for  0 
> > cdr_manager.so Asterisk Manager Interface CDR
> > Backend   0 
> > app_forkcdr.so Fork The CDR into 2 separate entities
> > 0 
> > cdr_csv.so Comma Separated Values CDR Backend
> > 0 
> > cdr_custom.so  Customizable Comma Separated Values
> > CDR  0 
> > 8 modules loaded
> > asterisk*CLI> module show like odbc
> > Module Description
> > Use Count 
> > 0 modules loaded
> > asterisk*CLI> module show like sql
> > Module Description
> > Use Count 
> > cdr_addon_mysql.so MySQL CDR Backend
> > 0 
> > app_addon_sql_mysql.so Simple Mysql Interface
> > 0 
> > res_config_mysql.soMySQL RealTime Configuration Driver
> > 0 
> > 3 modules loaded
> > 
> > modules.conf :
> > 
> > autoload=yes
> > noload=pbx_gtkconsole.so
> > load=res_musiconhold.so
> > load=cdr_addon_mysql.so
> > noload=chan_alsa.so
> > 
> > Why is there a res_mysql.conf and a cdr_mysql.conf ?? They both look
> > alike...
> > 
> > 
> 
> There's no other modules you need to load/unload. To disable CSV CDR
> recording just add what Tilghman told you into modules.conf.
> 
> cdr_mysql.conf is for MySQL CDR backend database settings.
> res_mysql.conf is for MySQL Asterisk Realtime Architecture (ARA)
> backend database settings.
> 
> Cheers,
> 
> -- 
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
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Re: [asterisk-users] Realtime extensions

[Andrew Nowrot]

Hi

Many thanks for your help.
I managed to solve the problem by rewriting the dialplan. I have split
the line _06[069] into three different extensions 060, 066 and
069 and now asterisk is matching the numbers as I expect it to do.
Maybe it is not the "clean" solution, but  ;).

Many thanks to Jared Smith for the exten matching explanation

Cheers
Andrew

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[asterisk-users] Minimizing downtime during updates

[Karl Fife]

I was about to ask this question when I figured out the answer by combing 
through the makefile.
I am posting this anyway because I think it's good to know, and I didn't 
find any threads that speak to it when I searched the list history.

My Question was:
When updating Asterisk, the sound tarballs for the selected codecs are not 
retreived until running make install.  This adds unnecessarily to the 
downtime when updating versions because Asterisk has to be stopped while 
running make install.  I wanted a simple way to pre-fetch these files to a 
local repository to speed up the actual install routine, instead of slowing 
it by the arbitrary duration of the fetch/download process which robs 
valuable NINES from uptime :-)

I discovered that after running make, you can run 'make sounds' before 
shutting down the service.  This cuts all of the download time from the 
install process minimizing service downtime to a fraction of what it would 
othewise be.

-Karl



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[asterisk-users] How to force TDM410 to alaw

[Alex Samad]

Hi

I am having some problem forcing my tdm410 to alaw over ulaw, I have
1.6.1.0 asterisk (debian i486)

dahdi1:2.2.0 built with the hardware echo canceller firmware

/etc/asterisk/chan_dahdi.conf
alaw=1-4


but I have this in the general section, before any channel definition


dahdi show  channel 1
Default law: ulaw

even when i have a call going it is still ulaw 

Thanks
Alex


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Re: [asterisk-users] Learn Asterisk

[Paul Hales]

I can definitely recommend the 'sit down and play with it' website.
Worked for me.

PaulH


David @ULC wrote:
>
> What the best website and book to start learning asterisk ?
>
>
> 
>
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