Displaying 20 results from an estimated 8000 matches similar to: "Spontaneous reboots on asterisk 1.6.2.11"
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
2010 Sep 13
3
doing dnsmgr_lookup
Hello list,
my CLI is spammed with :
[Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep
2010 Jun 28
3
Pickup a ringing Queue member
Hello.
I'm using asterisk 1.4.30.
I've found this patch for app_queue.c :
https://issues.asterisk.org/view.php?id=11700
Can I easily implement this by issuing : */wget
'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug'
-O - | patch -p0/* ??
Does this mean I have a "patched" asterisk ? (I ask this because some
applications require a
2010 Aug 01
2
# -key not to be 'transfer'
Hello list,
whenever I press the #-key I hear a voice saying 'transfer'. How can I
use the #-key without this voice-message or without having it the
function of unattended transfer ?!
The T or t option is not set in my Dial()-command so I don't know where
this transfer is coming from in the first place.
Kind regards,
Jonas.
-------------- next part --------------
An HTML
2010 Nov 03
3
How to make the sum of a ${VARIABLE} + 1 ??
Hello,
I have this in my dialplan :
exten => s,n,Set(vgLabel=vg(${number}+1))
exten => s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string :
[Nov 3 16:17:27] -- Executing [s at macro-f:43]
Set("SIP/test-00000002", "vgLabel=vg(1+1)") in new stack
[Nov 3 16:17:27] -- Executing [s at macro-f:44]
2014 Oct 30
2
${HASH(SIP_CAUSE,<channel-name>)}
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
*${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using
the destination channel, not the source channel.
But when I use this in my dialplan, this 'variable' is empty.
Dialplan :
exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten =>
2010 Oct 12
1
chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049
Hello,
what does this message mean ?
[Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401'
onto UDP socket destined for public_ip:2049
I find this in my debug log file when "core set debug 25".
Is something failing, or is this just informative ?
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2011 Aug 26
1
Asterisk spontaneous reboot
Hello,
Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could
no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds.
There is now a core dump present in /tmp :
-rw------- 1 root root 88M Aug 26 08:07
core.sip.pbx.tld-2011-08-26T08:07:35+0200
How can I get usefull information about what went wrong ? Because a
spontaneous reboot of Asterisk has never
2013 Nov 13
1
calendar.conf include
Hello,
can I use include-statements in the calendar.conf configuration file ?
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131113/8aaffcd8/attachment.html>
2018 Oct 04
4
Spontaneous reboot due to MySQL lookups ?
Hello
thank you for your answer.
If I read your (and others) reaction correctly I can conclude that this
is an Asterisk problem and not a problem of MySQL or dialplan logic ?
You should know that the MySQL database is heavily questioned :
mysql> show status like '%onn%';
+--------------------------+--------+
| Variable_name | Value |
2011 Feb 01
1
How to load new musiconhold classes ?
Hello,
I've defined some new musiconhold classes in musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
;
[908001]
mode=files
directory=/var/lib/asterisk/moh/908001
random=yes
;
[101001-1]
mode=files
directory=/var/lib/asterisk/moh/101001/1
random=yes
;
[101001-2]
mode=files
directory=/var/lib/asterisk/moh/101001/2
random=yes
But the new classes never show up
2018 Oct 04
3
Spontaneous reboot due to MySQL lookups ?
Hello
using Asterisk 1.8.32.
I notice that there is a spontaneous reboot of the Asterisk system from
time to time.
When I look in the logs (verbose file) I noticed that every time this
occurs it's at a moment that there is a MySQL action, be it a lookup or
an insert/update/delete.
I must say I do have some MySQL queries that occur in my dialplan when a
call comes in, to look up
2010 Sep 12
1
username mismatch with 1.6.2.11
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I
get the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have <329909006666>, digest has <3291119600>
[Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite:
Failed to authenticate device "0473990000"
<sip:0473990000 at
2009 Jun 26
1
registration failed, not a local domain
asterisk*CLI> sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
[Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889
handle_request_register: Registration from
'<sip:grandstream at 192.168.1.248>' failed
2009 Jul 01
2
Registrations problems to SIP-provider.
Hello List,
I'm having problems with registrating my Asterisk-server to the
SIP-provider. Yesterday all worked fine, this evening I cannot call out.
What can be wrong ?
This is my registration in sip.conf :
register => 092779077:XXXX at 85.119.188.3
This the output of SIP show peers :
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port
Status
2010 Jul 23
6
Asterisk 1.8.0-beta1 is Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
2010 Jul 23
6
Asterisk 1.8.0-beta1 is Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
2009 Jul 06
1
Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
I have installed gnutls and gnutls-devel from RedHat repositories
[root at asterisk asterisk]# yum install gnutls gnutls-devel
I have installed iksemel with gnutls support :
[root at asterisk asterisk]# cd /usr/src/iksemel-1.3/
[root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr
[root at asterisk asterisk]# make
[root at asterisk asterisk]# make check
[root at asterisk
2010 Sep 13
2
Correct queue agi syntax in 1.6.2.11
Hello list,
what is the correct syntax ?
exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL})
[Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to
execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a':
File does not exist.
Kind regards,
Jonas.
-------------- next part --------------
An HTML attachment was
2010 Sep 24
2
Debug compile fails
Somehow I can't get 1.6.2.13 to compile with DEBUG_CHANNEL_LOCKS.
Downloaded latest tgz and extracted
$ ./configure
$ make menuselect
(select the needed options from compiler flags)
$ grep DEBUG_CHANNEL_LOCKS menuselect.makeopts
MENUSELECT_CFLAGS=DONT_OPTIMIZE LOADABLE_MODULES DEBUG_CHANNEL_LOCKS
MALLOC_DEBUG
$ make && make install
$ asterisk && asterisk -rx "core show