If you are going to use call files don't write them directly to
/var/spool/asterisk/outgoing/
write them in some temp directory and then move them to
/var/spool/asterisk/outgoing/
Ish
On Thu, 2011-05-19 at 10:58 -0600, Alejandro Mejia Evertsz wrote:
You only need to tell your PHP script to
Post your cdr_mysql.conf and res_mysql.conf and we'll take it from
there.
Don't forget to remove any 'private' info first (like passwords).
Cheers
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans
On Friday 20 May 2011, Dovid Bender wrote:
I had issue with call files. They would lock up the system (this was 5
years ago so maybe things have changed.)
Whenever you open a file for writing, a link is created in the containing
folder's directory (which says where on the disk the file is
CPU utilization is constantly above 24% without any call activity..
*top - 05:53:09 up 1:28, 2 users, load average: 0.18, 0.27, 0.29
Tasks: 79 total, 1 running, 78 sleeping, 0 stopped, 0 zombie
Cpu(s): 9.7%us, 2.3%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.2%hi, 0.0%si,
0.0%st
Mem:
Maybe IO-Activity caused by intensive logging. Take a look at your
Log-Files. Maybe one or more log files a growing "rapidly"?
Am 20.05.2011 11:24, schrieb RSCL Mumbai:
CPU utilization is constantly above 24% without any
call activity..
top -
logger.conf is only set for:
full = notice,warning,error,debug
I have now removed debug.
On Fri, May 20, 2011 at 2:57 PM, Thorsten Göllner t...@ovm-group.com wrote:
Maybe IO-Activity caused by intensive logging. Take a look at your
Log-Files. Maybe one or more log files a growing rapidly?
This seems to be an interesting post:
http://forums.virtualbox.org/viewtopic.php?t=12903
As per OP's message, CONFIG_HG is indeed 1000
[root@e1 ~]# grep CONFIG_HZ /boot/config-2.6.18-194.3.1.el5
# CONFIG_HZ_100 is not set
# CONFIG_HZ_250 is not set
CONFIG_HZ_1000=y
CONFIG_HZ=1000
[root@e1 ~]#
Hi all,
I am using asterisk 1.4.25.1. when I am sending dtmfs the first digit is not
detected.
Do you know a workaround for this?
Besst regards,
Szabolcs Szasz
--
_
-- Bandwidth and Colocation Provided by
Ok thank you so much for all advice
2011/5/20 A J Stiles asterisk_l...@earthshod.co.uk
On Friday 20 May 2011, Dovid Bender wrote:
I had issue with call files. They would lock up the system (this was 5
years ago so maybe things have changed.)
Whenever you open a file for writing, a
Today, sessions 320-321 of the VoIP Users Conference will take place
at the usual time, 12 Noon Eastern [ http://vuc.me/next for local
times ]
We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You
can download or watch his accompanying slide presentation here:
Hi
When we use the *8 feature to pick up a call on another extension, the
phone will only display *8 and *8 is what is stored in the phones
memory. Is there anything we can do so that when we use *8 the incoming
caller's CLI will be presented on the screen of the phone and in the
phones memory?
Hello list,
I want certain devices to monitor certain extensions/SIPaccounts and
other devices to monitor other extensions/SIPaccounts.
Therefore I do the following :
[from-TEST1]
include = test1-blf
[from-TEST2]
include = test2-blf
[test1-blf]
exten = 10,hint,SIP/testcorp1
exten =
I think I managed to solve this issue
The problem lay in the VirtualBox setting for the VM.
I will post the exact setting tomorrow which should help others.
Sorry for being a trouble to others :(
Best regards have a great weekend.
Sans
On Fri, May 20, 2011 at 3:03 PM, RSCL Mumbai
On Friday 20 May 2011, salaheddine elharit wrote:
Ok thank you so much for all advice
This might help you a bit, too:
?php
$spool = /var/spool/asterisk/outgoing/; # outgoing callfile folder
$filename = asterisk- . date(U) . - . $_SERVER[REMOTE_PORT] . .call;
# this should end up being fairly
thanks a lot for your help and support
2011/5/20 A J Stiles asterisk_l...@earthshod.co.uk
On Friday 20 May 2011, salaheddine elharit wrote:
Ok thank you so much for all advice
This might help you a bit, too:
?php
$spool = /var/spool/asterisk/outgoing/; # outgoing callfile folder
Hi
Do many people use this?
Is it reliable and safe?
Tanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
2011/5/20 Ishfaq Malik i...@pack-net.co.uk
Hi
When we use the *8 feature to pick up a call on another extension, the
phone will only display *8 and *8 is what is stored in the phones
memory. Is there anything we can do so that when we use *8 the incoming
caller's CLI will be presented on
Hi out there
To play the correct announcement in app_voicemail I whould be able to read the
SIP Diversion Reason which ist sent by another PBX:
Invite contains:
Diversion: sip:+41315995003@157.161.10.190;reason=no-
answer;privacy=off;counter=1
Asterisk Logs:
RDNIS for this call is is
2011/5/20 Benoit Panizzon benoit.paniz...@imp.ch
Hi out there
To play the correct announcement in app_voicemail I whould be able to read
the
SIP Diversion Reason which ist sent by another PBX:
Are those PBXs directly connected to each other through a SIP trunk ?
--
On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
If you are going to use call files don't write them directly to
/var/spool/asterisk/outgoing/
write them in some temp directory and then move them to
/var/spool/asterisk/outgoing/
Ish
Make sure that your temp
Attach a debug[1] log so we can see what is happening.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
debug logs below:
Asterisk 1.8.4: http://pastebin.com/DFnKgSse
Asterisk trunk r319661: http://pastebin.com/B19tdbxJ
--
Great! Satish,
I am middle of migration 1.2 queue in 1.8 thats why i encounter there. if i add
SIP/XXX then my queue working fine. Also i don't understand relation between
agents.conf and member = at queues.conf
let me read that URL and see what i can find there.
-S
Date: Fri, 20 May 2011
Hi Olivier
Are those PBXs directly connected to each other through a SIP trunk ?
Yes, and the reason is definitely transmitted in the SIP header and also
parsed by asterisk and displayed in debug output.
After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is
just put
In other words : is it correct to say that hints need to be unique, even
if they are defined in different contexts ?
On 05/20/2011 12:07 PM, Jonas Kellens wrote:
Hello list,
I want certain devices to monitor certain extensions/SIPaccounts and
other devices to monitor other
- Original Message -
From: Chris Maciejewski ch...@wima.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 19, 2011 9:39:57 AM
Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to
satisfy
debug logs below:
Asterisk 1.8.4: http://pastebin.com/DFnKgSse
Asterisk trunk r319661: http://pastebin.com/B19tdbxJ
These show that a proper bridging tech module cannot be found to run
ConfBridge. The debug message showing that a capability for ulaw couldn't be
found was a buggy debug
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.
Leif.
--
_
-- Bandwidth and Colocation
On 11-05-20 10:39 AM, Benoit Panizzon wrote:
After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is
just put in a temporary variable __SIPDIVERSIONREASON but not in a variable
useable in the dialplan.
You could double check by using DumpChan() to see what channel
On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.
Leif.
The reasons I'm considering it are as
Hi,
On 05/09/2011 09:40 PM, Justin Sherrill wrote:
Anyone have some recommended equipment for alerting people to calls in a noisy
environment?
I have Polycom IP550 phones set up in some really noisy environments - our mine
hoists - and they tend to drown out the ringers. I'm using Clarity
- Original Message -
From: Chris Maciejewski ch...@wima.co.uk
To: asterisk-users@lists.digium.com
Sent: Friday, May 20, 2011 8:56:35 AM
Subject: Re: [asterisk-users] ConfBridge - Failed to find a bridge technology
to satisfy capabilities
Attach a debug[1] log so we can see what is
Building a web page which uses AJAX to get information from the AMI
every 10-30 seconds or so and not wanting to log on and off via AMI that
many times.
You could use HTTP AMI inteface. You would need to login only once,
start a session and send requests to Asterisk.
I've used this
On Fri, 20 May 2011, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.
I had to set it up once for some CRM plugin (Possibly Sugar)
On Friday, May 20, 2011 10:46:05 AM Ishfaq Malik wrote:
On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
On Fri, 2011-05-20 at 11:27 -0400, Jose P. Espinal wrote:
Building a web page which uses AJAX to get information from the AMI
every 10-30 seconds or so and not wanting to log on and off via AMI that
many times.
You could use HTTP AMI inteface. You would need to login only once,
start
Is this the same as AJAM?
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
+Manager+(AJAM)
Yes, but I would definitively look for information in
http://wiki.asterisk.org as voip-info is getting an paleontological feel
now.
If you have any doubts, let me know in order to
On Fri, 2011-05-20 at 11:58 -0400, Jose P. Espinal wrote:
Is this the same as AJAM?
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
+Manager+(AJAM)
Yes, but I would definitively look for information in
http://wiki.asterisk.org as voip-info is getting an
thanks a lot for your advice i really appreciate it :)
2011/5/20 Mark Deneen mden...@gmail.com
On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
If you are going to use call files don't write them directly to
/var/spool/asterisk/outgoing/
write them in some temp
Hi,
I want to add static agent in queue so how to do that it seem 1.8 has very
different approach. I have added SIP extension but they are not getting calls.
@queues.conf
member = SIP/blah
member = SIP/blah
--
On Thu, 2011-05-19 at 21:10 +, satish patel wrote:
How to get rid on following.. why its Invalid ?
holler*CLI queue show queue1
queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
Agent/7201 (Invalid)
I do have agents in agents.conf. I am not using agentlogin apps. I am using
AddQueueMember
agent = 7101,,Agent1
agent = 7102,,Agent2
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Fri, 20 May 2011 11:56:23 -0500
Subject: Re: [asterisk-users] Agent (Invalid) has
On Fri, 2011-05-20 at 17:57 +, satish patel wrote:
I do have agents in agents.conf. I am not using agentlogin apps. I am
using AddQueueMember
agent = 7101,,Agent1
agent = 7102,,Agent2
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Fri, 20 May 2011
Hi Guys!
This is strange issue with 1.8 I have restarted my asterisk and it destroy all
registered SIP peers now only solution is i manually reboot all phones to get
them register back. I have never seen issue like this before. Any idea what
would be the issue ?
Thanks
S
Hi -
I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are
based in the US, so would need an ITSP with US DIDs.
#1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to
receive faxes via T.38. Sending faxes is not a requirement. Does anyone have
a
Hi,
I would to send a message to an incoming call with no answer. My Asterisk
server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for
instance).
I do the command playback with option noanswer, Asterisk send 183 followed by
RTP and finish with 603. But the BRI gateway do not
On Fri, May 20, 2011 at 2:10 PM, satish patel satish...@hotmail.com wrote:
Hi Guys!
This is strange issue with 1.8 I have restarted my asterisk and it destroy
all registered SIP peers now only solution is i manually reboot all phones
to get them register back. I have never seen issue like
We have polycom 501 and i am waiting since last 5 min no registration require
appear.
-S
From: mden...@gmail.com
Date: Fri, 20 May 2011 14:56:20 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP
peers
On Fri, May 20,
On Fri, May 20, 2011 at 3:00 PM, satish patel satish...@hotmail.com wrote:
We have polycom 501 and i am waiting since last 5 min no registration
require appear.
-S
With Polycom 321 you can poke around the menus -- one of them has a
countdown timer which will show you when the next
Issue is we are running customer support queue and if by chance if i need to
restart asterisk then they will not able to get call until phone get register
:( Let me check polycom default timeout and set to min.
-S
From: mden...@gmail.com
Date: Fri, 20 May 2011 15:03:35 -0400
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Friday, May 20, 2011 3:10 PM
To: asterisk-users
Subject: Re: [asterisk-users] Restart asterisk destroy all
registered SIP peers
On 05/20/2011 01:20 PM, e...@erols.com wrote:
#1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to
receive faxes via T.38. Sending faxes is not a requirement. Does anyone
have a working asterisk 1.8.4 configuration and ITSP provider that they can
recommend? We have
Hey Eric,
I do have qualify=yes. Am i missing something ?
[seb-exten](!) ; Template
type=friend
host=dynamic
context=from-sip
qualify=yes
dtmfmode=rfc2833
nat=no
cc_agent_policy=generic
cc_monitor_policy=generic
[7022](seb-exten)
callerid=Rover Conference 7022
There is a fix https://issues.asterisk.org/view.php?id=19318
--
Sent from my iPhone
On May 20, 2011, at 4:40 PM, satish patel satish...@hotmail.com wrote:
Hey Eric,
I do have qualify=yes. Am i missing something ?
[seb-exten](!) ; Template
type=friend
host=dynamic
The AstLinux Team would like to announce the immediate availability of
the 0.7.8 release. This release includes either Asterisk 1.4.41 or
Asterisk 1.8.4. All current users are encouraged to upgrade to this
release to take advantage of bug fixes and other updates to Asterisk.
Please note that
Hello,
I've created a patch to correct error responses for the MeetMeList manager
action. Currently MeetMeList produces an error if no conferences are active,
success if any conferences are open. Requesting a conference that is not
active while other conferences are active does not produce
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