Logs attached.
Thanks in advance!
On Fri, May 30, 2014 at 11:54 PM, Prakash N prakas...@tevatel.com wrote:
Hi ,
Can you post cli log
With regards
N.Prakash
--
From: RSCL Mumbai rscl.mum...@gmail.com
Sent: 30-05-2014 11:16 PM
To: Asterisk Users Mailing
Hi,
I am using Asterisk 1.4 along with FreePBX.
My call flow is as follows:
Inbound DID Inbound Route Time Condition Queue.
My welcome greeting MP3 is setup under System Recordings its called
under Queue Join Announcement.
Not sure why, the MP3 audio file starts to play after a 5 sec
,Plaback(something)
exten s,n,Dial(agent)
exten s,n,Hangup duration and billsec end here
so billsec is 10 seconds less then duration
hope this will help you.
On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.comwrote:
I am using SIP.
I am still a bit confused about
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec duration is same.
With reference to the attached log, what does the 10 sec / 6 sec / 2
sec correspond
to:
(a) Time between call connection to asterisk and disconnection from
] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR
Hi,
Attached is a sample CDR.
I need some help to understand the billsec column.
PS: the time value in billsec
Hello,
I have Elastix ISO install (FreePBX 2.7.0.3)
My current Setup is as follows:
Inbound Route Queue (Dynamic Agents)
The queue distributes calls based on rrMemory.
I have been asked to redesign the call distribution as follows:
Calls will be delievered to Level-1 Agents (say 4 dynamic
Hello,
I am trying to construct MySQL query(s) to get a list of calls which lasted
for less than 5 seconds between a given date range.
Any help is appreciated.
Thank you in advance.
Regards,
Sans
--
_
-- Bandwidth and
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai
*Sent:* Friday, September 14, 2012 11:16 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] MySQL Query : Calls Answered for 5 sec
** **
Hello,
I am trying to construct MySQL query(s
@Raj
I tried your query and variation by using replacing duration with billsec.
In both cases, I get results including disposition NO ANSWER
On Fri, Sep 14, 2012 at 9:58 PM, Raj Mathur (राज माथुर)
r...@linux-delhi.org wrote:
On Friday 14 Sep 2012, RSCL Mumbai wrote:
I am trying
I need a list of calls Answered and Disconnected in less than 5 sec.
Thx
On Fri, Sep 14, 2012 at 10:07 PM, Warren Selby wcse...@selbytech.comwrote:
On Fri, Sep 14, 2012 at 11:33 AM, RSCL Mumbai rscl.mum...@gmail.comwrote:
@Raj
I tried your query and variation by using replacing duration
Hi,
Seeking recommendations for a good quality hosted predictive dialer service.
Low volume, single agent US dialing.
Thank you.
Best regards,
Sans
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
I would recommended FOP2
Its awesome.
On Fri, Nov 4, 2011 at 3:04 PM, Anthony Laudini alaudini.lo...@gmail.comwrote:
Hi Jean,
I suggest Queuemetrics. There are many out there but this one is good for
monitoring and reporting.
I know there's a free version you can try.
All the best
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in advance.!
--
Esteban L. Cacavelos de Amoriza
Cel:
Hi,
Using Asterisk 1.6.2.13
We are now starting to use *call transfer (patching) function.*
Call flow is as follows:
---
John Calls me and requests him to be connected to Nancy.
I place John's call on Hold
I dial Nancy and speak with her about John
I then patch the call
My favorite is didww.com ,
another one is ipcomms.net (not very prompt with their customer service)
Hope this helps..
On Sat, Oct 1, 2011 at 12:51 AM, amit mehta amit.magn...@gmail.com wrote:
Hello members,
I am looking for USA incoming DID which can be registered on softphone/IP
Phone/
On Sat, Sep 3, 2011 at 1:56 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
I tried and failed with VirtualBox too. Timing seemed impossible to
maintain, even on beefy hardware (hexacore)
with plenty of RAM (16G), and nothing else going on (single
Asterisk on VirtualBox ?
On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote:
My main interest of being on Virtual platform is portability / Backup.
In case of any h/w issues, or crashes, simply copy the VM on to
another box and you are up in minutes.
Sanjay
Bruce, that's exactly the command I was looking for.
Thx a ton.
Sans
On Thu, Sep 1, 2011 at 12:17 AM, Bruce B bruceb...@gmail.com wrote:
sip show channels is the command you are looking for.
On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai rscl.mum...@gmail.comwrote:
asterisk -rx core show
. However it does not track transcoding.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August 31, 2011 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Thursday, September 01, 2011 5:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cli command show codecs
Hi,
Does audio files
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Thursday, September 01, 2011 8:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cli command show codecs
Thx @Danny
I am feeling
Surprisingly, despite the error message, the files is uploaded in
/var/lib/asterisk/mohmp3 with correct permissions and ownership.
Its not showing in FreePBX MOH Screen.
I guess its a FreePBX issue.
Sans
On Thu, Sep 1, 2011 at 7:56 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Thanks again
Hi,
Anyone using Asterisk on Virtualbox.
I am using and facing CPU peaking issue.
Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores
and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of
now), 64bit CentOS 5.4.
Only SIP and softphones.
Max 10 simultaneous
On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
Hi,
Anyone using Asterisk on Virtualbox.
I am using and facing CPU peaking issue.
Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores
and 2 GM RAM
On Thu, Sep 1, 2011 at 9:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
Hi,
Anyone using Asterisk on Virtualbox.
I am using and facing CPU peaking issue.
Hardware is IBM
Hi,
Is there a CLI command which will tell me the codec used for active calls
and if transcoding is happening ?
Thx
Sans
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
* version.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai
*Sent:* Wednesday, August 31, 2011 10:44 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] cli command show
/8 RSCL Mumbai rscl.mum...@gmail.com
On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif fai...@vopium.com wrote:
If you take a bit deep analyses on SIP packet you will be able to
understand the issue,
** **
Iptables filter on layer-3 while SIP is on layer-7. It is easily possible
to generate
Hi,
(1) Since a few days, I am seeing unexpected (unwanted) calls reaching my
asterisk server.
Please see attached log files.
(2) I believe the source IP of these calls is the IP mentioned under the
CHANNELS column.
(3) But as per my firewall, these calls should not have reached Asterisk.
The
On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин anton.juga...@gmail.comwrote:
Hi,
Could you attach iptables-save output.
iptables-save output is blank -- no output.
Not sure why ?
Thx
Sans
--
_
-- Bandwidth and Colocation
On Mon, Aug 8, 2011 at 5:09 PM, Henrik sing...@common-hacking.org wrote:
**
Also you can set allowguest=no in sip.conf, if you didn't do it already
I will check sip.conf, but logically, the packets should not be reaching
Asterisk.
IP Tables should have blocked them.
Sans
--
For some unknown reason, the firewall script was not executed.
Now I get the output of iptables-save.
May be this is the reason why unwanted packets hit the system... a big
blunder.
Sans
On Mon, Aug 8, 2011 at 5:44 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
On Mon, Aug 8, 2011 at 4
2011/8/8 Антон Квашёнкин anton.juga...@gmail.com
lsmod | grep ipt
And what distribution do you use?
[root@e1 ~]# lsmod | grep ipt
ipt_REJECT 38977 1
iptable_filter 36161 1
iptable_nat40773 0
ip_nat 53101 1 iptable_nat
ip_conntrack
On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif fai...@vopium.com wrote:
If you take a bit deep analyses on SIP packet you will be able to
understand the issue,
** **
Iptables filter on layer-3 while SIP is on layer-7. It is easily possible
to generate a SIP packet with different
Hi,
I seem to be facing an intrusion issue, inspite of firewall (script attached).
What am I missing ??
Any suggestions / recommendation are welcome pls.
Best regards,
Sans
#!/bin/bash
echo 0 /proc/sys/net/ipv4/ip_forward
# Clear any existing firewall stuff before we start
/sbin/iptables
I want to secure my server from the hacker's. What is the case by which I
can protest it.
I have done security of Dialplan, Sip,IAX base security. For linux we are
working on Iptables. What else is left so that I will do it too...
Can you share the steps / scripts / settings done to
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:
?
Am 20.05.2011 11:24, schrieb RSCL Mumbai:
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
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 ~]#
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
Processor: Intel Dual Core Xeon 3.0GHz
- Host: CentOS 5.6 (64 bit)
-- Virtualbox 4 (64 bit)
--- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3
tail -f full shows the below:
[May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame
on SIP/voxbone.com-0139 of format ulaw since
Brummell te...@brummell.net wrote:
For 2 different hosts. SIP/voxbone.com and SIP/4420
--
*From:* RSCL Mumbai
*Sent:* Thu 5/19/2011 12:23 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dropping incompatible voice
Processor: Intel Dual Core Xeon 3.0GHz
- Host: CentOS 5.6 (64 bit)
-- Virtualbox 4 (64 bit)
--- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3
Anyone else facing high CPU usage problem with Asterisk 1.6.2.13 or any
Elastix 2.0.3 users here ?
With just 3 concurrent calls and none in queue, the CPU is
On Mon, May 16, 2011 at 6:19 PM, Pezhman Lali l...@lopl.net wrote:
check your running process, if you have more than one asterisk in your
top re install your asterisk.
On Sun, May 15, 2011 at 7:07 PM, Satish Patel satish...@hotmail.comwrote:
Check this out
http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/
Moving forward with the suggestion provided on the above link, I have the
activity dump of all asterisk processes when the load was 22%.
Need help in understanding the output.
What should I look for which
On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.comwrote:
Check if someone is brute forcing your asterisk accounts. It used to happen
to me before I install fail2ban. You can easily check the full log of
asterisk or with just a tcpdump -i any -n port 5060 or port 4569.
Thx
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing
Hi,
On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest.
Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly
1-2 concurrent calls. No other activity on server. Top shows asterisk on
top.
Its quad xeon server with 4 gb ram.
Any suggestion where should I
Hi,
I am using Trixbox 2.6.2.3, ISO install
I am getting the below error in `/var/log/asterisk/full`
Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Is there anyway to figure out which extension is causing this error ?
Thank you.
Best regards,
Sanjay
--
Thx Rudi. but this query results in *Empty set (0.32 sec)
src AND dst like number *seems to be the problem area.
*
*
Also, how can I get the hold time talk time as separate values OR may be
total call connect time talk time (the difference of the 2 will be hold
time).
Thx
Sans
On Wed, Aug 4,
Hi,
Can someone help me formulate MySQL Query(s) which will help me extract the
following details for a given DID (date range can be excluded for
simplicity).
Date-Time
DNID (I am recording this is `userfield`)
CLID
time-1 (when call was received)
time-2 (when call was answered by agent)
time-3
Hi,
I am looking for a Windows Desktop based application which will open a web
browser with the below url upon CALL RING on the softphone.
*http://192.168.1.4:3100/popup.php?did=DNID* (where DNID is the called DID
number)
Let me know for any help!!
Thank you.
Best regards,
Sanjay
--
Hi,
Looking for some reliable and quality providers of USA DIDs.
Any pointers ?
Thx
Sans
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick r...@readywire.com wrote:
Agreed! Didforsale.com is THE way to go.
--
Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
Anyone having experience with didww.com ?
Sorry, I forgot to mention I am looking for wholesale DID --
Hi,
I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4
I am looking for the following functionality:
``
I receive a call from Mr. A.
I put Mr. A on hold.
I dial Mr. B
I connect Mr. A's
08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read
08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread in the Asterisk Developer List, but I could
not figure out how to implement it ?
Is there a simple step-by-step.
Thx in advance.
Vai
--
_
--
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread
What is the command to log off the agents ?
Thx
On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
You could configure them as agents and have them log off automatically
after a while they're not responding.
l.
2009/10/14 Benny Amorsen
On Sat, Oct 10, 2009 at 7:59 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 10 Oct 2009, gergis.rasmy wrote:
can i use MP3 files as an IVR prompts directly without converting to
.gsm format?
You don't want to do this.
Asterisk will attempt to use prompts encoded with the same
On Sat, Oct 10, 2009 at 11:47 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Sat, 10 Oct 2009, RSCL Mumbai wrote:
How should I convert my .wav prompts into aLaw, uLaw, G729 ?
The standard Asterisk prompts are already available in a wide variety of
encodings.
Try googling
On Fri, Oct 9, 2009 at 2:18 AM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Thu, 2009-10-08 at 16:07 -0400, Michelle Dupuis wrote:
More specificallyI'm looking for a Linux package to allow shaping,
QoS, prioritization by port, etc.
snip
Spinning off from another
Hi,
I am using Trxibox 2.6 latest ISO install.
Following is the output of : sip show channels
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
Hold Last Message
212.53.40.40 0218245 6cfb845d050 09011/0 0x0
do you have that user 1006 defined by IP ?
*I have a user 1006.
Its not defined by IP.
*
does it have mailbox= also defined ?
*Yes. 1006 has a Mail box*.
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
*I will delete all
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
*I will delete all messages from the Mailbox and see if 1006 is removed
from the listing.*
Just checked, no messages in 1006.
Any other reasons!
Thx
Sanjay
Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?
Thx
Vai
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
Hold Last Message
192.168.1.126(None) MjkzYjNiMmY 00101/4 0x0
On Fri, Sep 25, 2009 at 10:27 PM, Philipp Kempgen philipp.kemp...@amooma.de
wrote:
RSCL Mumbai schrieb:
Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Use grep. (See `man grep`.)
I may
69 matches
Mail list logo
