e: ulimit [-SHacdfilmnpqstuvx] [limit]
> -
> How to set the ulimit command on in /etc/init.d/asterisk Since there is no
> parameter for ulimit in the file
>
> Thanks in advance
>
> Regards
>
>
&g
:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish
Patel
Sent: Monday, June 20, 2011 11:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk call limitation
It could be your OS limit try ulimit command.
--
Sent from my iPhone
On Jun 20, 2011,
It could be your OS limit try ulimit command.
--
Sent from my iPhone
On Jun 20, 2011, at 2:21 PM, "Kevin P. Fleming"
wrote:
On 06/20/2011 01:09 PM, Khaled W. Chehab wrote:
Dears,
i am using sipp to test asterisk(1.6.22) performance ,but when i
limit the
calls to 150 ,only 100 active
What company card you have? Copy paste your dahdi config and
chan_dahdi.conf
--
Sent from my iPhone
On Jun 15, 2011, at 6:53 AM, bilal ghayyad wrote:
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it UP, I have to
configure the two p
Problem solved. Just changed G1 to g1
--
Sent from my iPhone
On Jun 13, 2011, at 9:36 PM, James zhu wrote:
hi:
Please check the status of PRI, i think the channels keeps up and
down.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/
pri<->SIP)
k/voicemail/default/7000/tmp/L7WLl1 format: wav, 0x29e9ae8
[Jun 10 10:02:07] VERBOSE[7287] app_dial.c: -- DAHDI/i1/18004815122-1dc
answered SIP/7081-05f2
[Jun 10 10:02:07] DEBUG[7287] channel.c: setting peeraccount to "Pascal
Honscher" for SIP/7081-05f2 from data on channel D
Hi,
We having some PRI call drop issue on asterisk 1.8.x but we had no issue never
ever on asterisk 1.2. Anybody else having this issue ?
-S
--
_
-- Bandwidth and Colocation Provided by
Hi,
Anybody know how to set polycom 501 subscription expiry ?
-S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introduct
UDP 172.30.245.143;branch=z9hG4bK2b7c62c3FA125372
From: "Satish Patel" ;tag=9FBFC6B1-EE9095EE
To: ;tag=as65ea68d2
CSeq: 6 SUBSCRIBE
Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFE
172.30.245.143;branch=z9hG4bK2b7c62c3FA125372
From: "Satish Patel" ;tag=9FBFC6B1-EE9095EE
To: ;tag=as65ea68d2
CSeq: 6 SUBSCRIBE
Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143
Contact:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
Event: message-summary
isk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
>
> On 8 June 2011 17:20, satish patel wrote:
> > Interesting thing is when i reload sip.conf i got MWI lamp working on
> > polycom 501
> >
> > But its not working when anyone leave voic
Sure, but how to check which CA my iPhone using ?
--
Sent from my iPhone
On Jun 8, 2011, at 6:00 PM, Andrew Latham wrote:
On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel
wrote:
It not working on iPhone. It's saying not able to make secure
connection
--
Sent from my iPhone
Satish
dahdi/system.conf) for outgoing call.
(2) To dial from channel 25 , use DAHDI/25/XXX
[SATISH]
On Thu, Jun 9, 2011 at 9:39 AM, satish patel
wrote:
Awesome!!
Do you know if i want to use only specific channel for call out then
how do i write dialplan ? I want to use channel 25 specific f
Awesome!!
Do you know if i want to use only specific channel for call out then how do i
write dialplan ? I want to use channel 25 specific for my extension
DAHDI/25/ or DAHDI/i2/25/XXX
> Date: Wed, 8 Jun 2011 17:25:44 -0500
> From: rmudg...@digium.com
> To: asterisk-users@li
Hi,
We have two pri line and I want to see how asterisk distribute
outgoing call per channels
I meant it use first last channel 47 or it will use first channel?
Or it will allocate dynamically ?
--
Sent from my iPhone
--
_
It not working on iPhone. It's saying not able to make secure
connection
--
Sent from my iPhone
On Jun 8, 2011, at 4:54 PM, "Kevin P. Fleming"
wrote:
On 06/08/2011 02:27 PM, Andrew Latham wrote:
On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant
wrote:
A number of people are reporting
I'm using firefox and now it's works befrore after fill out
information submit I got blank page.
--
Sent from my iPhone
On Jun 8, 2011, at 4:54 PM, "Kevin P. Fleming"
wrote:
On 06/08/2011 02:27 PM, Andrew Latham wrote:
On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant
wrote:
A number of
Bad day today. Why this new JIRA system not working. I have created issue and
submit and i got blank page.. Please someone help me to create BUG!!!
--
_
-- Bandwidth and Coloca
Hey Guys!
Please help me to find out issue. I have two PRI
## Span 1: WPT1/0 "wanpipe1 card 0"
span=1,1,0,esf,b8zs
bchan=1-23
hardhdlc=24
echocanceller=mg2,1-23
## Span 2: WPT1/1 "wanpipe2 card 1"
span=2,2,0,esf,b8zs
bchan=25-47
hardhdlc=48
echocanceller=mg2,25-47
Sometime my calls got throu
message in a mailbox, does "voicemail show users" show
>>new messages for that mailbox?
Yes, I can see there are 10 voicemail
root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623
default7623 Satish Patel 10
> From
7;voicemail show users' | grep -i 7623
default7623 Satish Patel 10
> From: ewiel...@nyigc.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 8 Jun 2011 11:33:31 -0400
> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
>
>
&g
the [default] section?
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > satish patel
> > Sent: Wednesday, June 08, 2011 11:15 AM
> > To: asterisk-users
> > Subject: R
.8 tarball. Make
> sure your mailboxes specify a voicemail context on each mailbox= line.
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > satish patel
> > Se
I do have that
sip.conf
[7623](cam-exten)
callerid="Satish Patel" <7623>
accountcode="Satish Patel"
mailbox=7623@default
> From: ewiel...@nyigc.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 8 Jun 2011 11:03:24 -0400
> Subject: Re: [a
lists.digium.com
> Date: Wed, 8 Jun 2011 10:34:16 -0400
> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
>
> All major changes are listed in the UPGRADE.txt files included in the 1.8
> tarball.
>
> > -Original Message-
> > From: asterisk-users-boun..
.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > satish patel
> > Sent: Wednesday, June 08, 2011 9:57 AM
> > To: asterisk-users
> > Subject: [asterisk-users] Asterisk 1.8 broken MWI
> >
> > Hi ALL,
> >
> > After u
Hi ALL,
After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf.
Do i need to do anything else to fix my MWI on polycom 501 ? It was working
with 1.2 asterisk.
pollmailboxes=yes
--
_
e desired destination - for whatever
> reason.
>
> Am 08.06.2011 12:55, schrieb Satish Patel:
> > Thanks for reply,
> >
> > But I'm able to call those number from my cell phone and othere pri.
> >
> > I'm only having this issue on 2 pri line rest are wo
Thanks for reply,
But I'm able to call those number from my cell phone and othere pri.
I'm only having this issue on 2 pri line rest are working ?
--
Sent from my iPhone
On Jun 8, 2011, at 5:44 AM, Doug Lytle wrote:
satish patel wrote:
We are getting hangup cause
We have 2 PRI from AT&T
And all is well but only few numbers having following issue. We are getting
hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and
this issue raised
[Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got
hangup request, c
Hi ALL,
Is there any way i can reload chan_dahdi.conf without disconnecting active PRI
calls ?
I want to change pridialplan= option
-S
--
_
-- Bandwidth and Colocation Provided by ht
Hi ALL,
Is there any way i can reload chan_dahdi.conf without disconnecting active PRI
calls ?
I want to change pridialplan= option
-S
--
_
-- Bandwidth and Colocation Provided by ht
k-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:24:06 -0500
Subject: Re: [asterisk-users] PRI issue its BUSY
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, June 06, 2011 8:20
PM
To: aster
sion (from-sip, h, 1) exited non-zero on 'SIP/7328-0004'
From: ca...@usawide.net
To: asterisk-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:24:06 -0500
Subject: Re: [asterisk-users] PRI issue its BUSY
From:
asterisk-users-boun...@lists.digium.com
[mailto:aster
ed non-zero on 'SIP/7328-0004'
From: ca...@usawide.net
To: asterisk-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:24:06 -0500
Subject: Re: [asterisk-users] PRI issue its BUSY
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On B
-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, June 06, 2011 8:20
PM
To: asterisk-users
Subject: [asterisk-users] PRI
issue its BUSY
Hi all,
I just configures my PRI and incoming calls are working fine but outside
calling giving error PRI is BUSY :( any idea ? I have
Following is debug of pri
sfpbx1*CLI> pri set debug on span 1
Enabled debugging on span 1
== Using SIP RTP CoS mark 5
-- Executing [7076941815@from-sip:1] Dial("SIP/7328-0004",
"DAHDI/G1/17076941815") in new stack
1 -- Making new call for cref 32772
-- Requested transfer capability
Hi all,
I just configures my PRI and incoming calls are working fine but outside
calling giving error PRI is BUSY :( any idea ? I have same setup on other box
and that boxes works perfect.
-- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002
-- DAHDI/i1/6463279153-2 is
But still question is how it was working before? Did asterisk 1.8 has
some feature to manipulate polycom remotely ?
--
Sent from my iPhone
On Jun 6, 2011, at 5:12 PM, satish patel wrote:
look like we found issue in phone configuration files [2-9]xx
From: satish...@hotmail.com
To
look like we found issue in phone configuration files [2-9]xx
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:43:22 +
Subject: [asterisk-users] asterisk 1.8 issue with polycom dialplan
Hi all,
I have just upgrade asterisk 1.2 to 1.8 and we ha
Hi all,
I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from
_71XX. now what happen if i dial any 711X number my polycom just dial 711 and
say busy number look like my phone doing some regex itself. like 911 number..
Did you get what i am trying to say ? it was working be
Thanks but they should change svn revesion number change in file.
--
Sent from my iPhone
On Jun 5, 2011, at 7:13 PM, Barry Miller
wrote:
On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote:
Hey guys!
I have just download latest SVN Revision 322051 and compile and
install but
Hey guys!
I have just download latest SVN Revision 322051 and compile and install but my
asterisk -V showing still old version :( is it broken ?
/usr/sbin/asterisk -V
Asterisk SVN-branch-1.8-r321926
--
__
00
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] benefits of asterisk 1.8
No, it just means that the coredump will not have information that is as useful
Sent from my iPhone
On Jun 3, 2011, at 10:02 AM, satish patel wrote:
Sherwood,
I was wrong here
>>But unfortun
n 2011 09:53:01 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] benefits of asterisk 1.8
Message body
On 6/3/2011 9:49 AM, satish patel wrote:
But unfortunately i compiled with "DON'T OPT
terisk 1.8
>
> On 11-06-03 07:30 AM, Satish Patel wrote:
> > Yesterday my 1.8 got crashed and I have nothing in log or anywhere which
> > I can show you or submit bug. Kinda funny :(
> >
> Sounds like asterisk was not told to generate a coredump, add the
> following,
Hey Guy,
I want to implement Queue base custom ring tone so Agent will get aware of
incoming call for sale or tech etc.. I know its possible with SIPAddHeader
http://www.technicallyamusing.com/?p=44
I am confused here
We already have alertInfo set to "Ring Answer" how should i use both r
Yesterday my 1.8 got crashed and I have nothing in log or anywhere
which I can show you or submit bug. Kinda funny :(
--
Sent from my iPhone
On Jun 3, 2011, at 5:06 AM, Satish Barot
wrote:
If 1.8 doesn't panic for subset of PBX features for someone, you can
not say it is stable. You s
Hi Guys!
If i reload my asterisk it create /var/log/asterisk/* file with root
permission. I am running asterisk with asterisk user and group. Do you have
any idea ?
root@campbx1:~# ls -l /var/log/asterisk/
total 716
drwxr-xr-x 2 asterisk asterisk 4096 2011-05-06 15:38 cdr-csv
drwxr-xr-x 2
Is this available in current SVN ?
> Date: Thu, 2 Jun 2011 15:07:50 -0400
> From: asteriskt...@digium.com
> To: asteriskt...@digium.com
> Subject: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)
>
> The Asterisk Development Team has announced the release of Asterisk
> version
Hey,
Sometime i am getting following messaged on asterisk CLI console just wondering
what these messages are look like some codec related.
[May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping
incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our
native fo
Hi Guys!
We were using queuemetrics since long time with asterisk 1.2 but recently we
have install 1.8 asterisk and but there is a big different in queue_log its
saying SIP/ instead of Agent/ that is obvious behaviors. so do i need
to change Agent/ to SIP/ in queuemetrics ? or
I our setup we don't have DNS or Internet connectivity but we are good
no issue so far.
--
Sent from my iPhone
On May 31, 2011, at 7:24 AM, Hans Witvliet wrote:
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote:
On Mon, 30 May 2011, Sherwood McGowan wrote:
True, but with all due
Did you try different number in place of 5? I meant 1 2 etc..
Also check cli logs on console
Are you dialing from softphone or hardphone because some phone has
dialing regex for security.
--
Sent from my iPhone
On May 30, 2011, at 1:30 PM, salaheddine elharit > wrote:
Hello list
i have
ing source is the clock of the system. When a equipment is 0,
the other should be 1. The correct is: 0=slave, 1=master. The
default for private systems is "slave".
Att,
Rafael Saraiva
2011/5/27 satish patel
Hi There,
We have very old asterisk 1.2 running in production and it ha
Thanks also let me clear one thing this pri is PSTN connected to AT&T
techo.
So they are master.
--
Sent from my iPhone
On May 27, 2011, at 5:51 PM, Shaun Ruffell wrote:
On Fri, May 27, 2011 at 05:40:30PM -0400, Satish Patel wrote:
Got it but still confused. As per your example I sh
e but none of clear.
--
Sent from my iPhone
On May 27, 2011, at 5:41 PM, Edwin Lam
wrote:
On 5/27/11 2:20 PM, satish patel wrote:
Tell me in one word. We have 2 PRI line connected with sangoma card
what option
would be good for me?
0 or 1 ?
that would depends on what's the other en
Got it but still confused. As per your example I should go with
Port 1
Span=1,1,0
Port 2
Span=2,2,0
Correct me if I'm wrong.
--
Sent from my iPhone
On May 27, 2011, at 5:32 PM, Shaun Ruffell wrote:
On Fri, May 27, 2011 at 09:20:46PM +0000, satish patel wrote:
Tell me in one wor
>
> On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote:
> >
> > You mean say
> >
> > 0=Slave (Use PSTN clock)
> > 1=Master(generate Internal clock)
> >
> > So best option is 0 for all span if you connected on PSTN right ?
>
source
Hi
The timing source is the clock of the system. When a equipment is 0, the other
should be 1. The correct is: 0=slave, 1=master. The default for private systems
is "slave".
Att,Rafael Saraiva
2011/5/27 satish patel
Hi There,
We have very old asterisk 1.2 running in product
This has been submitted.
-S
> Date: Fri, 27 May 2011 16:05:28 -0400
> From: leif.mad...@asteriskdocs.org
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue
>
> On 27/05/11 03:18 PM, satish patel wrote:
> > In th
ueue
This is working great! Thanks a lot paul.
One more question before we have Agent/ configured in queueMetrics so i
need to change them in queueMetrics with SIP/ right ?
> Date: Fri, 27 May 2011 10:18:39 +0100
> From: p...@provu.co.uk
> To: asterisk-users@lis
One more question before we have Agent/ configured in queueMetrics so i
need to change them in queueMetrics with SIP/ right ?
> Date: Fri, 27 May 2011 10:18:39 +0100
> From: p...@provu.co.uk
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1..8 mu
Subject: Re: [asterisk-users] Asterisk 1..8 multiple queue
>
> On 26/05/11 23:18, Satish Patel wrote:
> > Thanks,
> >
> > I went through this example before. I was confuse and wondering how
> > should I add third queue in this picture?
> >
>
>
Hi There,
We have very old asterisk 1.2 running in production and it has following
setting in /etc/zaptel.conf. I have read on web about span and they told
span= [,yellow]
Just wondering why it has timing source 0 ? 0=master, 1=slave right ? Do you
think i should change it to 1 ?
#
digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > satish patel
> > Sent: Friday, May 27, 2011 10:42 AM
> > To: asterisk-users
> > Subject: [asterisk-users] DID for outbound PSTN call
> >
> > Hi There,
> >
> > We have single PRI wit
Hi There,
We have single PRI with multiple DID numbers and its working fine in receiving
call. And if you make outbound call it will send main-line CallerID (company
name). Now we want individual caller id for per extensions on outbound calls.
like if i call someone he will get my extension as
That's cool. I will give it a shot and let you guys know.
--
Sent from my iPhone
On May 27, 2011, at 5:18 AM, Paul Hayes wrote:
On 26/05/11 23:18, Satish Patel wrote:
Thanks,
I went through this example before. I was confuse and wondering how
should I add third queue in this pi
Thanks,
I went through this example before. I was confuse and wondering how
should I add third queue in this picture?
--
Sent from my iPhone
On May 26, 2011, at 5:43 PM, Leif Madsen
wrote:
On 26/05/11 04:20 PM, satish patel wrote:
Actually right now i have very big AddQueueMember
recall correctly you can even connect SQLite and DB2.
>
> However, let me ask you this...what trouble are you having with
> AddQueueMember and it's related applications that is making it hard for you?
>
> Sent from my iPhone
>
> On May 25, 2011, at 7:20 PM, Satis
Thanks for reply but is there any alternative way? Because we don't
have mysql and we dont want to use mysql.
--
Sent from my iPhone
On May 25, 2011, at 6:43 PM, Sherwood McGowan > wrote:
On 5/25/2011 12:32 PM, satish patel wrote:
Hey Guys!
We had migrate asterisk 1.2 to 1.8
Hey Guys!
We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had
3 queues and we were using AgentCallbackLogin but now its quite difficult to
use AddQueueMember.
Is there any easy way to logged into multiple queue using AddQueueMember ? and
restrict agent for speci
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 wrote:
Hey Eric,
I do have qualify=yes. Am i missing something ?
[seb-exten](!) ; Template
type=friend
host=dynamic
context=from-sip
qualify=yes
y 2011 15:15:45 -0400
> Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP
> peers
>
>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > s
-users@lists.digium.com
Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP
peers
On Fri, May 20, 2011 at 3:00 PM, satish patel wrote:
We have polycom 501 and i am waiting since last 5 min no registration require
appear.
-S
With Polycom 321 you can poke
, 2011 at 2:10 PM, satish patel 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 this before. Any idea what
would be
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
d) has taken no calls yet
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
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
--
__
to read 'The agents.conf File' section from given link for
more information.
[SATISH]
On Fri, May 20, 2011 at 2:40 AM, 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'
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) has taken no calls yet
Agent/7202 (Invalid) has taken
risk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, 16 May, 2011 3:48:05 PM
> Subject: Re: [asterisk-users] dahdi command not available
>
> Run Service dahdi start
> -Original Message-
> From: satish patel
> Sender: asterisk-users-boun...@li
agents.conf
agent => 7101,1234,Agent1
agent => 7102,1234,Agent2
queues.conf
...
...
member = Agent/7201
member = Agent/7202
CLI output
holler*CLI> queue show queue1
queue1 has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s
talktime), W:0,
How much memory have allocate to VM ? and send top or ps command output.
Date: Thu, 19 May 2011 22:44:58 +0530
From: rscl.mum...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk-cpu utilization > 60 %
Processor: Intel Dual Core Xeon 3.0GHz
-> Host: CentOS 5.6
I am reading at http://www.asteriskguru.com/tutorials/queues.html
They are using member in both static and dynamic method.
member => /
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Sometime reboot does help.
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Sent from my iPhone
On May 19, 2011, at 8:09 AM, vip killa wrote:
I'm sure it's not nagios. I'm not running "check_sip" and i'm
running nagios' NRPE on several other machines that do not have
asterisk running.
On Wed, May 18, 2011 at 4:43 PM, Alex Balashov >
Holy cow! you made my day
Thank you so much... It works great!!!
S.
From: mden...@gmail.com
Date: Tue, 17 May 2011 17:02:55 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] script to trim sip.conf
On Tue, May 17, 2011 at 4:21 PM, satish patel wrote:
Hey
Hey Guys!
Sorry i am posting scripting question in asterisk forum but i had no choice.
also i am not script expert so i though anyone here might help me.
following is my example sip.conf now i want to add
accountcode="" for example accountcode="Katie Wilson" in
entire file. we have arou
Hi All,
I have just latest branch of asterisk 1.8 and i didn't found dahdi command in
CLI everything seem fine. am i missing something ?
campbx2*CLI> dahdi
No such command 'dahdi' (type 'core show help dahdi' for other possible
commands)
campbx2*CLI>
root@campbx1:/etc/wanpipe# wanrouter h
First grab LWP thread ID which is eating more CPU
ps -LlFm -p `pidof asterisk`
Now look into your asterisk.stack.txt and search particular LWP thread ID see
following example
Thread 10 (Thread 0x41d8f940 (LWP 3406)):
#0 0x0033ce2ca436 in poll () from /lib64/libc.so.6
#1 0x
Sorry fro hijacking thread. I have following process running on my asterisk
eating around 2 or 3% CPU constantly. I knew events0/1 is CPU queue but why
only single queue is busy ? I have kernel running preemtive with 1000Hz
satish@campbx1:~$ ps aux | grep events
root 9 1.7 0.0 0
Thanks Leif,
I had changed it to res_timing_dahdi and since last few days it seem good.
-S
> Date: Sun, 15 May 2011 15:48:03 -0400
> From: leif.mad...@asteriskdocs.org
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so
>
> On 11-
dahdi timing is
selected and system become stable.
Regards,
tbskyd
2011/5/14 satish patel :
You mean say i don't use res_timing_dahdi.so ? I guess this is
just timing
module nothing related to Card.
_S
From: tu...@canistec.com
Date: Fri, 13 May 2011 18:
Check this out
http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/
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Sent from my iPhone
On May 15, 2011, at 4:08 AM, Tzafrir Cohen
wrote:
On Sun, May 15, 2011 at 08:24:08AM +0200, Leandro Dardini wrote:
2011/5/15 RSCL Mumbai
On Sat, May 14, 2011 at 1
o not use dahdi...
13.5.2011 v 17:16, satish patel :
Thank you so much!! I found following (res_timing_timerfd.so in USE). But we
have asterisk dahdi install and sangoma A102D pri card configured. Do you
think i should use res_timing_dahdi.so ?
campbx1*CLI> module show like timin
I found asterisk using res_timing_timerfd.so do you think i should use
res_timing_dahdi.so ?
campbx1*CLI> module show like timing
Module Description Use
Count
res_timing_pthread.so pthread Timing Interface 0
terisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of satish patel
Sent: 13 May 2011 16:03
To: tbs...@gmail.com; asterisk-users
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
Thanks for reply,
How do i find asterisk using which
Hi All,
We have asterisk 1.8 with Sangoma A102D PRI card and issue is sometime asterisk
process doing beyond 100% CPU load and only solution is kill. I did google and
many people talking about timing issue in asterisk i have check in asterisk and
i have three timing module are loaded in asteri
of the default res_timing_timerfd. I don't know if these are related
> to you problem. hope you can find the key point to make a stable
> asterisk.
>
> Regards,
> tbskyd
>
> 2011/5/13 Satish Patel :
> > Glad you solved it. Now I'm having high CPU load issue. I
dware sip phone and found that "prematuremedia=no" is
still necessary.
Regards,
tbskyd
2011/5/11 satish patel :
I am sorry about that but its interesting it doesn't work with
1.8 SVN
I would say please report this bug so that way you can track
issue, And may
be in future it help us :
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