in advance
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel
Sent: Tuesday, June 21, 2011 12:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
It could be your OS limit try ulimit command.
--
Sent from my iPhone
On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com
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
-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
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 bilmar...@yahoo.com wrote:
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it UP, I have to
Problem solved. Just changed G1 to g1
--
Sent from my iPhone
On Jun 13, 2011, at 9:36 PM, James zhu zhulizh...@live.com 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,
span 2 (/etc/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 satish...@hotmail.com
wrote:
Awesome!!
Do you know if i want to use only specific channel for call out then
how do i write dialplan
Sure, but how to check which CA my iPhone using ?
--
Sent from my iPhone
On Jun 8, 2011, at 6:00 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel satish...@hotmail.com
wrote:
It not working on iPhone. It's saying not able to make secure
connection
@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
Interesting thing is when i reload sip.conf i got MWI lamp working on
polycom 501
But its not working when anyone leave voicemail. Do you know its some
timeout
From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE
To: sip:7...@laverne.east.ora.com;tag=as65ea68d2
CSeq: 6 SUBSCRIBE
Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143
Contact: sip:7623@172.30.245.143
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY
=z9hG4bK2b7c62c3FA125372
From: Satish Patel sip:7...@laverne.east.ora.com;tag=9FBFC6B1-EE9095EE
To: sip:7...@laverne.east.ora.com;tag=as65ea68d2
CSeq: 6 SUBSCRIBE
Call-ID: 739c15bd-75f452ef-dcd95504@172.30.245.143
Contact: sip:7623@172.30.245.143
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE
Hi,
Anybody know how to set polycom 501 subscription expiry ?
-S
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
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 supp...@drdos.info wrote:
satish patel wrote:
We are getting hangup cause 18
- 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 working ?
--
Sent from my iPhone
On Jun 8, 2011, at 5:44 AM, Doug Lytle supp
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
--
...@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 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
MWI
All major changes are listed in the UPGRADE.txt files included in the 1.8
tarball.
-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 9:57 AM
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: [asterisk-users] Asterisk 1.8 broken MWI
Starting on line
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
Sent: Wednesday, June 08, 2011 10:44 AM
To: asterisk-users
Subject: Re
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: Re: [asterisk-users] Asterisk 1.8 broken MWI
I do have that
sip.conf
[7623](cam
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
I assume you misspelled default in your e-mail and not voicemail.conf
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: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
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 through
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
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
--
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:
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
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
We have 2 PRI from ATT
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,
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
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
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 Behalf Of satish patel
Sent: Monday, June 06, 2011 8:20
PM
-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 Behalf Of satish patel
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: asterisk-users
Subject: [asterisk-users] PRI
issue its BUSY
Hi
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
--
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 asterisk-us...@notanet.net
wrote:
On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote:
Hey guys!
I have just download latest SVN Revision 322051
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 satish4aster...@gmail.com
wrote:
If 1.8 doesn't panic for subset of PBX features for someone, you can
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
alertInfo voIpProt.SIP.alertInfo.1.value=custome-ring
: 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 OPTIMIZED option do you
think
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
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
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 h...@a-domani.nl wrote:
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote:
On Mon, 30 May 2011, Sherwood McGowan wrote:
True,
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
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
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 salah.elharit...@gmail.com
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 p...@provu.co.uk 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
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
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 with multiple DID numbers and its working
fine in receiving call. And if you make outbound call it will
send main-line CallerID
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= span num ,timing source,line build out
(LBO),framing,coding[,yellow]
Just wondering why it has timing source 0 ? 0=master,
: [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?
From the example:
*CLI database put queue_agent 0001/available_queues
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 multiple queue
On 26/05/11 23:18, Satish Patel
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?
From the example:
*CLI database put queue_agent 0001/available_queues support^sales
support^sales
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 this book example
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 satish...@hotmail.com
Hi There,
We have very old asterisk 1.2 running
, 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 ?
Not really. Looking in system.conf.sample in dahdi-tools [1]
Choose 1 to make
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 sruff...@digium.com wrote:
On Fri, May 27, 2011 at 09:20:46PM +, satish patel wrote:
Tell me
.
--
Sent from my iPhone
On May 27, 2011, at 5:41 PM, Edwin Lam edwin@officegeneral.com
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
Thanks also let me clear one thing this pri is PSTN connected to ATT
techo.
So they are master.
--
Sent from my iPhone
On May 27, 2011, at 5:51 PM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, May 27, 2011 at 05:40:30PM -0400, Satish Patel wrote:
Got it but still confused. As per your
:
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 satish...@hotmail.com
Hi There,
We have very old asterisk 1.2 running
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, Satish Patel satish...@hotmail.com wrote:
Thanks for reply but is there any
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
leif.mad...@asteriskdocs.org wrote:
On 26/05/11 04:20 PM, satish patel wrote:
Actually right now i have
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
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 sherwood.mcgo...@gmail.com
wrote:
On 5/25/2011 12:32 PM, satish patel wrote:
Hey Guys!
We had migrate
'The agents.conf File' section from given link for
more information.
[SATISH]
On Fri, May 20, 2011 at 2:40 AM, satish patel satish...@hotmail.com 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
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
--
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:0, SL:0.0% within 0s
Members
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
, 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 this before. Any
-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 satish...@hotmail.com wrote:
We have polycom 501 and i am waiting since last 5 min no registration require
appear.
-S
With Polycom
registered SIP
peers
-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
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
context
Sometime reboot does help.
--
Sent from my iPhone
On May 19, 2011, at 8:09 AM, vip killa vipki...@gmail.com 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
I am reading at http://www.asteriskguru.com/tutorials/queues.html
They are using member in both static and dynamic method.
member = technology/
--
_
-- Bandwidth and Colocation
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
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,
3:48:05 PM
Subject: Re: [asterisk-users] dahdi command not available
Run Service dahdi start
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 16 May 2011 18:41:01
To: asterisk-usersasterisk-users
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
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=callerid_name for example accountcode=Katie Wilson in
entire file. we
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 satish...@hotmail.com
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 11-05-13
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
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
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 tab tab
No such command 'dahdi' (type 'core show help dahdi' for other possible
commands)
campbx2*CLI
root@campbx1:/etc/wanpipe#
Check this out
http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/
--
Sent from my iPhone
On May 15, 2011, at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Sun, May 15, 2011 at 08:24:08AM +0200, Leandro Dardini wrote:
2011/5/15 RSCL Mumbai
res_timing_timerfd, then dahdi timing is
selected and system become stable.
Regards,
tbskyd
2011/5/14 satish patel satish...@hotmail.com:
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
hardware sip phone and found that prematuremedia=no is
still necessary.
Regards,
tbskyd
2011/5/11 satish patel satish...@hotmail.com:
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
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 satish...@hotmail.com:
Glad you solved it. Now I'm having high CPU load issue. I don't know why but
sometime my asterisk process reached ~150% CPU
not use dahdi...
13.5.2011 v 17:16, satish patel satish...@hotmail.com:
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
Hey Guys!
I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and
install with 1.8 ?
-S
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Holly Cow! Its there already sorry i thought it will only comes with 1.10. We
are using meetme since last 5 year do you think confbridge is better then
meetme ? just need your suggestion
/usr/lib/asterisk/modules/app_confbridge.so
From: satish...@hotmail.com
To:
Subject: Re: [asterisk-users] ConfBridge for 1.8 ?
On 05/12/2011 09:37 AM, satish patel wrote:
Holly Cow! Its there already sorry i thought it will only comes with
1.10. We are using meetme since last 5 year do you think confbridge is
better then meetme ? just need your suggestion
/usr
Check out
http://kb.smartvox.co.uk/index.php/asterisk/sip-extensions/shared-voicemail-part2/
Date: Thu, 12 May 2011 14:38:46 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
Eric Wieling wrote:
pbx*CLI
How to reload only agents.conf ?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Guys!
I am running 1.8 on production we have one PRI and 50 extensions. since last
few days its working fine but today some how server load get high 194 % CPU and
when i did asterisk -r i got CLI but no out put for any command. I check logs
and nothing interesting there.. I am not using any
...@gmail.com:
hi:
thanks a lot for your quick reply. I saw that patch and think that
it was already included in 1.8.3.
now I know it will be included in 1.8.5.
I will try it and thanks again for your kindly help!!
2011/5/10 Satish Patel satish...@hotmail.com:
Apply this patch https
it and thanks again for your kindly help!!
2011/5/10 Satish Patel satish...@hotmail.com:
Apply this patch https://issues.asterisk.org/view.php?id=18868
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On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote:
hi:
our current connection is below:
sip
We have IAX2 peer between two asterisk and I am getting following error
following IAX2 WARNING. IAX calling is functional
[May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6,
subclass = 11,
campbx1*CLI iax2 show netstats
LOCAL -
REMOTE
Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost
% Drop OOO Kpkts FirstMsgLastMsg
IAX2/orasebcam-612 83 -10
Thanks to all for reply,
I have already put 1.8 in production. Actually we are using basic
function so I hope we are good and fingurs cross.
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On May 9, 2011, at 7:18 AM, Alec Davis siva...@paradise.net.nz wrote:
Are you not seeing issues with *8 call pick up then ?
Which release are you running as this is still open
https://issues.asterisk.org/view.php?id=18654
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Thanks, Phil
I am using current SVN branch 1.8 and We aren't using above call pickup
features.
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Hey guys!
I have issue between iax vs iax2 following is my setup
asterisk-1.2 --IAXAsterisk-1.8
I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8
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