Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has
Anyone? Please advice. Thank you.
On Sun, May 8, 2011 at 8:59 AM, GNUbie gnu...@gmail.com wrote:
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be
On Thu, May 19, 2011 at 3:19 AM, GNUbie gnu...@gmail.com wrote:
Anyone? Please advice. Thank you.
That's WAYY too much info for me to go through right now, and I don't know
anything about TLS registration but what I would ask for is if you have the
following lines in your sip.conf
On Wed, May 18, 2011 at 9:39 PM, A E [Gmail] all.efor...@gmail.com wrote:
On Wed, May 18, 2011 at 9:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-05-18 08:01 PM, A E [Gmail] wrote:
boxb*CLI dialplan show Test
[ Context 'Test' created by 'pbx_config' ]
'' = 1.
Hi Guys
Using call files might be easiest. But I d also try out AGI scripting too. I ll
be sure to call back if I require any help.
For the sms bit,...let's say I want to send bulk sms to multiple mobile
devices.
Thanks a lot
Regards
Sent from my BlackBerry® smartphone from Vodafone
hello:
i think you can use php and get message from GUI and send by php AGI.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
To: asterisk-users@lists.digium.com
From: gadgetron...@gmail.com
Date: Thu, 19
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 abalas...@evaristesys.comwrote:
Are you sure it's Asterisk creating the zombie processes, not the
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
we are in a production environment and cannot reboot. besides, these zombie
processes appear minutes after asterisk starts taking calls.
On Thu, May 19, 2011 at 8:59 AM, Satish Patel satish...@hotmail.com wrote:
Sometime reboot does help.
--
Sent from my iPhone
On May 19, 2011, at 8:09 AM,
Actually not sure if it is asterisk generating these zombies... i'm starting
to believe it's the enswitch_routed daemon, anybody familiar with enswitch?
On Thu, May 19, 2011 at 9:02 AM, vip killa vipki...@gmail.com wrote:
we are in a production environment and cannot reboot. besides, these
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale
operation, so I configured A2Billing for not to answer the call nor play any
greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
Hi,
I am trying to use ConfBridge application, but it throws Failed to
find a bridge technology to satisfy capabilities 0x4 (ulaw) error.
Please see console output below.
-- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005,
1001) in new stack
[May 19 13:36:05] DEBUG[7452]:
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has
On 11-05-19 09:39 AM, Chris Maciejewski wrote:
Hi,
I am trying to use ConfBridge application, but it throws Failed to
find a bridge technology to satisfy capabilities 0x4 (ulaw) error.
Please see console output below.
-- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005,
1001) in
Hi Alex,
dunno, it changes all moh (moh and queues music) on the channel, haven't
tried with other people already in the queue I was told to stop
testing when I found out I cannot achieve my goal. :|
Giorgio Incantalupo
On 05/18/2011 05:41 PM, Alex Balashov wrote:
On 05/18/2011 11:34
Hi
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Regards
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX in
1.8.4, going back to 1.8.3.3 everything works. I did open
https://issues.asterisk.org/view.php?id=19264 and included a SIP trace.
Sorry all, I did not follow up adequately. Definitely a problem with 1.6.2.18
What version of Asterisk are you using? ConfBridge was rewritten in
trunk and would be good to see if you have the same issue.
Hi Paul,
I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661)
and it still doesn't work, this time throwing error as below:
-- Executing
On 05/19/2011 12:05 PM, Ishfaq Malik wrote:
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Hi Ishfaq,
I think that you might use a proxy, which connection is always active
(see Astman Proxy), and send
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
On Thu, 2011-05-19 at 12:15 -0400, Jose P. Espinal wrote:
On 05/19/2011 12:05 PM, Ishfaq Malik wrote:
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Hi Ishfaq,
I think that you might use a proxy,
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
Hello,
i have asterisk 1.4 installed and i want to use click to call in order to do
an outbound call
if there is any php code in order to do this operation
thanks and regards
--
_
-- Bandwidth and Colocation Provided by
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 frame
Processor: Intel Dual Core Xeon 3.0GHz
- Host: CentOS 5.6 (64
You only need to tell your PHP script to write a .call file on
/var/spool/asterisk/outgoing/ directory using the syntax described here:
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
I'm not a PHP programmer, so the PHP part is up to you hehe.
There are other methods like using
ok thank you i will test this solution and i will update you :)
2011/5/19 Alejandro Mejia Evertsz ame...@gua.net
You only need to tell your PHP script to write a .call file on
/var/spool/asterisk/outgoing/ directory using the syntax described here:
But why does *our *native format keep changing :)
Going by layman terms, if native format is alaw and someone speaks to me in
uLaw, I will say *format changed*.
But if native format is alaw and someone is talking with me in alaw, I
should be happy.
On Thu, May 19, 2011 at 10:28 PM, Terry
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
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Thursday, May 19, 2011 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropping
I updated my phones to the UCS 3.3.1 firmware a few months back. The
scenario is I place a call and receive an incoming call. With 3.3.1
the screen will show call 1/2 and I have to press the down arrow to
see the caller name / number. Has anybody else noticed this with
3.3.1? I had thought with
Hi
I change the chan_dahdi.conf and restart dahdi:
prilocaldialplan=private
pridialplan=private
But, in debug i see the following informations:
1 Calling Number (len= 8) [ Ext: 0 TON: *National Number (2) * NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1
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,
On 11-05-19 12:13 PM, Chris Maciejewski wrote:
What version of Asterisk are you using? ConfBridge was rewritten in
trunk and would be good to see if you have the same issue.
Hi Paul,
I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661)
and it still doesn't work, this time
perhaps you forgot to run make config _after_ installing dahdi drivers
---
Marcelo Ellmann
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016
- Original Message -
From: isr...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
also, make sure that when you installed asterisk, the option to load the dahdi
module was select.
when you run a ./configure it scans your system and when you run make
menuselect, the resource module dahdi will be marked to be compiled and
installed :)
---
Marcelo Ellmann
Freeddom
Thanks for reply Marcelo,
I don't know what was the problem but after reboot machine it works! I am
pretty sure i did service dahdi start/stop but that didn't work.
-S
Date: Thu, 19 May 2011 16:44:18 -0300
From: ellm...@freeddom.com
To: isr...@gmail.com; asterisk-users@lists.digium.com
I'm glad you got it right! :)
cheers,
---
Marcelo Ellmann
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016
- Original Message -
From: satish patel satish...@hotmail.com
To: asterisk-users asterisk-users@lists.digium.com
Sent: Thursday, 19 May, 2011 5:13:03 PM
Subject:
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
Ok, i tried the suggestion:
Instead of:
sippuser = resource, database_name, table_name
sippeer = resource, database_name, table_name
I put in:
sippuser = resource, context, table_name
sippeer = resource, context, table_name
Unfortunately, with the same results.
btw i tried both general as
I had issue with call files. They would lock up the system (this was 5 years
ago so maybe things have changed.)
- Original Message -
From: Alejandro Mejia Evertsz
To: asterisk-users@lists.digium.com
Sent: Thursday, May 19, 2011 19:58
Subject: Re: [asterisk-users] click
If you go for 1.8,Don't read from
http://www.asteriskguru.com/tutorials/queues.html. It is bit backdated
information. Rather I would suggest you to check
http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html.
Queue members are considered INVALID, if their device status is Invalid.
This
If you don't like callfiles, another option is AMI. Check the sample code
from
http://tycoontalk.freelancer.com/php-forum/156207-click-to-call-using-php.html,
do some changes as per your requirements.
I would love to use callfiles as it gives more flexibility(as per my
understanding) compared to
Hi,
I've defined a feature using a macro in features.conf :
special = #2,peer,Macro,special
Everything is working if the user use the phone key.
But i would like to call the feature (or the Macro on the peer
channel) from AMI or CLI. First i thought i would be simple, but i did
not find
On 19/05/11 16:04, vip killa wrote:
Actually not sure if it is asterisk generating these zombies... i'm
starting to believe it's the enswitch_routed daemon, anybody familiar
with enswitch?
Hello,
I am the lead developer of Enswitch. Enswitch comes with commercial
support as standard, so if
Yeap, I couldn't set Private TON too. Try to set all _prefix variables
in chan_dahdi.conf and use dynamic prilocaldialplan.
On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote:
Hi
I change the chan_dahdi.conf and restart dahdi:
prilocaldialplan=private
pridialplan=private
But, in debug i
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