Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten = _X., n, Dial(SIP/1001,30)
exten = _X., n,
Hi All
We're getting to the point where we need to start increasing capacity on
our asterisk servers.
I've had a look at the DUNDi J Richardson white paper and it seems
pretty straight forward.
My question is have any of you implemented this solution in a production
environment?
Regards
Ish
Ircd is not installed and cant be located in all system ,any one know or
have an idea how do they infect my system,
Any bug in asterisknow?
How to find the script that initiates this invites ?
135.307281 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=36
Ack=111 Win=5840 Len=0
135.307434
Hi All,
I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend.
My dialplan:
exten = _,1,Dial(SIP/${EXTEN},60,rt)
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
d.nik...@cem-solutions.net wrote:
Does anyone ported Asterisk to Android OS .please give details
www.servalproject.org
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Hello All,
i have asterisk installed in my call centre without any issue I would like
to ask you some questions related to services.
i want to start asterisk and httpd and aheevacti automatically when the
server centos reboot or shutdown
becouse i must start all services manually
netstat -anp |grep 6667
Best Regards,
Muhammad Nuzaihan Kamal
Network Consultant
Mobile: +65 97473874
Asfa Systems Pte Ltd
91, Alps Avenue. #03-10. Singapore 498787
Tel: +65 62538211
Fax: +65 62504814
www.asfasystems.com.sg
pub 4096R/36630777 2010-07-10
Key fingerprint = 670A 4D60
So simple - great, thank!!!
Am 17.12.2010 13:07, schrieb Vincius Fontes:
You probably want "core show channels
verbose".
Atenciosamente,
Vincius Fontes
Gerente de Segurana da
So simple - great, thank!!!
Am 17.12.2010 13:07, schrieb Vincius Fontes:
You probably want "core show channels
verbose".
Atenciosamente,
Vincius Fontes
Gerente de Segurana da
Will someone help/direct me find a way to implement this?
Or you can suggest some other method.
On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.comwrote:
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible or
not.
i believe there is a way to do it using asterisk and flashphoner
++
2010/12/20 Gilles codecompl...@free.fr
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
d.nik...@cem-solutions.net wrote:
Does anyone ported Asterisk to Android OS .please give details
www.servalproject.org
--
When installing asterisk you should type make config to have
asterisk create init script automatically.
For http chkconfig httpd on
Adolphe Cher-aime
From my Iphone
On Dec 20, 2010, at 5:48 AM, salaheddine elharit salah.elharit...@gmail.com
wrote:
Hello All,
i have asterisk
salaheddine elharit wrote:
becouse i must start all services manually (service asterisk start
,service httpd start
chkconfig httpd on
chkconfig asterisk on
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of one way audio on this outbound routes only,
My voice is going to outside pstn number but
On 10-12-19 06:23 AM, Daniel Knoll wrote:
In which Version of Asterisk is EventFilter: in manager.conf working?
Higher than 1.6.2.10 or from the 1.8.0 Version?
Always refer to CHANGES[1] or UPGRADE.txt. It was added in 1.8
[1] http://svn.digium.com/svn/asterisk/branches/1.8/CHANGES
--
Without reading too much into your description, I can tell you that being an
inband sound, and as long as the dtmf tone is heard by everybody during the
conference, and being the ivr gateway one of the parties of the conference, I
don't see a reason why the ivr gateway wouldn't act upon hearing
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of one way audio on this outbound routes only,
My
ok thank you so much for your help
2010/12/20 Doug Lytle supp...@drdos.info
salaheddine elharit wrote:
becouse i must start all services manually (service asterisk start
,service httpd start
chkconfig httpd on
chkconfig asterisk on
Doug
--
Ben Franklin quote:
Those who would
Hi Jonathan,
I already looked at their product a few weeks ago, but because Alcatel
wasn't on their list of compatible devices, I left it alone.
Because of your email, I went looking on their site for a second time
and noticed on their blog that they're experimenting with Alcatel
devices.
So
when i make chkconfig httpd on and chkconfig asterisk on
with chkconfig --list i found
httpd 0:off 1:off 2:on3:on4:on5:on6:off
asterisk0:off 1:off 2:on3:on4:on5:on6:off
the 0,1, and 6 are OFF just the 2,3,4,5 are ON ,
and when i reboot
I'm not certain what you mean by needing to setup up a SCADA solution? I
assume you want to connect an industrial data acquisition and control
system to Asterisk. We have a SCADA system interfaced with Asterisk in our
facility. The SCADA hardware we use is the SNAP PAC system from
No problem. We've had good luck with them so far. Support is also VERY
responsive (had a work around in a few hours, and a firmware upgrade to fix the
issue within a day or two).
-Jon
- Original Message -
From: Sander Naudts s.nau...@intersui.be
To: Asterisk Users Mailing List -
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it is working for calls too,
But there is issue of
Am 20.12.2010 16:00, schrieb Thorsten Göllner:
Am 20.12.2010 15:11, schrieb Shaun Ruffell:
On 12/20/10 6:50 AM, Max Alex wrote:
Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card: Digium, Inc. Wildcard AEX800 8-port analog card
I have configured this card properly and it
On Mon, Dec 20, 2010 at 02:34:23PM +, salaheddine elharit wrote:
the 0,1, and 6 are OFF just the 2,3,4,5 are ON ,
and when i reboot the server i found that the service httpd is off with
command service httpd status and service asterisk status
please advice
This is just one of many problems
On Fri, 17 Dec 2010 17:54:00 +, Roger Burton West
ro...@firedrake.org wrote:
How would you _expect_ to be able to specify a destination server from a
telephone keypad?
Thanks guys for the infos. My goal was to learn how to configure
Asterisk so it could call SIP URI (u...@domain) using XLite,
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote:
Now, when I Dial extension 1050, and there is no 1050 peer registered I got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
You haven't done anything wrong; I have the
On 10-12-20 04:41 AM, Jarek Jarzebowski wrote:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
It looks to be a regression with the IPv6 code added to chan_sip. Which
version of 1.8 are you using? I'd also be good to
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating files of
the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension: $OUTSIDE_NUMBER
Priority: 1
CallerId: $INSIDE_NUMBER
in
Anyone going to remove this spammer/scammer?
2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com:
http://www.barenakedbabies.com/shop/images/images.html
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-- Bandwidth and Colocation Provided by
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)})
Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO).
I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.
In both cases, asterisk rejects the call with a 420.
In this case, it’s a call
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating files
of
the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension:
On Monday 20 December 2010 10:33:33 A J Stiles wrote:
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating
files of the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension:
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2.
On 12/17/2010 06:25 AM, Gilles wrote:
On Thu, 16 Dec 2010 11:54:31 +0100, Gillescodecompl...@free.fr
wrote:
Now, I'd like to be able to call any number on the Net that is
advertised as sip:u...@domain.com, such as those:
I mean: Do I really have to first create a section in sip.conf each
time
Thanks for this info. It seems like good hardware and software solution
provider. I'll explore it a bit more and see if it fits my client's need.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com
On 2010-12-20 9:56 AM, marvin horst fivehor...@gmail.com wrote:
I'm not certain what you
On Mon, Dec 20, 2010 at 9:46 AM, Dovey Forman dovey.for...@idt.net wrote:
I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.
In both cases, asterisk rejects the call with a 420.
In this case,
On 12/20/2010 11:46 AM, Dovey Forman wrote:
Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO).
I am trying to initiate a call FROM a softphone client to asterisk
(either an internal 4 digit extension call) or an outside line via a SIP
trunk.
In both cases, asterisk rejects the call
On Monday 20 December 2010 11:35:21 Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
Any device that you can talk to and be used in Linux can be interfaced into
asterisk with the power of AGI.
I have some WebRelay modules that I can remotely control via asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID space (happens within hours) and brings the
On Mon, Dec 20, 2010 at 12:27:46PM -0600, Jason Parker wrote:
'_*[0-9a-zA-Z].*0.'
'.' stops further matching. Your extension ends up being (effectively)
shortened to _*[0-9a-zA-Z].
That explains a lot, never read it this way before. Thanks for the eye
opener.
What I'm looking for is a
Hello,
I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools
(Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward.
Everything is working.
Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a
backup of configuration files, codec
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to mysql. I did not change any configs. I checked to make sure that
the
Are you making progress?
Dear Friend,
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Come and have a look:www.topeleczone.com
We sell the famous brand-new and original electronic at the wholesale price on line and we are having
2010/12/20 Jeremy Kister asterisk...@jeremykister.com:
On 12/20/2010 4:41 AM, Jarek Jarzebowski wrote:
Now, when I Dial extension 1050, and there is no 1050 peer registered I
got:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1:
2010/12/20 Paul Belanger pabelan...@digium.com:
On 10-12-20 04:41 AM, Jarek Jarzebowski wrote:
[Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len
843) to 0.0.4.26:5060 returned -1: Invalid argument
It looks to be a regression with the IPv6 code added to chan_sip. Which
On 10-12-20 05:51 PM, Jarek Jarzebowski wrote:
OK, so I have attached debug log.
I am using:
*CLI core show version
Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on
2010-12-17 23:03:58 UTC
Definitely a bug, ran into the same issue with chan_iax2 and DNS
lookups.
Thanks Kevin.
Did it work with Asterisk 1.2 because it ignored it?
Why now?
On Dec 20, 2010 3:28 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 12/20/2010 11:46 AM, Dovey Forman wrote:
Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO).
I am trying to initiate a call FROM a
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