anyone?
From: hin lee hi...@yahoo.com
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Fri, March 12, 2010 10:08:53 AM
Subject: Polycom not updating the directory list
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be changed
The very obvious thing to check is the permission of the
mac-addr-directory.cfg.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Thursday, 18 March 2010 4:56 PM
To: Asterisk Users
hi,all
one problem confuse me these days. i want to sequence dial three PSTN
number(a,b,c)
first, if i dial number a, if a is busy , i will dial number b. if b
is busy, i will dial number c.
Dial(SIP/a...@ip,30)
Dial(SIP/b...@ip,30)
Dial(SIP/c...@ip,30)
i want to know before i dial number a,
Am 18.03.2010 05:11, schrieb Olivier:
2010/3/17 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Am 17.03.2010 10:40, schrieb Peter den Hartog:
Hello,
I was wondering if the following was possible:
When somebody sends a fax
Am 17.03.2010 19:31, schrieb Matt Watson:
On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion
klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote:
Attached is an untested (I did not had the time yet) port to
Asterisk 1.4.29.1 (DAHDI). Maybe the modules need some
Hello,
Please have a look to DIALSTATUS variable. here
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUSI hope it
helps
On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun bit...@gmail.com wrote:
hi,all
one problem
Hello all,
I would like to know if any one have experience with live audio streaming
like
1. Streaming from an online resource
2. Streaming from sound card AUX interface..
What i want to accomplish is that on receiving a callers call i play back a
live audio stream or stream from sound card AUX
Thanks. However, I discovered a guide on doing this at the following url:-
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Example 2 shows to use a macro to present a menu to the member of staff before
the call is bridged.
Many thanks
Dan
-Original Message-
From:
Yes it does.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Wednesday, March 17, 2010 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Door
Hi,
I'm trying to set up remote voicemail pickup. I've created the following
dialplan, but when I press *, I am not sent to voicemailmain. The unavailable
message just continues to play as normal.
exten = 234555,1,Set(MAILBOXID=1)
exten = 234555,n,Set(MAILBOXCONTEXT=company3)
exten =
On 3/17/2010 6:25 PM, Jeff Brower wrote:
Steve-
On Wed, Mar 17, 2010 at 6:02 PM, Jeff Brower
jbro...@signalogic.commailto:jbro...@signalogic.com wrote:
Steve-
2010/3/17 Vinícius Fontes
vinic...@canall.com.brmailto:vinic...@canall.com.br
- Kevin Sandy
Em 17-03-2010 20:51, Vinícius Fontes escreveu:
- Joao Gomes Pereiragomespere...@startel.pt escreveu:
Hello
Im trying to receive FAXes with my Asterisk with rxfax command.
To do that, Im trying to load the app_fax.so module but asterisk
says:
[Mar 17 20:06:04] WARNING[11907]:
This is a longshot, but the FXS indication tells me you're using DAHDI. Put
an answer at the start of the custom context and see if that solves your
problem.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon
This is just my approach, but I would run call 1 into an AGI that produced
the second call through AMI, then proceeded based on the return.
- exten = 123,1,answer
- exten = 123,2,AGI(callproc.agi)
- exten = 123,3,Gotoif($[${PROC} = VM]?voicemail)
- exten =
Do a link to the image as URL on the dial command?
- exten =
s,1,Dial(SIP/12345,20,KkTT,http://www.yahoo.com/image.jpg)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta
Sent: Wednesday, March 17,
Em 17-03-2010 20:28, Doug Lytle escreveu:
Joao Gomes Pereira wrote:
What could be missing?
Running ldconfig as root
Thanks, thats it!!!
Now the module is loaded.
I just hope the FAX code works:
[macro-faxreceive]
exten =
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323.
Trying to find a way I could use my laptop to send a fax over H323 to the
BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface
on a Cisco router and setup a h323 dial peer to the BrookTrout, but I
- Joao Gomes Pereira gomespere...@startel.pt escreveu:
Em 17-03-2010 20:51, Vinícius Fontes escreveu:
- Joao Gomes Pereiragomespere...@startel.pt escreveu:
Hello
Im trying to receive FAXes with my Asterisk with rxfax command.
To do that, Im trying to load the
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still having doubts about the problem :(
Thanks in advance
Message: 10
Date: Thu, 18 Mar 2010 00:21:06 -0400
From: Zeeshan Zakaria zisha...@gmail.com
Dan Journo wrote:
Hi,
Any ideas?
I'd be helpful to see the console output.
Doug
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Its ok, I discovered the issue.
The DTMP signals weren't being received.
All sorted now.
Thanks
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 18 March 2010 14:10
To: Asterisk Users
Hello,
This Friday on VUC, the SIP Router Project, Kamailio 3.0 will be
discussed with a couple experts. Your questions are welcome, as
always.
See the site: http://vuc.me for ways to phone in. For the best sound,
use g722 and call 200...@login.zipdx.com at 12 Noon Eastern.
See
Do you properly hang up the calls. Does 'zap show channel channel number'
shows that the channel is 'on hook' after its hang up?
On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote:
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told
Was there any hardware upgrade in December after which you recompiled
libpri?
On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote:
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still
Hi Jeff!
Looks like the term native bridging is a bit overloaded.
Some text from channel.h:
-# When the call is answered, Asterisk bridges the media streams
so the caller on the first channel can speak with the callee
on the second, outbound channel
-#
Klaus-
Looks like the term native bridging is a bit overloaded.
Some text from channel.h:
-# When the call is answered, Asterisk bridges the media streams
so the caller on the first channel can speak with the callee
on the second, outbound channel
Hello,
here my achitecture:
client1--Asterisk1ser1---centile
client2--
client1 do a call to centile.
centile do a forward to client2 (Diversion) and then use the same CALL-ID!
when asterisk1 receive the call with the same CALL-ID, it screen Now
forwarding SIP/ -02f6 to
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
I'll see if
On Thursday 18 March 2010 11:24:18 am Karl Fife wrote:
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re:
I'm not having problems with hanging up the calls, my problems is that i
asterisk dies, i'm using a pri and always zap show channel X will always
show Hookstate (FXS only): Onhook beacuse it only applies to FXS and i'm
using digital e1 trunk
Or am i wrong?
Message: 1
Date: Thu, 18 Mar 2010
E1 channels are also zap channels. Zap show channels doesn't differentiate
between them.
On 2010-03-18 2:05 PM, Danny Dias ing.diasda...@gmail.com wrote:
I'm not having problems with hanging up the calls, my problems is that i
asterisk dies, i'm using a pri and always zap show channel X will
Somebody has 5.1.7 firmware for SPA3102?
I'm having issues with inbound/outbound calls using asterisk through SPA3102
with firmware 5.1.10. I've read it has a codec bug, since it doesn't care
about what you set up in Preferred Codec.
Any help will be appreciated.
Sebastian
--
On 03/18/10 16:22, Sebastian Milioto wrote:
Somebody has 5.1.7 firmware for SPA3102?
I'm having issues with inbound/outbound calls using asterisk through SPA3102
with firmware 5.1.10. I've read it has a codec bug, since it doesn't care
about what you set up in Preferred Codec.
Any help will be
Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
http://www.venturevoip.com/news.php?rssid=2371
--
Cheers,
Matt Riddell
Managing Director
___
I would like to know if any one have experience with live audio
streaming like 1. Streaming from an online resource
Look at app_ices and icecast.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices
Philipp
--
_
On 15/03/10 11:23 AM, Thomas Perron wrote:
I want callers to enter a queue and then hear music on hold.
does anyone have notes on how to integrate queuing to a dial plan that uses
moh?
You can just set the music on hold class for the Queue in queues.conf -
you actually have to provide an
Hi David!
Thanks very much for helping me out will all !
Ok i try your tip and @ the moment i still have the same problem but now i have
the kernel and the kernel devel the same but wend i try to run make i still get
the same erro, do you guys have any idea how to fix it?
-bash-3.2# rpm -qa
Thanks Matt. This should be useful. I'll give it a try on my Motorola
Droid/Milestone.
On 2010-03-18 5:31 PM, Matt Riddell li...@venturevoip.com wrote:
Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
On 2010-03-18 5:31 PM, Matt Riddell li...@venturevoip.com wrote:
Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
http://www.venturevoip.com/news.php?rssid=2371
--
Cheers,
Matt Riddell
Managing
On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote:
Hi David!
Thanks very much for helping me out will all !
Ok i try your tip and @ the moment i still have the same problem but now i
have the kernel and the kernel devel the same but wend i try to run make i
On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote:
On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu
dlab...@gmail.comwrote:
Hi David!
Thanks very much for helping me out will all !
Ok i try your tip and @ the moment i still have the same problem but now i
have
Hello,
I'm looking for some advice on securing Asterisk.
Recently my servers been under several brute-force SIP attacks.
I have several remote sites, as well as many roaming users, who may have
PC softclients and/or SIP based hardphones.
My first step will be to strengthen the
On 19/03/10 1:19 PM, Adrian Marsh wrote:
Hello,
I’m looking for some advice on securing Asterisk.
Have a look at fail2ban:
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
--
Cheers,
Matt Riddell
Managing Director
___
On Fri, 19 Mar 2010, Adrian Marsh wrote:
I’m looking for some advice on securing Asterisk.
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate
Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put your
own ip address in ignore list otherwise it can block you too.
On
Hi List,
How can I define an array of sip number in sip.conf ?
I want to define an array of sip number from 1000 to 2000, so I can make a
performance test on Asterisk using sipp.
Thanks in Advance,
Giangnh
--
_
--
You'll have to type them all in manually. Or do what I did several times,
write a script in php which will generate the sip.conf with that many
extensions. Even better look into using realtime architecture, where you can
quickly generate as many extensions as you like.
On 2010-03-18 10:09 PM, huu
Hey hey!
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate between a hard phone and a soft phone.
Although: One could think about enhancing
Philipp, remembering sip user agent is a wondeful idea, and if you goggle
it, somebody had made a patch for it, so that one could identify sip devices
by their sip user agent names. Surprisingly the decision makers didn't like
to put it in the production branch of asterisk at that time, however it
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
ConfBridge seems to be a good fit but I can't get it going. The document
sounds like an easy to use app. Am I missing any bridge_ modules?
Asterisk 1.6.2.0~rc2-0ubuntu1.2
-- Executing [...@outbound:1]
Thanks! but if i use Queue to call out not Dial.
how should i know the status like busy or free?
for now . i know asterisk have QUEUESTATUS variable，
QUEUESTATUS The status of the call as a text string, one of TIMEOUT
| FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL
but the
What does the source code tell you about the circumstances in which
that particular error string is produced?
--
Sent from mobile device
On Mar 18, 2010, at 11:20 PM, Kelvin Chan kelv...@positronics.com
wrote:
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
Hi!
Did a quick test, worked as a clock:
exten = 0317998959,1,Set(CHANNEL(language)=se)
exten = 0317998959,n,Answer()
exten = 0317998959,n,ConfBridge(1001,s) 0317998959,n,Hangup()
On Thu, 18 Mar 2010 20:20:35 -0700, Kelvin Chan wrote:
Hi guys,
I'm trying to move away from meetme to
Thanks I will look into it.
On Fri, Mar 19, 2010 at 2:26 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
I would like to know if any one have experience with live audio
streaming like 1. Streaming from an online resource
Look at app_ices and icecast.
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