Johan Dindaine - Asterisk schrieb:
Hi every all,
since a few weeks I came back to asterisk and tried to install version 1.6.
The installation went fine so I decided to buy new dids on Voxbone.
I have added the sip peers of Voxbone Belgium1 like this in the sip.conf
[81.201.82.39]
Hello, if you dont know iptables that much, and would like to see more user
friendly configuration method, i suggest you to use Shorewall, which is
very flexible, has some clear logs, and generates same iptable rules behind.
2009/2/8 David fire ddf...@gmail.com
denay permit are in sip.conf and
On Fri, 6 Feb 2009, oumar ndiaye wrote:
Is there a way to restrict connection to my asterisk server to users based
on their IP addresses, and not just password. I have some hackers who
connect to my server to make illegitimate solicitation calls to people. I
had to shutdown the server for now
what about something along the lines of...
iptables -A INPUT -p udp --dport 5060 -j DROP
iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -s 66.66.66.66 --dport 5060 -j ACCEPT
Cheers
On Mon, Feb 09, 2009 at 11:09:34AM +, Geraint Lee wrote:
what about something along the lines of...
iptables -A INPUT -p udp --dport 5060 -j DROP
iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT
iptables -A
Voxbone have many more IPs than that, probably your calls are coming
from another IP.
As these calls are just for your internal calls, just remove this entry
in your sip.conf
Olivier (another Begian)
Johan Dindaine - Asterisk a écrit :
Hi every all,
since a few weeks I came back to asterisk
Hello,
I'm just wondering if anyone has fixed the 'InUseRinging' problem.
* v1.4.23.1
Ta
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Andrew Thomas schrieb:
I'm just wondering if anyone has fixed the 'InUseRinging' problem.
* v1.4.23.1
What's the InUseRinging problem?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
well, you got the general idea :)
2009/2/9 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Feb 09, 2009 at 11:09:34AM +, Geraint Lee wrote:
what about something along the lines of...
iptables -A INPUT -p udp --dport 5060 -j DROP
iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060
On Monday 09 February 2009 04:17:47 Gordon Henderson wrote:
On Fri, 6 Feb 2009, oumar ndiaye wrote:
Is there a way to restrict connection to my asterisk server to users
based on their IP addresses, and not just password. I have some hackers
who connect to my server to make illegitimate
core show hints is showing extens as InUseRinging, which in turn causes
the wrong 'hint' state to be shown and hang-up etc. which means BLF's actually
show the wrong condition(s)
http://bugs.digium.com/view.php?id=13238 gives more information (only seems to
relate to v1.6.x.x though).
I have
On Mon, 9 Feb 2009, Geraint Lee wrote:
what about something along the lines of...
iptables .
Well, whatever, but this isn't an answer to my question and I'm still
curious as to how the hackers are breaking usernames and passwords, as I
have servers which I can't firewall and if there
Michael Grave just posted a question about surround conferences.
http://www.facebook.com/notes.php?id=564633430#/note.php?note_id=5009726
3908id=564633430index=0
I didn't see it posted on the ast-list, what do you think? Does
something like this have potential?
I'd love to listen in
On 9 Feb 2009, at 14:20, Dean Collins wrote:
Michael Grave just posted a question about surround conferences.
http://www.facebook.com/notes.php?id=564633430#/note.php?
note_id=50097263908id=564633430index=0
I didn’t see it posted on the ast-list, what do you think? Does
something like
I've never used reinvite in systems I have installed to date, and I have
finally run across a situation where it would be preferred.
A remote office has a flaky Internet connection. With G729 encoding the
calls to the central office over the 'net are tolerable. One Linksys 2102
drives two
http://tinyurl.com/c4qbcj
is that better for you?
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Monday, 9 February 2009 9:47 AM
To: Asterisk Users Mailing List -
The problem in this particular case is that the actual monitor object is on
A's
channel. When A is no longer involved in the call, the monitor is gone, and
so
the call cannot be recorded further. One possible solution is to run the
Monitor
application on B's channel instead. This
Still doesn't work but i'm guessing it's to do with not being friends with
Michael?
2009/2/9 Dean Collins d...@cognation.net
http://tinyurl.com/c4qbcj
is that better for you?
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls and
this noise influences on the initial negotiation protocol, so modems
have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux
Geraint Lee schrieb:
http://tinyurl.com/c4qbcj
is that better for you?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve Howes
On 9 Feb 2009, at 14:20, Dean Collins wrote:
Michael Grave
I unwittingly started this on Facebook, which I don't user very much.
Here's the gist of it.
A Strange Brew: VoIP/Telephony Crossed With Surround Sound
With apologies to the McKenzie brothers. There appears to be an odd
cross between two of my passions in the works. As I get more into the
daily
On 9 Feb 2009, at 15:35, Dean Collins wrote:
http://tinyurl.com/c4qbcj
is that better for you?
No. He is not a 'friend' on Facebook.
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To
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to allow call takers to login
and before they can take calls. How
I moved this from Facebook to
blog;
http://www.mgraves.org/voip/2009/02/a-strange-brew-voiptelephony-crossed
-with-surround-sound/
or
http://is.gd/iVjd
If you'd rather.
Michael
--Original Message Text---
From: Dean Collins
Date: Mon, 9 Feb 2009 09:20:31 -0500
Anthony Francis wrote:
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to allow call takers to login
and
Anthony Francis wrote:
Anthony Francis wrote:
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to
It looks like a problem that I thought had once been fixed is broken again.
In manager.c line 2166 of both versions 1.6.0.3 and 1.6.0.5
if (!ast_strlen_zero(name)) {
The ! Should not be there.
I am not sure if there are other places there is a like problem but removing
the ! for sure fixes at
Hi people!
Asterisk PBX (version 1.6.5): I have Asterisk behind a NAT (192.168.1.2)
SIP Phone: A client behind NAT (192.168.1.3)
Softphone: One other client somewhere in the internet (also behind an NAT).
they want to speak with each other, and if they do, there is no sound.
if softphone in the
Hi List.
I have a small problem in using the transfer key transfer of IP Phone in
Asterisk 1.6, I think I spend some detail in the configuration but can not
find.
What happens is, when I do a transfer using the Transfer button, the
phone, does not play the music on hold, which is waiting on
How to make the Asterisk-GUI work with DAHDI..please??
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I upgraded my office PBX from 1.4.22 to 1.6.0.5 so we can start
evaluating and testing. I did not really test it over the weekend, just
made sure I could dial in and out. Today we are finding that incoming
calls to our POTS lines get dropped after a couple of minutes. All I
can see in
It seems as if there is a problem if one uses Local/somen...@some_context in
a Dial application. If the extension is changed to somename then things
work.
I have an extension SomeName defined. In another extension I try to dial
this extension and it does not work saying extension/context not
Assuming you're still in the 1.4 set, enable the Zap mirroring. I'm not
sure the fine folks at Digium have accounted for DAHDI in the current
interfaces (but I'm probably wrong).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
This problem only seems to occur when using Aastra phones. Calls to
Polycom never drop. Anyone know of a setting for Aastra that could
cause this?
On Mon, 2009-02-09 at 13:22 -0600, Carlos Chavez wrote:
I upgraded my office PBX from 1.4.22 to 1.6.0.5 so we can start
evaluating
It could be that the phone is trying to (re)register too frequently and
drops during the SIP negotiations.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, February 09, 2009 2:11 PM
I discussed my installation more with Tzafrir last week. He concluded that he
thinks I don't have an HFC card. I think it is somewhat a matter of semantics.
As far as I have been able to determine, there are at least three general
types of HFC cards. By far the most common are cards based
Greetings list-
I'd like the ability to hangup all calls for a particular extension from the
system CLI. I understand this can probably be scripted using the AMI but I'm
not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
A really, really quick question here!
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set anything up
to make it happen?
Thanks so much!
Chris
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A really, really quick question here!
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set anything up
to make it happen?
Thanks so much!
Chris
Sorry, I forgot to add when using the SIP protocol
Hi,
I would like to improve my understanding of T.38.
1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ?
voip-info.org implies one has to change values in chan_sip.c to make it
work.
Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in
global_t38_capability ?
Source code
Hi
I registered my asterisk box to my SIP provider as an UA. For every call I
receive on this trunk, I get the message That is not a valid conference
number. I'm using Asterisk version 1.4.22, I had install the dahdi-linux
and dahdi-tools and the conference is working between the phones
Chris Rowson wrote:
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set
anything up to make it happen?
The simple answer is 'yes', the correct answer is 'no' :-)
MD5 is not encryption, it is a digest (hash)
Hi,
My patton 4638 is sending :
*v=0
o=MxSIP 0 46 IN IP4 192.168.100.52
s=SIP Call
c=IN IP4 192.168.100.52
t=0 0
m=audio 4984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=image 4986 udptl t38
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
Asterisk
On Sat, 2009-02-07 at 15:51 -0500, Alexander Lopez wrote:
-Original Message-
From: Steve Murphy [mailto:m...@digium.com]
Sent: Saturday, February 07, 2009 1:59 PM
To: Alexander Lopez
Subject: RE: [asterisk-dev] 1.4 and CDRs -- The Breaking Point
On Fri, 2009-02-06 at 12:28
On Mon, Feb 9, 2009 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Chris Rowson wrote:
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set
anything up to make it happen?
The simple answer is
I'm usin both sets 1.4 and 1.6 in 1.4 is working with zaptel and 1.6 with
DAHDI, but Asterisk-Gui do not recongnize the TDM120 card, but is working
asterik fine, i want to make the asterisk-gui to work with DAHDI
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk
On Mon, 2009-02-09 at 23:25 +0200, Szasz Szabolcs wrote:
Hi
I registered my asterisk box to my SIP provider as an UA. For every
call I receive on this trunk, I get the message That is not a valid
conference number. I'm using Asterisk version 1.4.22, I had install
the dahdi-linux and
On Mon, Feb 9, 2009 at 4:43 PM, Olivier oza-4...@myamail.com wrote:
Hi,
My patton 4638 is sending :
v=0
o=MxSIP 0 46 IN IP4 192.168.100.52
s=SIP Call
c=IN IP4 192.168.100.52
t=0 0
m=audio 4984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Matthew Nicholson wrote:
On Mon, 2009-02-09 at 23:25 +0200, Szasz Szabolcs wrote:
Hi
I registered my asterisk box to my SIP provider as an UA. For every
call I receive on this trunk, I get the message That is not a valid
conference number. I'm using Asterisk version 1.4.22, I had install
Have you looked at soft hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, February 09, 2009 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have an Audiocodes MP-118FXO in production. When an outbound call is made and
the remote party hangs up, the Audiocodes hangs up the call immediately. But if
an incoming call is received and the remote party hangs up, the Audiocodes does
not hang up immediately.
I have tinkered with Current
I used the AMI show the current channels, grep out the ones I wanted to kill,
and then used Command: Hangup to get rid of them. And, all done with about 100
lines of nasty nasty bash. :-)
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Alexander Lopez
Hi all,
I'm looking into being able to send/receive SMS messages with my
asterisk box in the US. I've seen the SMS command as well as the Kannel
program. I'd prefer to do it from Asterisk.
I've tried something like:
exten = 999,n,sms(15551234567,s,This is a test)
in my dialplan, but when
On Jan 27, 2009, at 9:49 AM, Michael Higgins wrote:
Folks --
First, apologies for not lurking for weeks or months to get the
culture of the list. I read the recent post about improvement to the
quality of posts with some amusement and full agreement. The problem
is a big and very
You need a router with DSP modules for it to terminate voice, otherwise
it is just a data BRI and a router that has SIP. 1600's don't have them
and only the 1750-V/1751-V and 1760 support them on the 1700 series.
Pretty much every Cisco router from the 1600 and up supports a BRI
WIC/NM for
You might want to look into Cisco hardware, their WIC-1B-U cards work
fine in the US, or they did 10 years ago when I last used them for
VoIP. Used the WIC-1B-U is going for under $50 on eBay. An old 1600
or 1700 series router with an IOS that supports SIP wouldn't cost much
either.
On Mon, Feb 9, 2009 at 4:23 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I would like to improve my understanding of T.38.
I recommend you try out Asterisk 1.6 if you want to play with T.38.
I DID get asterisk-1.4 working with fax, but I was having a lot of
issues with faxes dropping in weird
You might want to look into Cisco hardware, their WIC-1B-U cards work =20
fine in the US, or they did 10 years ago when I last used them for =20
VoIP.
No, you didn't. :-) You might have used a VIC-2BRI-S/T-TE or one of
the other voice cards... but the WIC cards are WAN Interface Cards.
If
I'm quite happy to share what we've done with anyone who is interested
in our solution.
By the way, I'm also willing to work with any developers who want to do
US BRI work. The Adtran ports can play as either NT or TE, so it is
trivial to hook up a card to a spare port and pretend to be a CO
Can anyone help me determine where the problem lies and how to fix it?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: Mike Hammett
Sent: Thursday, January 15, 2009 1:00 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk - Trixbox
My
Is there anyone on this list with some advice about hosted voip services
offered by wholesale companies?
Good / bad or otherwise.
Wanting to do a bit of research into this market space and looking for
who is running the best (best either to end customers Or best because
they offer the best
2009/2/10 David Backeberg dbackeb...@gmail.com
On Mon, Feb 9, 2009 at 4:23 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I would like to improve my understanding of T.38.
I recommend you try out Asterisk 1.6 if you want to play with T.38.
I DID get asterisk-1.4 working with fax, but I was
yes，i conf the meetme.conf
[rooms]
conf = 1000
any other friends can give me some advices?
2009-02-10
邱磊
发件人： Giedrius Augys
发送时间： 2009-02-09 13:04:46
收件人： Asterisk Users Mailing List - Non-Commercial Discussion
抄送：
主题： Re: [asterisk-users] meetme application
2009/2/9 邱磊
On Tue, Feb 10, 2009 at 12:39 AM, Mike Diehl mdi...@diehlnet.com wrote:
I'm looking into being able to send/receive SMS messages with my
asterisk box in the US. I've seen the SMS command as well as the Kannel
program. I'd prefer to do it from Asterisk.
snip
Do you have SMS service on the
Hi all,
I built my first asterisk using the traditional (?) .conf files and
constructs.
I recall reading books at the time about AEL but it seemed new and
untested so I left it alone. Now, I'm interested to poll the audience
here to see if I should look into using AEL instead (or in addition
My working meetme.conf is like below.
[general]
[rooms]
conf = 101,,
conf = 102,,
Your original email says your meetme.conf is:
[rooms]
conf = 101;
If you don’t want to use passwords, I think it is better to use:
[general]
[rooms]
conf = 101
Hope this helps!
Of course you should be using AEL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Tuesday, 10 February 2009 6:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
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