Hello,
Anyone out there knows the steps to get a Sangoma UT50 or UT51 VoiceTimer
USB stick working with an OpenVZ instance of Asterisk?
I have Dahdi + UT50 driver installed on mother node running fine but not
sure what to do in OpenVZ which has Asterisk installed. Do I have to
install Dahdi on
Hi everyone;
Is it possible to provision lock Aastra phones to provider so that no soft
or hard reset can unlock them?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
, Jul 6, 2013 at 7:46 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 07/06/2013 08:15 AM, Bruce B wrote:
Hi everyone;
Is it possible to provision lock Aastra phones to provider so that no
soft or hard reset can unlock them?
Iirc you can use encrypted configs using an app
Hi everyone,
Has iCall gone belly or just having really lazy executives / support team?
They haven't placed a single long distance call for us since mid last
month. Have they run away with deposit money? Are they bankrupt?
I appreciate some feedback on this.
Thanks,
-Bruce
--
Hello,
I have 10 different routes with few different providers. When I place an
international call, I would like the system to try all those routes and
place the call through whichever possible. If there is any message but an
ANSWER the system should move on to next route. I know this is not the
Hi everyone,
We are getting cotinueous error messages over the past few days from iCall:
-- Called iCall/01144
-- Got SIP response 500 Server internal failure back from
72.249.14.242
Is this something everyone else is getting? They are very bad at support
and I am not sure if it's
Hey Zaf,
Just checking the Google Speech Recognition package again and I can't see
WolframAlpha.agi file. I check all of your projects on Git hub but can't
find wolframalpha.agi. Please let us know what the URL is.
Thanks,
Bruce
On Thu, Jan 12, 2012 at 2:49 PM, Lefteris Zafiris
Hello,
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM type of
attacks or others can't read the plaintext SIP packet info. VPN is not an
option. Looking for 2nd most secure to VPN.
P.S. Are both options part of
type
sniffing is not possible?
Regards,
On Fri, Jun 22, 2012 at 2:39 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 06/22/2012 12:56 PM, Bruce B wrote:
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM
Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file
cdr_mysql.so exists and I added it to modules.conf with load =
cdr_mysql.so. But the module doesn't show loaded when I do module show
like cdr.
Seems like some config is missing. Which file is responsible for this type
of
, Bruce B bruceb...@gmail.com wrote:
Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file
cdr_mysql.so exists and I added it to modules.conf with load =
cdr_mysql.so. But the module doesn't show loaded when I do module show
like cdr.
Seems like some config is missing. Which
asterisk using source TAR.GZ ? It will make
you learned where you have to do some setting... :D. Rather difficult but
fun... :D
** **
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* 18 Juni 2012
, Jun 17, 2012 at 11:13 PM, Bruce B bruceb...@gmail.com wrote:
This is not related to Asterisk Now but simply Asterisk as provided by
Digium repositories and documented in Asterisk Wiki. Source install is one
way to do this but that is not the issue in question.
I hope someone at Digium fixes
Hello,
I have done yum install asterisk18 freepbx and it has installed Asterisk
and FreePBX just fine. However, none of the CDR get recorded in
asteriskcdrdb table in MySQL. They are available
in /var/log/asterisk/cdr-csv/Master.csv. What configuration file sets the
setting for writing these CDRs
Hello,
I want to send out 1000 faxes. I have an excel sheet of numbers and I have
Asterisk 1.8 installed from repository. I don't want to use a fax machine
or any ATAs or analogue equipment. How would Asterisk help me with faxing
these? and what add-ons do I need to make this possible?
I can
:42 PM, Lee Howard fax...@howardsilvan.com wrote:
On 05/03/2012 01:28 PM, Bruce B wrote:
I want to send out 1000 faxes. I have an excel sheet of numbers and I
have Asterisk 1.8 installed from repository. I don't want to use a fax
machine or any ATAs or analogue equipment. How would Asterisk
do for me as
it's one channel limit like you mentioned. I probably don't need T.38 but
hey it won't hurt to have it.
Thanks again,
On Fri, May 4, 2012 at 12:37 AM, James Sharp ja...@fivecats.org wrote:
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running
I am looking for the same thing as Virendra, Easy to deploy open source.
Vicidial and Goautodial are hard to deploy and too blotted. Vicidial is
also ugly interface. No diss and I know that it's the best out there but
not the easy to deploy. Goautodial people didn't even show interest
configuring
virbh...@gmail.com wrote:
how many UDP ports is required for 1 call. and why .
If you mean a voice call, it appears that each host must open three
UDP sockets:
- One to send/receive SIP commands
- Two to receive sound (one for RTP, one for RTCP; The first port is
even, the other is odd)
On Thu, Feb 16, 2012 at 12:30 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/11/2012 06:59 PM, Bruce B wrote:
If your server is open to the internet and in SIP general section you
have nat=no and in peers you have nat=yes or vice versa then it's
possible to enumerate your extension
Wouldn't a shell script be a band-aid solution?
CLI verbose should have absolutely no effect on other loggings. I have been
saying this forever that Asterisk logging should be very strong and
separate of anything else including what we see on the CLI. This is
important for security reasons. You
Sammy,
Would you care to elaborate please. Have you had experience doing such a
campaign using AMI? Maybe you can share of the code.
Most appreciated,
On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.com wrote:
I'd definitely go with AMI !
On Sat, Feb 11, 2012 at 6:39 PM,
If your server is open to the internet and in SIP general section you have
nat=no and in peers you have nat=yes or vice versa then it's possible to
enumerate your extension. Because Asterisk responds with different messages
if the extension exists or not based on that difference in the nat setting
On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling ewiel...@nyigc.com wrote:
All screwing up with Asterisk is supposed to be documented in the
relevant UPGRADE*.txt files. Have you checked them?
is supposed to be but does NOT happen. There are many examples of
regressions introduced after many
Does sox have more features on a Debian system than RHEL? Is that why it
won't work on RHEL?
Cheers,
On Wed, Jan 4, 2012 at 6:42 PM, Lefteris Zafiris zaf@gmail.com wrote:
Fresh code is out! The use of sox can be now optionally enabled by the
user if the system has a recent version of the
:46:14 -0500
Bruce B bruceb...@gmail.com wrote:
Does sox have more features on a Debian system than RHEL? Is that why
it won't work on RHEL?
RHEL's 5 version of sox is really old and outdated. The command syntax
and the switches are totally different compared to recent versions of
sox
, Bruce B bruceb...@gmail.com wrote:
Thanks.
I have been testing Aastra phones with SIP and had great results. I am
testing my cell phone now and sometimes get -1 for id, status, utterance,
and confidence. What does that mean?
Cheers
On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris zaf
.
Updated script attached.
-Bruce
On Fri, Jan 6, 2012 at 11:03 PM, Bruce B bruceb...@gmail.com wrote:
NVM. I explored the code and see the logic. I had sox = 1 so it was
failing on RHEL.
To report, my cell phone from a PRI gets same confidence level just like
SIP. Building my control app
but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep introducing new
version and more bugs :-/
Wish not granted !!! :-) You will be the guinea pig to new features !!!
Same issue with A2Billing connecting to Asterisk. With older version this
problem is
Note to self: Never release anything asterisk related without testing
on RHEL/Centos 5
Thank you for reporting this. I have replaced sox with flac and it seems
to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
You can get the updated code here:
Very interesting. I just tried to get it to work but it complains about
sox. Probably you used a different version of sox?
*PBX-*CLI /usr/bin/sox: invalid option -- -*
*/usr/bin/sox: invalid option -- n*
*/usr/bin/sox: invalid option -- o*
*/usr/bin/sox: -r must be given a positive integer*
* --
And with recent version 14.3.2 I get:
/usr/local/bin/sox FAIL formats: no handler for file extension `flac'
-- speech-recog.agi: /usr/local/bin/sox failed: 512
-- SIP/-002eAGI Script speech-recog.agi completed, returning 0
Regards,
On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb
Hello,
Can anyone explain what this attack was trying to do? *19.19.19.19 *is my
server IP and it seems that they are trying to use my server IP to initiate
a SIP call to 199.16.208.29 or 199.16.208.30. Is that so?
*Call Date Channel Source CLID
On Sat, Dec 31, 2011 at 5:06 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Fri, Dec 30, 2011 at 09:03:34PM -0500, Bruce B wrote:
So, based on what you are saying if I issue the command core set verbose
0 and then exit the system Fail2Ban will stop working for Asterisk (this
is since
Had one of my systems hit this morning too. Asterisk 1.8 branch+FreePBX
2.9 no anonymous. 260 call attemps in 2 minutes. Here is part of the logs.
I am updating my filter to see if it helps, THANKS Bruce!!!
I am trying to get this working for FreePBX as I think they are more
vulnerable than
Hi everyone,
I am playing around with Asterisk 1.8.8.0 from Digium repository. This is
all there is to my logger.conf file:
*[general]*
*dateformat=%F %T*
*
*
*[logfiles]*
*full = notice,warning,error,debug,verbose,dtmf,fax*
*
*
However, when I do, core set verbose 0 at CLI, Asterisk ceases to
the
command logger mute and console will not get output but log file will.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 30, 2011, at 3:11 PM, Bruce B wrote:
Hi everyone,
I am playing around with Asterisk 1.8.8.0 from Digium repository
One can set the verbose level as well as the debug level. These control
how much log information is generated at all not where it is being written.
What do you mean by above? Can I see something in the logger.conf that
will keep it always at certain verbose level regardless of what command I
, 2011 at 8:36 PM, Jim Dickenson dicken...@cfmc.com wrote:
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 30, 2011, at 4:55 PM, Bruce B wrote:
One can set the verbose level as well as the debug level. These control
how much log information
:
On Thu, Dec 29, 2011 at 12:05 AM, Bruce B bruceb...@gmail.com wrote:
I have been running 1.8.7 with a few fixes back ported from the 1.8.8
release candidate for the last 2.5 months. The system processes around
4,000 calls per day over PRIs for 250 Polycom phones.
Previously I was running 1.6.1.18
Maybe your logger is not setup properly?! You should get the IP in logs. I
can't think of when you won't get the IP in your logs unless the SIP
packets are manipulated. That IP is from Voxel.net. You don't have a VPS or
service from them do you?
2011/12/29 Michelle Dupuis mdup...@ocg.ca
1. I
Interesting attack tonight fail2ban them Bruce B
mentioned it would be nice to have input from the Community to come up with
the best set of fail2ban filters. That's a great idea. So let's start with
Bruce's filters (thanks!) and take it from there. Anyone have any
improvements and/or additions
Hi everyone,
I see that there was a bug in version 1.8.5.x and people were advised to
move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem.
Here is the output:
*chan_sip.c: Asked to transmit frame type ulaw, while native formats is
0x100 (g729) read/write = 0x100 (g729)/0x100
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone,
I
I have been running 1.8.7 with a few fixes back ported from the 1.8.8
release candidate for the last 2.5 months. The system processes around
4,000 calls per day over PRIs for 250 Polycom phones.
Previously I was running 1.6.1.18 with a bunch of back ports for fixes and
features. Overall it
You mentioned the IP, 208.122.57.58, where did you get that from?
Following are the default for Asterisk 1.8 (It would be great to have
others input on this to strengthen this part of the filter):
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Hi everyone,
Since three weeks ago, we have been getting A LOT of 603 Declined calls
from iCall. I called a few times and their support is either non-responsive
(they never call back) or can't fix the issue. I am wondering if everyone
else is experiencing the same thing or is it because we
I could be wrong but this sounds like a NAT issue rather SIP related packet
issue. You are not receiving a response back is what I get a lot of times
when my NAT is not setup properly. Call goes on for 10 or 20 second (I try
the echo application and it hangs up before I get to talk) and then cuts
Can you register with Eyebeam to VSP and have it work? Make sure you are on
the exact same network as the ATA when making this test. This should
isolate the NAT issue.
On Mon, Dec 19, 2011 at 9:27 PM, William Scott will...@magicwilly.infowrote:
On 20 December 2011 12:51, Bruce B bruceb
You needed to do asterisk -g or amportal start after your install. The
configs didn't apply because Asterisk wasn't running so there was no
connection to AMI. But when you updated module you Fpbx did an amportal
restart or start automatically and hence it worked. Anyhow, but the FPBX
rpm is broken
I think it only works with certain soft phones. I tried Aastra and it
doesn't work. But EyeBeam soft phone receives messages.
-Bruce
On Mon, Dec 12, 2011 at 6:40 PM, Jay R. Worthington
jayrworthing...@gmail.com wrote:
Hiya,
SIP Messaging is implemented in asterisk-10...
The only
{tcpdump src port } in the
dialplan or something like this. And I want RTP traffic only of a certain
call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How
Paul,
These trolls are the people who put your kid to school and put food on your
table by giving valuable input and testing the open source software.
Are you sure Digium endorses this stand of yours? Does everyone at Digium
think the users who gives feedback that is not exactly what you like is
that be? Even with new features you
can still stick to certain principles if you plan it ahead. If you don't
know how to do it, ask the community for input and people will help.
-Bruce
On Sun, Sep 25, 2011 at 1:22 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-09-25 01:01 PM, Bruce B wrote
like you don't take offence but you did and you read as,
I am trying to be rude Well, suit yourself and keep sucking up Alex.
On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 09/25/2011 02:23 PM, Bruce B wrote:
Stop wishing for that. I like Asterisk and I
afford one afternoon
meeting to decide what the commands naming convention should be for the next
20 years.
On Sun, Sep 25, 2011 at 4:05 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-09-25 02:23 PM, Bruce B wrote:
Paul,
LOL...you are trying to change the subject. That's naive.
You
, Sep 25, 2011 at 9:23 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-09-25 08:57 PM, Bruce B wrote:
First of all, what the heck is this link you referenced:
http://lists.digium.com/pipermail/asterisk-users/2010-**
**April/247084.htmlhttp://lists.digium.com/**pipermail/asterisk-users
Hi everyone,
I don't mean to be rude but honestly which genius comes up with changing the
simple:
help
to
core show help
That's just an example. If it was only this or if this was only a two words
loss then I would be fine.
I think someone just loves to play around with the commands with
if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your
IVR announcement is not recorded in g729 and you see g729 on the channel
when you call into IVR then it's transcoding as well.
On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling ewiel...@nyigc.com wrote:
Assuming SIP sip show
sip show channels is the command you are looking for.
On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
asterisk -rx core show channels verbose does not provide transcoding
details.
Unless I have missed something.
Sans
On Wed, Aug 31, 2011 at 10:34 PM, Danny
Hello,
Is softhangup still there? It's unknown command to Asterisk
1.6.20-1..there is no mention of this in CHANGES files.
Also channel hangup request SIP/channel-name doesn't work for SIP.
Is there any other command I am missing?
Thanks
--
Hi everyone,
In USA when doing a CNAM search, what sort of information is provided back?
Does this information include carrier name? service address? service type
(public or private phone)? etc...?
Also, if you are not a CLEC do you have to purchase this service through a
mediator CNAM look
Hi everyone,
Pinging a phone set I get 0.529 ms round trip delay. Running sip show
peers in Asterisk CLI I see anywhere from 5 milli seconds to 280 ms. How
are both of these different and why are they so different? Is the latter
based on SIP packets return?
I have a paging device that shows
Can you please elaborate on how to apply the patch?
Also, is the repository updated with the new code?
Regards,
On Tue, Aug 2, 2011 at 7:34 PM, Richard Mudgett rmudg...@digium.com wrote:
Can you please point me to the patch that you just made?
The patch is committed to v1.6.2 SVN branch.
is the time
frame to fix such bugs?
On 7/30/11 7:39 AM, Bruce B wrote:
I think this should be a quick fix since it's rendering the latest
stable version useless and making the impression that it was released
just to break things and force people onto 1.8x. Just a thought...no
blame game
I would be very interested in iLBC. I even posted regarding this to this
mailing list and the thread died after no one was able to confirm it works.
I think there are others who would really like to see H.323 working from the
repo as well (I think that is not working as well).
Regards,
On Tue,
Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 31, 2011 3:03:07 AM
Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time
frame to fix such bugs?
On 7/30/11 7:39 AM, Bruce B wrote:
I think this should be a quick fix since it's rendering the latest
stable
asterisk-users@lists.digium.com
Sent: Sunday, July 31, 2011 3:03:07 AM
Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time
frame to fix such bugs?
On 7/30/11 7:39 AM, Bruce B wrote:
I think this should be a quick fix since it's rendering the latest
stable version
There is much more to installing and configuring OOH323 as it's not easy
breezy install. I think a professional developer help would be
more appropriate than users patching. Just my thought.plus it adds a
great deal of functionality to Asterisk to allow for all add-ons to be
install via
, Jul 31, 2011 at 4:03 AM, Vahan Yerkanian va...@arminco.com wrote:
On 7/30/11 7:39 AM, Bruce B wrote:
I think this should be a quick fix since it's rendering the latest
stable version useless and making the impression that it was released
just to break things and force people onto 1.8x. Just
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also noted
in 1.8x install. Is it not going to be taken care of because it's 1.6x ?
Thanks
--
things so badly on the last stable version.
Regards,
On Fri, Jul 29, 2011 at 6:23 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 07/29/2011 06:20 PM, Paul Belanger wrote:
On 11-07-29 06:12 PM, Bruce B wrote:
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/**jira
Hmmm, if alwaysauthreject is already breaking RFC rules then why not break
another rule for the greater good? It would only add another layer of
security.
Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous calls.
Keep it off by default and then
I would have to err on the side of CDR to say that the only difference in
analogy you provided (SSH vs Asterisk) is that people lose much more
in VoIP than they ever did in SSH hacking. So, if this is an
exceptional case bending a rule or two of RFC in favor of security won't
harm
, Jul 23, 2011 at 10:04 AM, Bruce B bruceb...@gmail.com wrote:
Robert thanks for weighing in.
So, you are saying that FreeSwitch on it's own can tackle issues like
this
without the need of OpenSIPs? Can you elaborate please?
Thanks
On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone rhuddles
Hello,
I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all.
Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any
...@evaristesys.comwrote:
On 07/22/2011 07:32 PM, Bruce B wrote:
Hello,
I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and
receive ACK or Declined rather that those inviting a call who are not
PEERs at all
wrote:
On Fri, 22 Jul 2011, Bruce B wrote:
1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS actually
give me the full capability to the SIP stack to do the sort of thing I was
asking for? And this can run on the same server as Asterisk is running?
Configure OpenSIPS
I can confirm as well that there is an issue with Asterisk crashing.
Asterisk 1.6.2.19 was installed using Digium repository. Probably some
module was enabled in the repository install that is causing this.
On Mon, Jul 18, 2011 at 12:13 PM, Lee Archer lee.arc...@thebigword.comwrote:
Hi Kevin,
.
*
*
- Bruce
On Wed, Jul 13, 2011 at 2:49 AM, Malvin Rito
mr...@mail.altcladding.com.phwrote:
**
Bruce,
Thanks. I already figured out the problem. It seems that a firewall issue.
Regards,
Malvin
On 7/13/2011 12:30 PM, Bruce B wrote:
Your trunk shows busy:
* -- Called CordiaVoIP/639285010430
Your trunk shows busy:
* -- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)*
Try this in the CLI (asterisk -r):
*core set verbose 0*
*sip set debug peer CordiaVoIP*
And then make a call and read why
Hi everyone,
Occasionally (with no set pattern), I get *SIP/2.0 407 Proxy Authentication
Required *from iCall when trying to termiate to their international
gateways. I have tried direct IP termination as well as SIP register but
both just fail with above message whenever they want. Specially in
Hi everyone,
I just lunched a CentOS VM in Proxmox and used the Digium repository to
install Asterisk using yum install asterisk16...and it works great. Runs
and it seems to have installed ztdummy as well without the need to touch the
host node. But when I try to compile Dahdi from source on the
Hi everyone,
When doing a sip show settings on Asterisk 1.6.2.18, I see the following:
Match Auth Username:No
Allow unknown access: Yes
Allow subscriptions:Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
What do each of above signify?
Hi everyone,
What is wrong in below asterisk application? The output should be content of
field booth_status from table booths:
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten = _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status
Hi everyone,
I want to issue the command:
iptables -F
and then rebuild everything from the beginning with a very limited scope and
then without locking myself block all other traffic. Can you suggest what I
should put in the shell that would get me this:
Allow traffic from subnet 172.16.0.0/24
to replace /etc/sysconfig/iptables with it and let
it accept all traffic from one subnet on my tun0 which is my VPN and block
all other traffic?
Thanks again
On Sat, May 14, 2011 at 8:14 PM, Hans Witvliet h...@a-domani.nl wrote:
On Sat, 2011-05-14 at 19:51 -0400, Bruce B wrote:
Hi everyone
Thanks Jeremy. But unfortunately no time to go over all this in detail.
Maybe in future. Also because as I repeatedly said I have OpenVPN setup so I
trust the VPN network there is no need for all this complication. Simply
allowing all traffic out and only allowing VPN traffic in from tun0 would do
the CALLERID data before dialing the SIP extensions and, if it is
empty or contains “asterisk,” reset it to something like “not available.”
Cheers,
~Brian
*From:* Bruce B [mailto:bruceb...@gmail.com]
*Sent:* Friday, May 06, 2011 10:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial
Hi Brian,
Did you find a solution to your problem? or at least got a working dial-plan
for it? I have the same problem again as well and want to know what to do
with the dial-plan to off-set the effect at least since Telco says it's not
their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53
can add this in extenssion.conf
exten = 223,1,Answer()
exten = 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 223,3,Dial(SIP/223)
exten = 223,4,Hangup()
i can record without any issue in /var/spool/asterisk/monitor
2011/5/4 Bruce B bruceb...@gmail.com
Thanks for the input
/asterisk/monitor/
*
On Tue, May 3, 2011 at 10:40 AM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually
shows the MixMonitor line just fine on the CLI. How can MixMonitor be
debugged for things like privilege issues
Hi everyone,
For some reason MixMonitor doesn't record when it should; It actually shows
the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for
things like privilege issues or filename issues?
**I had this working at one point and then stopped working. Not sure what I
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all other bad
quality things that can happen to a SIP trunk? I have plenty of bandwidth
and crisp clear lines so the only thing that I can think of is to limit
bandwidth but even that requires quite some scripting work.
as
well. I am going to try the the packet drop method now. I think that is the
right one for the situation.
Thanks again
On Thu, Apr 28, 2011 at 11:57 AM, Tony Mountifield t...@softins.co.ukwrote:
In article banlktim8w+vjjj87oyy1mvppsfwflut...@mail.gmail.com,
Bruce B bruceb...@gmail.com wrote
Hi Bilal,
Probably there is no open source tool or a good ones available. But few of
them I worked with provide up to 2 users free of cost license type of
reporting. Reporting for Call Centers can get very complicated. Once you
explore some of the commercial apps you will notice how extensive
Henning
*Sent:* Monday, April 11, 2011 8:47 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] Occasional call from asterisk
Bruce B said:
We experience exact same thing on DAHDI with Sangoma USB FXO device on
short circuited lines. Phantom calls
, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic.
This is the dialplan:
[a2billing]
exten = _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten = _X.,n,AGI(a2billing.php,1)
exten = _X.,n,Hangup
, and will be executed after hangup
On Mon, Apr 11, 2011 at 6:36 PM, Bruce B bruceb...@gmail.com wrote:
Thanks for the input but I am not sure if that answer my question of if
it's normal behaviour for AGI scrip to terminate after the h extension
rather than end of x extension even
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