before you load the module.
But you can find out easily by loading the module manually while keeping
an eye on the output in the console.
Regards,
Patrick
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lp ooh323' for other
possible commands)*
Same again. Do you have any experience with Asterisk? You may want to
first read an Asterisk book or just hire an experienced Consultant to
setup your Asterisk box with H.323.
If you use Asterisk version 1.8 have a look
ities like OpenSIPS or Kamailio.
Regards,
Patrick
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http://www.asteris
s :-)
Regards,
Patrick
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asterisk-users
On 02/17/2011 02:26 PM, ERIC HERRON wrote:
Yeah it’s the same thing; I think.
I think we have different config files…are you using the split?
Unfortunately I have no idea what "the split" means. Can you please explain?
Regard
3.3.1 :-)
Regards,
Patrick
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asterisk-users ma
st.2.atten="0"
se.pat.misc.messageWaiting.inst.2.param="0"
se.pat.misc.messageWaiting.inst.2.type="silence"
se.pat.misc.messageWaiting.inst.2.value="0"
se.pat.misc.messageWaiting.inst.3.type=&
P670_backlight_on.jpg
Regards.
Patrick
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configured).
Hope this helps.
Regards,
Patrick
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g or scrolling on a phone screen is
annoying, distracts and should be disabled. I find the blinking MWI led
annoying and afaict there's no way to change its behavior.
Regards,
Patrick
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dialing when
it connects the caller name and number jump 1 pixel higher, which
looks weird as it is close to the line. One 3.2.3 it didn't move up
and looked centered. However the scrolling caller id for incoming
calls make this minor annoyance worth the upgrade.
That's good to know.
usly big clue-by-4.
To me it seems like their 3.3.x branch could use a few bugfixes...
Have you tried an older or newer release?
Regards,
Patrick
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Ne
I also forgot to add that my bandwidth is highly used (mostly out
traffic) since I've detected the "attack"
On Wed, Nov 17, 2010 at 06:46, Patrick wrote:
> Dear asterisk users,
>
> A few weeks ago I've been attacked by a DOS on REGISTER that I've
> solved w
Rx: REGISTER
127.0.0.1(None) 17042946600101/1 0x0 (nothing)
No Rx: REGISTER
It is not a configuration issue causing loops because my config has
not changed since months.
Any help is appreciated
Best regards,
Patrick
--
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>>> As a podcaster I use Asterisk extensively and often have several people
>>> in
>>> a conference room. We'll record the calls via a SIP phone connected to
>>> a
>>> sound mixer. Is there an easy way to bump up the audio bitrate for all
>>> callers connected to the Asterisk server and improve the
>> As a podcaster I use Asterisk extensively and often have several people
>> in
>> a conference room. We'll record the calls via a SIP phone connected to a
>> sound mixer. Is there an easy way to bump up the audio bitrate for all
>> callers connected to the Asterisk server and improve the genera
Hello,
As a podcaster I use Asterisk extensively and often have several people in
a conference room. We'll record the calls via a SIP phone connected to a
sound mixer. Is there an easy way to bump up the audio bitrate for all
callers connected to the Asterisk server and improve the general sound
qu
actually
using AGI script.
How can I do this with asterisk ?
Best regards,
Patrick
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routes and trunks
my problem is i cant make inbound and oubound calls.
my settings are like -
inbound route;
DID NO: 1000 - no to show external incoming calls
CID NO: 020XX - Telkom line
Hi,
at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.
You can convert wav to mp3 on the recording server and then send it to
the central system.
Bye,
Patrick
On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
wrote:
>
> Hello,
&g
Hi,
you can do print the dialstatus to the console e.g.:
exten => s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
Bye,
Patrick
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Hi all,
I have tried to configure inbound routes in my elastix box 1.5.2 but i don't
get how to associate it to a Trunk i have created.
Also configuring outbound routes is proving to be a challenge. I have read the
Elastix without tears.pdf guide but seems not much information is provided in
th
Go to sites like digium.com, asterisk.org, asteriskguru.com,
trixbox.com,elastix.org for more understanding.
Goodluck.
From: "aster...@opensourcesolution.in"
To: asterisk-users@lists.digium.com
Sent: Tue, October 27, 2009 12:34:18 PM
Subject: [asterisk-users]
00:04:02.0 ystdm8xx+e159:0001 Yeastar YSTDM8xx
From: Tzafrir Cohen
To: asterisk-users@lists.digium.com
Sent: Mon, October 26, 2009 4:05:16 PM
Subject: Re: [asterisk-users] No tone, one way communcation.
On Mon, Oct 26, 2009 at 05:02:18AM -0700, PATRIC
1. When i connected my analog phone to fxs card, i cannot get dial tone what
could be the problem?
I am using elastix 1.5.2 based on centos 5.2 Final.
2. On my 2 sip softphones using x-lite linux versions, i get one way audio how
do i solve this?. This problem is also present when i use a windo
I want to interface asterisk with a legacy pbx that has around 23 extensions
through my 8 fxs card, how do i work around this?
Hint: I have already terminated 8 extensions from the legacy PBX, i was
thinking whether i can peer the extensions from the PBX i.e like 5 extensions
be peered to one ex
ly
that the AGI scripts continues.
I guess in this case that a new thread is the most convenient way, or,
by any chance, do you know if asterisk will stack commands if a new
agi command is received while the stream file is not finished ?
Thanks in advance,
Patrick
On Thu, Oct 22, 2009 at 17:
ll it stop the first one to execute the second one
?
If stops, is there any AGI library that handles this kind of behavior
or should I code it myself using callback methods ?
Thanks in advance for your answer
Patrick
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"tar.gz" of your kernel version and extract into
/usr/src/kernels/ directory
!
--
Regards,
Chandrakant Solanki
On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE
wrote:
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error;
>
>make[3]: L
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error;
make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a
mxml/libmxml.a -lncurses
make[2]: Leaving directory `/usr/src/zaptel-
Hello,
we are using vyatta, a linux based router. the software is more
focused on routing capabilities, than on firewall rules, but it works
fine an there is a very good support. for ha you can use it in a
cluster.
bye,
patrick
--
Niemann + Frey GmbH
Bischofstraße 80
47809 Krefeld
Tel. +49
r 3G ?
Thanks in advance
Patrick
On Sat, Oct 10, 2009 at 06:57, Frank Bulk wrote:
> There are two commercial vendors that come to mind, namely DiVitas and
> Agito.
>
> Frank
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-use
Hello guys,
I'm wondering what is required and involved in order to provide a
wifi/GSM handover to customers.
After googling I haven't found any product/vendor. Do you have an idea ?
Thanks in advance
Patrick
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e pbx)
Thanks anyway, I'll definately put this useful command on top of my
head. I'm sure I'll need it one day
Patrick
On Fri, Sep 18, 2009 at 02:11, C. Chad Wallace
wrote:
>
> At 7:16 AM on 17 Sep 2009, Patrick wrote:
>
>> I've one SIP trunk that support multiple
Hello Steve,
Thats what I was expecting :-(
I want to send an email in html format as well as sending an SMS to
the mailbox owner using clickatell's api
Any other ways to do this ?
Best regards,
Patrick
On Thu, Sep 17, 2009 at 09:26, Steve Edwards wrote:
> On Thu, 17 Sep 2009, Patri
Thank you Alex, I'll handle this programatically if there is no other way.
Best regards,
Patrick
On Thu, Sep 17, 2009 at 07:51, Alex Balashov wrote:
> You can set some kind of counter in the dial plan, call an AGI script,
> use func_odbc to make database calls, or otherwise a
e SIP configuration parameter
"limitonpeers" will limit at the trunk level, right ?
Thanks in advance
Patrick
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Reg
in advance
Best regards,
Patrick
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Hello,
I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.
Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
there any alternative/workaround ?
Best regards,
Patrick
_
Hello Danny,
I've also the same question :-)
I've tried to find more information on the "pup" mime enabled program
but I haven't find something on the internet (every search refers to
puppy linux :-( )
Can you give more info ? Where can I find it ?
Thanks in advanc
Thank you Jim, I'll check what I can find from the DumpChan() and keep
the mailing list posted.
Best regards,
Patrick
On Wed, Sep 9, 2009 at 16:57, Jim Dickenson wrote:
> Depending on version you are using you could use the M option on the
> Dial command. I use 1.6.0.x and it works
very
reliable in my case.
Is there any other way to retrieve the information ? Channel variable ?
What a CDR(dst) returns after the Dial ? All destination or only the
destination that has answered ?
Thanks in advance
Patrick
On Wed, Sep 9, 2009 at 16:15, Danny Nicholas wrote:
> You could
your help
Patrick
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Setting echocan for channel 6 to none
Bye,
Patrick
On Tue, Aug 18, 2009 at 3:08 PM, Jean-Yves Avenard wrote:
> Hi
>
> That was a fast answer, impressive !
> 2009/8/18 Kevin P. Fleming :
>>
>> Did you read the upgrade documentation that comes with DAHDI,
>> specifically fr
nts, as
gordon wrote.
You can also use a WLAN adapter to use usual sip phones without to many cables.
Bye
On Tue, Aug 18, 2009 at 1:20 PM, Olivier wrote:
>
> 2009/8/18 Patrick Plattes
>>
>> You can also use different identities.
>
> Yes, it's true but the trouble is to f
hi,
stunaddr = stun.exiga.net looks wrong ^^
in generally it looks like a nat problem.
bye,
patrick
On Mon, Aug 17, 2009 at 8:12 PM, Daniel Bareiro wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi all!
>
> I'm trying to connect to ekiga.net through a
hi,
you can use call-limit=1 in sip.conf or DEVSTATE()
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
bye
On Tue, Aug 18, 2009 at 9:03 AM, James Mut
Hi,
well there are differnt ways to do it. It depends on what you want.
The start-stop scripts in /etc/init.d/ are looking for a pid file, so
they can figure out if the server is running. You can change the
script to get a message if the server is going up or down by the
script.
If you want that
You can also use different identities.
On Tue, Aug 18, 2009 at 9:08 AM, Olivier wrote:
> Hi,
>
> I need to replace digital handsets in offices where there cabling is
> appareantly not Ethernet-compliant.
> Today's usage is to press a key to toggle between private ou public line
> before issuing an
Hi,
maybe you wan't to use '0' in front of you telephone number. eg.
intern: 261 -> 261
exten: 002151-5462 -> 021515462
Bye
On Tue, Aug 18, 2009 at 9:08 AM, Olivier wrote:
> Hi,
>
> I need to replace digital handsets in offices where there cabling is
> appareantly not Ethernet-compliant.
> Toda
e0:passw...@sipgate.de/8001187e0
>> [8001187e0]
>> type=friend
>> context=testing
>> secret=password
>> host=dynamic
>> caninvite=no
>> canreinvite=no
>> qualify=yes
>>
>>
>> extensons.conf:
>> [testing]
>> exten => 800
Hi Jonas,
that works fine, but I think its just a work arround and not a real
fix :-). For the moment it is okay and I'll try to fix the error next
days.
Thanks,
Patrick Plattes
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; [8001187e0] bit?
>
> I have this in my Sipgate setup and it works. Worth a try.
>
> Cheers
> Andy
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick
> Plattes
&
> What does dialplan show testing output?
[ Context 'testing' created by 'pbx_config' ]
'261' => 1. Noop(261) [SIP]
'262' => 1. Noop(262) [SIP]
'263' => 1. Noop(263)
Thanks for the fast reply, but it does not help :-(.
Bye, Patrick
On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashov wrote:
> Try prefix your extension in extensions.conf with "_", e.g.
>
> exten => _123,1,...
>
>
l(SIP/263)
I don't know whats wrong here :-( Does anyone see my (usually) stupid error.
Thanks,
Patrick
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Does it this link help?
http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold
On Fri, Aug 7, 2009 at 10:07 PM, Venkateshwarlu
Kakkireni wrote:
> I want to a place a call (SIP) on hold in asterisk? Is there any way to do
> it? If yes, please give me an example. We are using Asterisk 1.4.24.1.
Hello,
well let me explain one part of your question, the host parameter. if
you want to restrict the access to one ip you can say it here.
"host=192.168.2.13" means, that you can only use this account from
192.168.0.13, eg. for security reasons. i recommend so set it to
"dynamic" at the moment an
Hi,
Vyatta & Asterisk works fine here. We are using traffic shaping DynDNS and NAT.
Bye,
Patrick
On Tue, Aug 4, 2009 at 2:27 PM, Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Tarek Sawah wrote:
>>
>> First of all it acts like a firewal
;m asking because it's not clear to me
if I need mISDN or that Digium (you) has developed native support for
the B410P card BRI card in zaptel/dahdi/libpri. If there's native
support for BRI, which version(s) of zaptel/dahdi/libpri would I need to
install to
vice '/dev/zap/ctl'
[snip]
Besides Tzafrir's advice, have you made sure that the udev rules for
creating the /dev/zap/* devices are in /etc/udev/rules.d and are correct?
Regards,
Patrick
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Stefan Schmidt wrote:
> you could use mrtg to get stats of the overall usage of the server. or
Thanks for your suggestion. I found a script here:
http://karlsbakk.net/asterisk/
Regards,
Patrick
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> with an "Incompatible Destination" Cause code 88. I found that some phone
> lines/numbers just couldn't call my isdn line. I still haven't figured it
> out yet...
Thanks for the info Jay. Do you use bristuff by any chance?
Regards,
Patrick
_
Luis Morales wrote:
> Try with fop,
>
> http://www.asternic.org/
Thanks Luis. I'll give that a try.
Regards,
Patrick
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To UNSUBSC
are indeed maxing out their ISDN channels.
Thanks,
Patrick
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There is allready a SIPGATE client. Closed for use with sipgate only,
but there will be more shortly...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Samstag, 27. September 2008 04:11
To: Asterisk Users List
Subject: [aste
Probably its saving its calls in wav format
Just check your recording directory , probably a lot of wav files in
there.
rm *.wav -f will then do the trick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mr surfit
Sent: Samstag, 27. September 2008 12
Or you have paid licences fees for it off course
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SIP
Sent: Dienstag, 16. September 2008 19:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Streaming MoH on 1.4
O
Ice is the feature you're looking for I think
If two clients support ice, a direct link between them will be made
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al lists
Sent: Dienstag, 09. September 2008 23:40
To: Asterisk Users Mailing
th ' , so it gives an error.
It works by changing it to this:
include_once(dirname(__FILE__) . "/lib/fpdf.php");
Regards,
Patrick
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least for now)
Tnx anyway..
Reg
PM
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Montag, 08. September 2008 17:22
To: Asterisk Users
Subject: Re: [asterisk-users] Asterisk realtime MySQL clients from same IP
problem
Patrick
> Likely, for you, like me, it's not that his email client is indersting
> blank lines... it's that whatever you're using to render his HTML
email
> into text is doing it -- for me, it's lynx under Mutt.
No. I configured my email client not to render the text/html part.
(Not wanting to render a sp
: [asterisk-users] Asterisk realtime MySQL clients from same IP
problem
Patrick Maartense schrieb:
> Users are creeated in the sippers table with following Fields set
>
> Name : .unique
>
> Host : dynamic
>
> Nat : yes
>
> Type: friend
>
> Callerid: xx
Following setup :
Users are creeated in the sippers table with following Fields set
Name : .unique
Host : dynamic
Nat : yes
Type: friend
Callerid: .unique value
Context: autocreate
Secret : xx
Disallow: all
Allow : all
Username : unique : same as Name
actly the same config
file as the other working phones. If I stick SCCP firmware in them they
start registering again.
Thanks!
Patrick
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ng:
>
> londo*CLI> module load app_voicemail.so
> [Aug 15 12:45:24] WARNING[14459]: loader.c:363 load_dynamic_module:
> Error loading module 'app_voicemail.so':
> /usr/lib/asterisk/modules/app_voicemail.so: cannot restore segment prot
> after reloc: Permission denied
Hi Matt,
Thank you for your suggestion. Comment inline.
Matt Gibson wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Patrick
> Sent: Wednesday, August 06, 2008 7:34 AM
> To: asterisk-users@lists.digium.com
> Subject: [
an hour).
Anyone have an idea how I can fix this?
Thanks and regards,
Patrick
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aster
blic documentation how to do
it. I did not dig into the Symbian developer docs. Maybe those contain
the answer.
Regards,
Patrick
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Rob Hillis wrote:
> Patrick wrote:
>> Andrew Latham wrote:
>>
>>> Read my hacks on the Cisco phones in Oreilly's "VoIP Hacks" book
>>>
>> Why the sales pitch for a 3 year old book? Can't you just give some
>> informatio
Andrew Latham wrote:
> Read my hacks on the Cisco phones in Oreilly's "VoIP Hacks" book
Why the sales pitch for a 3 year old book? Can't you just give some
information?
Regards,
Patrick
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Hi,
I'm looking for an addressbook solution that works with Cisco 7961 (SIP
8.3.5 firmware) so it's available as a service by pressing the button
with the picture of the globe on it.
Suggestions most welcome.
Regards,
Patrick
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to give oneself some
piece of mind over the crapload of money forked over for the product and
another crapload of money for the support contract/SLA.
Not sure what the term is in English but I think it is positive
cognitive dissonance.
/me steps down from soapbox now :)
Regards,
Patrick
__
On Thu, 2008-07-17 at 19:34 +0200, nik600 wrote:
> Hi what version of openh323 and pwlib are suggested for asterisk
> 1.4.21.1.? Thanks to all
Iirc it is openh323 1.18.0 and pwlib 1.10.1.
Regards,
Patrick
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you will then get more info what's going on.
Good luck!
Regards,
Patrick
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aste
On Wed, 2008-06-04 at 18:01 -0500, Bob G wrote:
> sorry its http://1ezphone.com/download
Anyone ran wireshark on the box running this app? Who's to say this
binary swf is to be trusted? Is the source available somewhere?
Cheers,
the mISDN mailing
list archives[1]. If your kernel version is 2.6.25 or newer than mISDN
1.1.7.2 will not work and you will need to install mISDN from git cause
Christian committed some fixes[2].
The card does come with support from Digium. Have you tried calling
them?
Regards,
Patrick
w with these
settings):
rx-checksumming: off
tx-checksumming: on
scatter-gather: on
tcp segmentation offload: on
udp fragmentation offload: off
generic segmentation offload: off
Regards,
Patrick
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you need to prepend it with an underscore to
make the variable persistent.
Regards,
Patrick
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(closes issue #12682)
Iirc JerJer always told people to use openh323-1.18.0 with chan_h323. Is
that still the case or can openh323-1.19.0.1 also be safely used with
chan_h323 in 1.4.19 and later?
Thanks,
Patrick
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too ?
> >
> don't bother, i found the backport
Can you please tell me where you found the backport?
Thanks,
Patrick
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C6
has been end-of-line for a long, long time...
Regards,
Patrick
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Hi,
Does anyone have a Jitterbuffer patch for Asterisk 1.2.28? Iirc the JB
patch used to be available at asterisk-backports.org but that website is
now one big advertisement.
Thanks,
Patrick
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w hours only to discover there is a "minor" omission in the
> documentation.
Can you please explain what you mean with "zaptel is required for SLA to
work"?
Thanks,
Patrick
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On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
> Anyone seen anything on the IP670 & the Color Expansion?
Great timing. Yesterday I was looking at the IP650 and wondered when the
successor to the IP650 would arrive. Do you have a link or more info
about the IP670?
Thanks,
On Fri, 2008-04-25 at 08:21 +0200, Olivier wrote:
>
>
> 2008/4/24 Patrick <[EMAIL PROTECTED]>:
> Hi,
>
> I need to setup an Asterisk box with 4x ISDN BRI links.
> Looking at the
> specs of various cards I favo
t; This also doesn't apply to chan_misdn hardware ...
Afaik you are right. I don't think you can use HPEC with the Digium BRI
card. Please correct me if I'm wrong. If you don't mind using some
experimental stuff give OSLEC a try since mISDN/chan_
alog, that is a horse of a different color, also the phone on
> either side, but especially your side can be the culprit (older
> Grandstream for one) Polycom seems to eliminate much of this.
Point well taken. You get what you pay for.
Regards,
Patrick
> Thanks,
> Steve Totaro
>
>
ther cheap price.
That's a solution too but a bit of a risk since afaik the hardware echo
cancellation can perform better than the software one. The CEO at the
client has echo to certain POTS destinations and I want to make sure
everything on his side is top notch.
Regards,
Patrick
__
t even understand why Sangoma would make a
> version without the hardware echo cancel. You get some degree of echo
> on practically every call.
Thanks Andres. Your feedback is most helpful.
Regards,
Patrick
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