Thank you for the confirmation
Best Regards,
Le vendredi 21 janvier 2011 14:17:20, Kevin P. Fleming a écrit :
> On 01/21/2011 05:59 AM, Marc Leurent wrote:
> > Could you please give me a feedback regarding this issue, I'm not sure of
> > the answer I got browsing the web
Could you please give me a feedback regarding this issue, I'm not sure of the
answer I got browsing the web
Thanks and Best Regards
Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit :
> Good morning,
> I have a simple question,
> Is this problem would affect also an Aste
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if "Pedantic SIP
support: No" in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 201
Take a look at http://dev.leurent.eu/voip/MOS/
I'v done this a long time ago, hope it will help!
++
Le 08.03.2010 11:10, mosbah.abdelkader a écrit :
> Hello All,
>
>
>
> MOS and R factor are the two QoS parameters used to estimate VoIP call
> quality.
>
>
> I have found that they are calculated fr
I have the same result with Asterisk 1.4.21 on a Debian Lenny server
--
-- --
Marc LEURENT
lf...@leurent.eu
Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit :
> Hello, when I remove a peer from my sip.conf and just do a reload, the peer
> is still ping with SIP OPTIONS u
=default
;dtmfmode=info
;insecure=port,invite
;nat=never
;sendrpid=yes
;disallow=all
;allow=alaw
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Marc LEURENT
lf...@leurent.eu
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Thank you Klaus and Martin for your answers!
It's very helpful!
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Marc LEURENT
lf...@leurent.eu
Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit :
> You can call application Progress() from within dialplan and it will
> cause the Asterisk to send a SIP reply 183
> on
OTIFY
Supported: replaces
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Marc LEURENT
lf...@leurent.eu
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Thank you Shaun for your answer!
Indeed, I have made some basic tests to convert a file to g729 using the
software codec and it works!
Have a nice day!
--
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Marc LEURENT
lf...@leurent.eu
Le mercredi, 9 septembre 2009 19.32:30, Shaun Ruffell a écrit :
> On 09/09/2009 09:33 AM, Marc Leur
8000 Hz
service_notactivated.g729: empty
service_notactivated.gsm: data
I was able to create the gsm file with the command, but the g729 one is empty.
Have you got any idea how I can solve this?
Thanks
PS: I'm able to place call in g729 without problem and the TC400B works well
--
-- --
Marc LEURENT
lf.
Hello everybody,
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the time
being.
Best Regards,
-- --
Marc LEURENT
lf
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
> Could you share with us y
ne in X-number-to-dial SIP Header
subst('/^(To|t):(.*)sip:[...@]*@(.*)$/\1:\2sip:
$(hdr(X-number-to-dial))@\3/ig');
}
Have a nice day!
-- --
Marc LEURENT
Le Monday 23 March 2009 13.41:59 Marc Leurent, vous avez écrit :
> I have spoken to quickly,
> Usually Asterisk on an incoming ca
y
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-te
erfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
"ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=20734;rlp=0;rtt=0.094000"
t; > ... will generate a Request URI of
> > 1...@host.or.ip.of.sip.conf.peer.named.peer_name.
> >
> > It is also possible to send requests to hosts that are not explicitly
> > defined in sip.conf, with the caveat that only background [general]
> > sip.conf se
explicitly
> defined in sip.conf, with the caveat that only background [general]
> sip.conf settings will then apply:
>
>Dial(SIP/1...@ip.of.peer.not.in.sip.conf)
>
> Marc Leurent wrote:
>
> > Hello,
> > it is not an OpenSIPs problem I have, it's an A
gt; >
> > Thanks for your time!
> > ++
> >
> >
> > On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent wrote:
> >> Hello All,
> >> I have a little complicated question about the Dial command.
> >> I use OpenSIPs to loadbalance Asterisk Server
have an APT (for debian) repo up and running (still beta). We
> could do the same for RPMs or, in the worst case, to generate the
> packages for download.
> Also, there are some RPMs (for suse) - see
> http://www.opensips.org/index.php?n=Resources.Downloads
>
>
> Regards,
e dependencies. I can get an RPM from for libxml2 from
> ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me
> headaches. Any suggestions would be helpful. Thanks.
>
--
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une
rying use the Dial Command with
Dial(SIP/0123400010/0123400019@"Reg. Contact of the main number") but it
doesn't work
Have you got any idea how to rewrite the IP of the URI sent?
Thanks!
--
-- --
Marc LEURENT
lf...@leurent.eu
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Good morning,
Is it possible with asterisk to allow to share the same account on 2 different
devices, for example I want both my fix phone and my wifi phone to ring
in the same time.
I want to do it without making ringroups...
Any idea how to do it?
ay
Jared Smith a écrit :
> On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote:
>> It's just that I received SIP notify message saying that there is
>> nothing in the voicemail even when there is a message...
>
> Do you have a mailbox defined for the SIP device in sip.c
gt; On Mon, 3 Dec 2007, Marc LEURENT wrote:
>
> Good evening, I have something strange,
> I have unread message in my voicemail box but the SIP NOTIFY that are
> received by my telephone are like:
> whereas there is voice messages inside!
>
> Any idea how to solve that?
-BEGIN PGP SIGNED MESSAGE-
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Good evening, I have something strange,
I have unread message in my voicemail box but the SIP NOTIFY that are
received by my telephone are like:
whereas there is voice messages inside!
Any idea how to solve that? Thanks
PS: I'm using asterisk 1.4.13 +
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Good morning,
I would like to find a simple PCI express card with only one FXS module,
do you know where I can find such a card?
Thanks
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Comment: Using GnuPG with Mozilla - http://enigmail.moz
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Just download the g729 module that fits your hardware at
http://downloads.digium.com/pub/telephony/codec_g729/ and follow the
README: http://downloads.digium.com/pub/telephony/codec_g729/README
PS: do a 'cat /proc/cpuinfo' to know what it your process
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Hello,
Is the asterisk B2BUA patches useful anymore??
I'm trying to set a prepaid SIP network and the only way seems to get
through a patched asterisk with B2BUA functions..
The patches failed, Hunk + problems: I have repaired them, but is it
very use
//www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func,
> fileinfo/id,25/
>
> And yes - it's FREE as name suggests.
>
>
> Regards/Pagarbiai,
> Mindaugas Kezys
> Advanced Billing for Asterisk PBX
>
>
> -Original Message-
> From:
Good evening,
Have you got any idea which prepaid application will be the best to do
simple prepaid calls with a MySQL storage...?
PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch
Thanks
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in the Mailing list is
> able to help u out;)
>
> Best regards
> MoutaPT
>
> On Nov 13, 2007 6:14 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote:
>
>> Good evening!
>> I was wondering one thing,
>> I'm using freepbx to configure my asterisk server and I
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:[EMAIL PROTECTED] The
> codec_g726.so ITU G.726-32kbps G726
> Transcoder 0
> format_g726.so Raw G.726
> (16/24/32/40kbps) data 0
> 4 modules loaded
>
> So can u advise?
>
> Regards
> Bilal
>
>
>
>
>
&
ranscoder
> 0
> format_g729.so Raw G729 data
>
> 0
> format_g726.so Raw G.726
> (16/24/32/40kbps) data
> 0
> format_g723.so G.723.1 Si
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Good evening,
I have something strange, when I add an ALERT_INFO variable to a ring group,
the invite generated contains 2 lines with Alert-Info and my phones return a
400 Bad Request...
I've checked in my config files, there is only one line with
Se
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I have license for g729a audio codecs and I would like user to use them and
when the limit of 10 is reached, I would like the others to use ulaw...
Do youu know how to do it...
I have put:
allow=g729,ulaw
disallow=all
But ulaw is always chosen
H
726.so Raw G.726 (16/24/32/40kbps) data 0
> format_g723.so G.723.1 Simple Timestamp File Format 0
>
> The codec_g729a.so doesn't appear..
>
>
> Any idea how to solve the problem.
>
> Thanks
>
> Best Regards,
&
16/24/32/40kbps) data 0
format_g723.so G.723.1 Simple Timestamp File Format 0
The codec_g729a.so doesn't appear..
Any idea how to solve the problem.
Thanks
Best Regards,
Marc LEURENT
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