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 Asterisk 1.4.38
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
Thanks and Best Regards
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 2011
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 from other
I have the same result with Asterisk 1.4.21 on a Debian Lenny server
--
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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 until I restart
=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 the call that came
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 Leurent
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
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Marc LEURENT
lf...@leurent.eu
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
://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-telecom.ch
:\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 call sends an INVITE
Reg.Contact
Number@Reg Contact IP to the Peer IP
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 your
(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
codec used: g711a
--
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Marc
from the [Open]SER family.
lftsy wrote:
Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent by
Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent
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 Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le
:55 Marc Leurent, vous avez écrit :
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername)
Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated
. 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!
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lf...@leurent.eu
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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.
--
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Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
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,
Bogdan
Marc Leurent wrote:
Hello Darrin,
Maybe you should ask this question
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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?
-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.conf? If you
don't, Asterisk has no way of matching up a mailbox
, 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? Thanks
PS: I'm using asterisk 1.4.13 + Freepbx
#
U
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (Darwin)
Comment: Using GnuPG with Mozilla -
/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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT
Sent: Friday
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
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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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 have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound
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]
--- Marc LEURENT [EMAIL PROTECTED] wrote:
To know your architecture, use the cmd: cat
/proc/cpuinfo
After try to start to use the version below (i686):
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v32_i686.tar.gz
Good luck
bilal
.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
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
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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
/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
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using
appear..
Any idea how to solve the problem.
Thanks
Best Regards,
Marc LEURENT
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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
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