Hi,
> There is of course one obvious issue, that of powering your equipment
> at 220V/50~ and the plug convertors if your are lucky enough to have
> power supplies that do 100-250v.
and the plug format is different
(UK, germany+NL, France+Belgium, Italy, ...)
there are some 'universal' plug chan
Hi,
Try to cancel 'silence suppression' from
the 'other' source.
Asterisk does not support 'silence suppression'
(yet ?) (as far as I know)
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTE
Hi,
Try WITHOUT silence suppression.
* does not seem to support that
regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Asterisk guy [mailto:[EMAIL PROTECTED]
Sent: mardi 3 mai 200
Hi,
FXS card = plug to regular phone
FXO card = plug to phone line
The trick I use is:
FXO with a 'O' as in Office.
This is where you plug your phone
A FXO card emulates a phone (receives power)
FXS with 'S' as in (public) Switch
This is the part that gives power
A FXS card emulates a swi
Hi,
First, do forgive any syntax or language errors as English is not my mother
tongue.
To make a 'long' story short,
- let's look at a phone home.
The public switch (the very big machine at your telco) provides power to the
line. This power gives energy to your phone.
This is how a 'standard' p
Hi,
What do you have h323 or oh323 ,
(open h 323)
I think you have the latest.
You must use the SPECIFIC files.
Check http://www.inaccessnetworks.com/projects/asterisk-oh323
Also PATCH the file BEFORE compilation
It should run then.
Good luck
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Te
Hi,
The legal way is to buy a smartnet (support contract) for the soft.
That way you can download it from Cisco's
web site.
Try to contact your reseller.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 8
Hi,
Look into your "*.conf" files
Res_mysql.conf is a good start
Check for user-id & password
Also check the dbsock=.
(the default
value did not correspond to my 'default' installation of sql).
I have now dbsock=
/var/lib/mysql/mysql.sock
Look for THAT file in your system.
Hi,
My users use sip phones (grandstream 286 / 486).
No echo between sip calls (g729 too).
Calling the 'world' though an h323
VoIP provider, I have a very high echo level.
(I do not have this problem calling
through sip)
The connectivity to this partner is rather
good:
No pac
Hi,
I also 'spent' some times there banging my head on the wall.
Please read CAREFULLY :
http://www.inaccessnetworks.com/projects/asterisk-oh323
use only the mentioned version
the compilation & linking seem to be rather sensitive
(for info, I use chs 29 march 05 & their module 0.7.2)
read also
Hi,
I am using real-time, oh-0.7.2, G729
Calling from (SIP)UA through asterisk towards h323 devices or the other way
round, I get only one-way audio.
Called party can only talk, caller can only listen.
Calling SIP to SIP is ok.
All devices are on official IP addresses.
(no NAT)
Regards,
PROTECTED]
Sent: lundi 4 avril 2005 18:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Realtime - extensions
configurationhelp
Shaoul Jacobson - TELLINK wrote:
> I still do not know how to 'translate' (from old extension.conf)
Hi,
> 0: 184892 XT-PIC timer
> 1: 5 XT-PIC keyboard
> 2: 0 XT-PIC cascade
> 8: 1 XT-PIC rtc
> 11:3585589 XT-PIC wcfxo, ztdummy, usb-uhci, eth0
> 14: 5360 XT-PIC ide0
> 15: 0
Dear Matthew,
Btw, thanks for your active presence on the list.
> You just answered your own question in the same post.
> so ... why did you even post this question
> if you answered it 2 lines later?
I mentioned I found out about the ',' that must be translated into '|'.
I was suggesting addi
Hi,
The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions shows a
very trivial sample:
INSERT INTO `extensions_table` VALUES
(1, 'mycontext', '_574555', 1, 'Wait', '2');
but how would you 'translate' an old definition as :
exten => _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30
Hi,
> I'm not sure I totally agree.
Good, we do agree on some :)
I also agree with some of your remarks
(no flame war)
> It is also useful if you control the narrowest pipe.
I agree. But I disagree about the definition of the narrowest pipe.
> A well configured router there will slow outgoing em
Hi,
QoS is nice (and important) but only works within a FULLY controlled end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own MPLS end to end.
Going from one provider on MPLS to anot
Hi,
(re-posted since I did not see my original one after some time)
In short : cannot register SIP phone (403 forbidden)
In long :
I am rather new to asterisk (and linux)
One month experience fighting my way in the doc & wiki.
I worked before with the static '*.conf' files.
That worked but I
Hi,
In short : cannot register SIP phone (403 forbidden)
In long :
I am rather new to asterisk (and linux)
One month experience fighting my way in the doc & wiki.
I worked before with the static '*.conf' files.
That worked but I need real-time.
I did compile a cvs head 29 mach 2005.
MySQL is
Hi,
Welcome.
Read the samples *.conf files
(in /etc/asterisk)
extension.conf, sip.conf are
some good places to start.
Read & search the wiki.
Many info there (also not always very clear)
success
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :
Hi,
Some time I did not touch a cisco.
At a previous job, I managed a 53xx
If I remembered well, you can define dial-peers at ingress and outgress.
The trick is the add a very specific header at ingress and remove it at
outgress.
Also, by then, not all traffic directions where possible on the 53
Hi,
I have the same problem with cvs head. (1.0.6)
See http://www.inaccessnetworks.com/projects/asterisk-oh323
And https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php
(issue 00...008)
some 'patch' files are included.
I am a newbie to linux and asterisk.
I do not want to blow my con
Hi,
pwlib 1.6.6 &
downloaded & ./configure & make it as written
The same with openh323-1.13.5
Downloaded & patched make & ./configure & make it as written
Then with asterisk-oh323-0.7.1
Downloaded (I used u file there to patch openh323)
Made some changes in the Mak
> Recently, I've been getting these messages:
> Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed:
> Dropping extra frame of G.729 since we already have a VAD frame at the end
Well I got the same when I started to use g729.
I did some search crawl in the archive and fount in the 'dev'
Hi,
There seem to be some codec incompatibility.
On *, you define alaw and you set ulaw on the Cisco.
Set both to same or add the other codec on (at least) one side.
Try if that solve it
Ex:
Add "allow ulaw" on * after the "allow alaw"
And / or
Add "codec g711alaw" on Cisco above the "codec g711
Hi,
Other basic settings that must be right:
- framing with crc4 or not
- clock source (mostly provided by the 'provider')
cabling: a straight cable will most probably be ok
but I have seen some strange settings over the time
LAYER 1 or 2 problems are often overloocked
Good luck
Shaoul Jacobso
Hi,
cisco's phones are VoIP only
polycom build (video-) conferencing devices.
One Cisco model (7930 I thnk) is a polycom in disguise.
The code is not 'cisco-like' (at least the version I had.
Both brands make very good quality equipments.
Good sound, good support, ...
Regards,
Shaoul Jacobson
Hi,
I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards)
1. the distro
I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing
(some C or C++ or python ...)
(buy the full version )
maybe the latest fedora is more complete ?
or easier to complete w
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