Hi List;
The output of cat /proc/cpuinfo giving a [Intel (R)
Pentium (R) D] so what is the g729 version I have to
download to work with my machine?
Any help?
Regards
Bilal
Looking for last minute shoppin
Hi List;
Anyone tried to use speex, ilbc and g729 and come back
with a preferred one in the quality?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
Dear Jared;
Thanks a lot, what is the website link (from where I
can buy this device)? Also, do u have any idea about
the prices?
Regards
Bilal
--
> Anyone can advise for a good IP Phone that has the
> ability to support SIP firmware and IAX firmware?
> Ofcourse, SIP there is a lot,
Hi All;
If one of my FXS port damaged at TDM22B because we
connected the Telephone Line cable to the FXS port
while it should be connected to the FXO port, then can
I order S110M FXS Module and fix it instead of the
damaged FXS? (This if we assume my problem that really
the FXS port damaged).
Rre
Hi Gordon;
They are able to receive calls? Origination is not a
problem I know, but what about receiving calls from
the Asterisk to them?
For example, how I can call the extension 200 that is
behind NAT? (Assuming that extension 200 is registered
on the Asterisk).
Regards
Bilal
---
Hi;
What is the difference between free version and
business version?
Are the differences at voice quality level or it is a
matter of add on features (luxury)?
Regards
Bilal
---
You can try Zoiper.follow the given link
http://www.zoiper.com/
--Keshav
bilal ghayyad <[EM
Hi All;
Did anyone try to use IAX IP Phone behind NAT, and let
it receive calls from Asterisk without doing port
mapping at the router existed at the site where the
IAX IP Phone existed? Is the need just to let the IAX
IP Phone that is NATed to register on the Asterisk and
at asterisk I set nat=ye
Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
Regards
Bilal
Be a better friend, newshound, and
know-it-al
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
Any advise.
Regards
Bilal
__
Hi;
How can I use SSH in that senario? Is there a link
that can help to understand what I have to install and
to configure?
Regards
Bilal
--
bilal ghayyad wrote:
> Hi;
>
> Via OpenVPN or port forwarding is known for me, but
> via SSH is new for me, how I can do it
Hi;
Via OpenVPN or port forwarding is known for me, but
via SSH is new for me, how I can do it and what is the
difference by SSH and OpenVPN?
Regards
Bilal
-
Good question. I have never tried tunneling IAX over
SSH but it seems
like
it should work just like anything
This is my configuration in the extensions.conf,
iax.conf at Site A and Site B, so anyone can help why
the call refused?
Site A:
[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.3
usename=IPLink
secret=password
canreinvite=no
nat=no
[SiteBInternal]
exten => _2XX,1,Dial(IAX2/[EMAIL PRO
Hi AK;
I would like to ask a question: where is the problem
if u record the prompted messages in ur voice and as u
need? Does not work?
Also, if that the situation: how can I determine the
needed voicemail language? For example I need ARABIC
language, so what should I do to have arabic prompts?
Hi All;
Did anyone tried to use dns name or ddns name with
host (host=abc.www.com) and it worked fine?
Regards
Bilal
Looking for last minute shopping deals?
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http://
Hi All;
I did an IP Trunk using IAX between two Asterisk
boxes, now Asterisk A can send a call for B but B
refuse it. The IAX type was configured to be "friend"
in the iax.con for Asterisk A and B, is there any
thing else need to be done to let B accept the call
from A?
Also, I used an static IP
Hi List;
With new technolgy, alot of mobiles now support Video
Call, so what is the possibility to have Asterisk
supporting Video so it support Video call at theie
Phones?
Regards
Bilal
Looking for last
Hi List;
Is there an Digium cards support GSM SIM cards so we
can fix an SIM card to be used for calls within
mobiles as it is less rate?
Or I have to use an FXS to SIM adaptor? If yes, then
anyone advise a models and prices?
Regards
Bilal
___
Hi Marc;
Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.
Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are faci
Hi Marc;
Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.
Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are faci
Hi List;
Is there any possibility to let asterisk support G723?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
http://tools.search.yahoo.com/newsearch/catego
Hi All;
Is there a method to use OR and AND operator with
GotoIf, so I can make better logical expression?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
htt
Hi All;
Is there a command that can let me execute the
Dial(.) if {CALLERIDNUM}= ..? Without using
GotoIf?
Any help?
Regards
Bilal
Be a better friend, newshound, and
know-it-all with Yahoo! Mobi
that
the appropriate UDP port (by default 4569) are
forwarded to your
Asterisk servers. Only this port is required - RTP
isn't used by IAX2.
bilal ghayyad wrote:
> Hi List;
>
> I heared that IAX is good for NATing issues, but I
do
> not know if it can help me in that senari
Hi List;
I heared that IAX is good for NATing issues, but I do
not know if it can help me in that senario:
I have two Asterisks machines in different sites and
both are behind NAT (both have private IP address), I
need to link these two asterisks with IAX trunk (if it
help really in such senario)
Hi List;
Can we use the Cubix on Asterisk with the feature of
checking who is online and to send text messages for
the others?
Regards
Bilal
Be a better friend, newshound, and
know-it-all with Yahoo! M
Thanks a lot for the help.
But if Asterisk has private IP address and the only
way to access it from remote sites is to have vpn
connection to the site that asterisk existed (the site
has vpn), then how that will happen from the Mobile to
be able to run the softphone from the mobile?
Any help?
Re
Hi Cohen;
Sorry in that question:
What is the OpenVox and that A800P01?
Regards
Bilal
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
http://tools.search.yahoo.com/newsear
Hi All;
I established h323 trunk using chan_h323 (between
asterisk and softswitch, already i did this using sip
and succeed, but now in h323 and i am facing a
problem).
The call reached to the softswitch, but it always
dropped when it send for destination, and we tried to
let softswitch to send t
Hi List;
In the h323.conf file, the parameter gatekeeper is
used to let asterisk work as h323 gatekeeper listening
at port 1719 by setting gatekeeper=DISCOVER or it is
used to let asterisk search for the gatekeeper to talk
with it and receive calls from it? But if just to let
asterisk talk with it
Hi All;
Is there any limitation on the number of users for
MeetMe Conference? In other words, how many parties
can join to the room and become a member of the room?
Is there any limitation?
Regards
Bilal
Hi All;
Does new asterisk version still requires g729 to be
bought or it required for some features and not for
others?
Also, how can I use g723 with Asterisk?
Regards
Bilal
Never miss a thing. Make Y
Hi List;
I was knowing when asterisk started, there was a
softphone that has an text messages feature, voice
calls, knowing who are online with u, look like
messanger. Where that softphone? I do not see it any
more in Asterisk.
Regards
Bilal
__
Dear Kiven;
Actually it is default and not degault. Also, I was
doing the compilation remotely via the Putty. Another
thing, I did another senario and got another thing, as
below:
I copied /usr/local/lib to /usr/lib and then I
restarted asterisk, but when I come back to run it,
then it was giving
Hi All;
Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)? I was think in AsteriskNow does
this? Any advise?
Regards
Bilal
Be
Hi All;
I am trying now to compile h323 to be able to use it,
I did the pwlib and openh323 successfully and I
exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the
OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need
to compile h323 as following:
cd /usr/src/asterisk-1.4/channels/h323
When I ty
Hi All;
For me, I am in Kuwait and using the TDM22B and I used
all the below settings and did not resolve my problem,
I do not know if there is any other settings, or if
there is a method to detect that no signaling is still
existed on the, so we can do Hanup, the below settings
used and did not r
/Main_Page).
CS
-Urspr?ngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im
Auftrag von bilal
ghayyad
Gesendet: Dienstag, 11. Dezember 2007 13:16
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] VPN Client with the IP Phone
and what its
VPNServer
Hi Oliver
Hi Oliver;
Thanks alot for your reply.
What the needed VPN server? Cisco, Juniper, Planet ?
Does it use IPSec or PPTP?
Regards
Bilal
-
Snom
2007/12/11, bilal ghayyad <[EMAIL PROTECTED]>:
>
> Hi All;
>
> Is there an IP Phones working with Asterisk that
come
&
Hi All;
Is there an IP Phones working with Asterisk that come
built in with VPN Client? And what the VPN server it
works with it fine?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http:
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is 193.111.194.111
IP Phone (A): 192.168.0.1 and its default gateway is:
195.2
Hi All;
Any one can advise for a good stable open source video
conference or video server?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
__
or more if I need to increase the voice
> volume?
>
> What if I have another TDM22B card, and I need to
> increase the voice volume at port 3 and 4 of that
> second card, what the attribute to be bypass for the
> above commands?
>
> Your kindly help is high appreciated.
&g
volume at port 3 and 4 of that
second card, what the attribute to be bypass for the
above commands?
Your kindly help is high appreciated.
Regards
Bilal
bilal ghayyad wrote:
> Hi List;
>
> Anyone knows a method (command) to increase the
voice
> volume at diguim card level?
Are you
Hi All;
It is digium analoge card (2 fxo and 2 fxs), so what
do I need to use? And where I can find a link for
that?
Also, is it possible to have a difference voice
volumes to be used each for each Trunk or each user?
Your kindly help is high appreciated.
Regards
Bilal
bilal ghayyad wrote
Hi All;
I installed one digium card of 2 fxo and 2 fxs, but
the following problems related to the voice are
happening:
1) Sometimes when I call to the PBX, I hear like modem
sound and after little it disapear.
2) There is a disturbance in the background (like the
channel radio disturbance that m
Hi List;
Anyone knows a method (command) to increase the voice
volume at diguim card level?
Regards
Bilal
Be a better pen pal.
Text or chat with friends inside Yahoo! Mail. See how.
http://overview.ma
Hi All;
I would like the needed hardware (MHz, MB, and GBI)
for the following:
1) Users: 30 IP Phones.
2) IP Trunk for maximum 10 concurrent calls, with g729
codec.
3) Analogue card of 8 lines FXO.
4) Softphone 5 and they use g729 codec.
5) Functions to be normal functions (call pickup, call
forw
Hi All;
I have an IP Trunk established between Asterisk and
the VoIP service provider, when call from my mobile to
the PBX and then enter the destination number to call
via the VoIP, I got a connection but the voice level
volume need to be increased, I am trying to find if
zaptel (diguim card) can
Dear Philipp;
If I used SVN, then if later I needed to do upgrade
using the make update and make upgrade, I will be able
to do it or it is a condition that I have to be used
CVS in the beginning?
Regards
Bilal
-
> Which is better (to have more stable or release
> version
Hi All;
Which is better (to have more stable or release
versions) of zaptel, libpri and asterisk: to use cvs
or svn?
In case of using cvs, why I need to type:
export
CVSROOT=:pserver:anoncvs:[EMAIL PROTECTED]:/usr/cvsroot
In other words: what is the use of pserver, anoncvs,
... with cvs checko
Hi List;
If I have a running Asterisk on one machine and I need
to have another Asterisk on another machine, can I
copy the files from the first running Asterisk machine
to the new machine or I have to do the ./configure +
make + make install?
If I can copy, then which directories (and files) ne
Hi List;
Is Digium the best telephony cards to be used with
Asterisk? The prices are some how high, any
suggestion?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/h
Hi All;
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
1) As SIP or H323 client, with the ability to add
button functionalities (call pickup, call transfer,
...) so if there is a wireless network, then it can
use it to connec
Hi List;
I am using Firefly, and when I am calling from my
mobile to the Zaptel (via pstn analoge lines), and
select the firefly extension, it starts ring at the
firefly, if I decided to hangup the mobile before
answering the call at the firefly, then it stays
ringing till being transferred for th
, correct?
About configuration files, they are only in /etc/ and
/etc/asterisk?
Regards
Bilal
If you want to be completely safe, copy everything
from
/var/lib/asterisk,
/var/spool/asterisk, and /var/log/asterisk as well.
On Fri, 16 Nov 2007, bilal ghayyad wrote:
> Hi List;
>
> I need to d
Hi List;
I need to do upgrade for Asterisk and Zaptel, so which
directories or files need to be copied to keep my
configuration? Is it only the /etc directory or there
is other directories?
Regards
Bilal
Hi List;
I have Asterisk installed on FC5 (Fedora Core 5), and
I am going to install FC 7 and that means I will
download new latest Asterisk and Zaptel versions. I do
not need to repeat writing the configuration again, so
what I need to do now is only copying the
/etc/asterisk/ files, and the zapa
Hi All;
As I understood that h323 module in asterisk does not
support the ability to let the h323 endpoints register
at asterisk (this registeration happens at 1719 port),
so how asterisk will be able to route the call for the
destination IP Phone if it is not registered (so the
IP is unknown)?
I
Hi Friends;
Actually I would appreciate if Vivek can advise if the
VPN resolved the RTP packets in the SIP Trunk between
Asterisk and another softswitch? In other words,
openvpn helpful in NAT cases in what exactly? As
without VPN, I was able to establish a call but
without voice or with complete
is
fairly simple to
use.
- Original Message -
From: "bilal ghayyad" <[EMAIL PROTECTED]>
To:
Sent: Saturday, November 03, 2007 10:43 PM
Subject: [asterisk-users] Asterisk versions and H323
> Hi List;
>
> Is there an Asterisk version that contains H323
>
Hi All;
nat=yes for example, it effects on the success of the
registeration.
What are the parameters that might let the
registeration fail when I need to register Asterisk on
a softswitch using register => ?
Any help?
Regards
Bilal
__
Do You Yaho
Hi List;
Is there an Asterisk version that contains H323
module, or still I have to download the h323 alone and
compile it?
Regards
Bilal
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
t: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is
busy/congested: IP Trunk
> No:
>
> register => abc:[EMAIL PROTECTED]
>
> [peer]
> host=zzz
>
> Its possible to make mistakes and typos you know.
Maybe you can post
> your config file
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Regards
Bilal
> Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route t
Dear All;
The codec problem resolved, I used the correct codec
should be codec_g729a_v32_pentium4m.tar.gz as the
output of the command cat /proc/cpuinfo giving:
Pentium(R) 4 CPU and my confusion was that I cared for
the output of the command core show version which
gives:
Asterisk SVN-branch-1.
Hi List;
I established an SIP IP Trunk between Asterisk and
another softswitch (asterisk registered on the
softswitch successfully) and I saw this on the
softswitch.
>From firefly softphone, I was need to do a call to be
via this softswitch (ofcourse, the softphone will send
for asterisk and aste
>
> 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 ghayyad a écrit :
&g
11? I beleive it should process the call with fail
(codec miss match), but I do not see the call.
Looking to hear from you.
Regards
Bilal
Bilal,
On Tue, 23 Oct 2007, bilal ghayyad wrote:
> This is if I need to let Asterisk register with
another softsw
rg/wiki-Asterisk+config+sip.conf
On Fri, 19 Oct 2007, bilal ghayyad wrote:
> Hi All;
>
> Alot of softswitches or PBX's does not accept to
> manipulate any SIP call without being registered
> firstly. So that means, I need asterisk to register
> firstly then I can route my calls
Dear Marc;
I readed your email about the codec G729a and I am now
also need to install the codec on my Asterisk.
I typed from Asterisk CLI:
core show version and I got the following:
Asterisk SVN-branch-1.4-r72556 built by root @
localhost.localdomain on a i686 running Linux on
2007-06-30 13:0
Hi All;
If I installed H.323 on asterisk, and the caller phone
was SIP endpoint while I need to route the call for a
destination via an H.323 trunk, so Asterisk will do
that SIP to H.323 translation automatically or I have
to do also a configuration to SIP to H.323
translation?
Regards
Bilal
___
Hi All;
Alot of softswitches or PBX's does not accept to
manipulate any SIP call without being registered
firstly. So that means, I need asterisk to register
firstly then I can route my calls to that SIP trunk.
In IAX2, we use the register => , so what shall we do
in Asterisk? And how its format
Hi List;
My Asterisk version is 1.4 and I am trying to use the
ResponseTimeOut() application to control the response
time of the Background function, but when the running
arrive for the ResponseTimeOut() then the call drop
and my debuging says:
No Application 'ResponseTimeout' for extension
(Test
Hi List;
I need to do IP Trunk between Asterisk and another
softswitch provider, the softswitch support SIP but
requires Asterisk to register for this IP Trunk (it
should appears as gatekeeper entity that does
registeration to another gatekeeper entity).
How can I configure this SIP trunk to do r
So still without the .conf files, the GUI will not be
enough to do the work.
And the .conf files come with GUI are all the .conf
files that existed in the CLI commands or it is a
matter of some files that usually will be used?
Regards
Bilal
bilal ghayyad wrote:
> Can I download and inst
Hi List;
Can I download and install AsteriskNow or AsteriskGUI
free or there are a kind of charges calculated based
on somethings?
Actually, I did not start with GUI but I started with
.conf file, but till now I do not know at what kind of
configuration level I have to go for .conf files as
GUI w
Hi ALL;
Any one knows a websites that has really a members
that use DUDNI wouldwide and ready to do route
exchanges?
I tried www.dundi.com but it look like still not
working, as most of its pages are not accessible
except the home page :) -
Regards
Bilal
___
Hi List;
Can someone advise me why in the below context, it
does not run the Background step? Once I dial 1000,
then it hangup and give congestion signal? If I
comment the ResponseTimeOut, then it run the
Background but it does not wait till caller enter the
digits, once the sound file finish, the
Hi List;
When I readed about DUNDI, I though there is a website
and we can buy and sell routes there using DUNDI, but
when I tried to browse http://www.dundi.com) then I
found a strange page that contains only this text "It
Works"!
So, what is the market of DUNDI really? Is there a
method to buy
Hi List;
Any one can advise me to a good link to see and buy
Polycom IP Phones?
Also, if I need support (in case the Phone was damaged
and need to replace, so the warantee), so which web
can provide that? I do not need to buy from one and he
is not responsible for support.
Regards
Bilal
Hi List;
I am trying to find a link to see the polycom IP
Phones that work with Asterisk, but not able to find
until now.
I checked this link, but did not find any thing
related to Polycom IP Phones:
http://www.voip-info.org/wiki/view/Asterisk+phones
So any advise where I can find a link to see
Hi List;
>From where I can buy the G.729 and G.723 licenses, and
how I can install it on Asterisk so I can use it?
Anyhelp?
Regards
Bilal
Don't let your dream ride pass you by. Make it a reality with Ya
Hi List;
How to let context3 does not use context1 in the below
senario:
[context1]
.
.
[context2]
include => context1
..
..
[context3]
include => context2
.
.
Regards
Bilal
___
Dear Cohen;
Your help was great, now it is loading.
But as a favourite, could u please explain for me what
was you mean by the following:
FATAL: Error running install command for wctdm
This is caused by the silly post-install command for
wctdm in
/etc/modprobe.d/zaptel or /etc/modprobe.conf ,
Hi list;
I need to run the command modprobe wctdm and whenever
I write it, then it gives me the following message:
FATAL: Module wctdm not found
FATAL: Error running install command for wctdm
So, do I have to run that command from specific path?
Or what is the problem?
Any help?
Regards
Bilal
tone by
placing a , in the
> phone's dialplan. SIP phones have their own
internal dialplan that
is
> not part of Asterisk's dialplan. You would have to
check the docs
for
> your phone. Not all SIP phones can continue
dialtone.
>
> bilal ghayyad wrote:
>
>>I n
EN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group
4...
bilal ghayyad wrote:
> I need to select a line from the Zap group channel
> using the SIP Phone (not FXO and not FXS ports).
>
> ignorepat does not work?
>
> Also, what is the method to let the second dial
repat help this guy?
>>
>> Al lists wrote:
>>> ignorpat is your friend
>>>
>>> On 9/30/07, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>>>> On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
>>>>> Dear List;
>>
lp this guy?
>
> Al lists wrote:
> > ignorpat is your friend
> >
> > On 9/30/07, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> >> On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
> >>> Dear List;
> >>>
> >>> How c
Dear List;
How can I place a call via Zap/g1 (group) but need to
determine the line (FXO port)
that will go via it?
Also, how it will be possible to assign an dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be
Hi List;
In the outbound, I read in the documents the Wildcard
match "by using the . (period)", but I did not
understand how Wildcard will work (like what)? As I
know that Wildcard is a term used with the Diguim TDM
card (FXO and FXS), so what is the relation between
such cards and the matching in
Hi list;
While I am writing my configuration on the .conf
files, I would like to know if I wrote the command in
write syntax (form), there is not any way to check if
I am writing correct or not (other than checking my
documentation)?
Also, is there any method for searching on specific
topic about
Hi Cohen;
And do I need to run #modprobe wcfxs / #modprobe wcfxs
or I have to run #modprobe wctdm? What is the
difference?
Regards
Bilal
> Hi List;
>
> If I am configuring Diguim Analoge card, then I need
> to run #modprobe wctdm, but the question why I need
to
> run #modprobe zaptel also?
N
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
Any help?
Regards
Bilal
_
Dear Cohen;
But as I remembered, they told you something about the
syslog, I did not understand it, could u please
explain it for me and it relation with our problem?
Regards
Bilal
> Dear Benjamin;
>
> OK friend, things are clear. But now I came to the
> same original issue that you asked about
Dear Benjamin;
OK friend, things are clear. But now I came to the
same original issue that you asked about it, which is
the ability to stop the log/debug messages into
/var/log/messages.
Same like your situation, the messages is comment (;)
and even the logges are written to the
/var/log/messages
ok into /etc/asterisk/manager.conf for the
> required
> directories where Asterisk stores its various
> files/directories.
> Then read up logger.conf and look at some examples
> on the net as well.
>
> cheerz
> - Ben.
>
>
> bilal ghayyad wrote:
>
> >Hi Benja
Hi Benjamin;
I am also interested in the same issue, but I would
like to know how you can know where these logs are
stored (in which file and path)?
I readed that syslog, can you please help me about
that?
Regards
Bilal Ghayad
Mobile: 00965 9849460
---
>When you access the A*k console,
different codecs, capture DTMF or
different
protocols it will stay in the path.
On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi List;
>
> If I need traffic to be directly between the
> endpoints, then I have to set the canreinvite = yes?
>
> If I did not configure the ca
not operate
>properly if set to yes when they are in fact local.
>
>On 9/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
>
>>Hi List;
>>
>>If I set nat=yes, then asterisk will send the
packets
>>to the public IP address or to the private IP
address
>&
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