Dear All,
I am not sure if this is the right place to ask my question but I can't
find a newsgroup or support for this app_RPT concept so I hope if some
one in this community who have tried it out could help me out.
I studied this application requirments and saw the hardware needed they
describe
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Dear Khaled,
What is the softphone u r using?
Thx
MAG
Khaled wrote:
I am using firmware version pos3-07-500
Kindly can you provide me with the basic configuration for cisco ip
phone and asterisk config file
*I have nat=never at my asterisk config file and nat enabled N0 at
cisco phone
to call the softphone you just need to focus on
the debug and check the configuration.
Thx
MAG
Khaled wrote:
Softphone Eyebeam v 1.5.2
---
From:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed
A. Gombolaty
Dear Mike,
I had wanted to do something that is similar to your need as I wanted to be
able to add one active channel in multiple groups, it worked with The Ramon's
example in the link below which uses categories beside the set command, note
there are two examles depending on the asterisk version
Dear Khaled,
The way I would go to do so is to put the group of people you want to
call each other in one context and the other people in an another
context. That's one way to do so.
Thx
MAG
Khaled wrote:
Dears
Please how can create an independent group of users on asterisk ,in
which user
Dear All,
I am currently very stumped on the subject of Active Directory listing,
as I am unable to find any documents regarding this feature thus I am unable
to configure it or know how to use it. Does anyone have any useful info
or documents regarding this feature in terms of how to or guides
Dear All,
We need to do the following crazy scenario which is really stupid but
wanted :-((, I need to make the sip server initiate a call on zap
channels and once the phone answers, it should play an IVR and according
to the choice of the called he will be moved to other extensions, we plan
to
Dear All,
I am have experimented asterisk long before any gui was available and
also currently working with trixbox, ofcourse working with asterisk directly
makes you more aware but when you start deploying the system you will face
management issues for asterisk, as anyone who deals with
Dear All,
I am trying to find a way to stop people who use phones after
business hours (a policy the company wants to implement), we have cisco
7940 and 7910 phones and sadly they don't have a phone lock password system
(on these ciscos it locks config menu changes but not the calls but the
call restricted"
priority 11: hangup
The next move in your text adventure might be "Show Application
GotoIfTime" from the CLI :)
Moj
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I am trying to find a way to stop people who use phones after
business
> hours (a policy the compa
Dear Lacy
Thx Lacy for this important reminder we engineers do tend sometimes
to forget about all the law part, indeed while I was putting down the implementation
we do have exceptions we have a 24x7 call center and ofcourse the emergency
number.
Thx
MAG
Lacy Moore - Aspendora wrote:
So
I was
Dear Rich,
It seems that my question is very general I apologize for that, but
I am glad to see others like yourself pointing me in different directions,
it seems all around the world we have problems with the cleaning folks.
What I have in mind is to make the phone user lock his phone when he
le
for emergency calls without this feature so you can only call the police,
ambulance, power and fire departments and internal extensions but not the
costly outside calls
Thx
MAG
Benjamin Jacob wrote:
Mohamed A. Gombolaty wrote:
> Dear Rich,
>
> It seems that my question is very genera
Hi All,
I am facing a problem with makeing asterisk work realtime with mysql,
after following the tiki steps which are:
uncommented the lines sipuser and sippeers from extconfig.conf
copied the res_mysql.conf and configured it with the right parameters
checked that mysql is working
added the
Dear Matt,
Yes indeed I did I have used cvs to download asterisk and it's addon
from CVS.
Thx
MAG
Matthew Boehm wrote:
Did you install res_config_mysql.so from asterisk-addons?
-Matthew
> From: "Mohamed A. Gombolaty" [EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing Li
Dear All,
I was trying to use Realtime Asterisk option for Sip users and peers
+ Heartbeat + Mysql Replication in order to make a failover system, so
that if Ast1 went down for any reason, Ast2 server will have the same data
that Ast1 used in the Mysql database and don't need to make the phones
Dear All,
I was trying to use Realtime Asterisk option for Sip users and peers +
Heartbeat + Mysql Replication in order to make a failover system, so
that if Ast1 went down for any reason, Ast2 server will have the same
data that Ast1 used in the Mysql database and don't need to make the
phones
Hi Wasim,
Check out the x-lite softphone
http://www.xten.com/
As for linux check this page there are two softphones type available
:
http://www.iptel.org/products
Thx
MAG
wassim Darwish wrote:
ok what softphone i should use to fit windows and
linux supporting
iax,thanks in advance.
Dear All,
I was trying to load balance between two asterisk servers using vovida.org
loadbalancer, but when I was running it i faced the following problems:
-When phones try to register the lpproxy gives the following message for reach
phone trying to connect:
Sticky header data is: Call-ID:
Dear All,
I am currently working on asterisk cvs-head version in order to use
realtime with mysql, 2 asterisk servers with duplicate mysql databases,
one asterisk server is serving the sip phones and the data is logged to
the database and replicated to the other asterisk database, when the first
Hi Angus,
I don't believe it can be the root password of mysql, I used to install
the addons without even haved installed mysql server yet, I guess we need
to know which platform are you working on and which version you are trying
to install.
Thx
MAG
Angus Comber wrote:
Hello
I have
Dear Kib,
As I believe the Realtime options concerning the mysql database can
only be used with the Asterisk CVS-HEADversion it's still not implemented
on Asterisk v 1.0.* .
Thx
MAG
Kib Eki wrote:
Hi,
I configured cdr_mysql (addons 1.0.9) to write the cdr records to the
mysql db.
The problem is
Dear All,
I am totally new in this arena and I am still waiting for my installation
process on freebsd to finish, but I wanted to make sure of the following:
- Call routing between IP telephones can be done regardless of who made
the phones?
- Asterisk does support SIP V2?
- it does act as SIP
Dear All,
I have installed Asterisk everything is OK until I tried to configure
meeting room, configuration was simple enough when I try I get a message
that it's not a valid meeting room, Now I don't have a Zaptel device on
my machine, so I found that you will have to use ztdummy to make
a
Dear Ghassan,
I never used fedora but in the link below you will find a step by step
installation for fedora platform check it out and see if you are missing
anything.
http://www.voip-info.org/wiki-Asterisk+Linux+Fedora
Thx
MAG
Ghassan Lama wrote:
Hi;
It is
my first time installing an
Hi Chris,
Did you try the echo test, this will help us to better test the latency
between the two distance phones, the link below should guide you through
the echo cmd.
http://www.voip-info.org/tiki-index.php?page=Asterisk cmd Echo
Thx
MAG
Giles Coochey wrote:
>
> Has anyone seen a situation
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten =>
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten =>
Dear Peter,
here is my 777 conf in extensions.conf:
[Internal-sip]
exten = 777,1,Dial(SIP/777,7,tr)
exten = 777,2,Dial(SIP/777SIP/888,10,tr)
exten = 777,3,voicemail,u777
exten = 777,104,voicemail,b777
As for the stdexten macro I really don't know what you mean by using it do you
mean
by doing
Hi Peter,
You are totally right it worked, and I really loved the macro idea I
have mostly grasped it now and will use it more extensivley in the future.
Thx
MAG
Peter Bowyer wrote:
On 02/06/05, Mohamed A. Gombolaty [EMAIL PROTECTED]>
wrote:
> here is my 777 conf in extensions.conf:
>
Dear All,
I was trying to make call confrence available but all the asterisk documents
use the meeting room concept, where those who wanna meet have to dial an
extension corresponding to the meeting room, while call conference actually
means that I am on exten 100 I can dial exten 200 and add it
Dear All,
I was trying to configure Asterisk to work with Cisco Softphone version
1.3(4a) and I am having a problem, the Softphone when is started asks
for a Line to use, all documents I found specify this is something to be
done from t Cisco Call Manager, has any one worked on this before?
--
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
Hi Shahan,
yes both are in the same LAN
Thx
MAG
Shahan Kalutanthri wrote:
HI..!!
Is you windows PC the Asterisk in the same LAN.
-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject
Hi Wilson,
yes I am leaving it blank although I did try to use a username in the
sip.conf but with the same result also I have tried to put the extension
881 but the same result.
Wilson Pickett wrote:
> Enabled: yes
> Display Name:
> Username:
> Authorization User:
> Password:
>
Dear All,
I am trying to make my sip phones register with SER and make use of
Asterisk capabilities such as voicemail and parking calls for example.
on SER side
the ip of the server is 192.168.99.170 and uses port 5060
in my ser.cfg I added the following lines :
if (uri=~"sip:[EMAIL
Dear All,
I am trying to make the phones always talk to each other (peer to peer)
using SER as a sip proxy, and incase the call is not answered we will
use the voicemail of asterisk and other feautures, I have done that
already, but in order to do so I found that I have to make the users
dial
On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am trying to make the phones always talk to each other (peer to peer)
using SER as a sip proxy, and incase the call is not answered we will
use the voicemail of asterisk and other feautures, I have done
the two IP Phones without having asterisk
in the middle which will save bandwidth on the wan link.
As for SER when you perform it after this step it shoild work fine with you.
Thx
MAG
Mohamed A. Gombolaty wrote:
Dear Yair,
Actually what happens is that from SER debug I can see the call is looping
Hi Erdem,
Can you try to put another dial command that points to the trunk afetr
the dial command to the SIP?
fro example:
exten => XXX,1, dial(sip/,20,r)
exten => XXX,2,dial(zap/) ->
note here that I am not sure if the order number should be 2 or 102 but
if this didn't work
Dear All,
I was searching voip-info for Failover and load balancing for
Asterisk, my goal here is to have a system where the SIP traffic is being
divided on five central servers with Asterisk on, and if an asterisk server
fails another asterisk server will assume it's place , from my readings
I
Dear All,
I am using Linux-High Availability between two Asterisk servers, everything is
fine but I do have one problem with this, When a server fails and the other
assumes the ip address and start asterisk on server 2, the ip phone must
re-register themselves again, otherwise the phones won't
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with this, When a server fails
and the other assumes the ip address and start asterisk on server
2, the ip phone must re-register themselves again, otherwise the phones
are
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with
this, When a server fails and the other assumes the ip address
and start asterisk on server 2, the ip phone must
re-register themselves again, otherwise the phones are
Dear All,
I read your notes and was very glad, it was a healthy and useful debate,
I have set my mind on implementing Realtime for sipusers and peers with
mysql database and either use the Mysql replication process or mount the
database on both servers.
I will write a document of this trial and
Dear Eric,
You are totally right, I already know the information below
but I don't know why I couldn't see them, I certainly need a vacation,
anyway it worked like charm.
Thx
MAG
Eric \"ManxPower\" Wieling wrote:
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I have this ques
Dear All,
I was trying to limit the number of calls between different located sites in
order to avoid congestion of the bandwidth, but as I found from the mails and
testing that it is easy to do it for the incoming calls by the setgroup() and
group_count while it is the outgoing is hard to track
the
correct count of users dialling out.
As u said I am using CVS-Head and used the group_count() with gotoif
statements so I am clear of the checkgroup() bug.
Thx
MAG
trixter aka Bret McDanel wrote:
On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty
wrote:
> Dear All,
>
> I was trying
Dear All,
I am facing a problem in compiling the add-ons for the mysql, though the files
are downloaded correctly and checked and I tried different mirrors even the cvs
but yet I get those errors :
app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory
cdr_addon_mysql.c:38:19: mysql.h:
Hi all,
In case you have a number of trunks there is a software named
astbill (www.astbill.com) in which you can configure the trunks and decide
their costs and it will automatically choose the most suitable trunk.
Thx
MAG
Pikoro wrote:
By "trunk" I mean each trunk is a different account
on
Dear All,
I have this question regarding goto command, I amusing Asterisk cvs
head version, and I am trying to put a goto statement to send the user
to another extension that contains the extension he is dialing here
is how I am doing it :
exten => 2x.,1,setgroup(outgoing)
exten =>
Dear All,
I am having a problem in a scenario I am doing, I have two branches,
every branch has has an [EMAIL PROTECTED] that deals with each branch locally
and a trunk connected to a central asterisk, now if any branch wants to
call another branch it goes from the local asterisk@ home --> to
Dear All,
I have bought a digium TE205p in order to move our E1 pri from a siemens
pbx to an asterisk server platform, I have already gathered the data needed
to configure the card but I am troubled by one thing that seems unclear
on all the documents I read.
The E1 is currently inserted in a
Dear Steve,
Yes I did mean a csu/dsu I will try your suggestion and update the results.
Thx
MAG
Steve Totaro wrote:
Mohamed A. Gombolaty wrote:
Dear All,
I have bought a digium TE205p in order to move our E1 pri from a
siemens pbx to an asterisk server platform, I have already
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well, but when a call comes I can see
the call along with the caller ID but then I get this strange message which
make the call hungup:
error msg: 'zap-in' from '0109687348' does not
Dear Steve,
The line has worked like charm, but now I am facing a new problem with
recieving the call, I have sent another mail with this issue.
Thank you very much for your support
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Steve,
Yes I did mean a csu/dsu I will try your suggestion
Anthony Rodgers wrote:
This looks like a dialplan problem - do you have
a match for
0109687348 in the zap-in context in your dialplan?
A.
On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote:
> Dear All,
> I have a strange problem in recieving calls on the pri the
zaptel
> is green and e
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file extensions.conf but not the
correct group I configured so I changed it and will test again.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
The
Dear All,
I just wanted to comment on this point of the discussion:
> In a PRODUCTION environment, you can't be running a sip debug to your
> console.
In a PRODUCTION environment you have all of these issues
worked out in your
test lab before deploying to production.
I do agree with Douglas
Dear All,
After doing the test everything went fine, Thanks Anthony for putting
me on the right direction.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file exten
Dear Storm,
I have two guesses
One could be something in the ubuntu make which makes it unable to understand
some regx in the scripts used or
I am not quite sure but check the kernel version you are having (i
do that by uname -a ) I believe you will find something there, if it is
not the same as
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox
and I am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
Zapata-auto.conf:
callerid=asreceived
Dear All,
The resolution to the problem below was very easy and I guess that what
made it very hard:
callerid=asreceived
signalling=pri_cpe
switchtype=> euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear All,
I have an
Dear All,
I have this problem which is preventing me from switching to
voip system andstill working on that old siemens pbx, we have fax machines
that we attached to ATA called planet and when we try to send a fax locally
between the fax machines it works great but when we try to get a fax
Dear All,
I am facing a strange problem that I can't find any matches for while
googling, sometimes while a call initiated from asterisk to the PSTN is
answered and the answering party say the receiptionist tries to transfer
the call to someone else, the call dies, the full log shows nothing
66 matches
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