) -- I'm not sure parentheses are
allowed.
yonoko molomo wrote:
Now I update the extensions.conf file accordingly.
exten = clientA_Number,1,Dial(sip/$(exten),10)
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Hi,
I have some problems and doubts connecting two asterisk servers.
I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with another client (clientB).
Now I want to call from client A to B and from B to A.
Searching in google i find many
examples, but i do not know what is it.
any ideas?
2007/9/27, Gordon Henderson [EMAIL PROTECTED]:
On Thu, 27 Sep 2007, Anthony Messina wrote:
On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
Hi,
I have some problems and doubts connecting two asterisk servers.
I have one
Hi,
i bought this device and the cost of the 7040G itself was similar to
the license. if im not wrong, the telephone cost around 80€. the sip
license was around 80€ as well
however, i am quite annoyed because the phone did not come with sip,
but callmanager so i cant use it as i planned.
i have
asterisk, probably nagios.
thanks
Tim.
- Original Message -
From: yonoko molomo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 4:51:11 PM (GMT) Europe/London
Subject: [asterisk-users] res_snmp
Hi,
I have problems compiling asterisk 1.4.11
Hi,
I have problems compiling asterisk 1.4.11 with res_snmp.
I do 'make menuselect', and I see that this resource module depends on netsnmp.
I am using centOS 4.5.
I do:
yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs
I don't know if i am missing something.
I go to the source
Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put some wait (2) functions after the answer( ) or playback( )
functions, it think that asterisk takes some
hi,
how to add radius support to asterisk 1.4.5?
i do make menuselect and i do not see any module or option related to
radius, pam, authenticacion or similar.
any ideas?
thanks
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Hi,
http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html
Thanks, I have already seen that document before but it did not help
much to have a better understanding to set up radius with asterisk.
In 4.3 it is written: Asterisk has been patched along
Hi,
I have a question which might be a bit simple for advanced users.
Where is the correct config file to define users in asterisk 1.4?
I was using sip.conf file and it was working fine.
Recently I installed an asterisk web interface and I see that new
users created using this web interface are
problem using Voicemail. I should hear a message asking for
my mailbox number, but normally I do not hear anything.
Do you think the patch I will fix the problems?
I will try later, thanks
2007/7/19, Russell Bryant [EMAIL PROTECTED]:
yonoko molomo wrote:
[Jul 17 11:19:22] WARNING[23645]: src
.
- Original Message -
From: yonoko molomo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, July 16, 2007 6:13 PM
Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323
Hello all,
I have seen some people asking how to configure asterisk to work with
h323 but i
hi,
i fixed the problem.
as i thought it was a configuration problem, i was not defining the
asterisk users at ooh323.conf.
now it seems to work,
thanks
2007/7/17, Dovid B [EMAIL PROTECTED]:
What output do you get from the CLI ?
- Original Message -
From: yonoko molomo [EMAIL
Hi,
I am using asterisk 1.4.
I have confgured the musiconhold.conf file.
However, when i make a call and then hold the call it does nothing.
in the CLI i do not see the starting/stopping musiconhold messages.
i am making calls from sip to h323 using asterisk assip/h323 gateway
(with gnugk and
Hello all,
I have seen some people asking how to configure asterisk to work with
h323 but i did not manage to do fix it yet (i am not an asterisk
expert).
Can someone help me configuring asterisk?
It is already compiled asterisk 1.4.5 with H323 support.
Everything looks fine.
Then i understand i
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