Displaying 6 results from an estimated 6 matches for "komarek".
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komanek
2006 Mar 27
2
registration with different username
Hello,
I am trying to register to the asterisk with different phone number,
login and password. This is my setting in the sip.conf:
[246079011]
type=friend
context=cisco
secret=XXX
host=dynamic
username=tomas
allow=alaw
nat=yes
canreinvite=no
mailbox=246079011
but I get this reply:
Mar 27 13:17:00 NOTICE[5144]: chan_sip.c:10889 handle_request_register:
Registration from
2006 Jan 11
1
a2blling billing system
Hello,
I am trying to setup a2billing system for asterisk. I have installed it
corectly, but I have not found any users manual. I do not understand the
whole structure. How do the parts like calling cards and sip friends
cooperate together?
I simply need to know how to make a call through it. With all the
features like CDR's, etc.
Can anybody help me with this?
Thanks in advance.
2006 Oct 10
1
read.table versus read.csv (PR#9284)
...rs on R 2.4.0 running on Linux Debian system (see =20
below full information concerning the platform). However, according to =20
my information, exactly the same problems happens with R running on =20
Windows XP Professional, Windows XP Home Edition and Linux Mandrake =20
10.1.
Kind regards
Arnost Komarek
--=20
Arnost Komarek
Biostatistisch Centrum K.U. Leuven
U.Z. St. Rafael
Kapucijnenvoer 35
B-3000 Leuven
Belgium
Tel : +32 - 16 - 33 68 86
Fax: +32 - 16 - 33 70 15
Web: http://www.med.kuleuven.be/biostat/
--please do not edit the information below--
Version:
platform =3D i686-pc-linux-gnu
ar...
2005 Aug 11
1
Ignoring the called number in the INVITE message
Hello,
I've got such a problem. I'm configuring Asterisk as a backup server, if
call to the first one fails.
My problem is, that the redirection from the sending machine work so,
that in the INVITE line of the INVITE message is the presentation number
of the Asterisk server and in the To line is the real called number.
So I need to setup Asterisk so, that it will ignore the number in
2005 Aug 17
1
SIP message 183 and in band info
Hello, I have such a problem. I have an * configured as a peer connected
to the gateway to PSTN.
While calling to the switched off cell phone, the gateway sends to the *
the SIP message 180 with the SDP part, and also a lot of rtp packets
containing the operator's in band info.
But * forwards the 180 to the UAC without the sdp part and also without
the rtp stream.
Is there any way, how
2005 Aug 17
0
chan_sip2.c compiling
Hello, I've tried to compile the new sip channel, sip_chan2.c but I am
not succesfull. When I make * I get error messages, some of them also
considering syntax error in the code.
Does anyone use this channel? Wuld you please give me some advices how
to compile it?
Or do you have the source code that works???
Thanks for answers.
Tomas