Dear all,
Thank you very much for this chance. Thank you for
your time to read this message.
All, I have read document about tls from openser.org
In that document, I read that to create CA root and
client certifacate, we just must run script
./gen_rootCA.sh and ./gen_usercert.sh where is in
tls/tools. It will produce rootCA directory and user
directory.
Then, I put the rootCA directory and user directory
into my public webserver in order the client can
download it.
But, when I try to download the certificate from the
server (using Redhat 9.0), to the client computer
(using Windows XP), why I can not download it?
Do, I have to convert the extension form .pem to .crt?
How can I convert it?
Please help..Please..
Regards,
Ferianto
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Hello guys,
I want to install SER on a SERVER so please suggest me some servers
(prefererably with configuration like HP, or some thing)
because basically i dont have any knowledge on server types .
so, any comments and suggestions are appreciated .
Thank You.
Regards,
Ravi.
Hello,
maybe someone can help me with the routing.
here the schema.
Firewall with 3 Network interfaces 2x interfaces intern 1x interface
extern
the default provider is sipgate.de.
ser shoult route all registration default to sipgate and listen on 2
intern interfaces.
ser shoult handel all intern calls and route direct intern.
user auth i'dont need or radius i'dont need.
all inbount calls from extern shoult route automatic to right sip-
phone.
current i use the default configuration with 2 options more.
listen 212.21.68.62
listen 212.21.69.110
maybe someone can help me to with the routing
thanks a lot for your time, if you can help me.
regards
mario roeber
Hi Everybody,
I was wondering if I would be able to display the CallerID in the display of
the callee's phone rather than it displaying anonymous, maybe it can be done
if I used something like this from the tm module:
if(search("From:[0-9]{10}@.*"){
append_hf("remote-party-id:<sip:2134445555 at mydomain.com>")
)
or would i use openser auth module with append_rpid_hf()
in some form or fashion? If so, how can I do this?
Just need a little input, guidance and assistance and to know if it actually
can be done. If not this method, then what can I use to display the callID
of the calling party to the caller display. Some providers are known to
block the call if a certain amount of digits are not displayed.
Thanks in advance.
Tracy
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thanks
when i install mysql and i type this command
./configure --prefix=/usr/local/mysql --enable-large-files --enable-shared=yes
it show me some checking and show me this error
checking "LinuxThreads"... "Not found"
configure: error: This is a linux system and Linuxthreads was not
found. On linux Linuxthreads should be used. Please install Linuxthreads
(or a new glibc) and try again. See the Installation chapter in the
Reference Manual for more information.
and i am not able to install mysql
sip <sip(a)arcdiv.com> wrote: Why not? That's a little vague.
Can't add them because you don't have it? Can't add them because of permissions problems? Can't add them because the directory doesn't exist? Can't add them because you don't know the unix copy command?
A little more specific information goes a long way.
N.
On Fri, 29 Sep 2006 07:30:18 -0700 (PDT), vijay tiwari wrote
>
> hi all
>
>
> i am not able to add the mysql in that locations /usr/local/lib/ser/modules
>
>
> thanks
> vijay
>
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I wonder if anyone on this list is open to a thought exercise?
I'm trying to put together a cogent message about this,
but in the meantime:
I've got a situation where my openser box seems to route stuff to itself
and it loops around and around until I get the classic "Too Many Hops" error.
This seems to happen most often on BYE, although I've seen an ACK do it too.
Here is a small ngrep snippet, after
PHONE has called OPENSER-ONE who statelessly relays to OPENSER-TWO
who t_relays to PSTNGW, and then the pstn user hangs-up:
U 23:03:53.254382 PSTNGW:5060 -> OPENSER-TWO:5060
BYE sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone SIP/2.0
Call-ID: callid@PHONE
From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674
To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag
Content-Length: 0
CSeq: 1 BYE
Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch
Route: <sip:JoeUser@OPENSER-ONE:5060>
Max-Forwards: 70
#
U 23:03:53.255013 OPENSER-TWO:5060 -> OPENSER-ONE:5060
BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0
Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674>
Call-ID: callid@PHONE
From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674
To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag
Content-Length: 0
CSeq: 1 BYE
Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0
Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch
Max-Forwards: 69
Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone>
Right here OPENSER-ONE freaks out, and the delay causes
OPENSER-TWO to re-transmit:
#
U 23:03:53.611227 OPENSER-TWO:5060 -> OPENSER-ONE:5060
BYE sip:JoeUser@OPENSER-ONE:5060 SIP/2.0
Record-Route: <sip:JoeUser@OPENSER-TWO;lr=on;ftag=26674>
Call-ID: callid@PHONE
From: <sip:18005551212@OPENSER-TWO.example.com:5060>;tag=26674
To: "Joe User" <sip:JoeUser@OPENSER-ONE.example.com>;tag=thetag
Content-Length: 0
CSeq: 1 BYE
Via: SIP/2.0/UDP OPENSER-TWO;branch=otherbranch.0
Via: SIP/2.0/UDP PSTNGW:5060;branch=thebranch
Max-Forwards: 69
Route: <sip:JoeUser@OPENSER-ONE.example.com:5060;user=phone>
OK, so who on this list is good enough to be able to suggest plausible
theories about what my openser.cfg flaw is? If you don't want to
speculate openly on the list then please send me a direct email.
Thanks,
-mark
Hi all,
I'm facing a cenario where an INVITE transaction goes through one
(Open)SER more than one time, and the ACK to a 480 Temporarily
Unavailable is wrongly being matched as an e2e ACK.
I dug through tm code and found this comment in t_lookup.c:
/* here we do an exercise which will be removed from future code
* versions: we try to match end-2-end ACKs if they
appear at our
* server. This allows some applications bound to TM
via callbacks
* to correlate the e2e ACKs with transaction context, e.g., for
* purpose of accounting. We think it is a bad place here, among
* other things because it is not reliable. If a
transaction loops
* via SER the ACK can't be matched to proper INVITE transaction
* (it is a separate transactino with its own branch ID) and it
* matches all transaction instances in the loop dialog-wise.
* Eventually, regardless to which transaction in the loop the
* ACK belongs, only the first one will match.
*/
Does anyone have a solution to this problem?
Thanks,
JF
> Hi,
> I want to configure SER as a proxy server. I would like SER to
> proxy presence messages and I have done that with the help of
> record_route and t_relay_to_udp.
> I would like to have SER forwarding the register message to another
> server to authenticate and after successful authentication SER store
> the location to route the requests. I observed SER is not allowing URI
> getting registered.
>
> I tried the following but did not work..
>
> if(!uri==myself) {
>
> if( method=="REGISTER") {
> record_route();
> save("location");
> t_relay_to_udp("132.146.109.243","5060");
> break;
> };
> };
>
>
> Can anyone let me know how can I enable other domain URI passing
> through SER to reach other server which authenticates them?
>
>
> Thanks and Regards,
> Senthil Kumar,
>
hi all
i am not able to add the mysql in that locations /usr/local/lib/ser/modules
thanks
vijay
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Can anybody explain me this, please.
How is the rtp stream with video? Which ports are usually used for that?
Following thoughts I have how it could be:
1. video and audio are two indipendent different rtp streams
2. audio starts with the lower free udp numbers, video with the higher ones.
Only that could give me an explanation, why audio is ok in our test, but
no video, and only if we want to call to a NAT, but worked from it.
Where can I get more info?
bye
Ronald