HI,
The log of xlog are generated in messages. So i would like to know, what
i must do if i want that xlog send its information in an other file for
only one request L_WARN(...)
I want to use this request in order to do stats of connection of UA
Thx for your help.
Hello Users,
When I working account part and Nat with OpenSER,
The Transaction of route record entered into the Loose route Section.
if(loose_route())
{
acc_db_request(" ","");
t_relay();
exit;
};
# call type transaction taking place here.........
..............
.....................
...................
......................
.....................
..................
All the Accounting and voice part are working ...
but i'm doubt in loose route taking place all the call transcation....
--
Thanks and Regards
Ravi Prakash Sunkara
ravi.sunkara(a)hyperion-tech.com
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
ravi.sunkara(a)hyperion-tech.com
www.hyperion-tech.com
Dear all:
I am new in ser. I install ser 0.9.6. And make a call
using halaphone..
Now i want to use radius server for authenticate.
I already install Radius server freeradius-0.9.0 and
radiusclient-ng-0.5.3 ..
Now the problem is auth_radius module is not
installed...
>From the scratch ... how i solve the problem.
Regards
John
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Hi all,
I'd like to ask some questions about MSILO module..
I have setup server and it works well, then I add msilo configuration into openser.cfg
For addition, my clients are using MS Windows. I used 2 clients to test MSILO feature.
I try to call the offline client, but my x-lite display that user not found.
Please tell me what's the problem.
Thanx
Regards,
Aldi
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Dear all,
I am am using SER to provide SIP service to my clients, but I would like to
allow them to make outbound calls. This is where a SIP VoIP provider comes
in. As far as I know, I have 2 possibilities:
- using IP authentication
- using client authentication
As for the first one, I only have to forward the INVITE messages, right?
What I would like to know, is how to use "user authentication". In this
case, it would be transparent to the VoIP provider since it should be
treated as a regular SIP user agent.
Can anyone help me out?
Jose Simoes
hi,
When calling from Sipura to softphone (either SJphone or the sip client in my
e61 nokia phone), with the Sipura set to
Prefered codec: g723
Use prefered codec only: no
and the softphone only supporting g711, I get this reply from the softphone:
U xxxx:11474 -> xxxxx:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP
xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001.
To: <sip:1@test.com>;tag=ujk6mpqhbhhc6kj68irr.
From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0.
Call-ID: e5052808-575f7746(a)192.168.1.52.
CSeq: 102 INVITE.
Warning: 304 192.168.1.3 Media type not available.
Content-Length: 0.
the invite was:
U xxxx:5060 -> xxxx:11474
INVITE sip:1@test.co SIP/2.0.
Record-Route: <sip:xxxx;lr=on;ftag=d0404120874d710eo0>.
Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1.
Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001.
From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0.
To: <sip:1@test.com>.
Call-ID: e5052808-575f7746(a)192.168.1.52.
CSeq: 102 INVITE.
Max-Forwards: 69.
Contact: sipura line1 <sip:2@xxx:10001>.
Expires: 240.
User-Agent: Sipura/SPA2002-3.1.5.
Content-Length: 419.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: x-sipura.
Content-Type: application/sdp.
.
v=0.
o=- 36143 36143 IN IP4 xxxx.
s=-.
c=IN IP4 xxxxxx.
t=0 0.
m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101.
a=rtpmap:4 G723/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729a/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:100 NSE/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.
If I set prefered codec to g711 on the Sipura it works normally.
What is the best way to handle this on the sip proxy?
Thanks,
Richard
Dear all:
I setup SER and use radius server to accounting. However, the record
after I
test are not complete.
A(Caller) <------> B(Callee)
Suppose A is caller and B is callee. The record will be only one and
correct if
A hang up first. Unfortunately, if B hang up faster than A, the record
will be
divide into two records like table1(for example.).
Table 1.
|-----------------------------------------------------------------------------------------------------------------|
|AccSessionID AccUniqueId ...<skip>... Start Time Stop Time
...<skip>... |
|-----------------------------------------------------------------------------------------------------------------|
|AHandUpFirst 001 2006-10-02
2006-10-02 |
| 12:40:03
12:45:32 |
|-----------------------------------------------------------------------------------------------------------------|
|BHandUpFirst 002 2006-10-02
0000-00-00 |
| 12:58:40
00:00:00 |
|-----------------------------------------------------------------------------------------------------------------|
|BHandUpFirst 003 2006-10-02
2006-10-02 |
| 13:05:03
13:05:03 |
|-----------------------------------------------------------------------------------------------------------------|
How can I solve this problem? Please give me a hand.
Test User Agent: Window messenger 5.1 and X-lite
Regards,
Caxton
hello group,
i am searching for a sip-softclient (windows or linux - opensource,
freeware preferred), which is able to send an INVITE completely without
any SDP information (no message body).
does someone of you know something like this?
regards
thomas
--
thomas balsfulland tbals(a)ctrl-c.de
zwischen mut und dummheit liegt nur ein grat
der sich durch das vorbereitetsein unterscheidet
Dear all,
Another question that I have to do is:
Imagine that I have a connection to a VoIP provider using a username and a
password (so that SER register in the VoIP provider as a normal user). Once
it is registered, do you think it possible to receive calls? That is, make
it ring in one or all of the SIP users configured in SER?
Any help would be appreciated.
Jose Simoes
Hello,
I run ser.0.9.6 and I can use
either
rewritehostport("1.2.3.4:5060");
or
rewritehostport("pstngw.foo.com:5060");
with t_relay() command on INVITE/PSTN handler in order
to route calls to a Cisco PSTN GW.
This works fine.
Is "DNS SRV queries" supported on ser.0.9.6 and is there a specific conf ?
thanx
Kostas
