Hi!
I've checked out branch 1.2 using:
svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2/
openser
...
...
A openser/radius.h
A openser/qvalue.c
U openser
Checked out revision 1836.
Why has it checked out revision 1836? commit 1836 was Juha's LCR AVP
changes which were applied to trunk.
Taking a look at Makefile.defs I see that I've checked out 1.2, but why
does svnversion report 1836?
Maybe someone can shortly explain this.
thanks
klaus
--
Klaus Darilion
nic.at
Hi,
I have got the following strange problem: after creating a user with
openserctl add alice alice alice(a)netx.test
database engine 'MYSQL' loaded
Control engine 'FIFO' loaded
is_user: user counter=0
check_db_alias: alias counter=0
MySql password for user 'openser@localhost':
new user 'alice' added
root@sip:/#
the password column in the database of user alice is empty, and I only can
register without any password.
The same happens with 'bob' etc...
I am using plaintext passwords:
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
Any ideas what could have gone wrong?
cheers
Franz
Hello,
I installed 1.2 and am trying to add user accounts using the following
command:
openserctl add Bob xyzzy Bob(a)gigef1.net
It let me add one user account with no problems. When I tried adding
the second account i get the following error:
ERROR 1062 (23000) at line 1: Duplicate entry ' ' for key 3
Any clues would be appreciated.
Thanks,
Charlie
Has anyone been successful sending/receiving T.38 faxes with the
latest version of MediaProxy? I have read that there were some issues
with NAT previously. Is this still the case?
Thanks!
Daryl
Hi,
I'm trying to understand how the SST handles the case of a UA that is
off the network when the session timer expires.
I have the following configuration:
x-lite has session timer support, so using that to test the module.
min session timer is set to 91 seconds.
x-lite (149) - > openser --> x-lite ( 101)
INVITE --->
-> INVITE
<--100 trying
100 trying <-----
<-180 ringing
180 ringing <--
<-200 ok
200 ok <---
ack --->
-->ack
Call is setup at this point.
I then remove the power supply from 101.
some seconds later we see this:
INVITE -------> --> INVITE
100 trying <------
...... ( no response from 101)
<--- 408 timeout
BYE ---------->
....
Ok at this point openser never sends a 200 ok to the BYE.
>From what I can see in the SST module, that it deletes the dialog.in
memory, but not sure how that will help here.
--- debug logs -- where the FIRST bye comes in from 149:
0(5526) SIP Request:
0(5526) method: <BYE>
0(5526) uri: <sip:101@192.168.0.104:5061>
0(5526) version: <SIP/2.0>
0(5526) parse_headers: flags=2
0(5526) Found param type 232, <branch> =
<z9hG4bK-d87543-8f727a53ca51673e-1--d87543->; state=6
0(5526) Found param type 235, <rport> = <n/a>; state=17
0(5526) end of header reached, state=5
0(5526) parse_headers: Via found, flags=2
0(5526) parse_headers: this is the first via
0(5526) After parse_msg...
0(5526) preparing to run routing scripts...
0(5526) parse_headers: flags=100
0(5526) DEBUG:maxfwd:is_maxfwd_present: value = 70
0(5526) parse_headers: flags=10
0(5526) DEBUG: add_param: tag=faea2a83cd5fd6c1
0(5526) DEBUG:parse_to:end of header reached, state=29
0(5526) DBUG:parse_to: display={"101"}, ruri={sip:101@192.168.0.101}
0(5526) DEBUG: get_hdr_field: <To> [51]; uri=[sip:101@192.168.0.101]
0(5526) DEBUG: to body ["101"<sip:101@192.168.0.101>]
0(5526) DEBUG: add_param: tag=5077431e
0(5526) DEBUG:parse_to:end of header reached, state=29
0(5526) DBUG:parse_to: display={"149"}, ruri={sip:149@192.168.0.101}
0(5526) parse_headers: flags=200
0(5526) is_preloaded: No
0(5526) grep_sock_info - checking if host==us: 13==13 &&
[192.168.0.104] == [192.168.0.101]
0(5526) grep_sock_info - checking if port 5060 matches port 5061
0(5526) grep_sock_info - checking if host==us: 13==13 &&
[192.168.0.104] == [192.168.0.101]
0(5526) grep_sock_info - checking if port 5060 matches port 5061
0(5526) DEBUG:check_self: host != me
0(5526) grep_sock_info - checking if host==us: 13==13 &&
[192.168.0.101] == [192.168.0.101]
0(5526) grep_sock_info - checking if port 5060 matches port 5060
0(5526) after_loose: Topmost route URI:
'sip:192.168.0.101;lr;ftag=5077431e;did=5ff.8e4f3775' is me
0(5526) parse_headers: flags=200
0(5526) get_hdr_field: cseq <CSeq>: <3> <BYE>
0(5526) DEBUG: get_hdr_body : content_length=0
0(5526) found end of header
0(5526) find_next_route: No next Route HF found
0(5526) after_loose: No next URI found
0(5526) DBG:rr:run_rr_callbacks: callback id 0 entered with
<lr;ftag=5077431e;did=5ff.8e4f3775>
0(5526) DEBUG:dialog:dlg_onroute: route param is '5ff.8e4f3775' (len=12)
0(5526) DEBUG:dialog:lookup_dlg: dialog id=1467217128 found on entry 4085
0(5526) DEBUG:dialog:run_create_callbacks: dialog=0xb5c65280, type=16
0(5526) DEBUG:sst_handlers.c:sst_dialog_terminate_CB:403: Terminating
DID 0xb5c65280 session
0(5526) DEBUG:sst_handlers.c:sst_dialog_terminate_CB:410: Freeing the
sst_info_t from dialog 0xb5c65280
0(5526) DBUG:dialog:unref_dlg: unref dlg 0xb5c65280 with 2 (delete=1)-> 0
0(5526) DBUG:dialog:destroy_dlg: destroing dialog 0xb5c65280
0(5526) loose_route() succeeded - M=BYE
RURI=sip:101@192.168.0.104:5061 F=sip:149@192.168.0.101
T=sip:101@192.168.0.101 IP=192.168.0.102
ID=NTUzNzA5NWNjYmM5YWQ2MjQxYzgzZDdkZTRlMmQwODk.
0(5526) ERROR:dialog:dlg_status: res->ri = 5
0(5526) comp_scriptvar: int 20 : 5 / 0
0(5526) DEBUG: t_newtran: msg id=8 , global msg id=7 , T on entrance=0xffffffff
0(5526) parse_headers: flags=ffffffffffffffff
0(5526) parse_headers: flags=78
0(5526) t_lookup_request: start searching: hash=10869, isACK=0
0(5526) DEBUG: RFC3261 transaction matching failed
0(5526) DEBUG: t_lookup_request: no transaction found
0(5526) DBG: trans=0xb5c67b00, callback type 1, id 1 entered
0(5526) DBG: trans=0xb5c67b00, callback type 1, id 0 entered
0(5526) parse_headers: flags=78
0(5526) DEBUG: mk_proxy: doing DNS lookup...
0(5526) check_via_address(192.168.0.102, 192.168.0.102, 0)
0(5526) DBG:check_against_rule_list: using list dns
0(5526) DEBUG:tm:set_timer: relative timeout is 500000
0(5526) DEBUG: add_to_tail_of_timer[4]: 0xb5c67c4c (61500000)
0(5526) DEBUG:tm:set_timer: relative timeout is 30
0(5526) DEBUG: add_to_tail_of_timer[0]: 0xb5c67c68 (91)
0(5526) DEBUG:tm:t_relay_to: new transaction fwd'ed
0(5526) DEBUG:tm:UNREF_UNSAFE: after is 0
0(5526) DEBUG:destroy_avp_list: destroying list (nil)
0(5526) receive_msg: cleaning up
---
thanks,
Andy
Hi Everyone,
I just installed OpenSER w/MySQL and noticed that whenever MySQL
became unavailable I could no longer place calls. Is this the correct
behavior? I thought OpenSER's in-memory database would still allow new
calls to go through.
I'm using db_mode 1 in my openser.cfg on SVN 1.2.0 version of OpenSER.
I'd really like to have OpenSER be able to continue working in the
event MySQL is unreachable temporarily, for example, if the MySQL
server crashes and needs to be rebooted or repaired. How can I
accomplish this?
Thanks!
Daryl
Guys,
Is it possible that permission module doesn't allow requests from a
host which is sending from a port other than 5060 (my case 5061), even
if this host is in trusted ip table?
All other trusted hosts are working fine, but are sending from 5060.
Thxs in advance,
Dan
Hi all,
I'm using exec_dset to run a script which takes the Request-URI and
returns a new Request-URI based on the number dialed.
In ser.cfg I'm doing this:
if (exec_dset("lookup.sh \"$SIP_RURI\"")) {
# Route call
} else {
# Reply with a cause
}
Recently a customer started to use SIP-X, and his INVITES with
Request-URIs looks like this:
sip:1001@192.168.0.1:5060;sipx-noroute=Voicemail
Ser responds with "500 Server Internal Error" to these INVITEs, which is
due to that my script fails.
If I run SER in the foreground with debugging I can see this:
"sh: line 1: sipx-noroute=Voicemail: command not found"
so clearly the ";" in the INVITE causes a problem.
How do I correctly escape the SIP_RURI in my ser.cfg to prevent my
script from failing? If I run my script from commandline with the
request uri, everything is fine.
Br,
/Tobias
Hello
I need help in configuration.
Lets say I already have a SIP Server (anyone) and I use it to handle
VoIP in my company.
Now I want to use OpenSER as a SIP proxy between the UAs and the SIP
Server. I want all SIP messages to pass via OpenSER and the OpenSER
shall be able to reject an INVITE message in both direction (incoming
call and outgoing call). That's all.
All the rest shall be handled by the SIP Server (forking, call
forwarding, PSTN Gateway connectivity, etc...).
Actually I just need the OpenSER receiving the requests, adding itself
on route and forwarding to the SIP Server for outgoing requests and to
the UA for incoming requests.
Is there an example of config file that can do that?
Thanks.
Regards,
Michel.
