Dears,
Any one can help on this?
Thanks,
BR,
Dears,
I am trying to use OpenSIPS with TLS but didn't work till now :(
I am getting the following error:
Mar 25 14:09:49 [16855] DBG:core:print_ip: tcpconn_new: new tcp connection
to: 192.168.20.19
Mar 25 14:09:49 [16855] DBG:core:tcpconn_new:
On 29/3/11 6:21 AM, n...@uni-petrol.com wrote:
Yes, ip_tables loaded:
ip_tables 57440 3 iptable_nat,iptable_mangle,iptable_filter
But media-relay exception still present on every calls.
Was it loaded before or after starting the media-relay?
Is it possible that error because of python
Hello Brett,
The only db_mode that doesn't write to DB at shutdown is 0 ( NO_DB ).
For all the other db_modes, dialog info is flushed to DB at shutdown.
Regards,
--
Vlad Paiu
OpenSIPS Developer
/
/
On 03/29/2011 07:25 AM, Brett Nemeroff wrote:
All,
on dialog db_mode 2 - DELAYED, does it
Because of media-relay running in openvz virtual container it inherit
kernel modules like ip_table, ip_conntrack, nfnetlink from host node,
so
yesm ip_tables module like others loade before media-relay start.
Below is a list of running modules.
Unfortunately it is very hard to migrate to
Hi Vlad,
Thanks for the reply, below is the output of bt full.
Mark
#0 0x7fb9cd8cde57 in db_mysql_get_columns (_h=value optimized out,
_r=0x796490) at res.c:71
col = 1
fields = value optimized out
__FUNCTION__ = db_mysql_get_columns
#1 0x7fb9cd8c7e36 in
Hi all!
Please tell me know if this behaviour is intentional:
Problem with proxying rtp:
UAC receives ip in the UAS' subnet while UAS receives the ip of UAC's
subnet of rtp proxy by default.
IP-Phone is on 10.200.10.195.
Network configuration:
ast1.local --- opensips+rtpproxy
Hi David,
Have you configured OpenSIPS to check clients certificate (have you set
tls_require_client_certificate = 1) ? Then you have to configure the
accepted certificates:
http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html#AEN264.
Regards,
--
Anca Vamanu
OpenSIPS Developer
On
On 29/3/11 11:42 AM, n...@uni-petrol.com wrote:
Because of media-relay running in openvz virtual container it inherit
kernel modules like ip_table, ip_conntrack, nfnetlink from host node, so
yesm ip_tables module like others loade before media-relay start.
Below is a list of running modules.
OK.
I will try:
1. Install python 2.6 in virtual environment and test it.
2. Run media-relay on host environment with python 2.4.
3. Run media-relay on host environment with python 2.6.
I have new periodic error in logs:
Mar 29 14:22:46 media-relay[1983]: error: Connection with dispatcher at
Hi thrillerbe,
I think that if you only want to build the list of selected
destinations, you can just call use_next_gw and add the uri in RURI to a
destination string ( because use_next_gw sets the RURI to the
destination-
Hello,
I have installed opensips 1.6.4 + mediaproxy + cdrtool + freeradius + mysql,
and I have followed the install guide from
http://cdrtool.ag-projects.com/wiki/Install.
I am able to find accounting record in the freeradius server log file at
/var/log/freeradius/radacct/, but when I trying
adrian,
thanks for the new version of sylkserver. i built it myself on debian
squeeze by first getting it from your repo with command
apt-get source sylkserver
build and install went fine, but when i try to start sylkserver, i get
the error below.
any idea what goes wrong? my
Hello there,
I am trying to create a seperate Presence server to use with my existing
OpenSIPS proxy. My main proxy/registrar listens at 10.1.1.1 port 5080, while
the presence server listens to 10.1.1.1 5061.
The registrar/proxy only makes use of pua so as to change the status of
non-presence
Hello there,
I am trying to create a seperate Presence server to use with my existing
OpenSIPS proxy. My main proxy/registrar listens at 10.1.1.1 port 5080, while
the presence server listens to 10.1.1.1 5061.
The registrar/proxy only makes use of pua so as to change the status of
non-presence
Hello.
Sometimes I see in a log following lines:
[89566]: DBG:dialog:dialog_update_db: inserting new dialog 0x80417c568
[89566]: DBG:db_mysql:db_mysql_do_prepared_query: conn=0x731538
(tail=7530824) MC=0x72e9a8
[89566]: DBG:db_mysql:db_mysql_do_prepared_query: new query=|insert into
dialog
Hi,
Just took a look at the code and it seems to be a problem somehow
related with the mysql library on your machine.
At startup, OpenSIPS tries to fetch all dialog info from DB, and it
first gets the column names and column types from the dialog table. It
seems that in your case, the mysql
Hello
Everyone
I want to replace the Asterisk (being used as a SIP Server for
registration, authentication and call routing) with OpenSIPS in OpenBTS
project, as i am planning to have an Asterisk cluster for dedicated services
and OpenSIPS will be forwarding the SIP calls
Hello,
The '?' sign is just a placeholder in the case of prepared statements.
When the query will be issued to the DB, all '?' signs will be replaced
with the appropriate values for the query.
The to_tag column in the dialog database is marked as NOT NULL because
dialogs are pushed to DB
Hi Vlad,
Yes, I'm using dialog module with B2B top hiding scenario. In case of such
problem, should I rewrite config to avoid using them together? This will not be
easy...
Also, in a MySQL log I see exactly that requests with '?'...
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Hi,
First of all OpenSIPS is a sip server so it works only with SIP.
Secondly, by default opensips is SIP proxy, so it cannot do handover.
But using the Back2Back User agent module, you may be able to play with
the ongoing calls and move them between different termination points.
I can help
Hi David,
David Chedid wrote:
I need to test the SIP over TLS using the OpenSIPS.
· I need to know what is the best stable version? so I can
install it and start testing.
use 1.6.4:
http://www.opensips.org/Resources/Downloads
· Do I need to generate certificate and
Probably you're looking for:
http://www.opensips.org/Resources/DocsTutLoadbalancing
BTW, do you have OpenBTS running in a production environment?
Regards,
Erik
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org]
Namens ALICOMPUTECH
Hi Boris,
are you sure you properly did the relation between the two interface in
RTPproxy and the i and e flags in nathelper - maybe you simply
swapped the interfaces (as meaning) between the definition in rtpproxy
and usage in nathelper.
Regards,
Bogdan
Boris Ratner wrote:
Hi all!
Hi,
Another tricks:
1) you can read the pending destinations directly from AVPs, without
calling the use_next_gw() function. See:
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id293166
2) as append_branch() does not accept variables as params, use the
$branch variable to
Hello
bundle of thanks for the reply,
i am sorry for not explaining the problem in a proper way, actually OpenBTS
does not support handoff of calls and i want to control it via OpenSIPS
OpenBTS does not offer handoff between base stations during a call. Handoff
between calls can be done
Hi
Bogdan
thanks for the prompt and quick reply
i will be using Multi Criteria
Decision Theory (MCDT) to take the handoff decision between base stations
during a call
the possible scenario might be
e.g. if the Signal strength is not good
Hi Denis,
Indeed, the SIP capture looks like opensips is not matching the received
BYE to the dialogThis is why the timeout is fired. But is strange, I
do no see any obvious reason for the the matching to fail...
If you can reproduce this case, could you enable full debug in opensips
Hi Dani,
See the radiusclient.conf (conf of the libradiusclient lib) - relevant
part:
quote
# time to wait for a reply from the RADIUS server
radius_timeout 10
# resend request this many times before trying the next server
radius_retries 3
/quote
Regards,
Bogdan
Dani Popa wrote:
Hi
Hi Chris,
the avp_db_load() loads the value into avp(s:maxtime) as string value
and you pass it as string to $avp(i:10).
The dialog module expects an integer value in the $avp(i:10) variable,
so do:
If (is_method(“INVITE”)){
avp_db_load(“$ru”,”$avp(s:maxtime)”);
Hi,
Thanks for sharing your experience!
I tried media-relay within a OpenVz container. As far as I can remember, I got
same exceptions during call establishing:
mediaproxy.interfaces.system._conntrack.Error: Table does not exist (do you need to
insmod?)
OS: debian/ubuntu
Bogdan,
When I configure:
$(branch(uri)[0]) = $ru;
$(branch(q)[0]) = 100;
xlog(L_INFO,branch 0 = $(branch(uri)[0]) with q-value
$(branch(q)[0])\n);
I get this debug:
ERROR:core:pv_set_branch_fields: SCRIPT BUG - inexisting branch assigment
[0/0]
ERROR:core:do_assign: setting PV failed
Need some guidance on how presence should be configured.
Our setup consists of two SIP registrars with opensips in between them (mainly
for codec manipulation purposes).
I'm trying to get presence to work. I see SUBSCRIBE messages coming from both
SIP registrars, but I don't see that they are
Hopefully my last question:
Using append_branch() and $branch allows me to add all destinations as
branches with q-values. However, I am unable to remove/edit the initial
entry in $ds as set by do_routing():
Contact: *sip:15552345678@1.1.1.1, *sip:15552345678@1.1.1.1;q=1,
Hi Razvan,
I've turned on DBUG, although not many output in syslog.
Mar 29 22:12:05 /usr/sbin/opensips[9336]: INVITE Received -
RURI=sip:x
Mar 29 22:12:05 /usr/sbin/opensips[9336]: Alias Found, New
RURI=
Mar 29 22:12:05
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