Hi Alejandro,
I'm aware of the incapacity of Asterisk of properly answer to OPTIONS
pings, so we made a fix on the opensips side - the LB module has a
parameter that allows you do define an extra list of reply codes to be
considered as success (aside 200 ok). See:
Hi Andrew,
you can achieve what you need by using the sort_order option:
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id272689
Probably alg. 1 is what you need:
/1/ - the destinations from each group are randomly arranged (only the
two elements are randomly selected);
Hi,
Had the same problem, change $config-admin_passwd_mode=1; to
$config-admin_passwd_mode=0;
in opensips-cp/config/globals.php and it worked. I dont know is this is the
right way.
regards
On Thu, Mar 11, 2010 at 2:29 AM, er...@netcrc.net wrote:
I have the following problem
With install
Hi,
From your post I understand that you are trying to access the Control
Panel using a SIP user username + password.
You can only log in into CP using administrator account + password. (You
can add one or more admin accounts if you want).
If you read the INSTALL file you will see how to add
Hi,
Did you add the first admin account using the query provided in the
INSTALL file ?
Because if you did it should have worked without changing the value of
*$config-admin_passwd_mode* from 1 to 0.
Regards,
Alex Ionescu
On 3/11/2010 13:51, TCB wrote:
Hi,
Had the same problem, change
Hola Iñaki
Here is the SIP flow you asked me.
Hope it helps to spot the problem. Observe the watcher is behind NAT.
SUBSCRIBE sip:present...@presence.net sip%3apresent...@presence.netSIP/2.0
Record-Route: sip:xxx.xxx.xxx.xxx:;lr=on
Call-ID: d041f6cae786fea76503d43658570...@10.1.1.235
CSeq: 1
Great help, thank you both!
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Hi Liu,
Ok - the bug is fixed now :)
Regards,
Bogdan
liuf wrote:
Yes, you are right. They're the same.
Had this bug been fixed yet? It seems still open now.
Best Regards,
Liu Fan
--
Bogdan-Andrei Iancu
www.voice-system.ro
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Users
Hi Liu,
Right - thanks for the report. I also fixed these tables.
Regards,
Bogdan
liuf wrote:
Thanks.
Base svn revision 6624, I check files in db_berkeley/opensips/ again. I
found the files which have no METADATA_KEY value as below:
dialplan
dr_gateways
dr_groups
dr_rules
Hi,
I don't know if anyone could point me in the right direction, but I am trying
to get Presence, BLF and BLF dialog working so that when a call is made the
lamp lights show the staus on the Snom.
At present it all works for a few minuites and then either just stops working
or the
Hi Pascal,
I have just tested myself - with a Subscribe sent on udp and contact
with trasport=tcp and the Notify is sent over tcp.
I would suggest to run opensips in debug mode and paste the log messages
when starting the processing for sending the Notify ( starting with
Thank you, Bogdan
mhomed solved my problem. OpenIMS is based on SER but obviously the
parameter is available also there.
Regards
Kenneth
Bogdan-Andrei Iancu skrev:
Hi Kenneth,
not sure what this openims is, but if something from the OpenSIPS
family, the normal behaviour is to preserve the
Hi, I have a question about mediaproxy port usage, mediaproxy is a very good
friend of opensips so I guess I can post the question here.
Our mediaproxy, is using RPT port 5 it this optimal, shall I configure
it differently ?
I am concern that some equipement may implement QoS efforts
Hi Julien,
On 11/3/10 2:55 PM, Julien Chavanton wrote:
Hi, I have a question about mediaproxy port usage, mediaproxy is a very
good friend of opensips so I guess I can post the question here.
Our mediaproxy, is using RPT port 5 it this optimal, shall I
configure it differently ?
I am
Anca
I got the following logs when subscribing:
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:presence:build_dlg_t: CONTACT = sip:watc...@79.149.10.246:5060
;transport=tcp
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:t_uac:
next_hop=sip:213.151.105.170:;lr=on
Mar 11
Hi Pascal,
Only now I observed that there is a Route header in the Notify - so the
first hop is in fact the address in the route and not the RURI. So it is
correct to send the Notify on udp, and the proxy at the next hop should
switch to tcp.
Regards,
--
Anca Vamanu
www.voice-system.ro
Ok, good comment. Yes indeed we have a SIP proxy between the subscriber and
opensips (which is only acting as presence server in this scenario).
I thought there were two problems (one at proxy and one at presence server
side). We will work on proxy side now to solve this issue.
Many thanks for
Hi Dmitri,
found the bug and fixed it :). If you update from SVN, it should work as
a charm
Thanks for your report,
Bogdan
Dmitri G. wrote:
Hi Bogdan,
I'm running OpenSIPS 1.6.1
version: opensips 1.6.1-notls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
Hi all,
OpenSIPS 1.6.2 is the first release following the new release policy
after the 2.0 work started (see
http://www.opensips.org/Development/Development).
1.6.2 release brings both new features / enhancement and a lot of fixes.
The listing with all additions and fixes is available under:
Thanks for your help, It's working, I will continue testing.
Regards
Erick Ch.
Hi,
Had the same problem, change $config-admin_passwd_mode=1; to
$config-admin_passwd_mode=0;
in opensips-cp/config/globals.php and it worked. I dont know is this is
the
right way.
regards
On Thu, Mar 11,
Greetings folks;
I've been attempting to get a working db_virtual failover setup going and am
having some very basic functionality problems while in testing, I was
wondering if someone could shed some light one what I, presumably, am doing
wrong.
Appropriate sections of config look something
Hi,
When compiling the perl module on version 1.6.2-notls on Centos :
make prefix=/ all
I receive :
...
make[1]: Entering directory `/usr/src/opensips-1.6.2-notls/modules/perl'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall
-minline-all-stringops -falign-loops -ftree-vectorize
I have with the following problem:
The file /tmp/opensips_fifo for any reason disappears
for example if send the command for reboot the server.
The opensips using mi_fifo module.
With the command opensips -c said is ok
opensipsctl moni said (if not find the file)
ERROR: Error opening
Dear All,
I'm having the following error with mediaproxyx 2.3.8, and all the call
will have no voice after running mediaproxy for a few days in a high
loading (40 calls) server.
I check the new mediaproxy with the same line in send, there is no
such getHost call.
Is this a fixed issue in the
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