Hi
I have problem with force_rtp_proxy. Kamailio server have multi network
interfaces. I have two PSTN GW-s. And If is something wrong with first
one, the call is send to the second via PSTN failover route. When I send
call to the first GW I'm calling force_rtp_proxy(oc,10.10.10.1); But
the
Sorry wrong user list.
Ernest
On 15. 09. 2010 08:18, Ernest Mavrel wrote:
Hi
I have problem with force_rtp_proxy. Kamailio server have multi network
interfaces. I have two PSTN GW-s. And If is something wrong with first
one, the call is send to the second via PSTN failover route. When
Hi
I am using uac_replace_from() in the request route of my opensips
script (using opensips 1.6.3) which is working very well.
I notice there is also a uac_restore_from() function which can be
used. Can anyone advise when (or if) this function should be used? I
have it at the moment in the
Hi Dave
Thanks for the fix.
I went through your initial script and it looks very interesting.
I manage to get monit, Smonitor and MI to work without the script.
Misinterpreting the configurations ;-).
This is OK for now, but I am very interested in your solution.
I will go through the new
I forgot to add the opensipsctlrc file configuration:
## control engine: FIFO or UNIXSOCK
## - default FIFO
# CTLENGINE=xmlrpc
## path to FIFO file
OSIPS_FIFO=/tmp/opensips_fifo
## MI_CONNECTOR control engine: FIFO, UNIXSOCK, UDP, XMLRPC
MI_CONNECTOR=FIFO:/tmp/opensips_fifo
#
I am also getting this output when checking the opensips.cfg file.
opensips:/var/log# opensips -C /etc/opensips/opensips.cfg
Sep 15 15:55:29 [3090] DBG:core:yyparse: loading module /lib/opensips/
modules/db_mysql.so
Sep 15 15:55:29 [3090] DBG:core:yyparse: loading module /lib/opensips/
Hi all,
I return on an argument proposed a couple of weeks ago.
I have an installation of opensips binding on two ports (5060 and 5061).
Some client registers itself on 5060, others on 5061.
My wish is that opensips uses as from port the same port used by client
for registration.
Is it possible
Hi, I am looking for a way to monitor RTPProxy from Nagios
I did monitor the traffic between opensips and rtpproxy
ngrep -d lo port 9000
Then I was able to retreive this identifier 21756_365, how do I find one ?
osip2:~# echo 21756_365 I | nc -u 127.0.0.1 9000
21756_365 sessions
All of the sudden yesterday and today my OpenSIPS kept crashing every every
few hours before of out of mem messages. I tried to increase the PKG
4*1024*1024 and SHM to 256 and recompile and still having the same problem.
There is no core dump generated for the crash and there is out of memory for
Hi,
I had the same problem using large carrier route tables. Which modules do
you use in your setup ? What version of opensips ?
Eric LI DESTRI
2010/9/15 k1028 mrprotoc...@gmail.com
All of the sudden yesterday and today my OpenSIPS kept crashing every every
few hours before of out of mem
I too encountered the same problem in our production environment, but it was
resolved once we increased the SHM and PKG memory to some higher value. We
are using OpenSIPS version 1.6.2. Can you place the log here and also total
number of OpenSIPS children process ?
Regards
Rajib
On Wed, Sep 15,
Now I have the time to look into the log file more. The the memlog is very
big. What specific should I look into it?
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation
ERROR:tm:new_t: out of mem
ERROR:tm:t_newtran: new_t failed
ERROR:tm:store_reply: failed to alloc'
I am using opensips 1.6.3 with very only few modules loaded. The OpenSIPS is
up for over a months with up any issue until now. I also tried to increase
the PKG to 4mb and SHM to 256 mb still crash.
loadmodule db_mysql.so
loadmodule sl.so
Seems like when I try to transfer a call from one user to the next, it
does not do anything, so I am guessing we have to handle the REFER message?
What is the best practice for handling REFER messages?
Thanks.
___
Users mailing list
Call transfer using SIP require cooperation between the 3 end-points involved.
On the public internet, which is probably why you want to use OpenSIPS, this
does not work primarily because of accounting reasons. You can configure the
SIP proxy to route the Refer and Notify alright but as an
Look at the b2bua module.
-tr
On Wed, Sep 15, 2010 at 12:02 PM, David J. da...@styleflare.com wrote:
Seems like when I try to transfer a call from one user to the next, it
does not do anything, so I am guessing we have to handle the REFER message?
What is the best practice for handling
Adrian,
So in that case I would need to have a PBX, such as Asterisk, or
FreeSWITCH to handle this case, which would mean the PBX has to be in
the call path the entire time?
Thanks.
On 9/15/10 12:17 PM, Adrian Georgescu wrote:
Call transfer using SIP require cooperation between the 3
Yes, this is the only way to be able control the success of the call
transfers, actually the PBX administrator will make sure that the phone is
trusted and can honor all transfers request correctly.
It may be possible to achieve this with the B2BUA module as suggested, as long
as you control
I ran into out of memory and adjusted the shared memory opensips uses with
the -m option.
It tells opensips how much shared memory to use in MB. eg:
.../sbin/opensips -m 500 -l eth0
gives it 500MB of shared memory and listen on eth0
I believe by default it uses 100MB if -m is not specified.
Dave
I even increased the SHM to 1024 still haven't the same problem. I believe
identified the problem now and will share with everyone once I confirm this
is the fix. Thank You
--
View this message in context:
Hi,
I have installed Opensips, opensips-cp, Free Radius (accounting), CDRTool
and is weoking fine.
Recently i have installed CALL CONTROL application for prepaid accounting
purpose. Can someone pls brief where and how to define the customers with
their balances and how do they relate with their
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