Hi Bogdan,
The exact version used is 1.7.1 stable release compiled from src. Output of
opensips -V is:
version: opensips 1.7.1-notls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=10
Hi Duane
But, I don't used mediaproxy. I used RTPPRoxy.
Nick
On 2011年12月02日 11:24, duane.lar...@gmail.com wrote:
If you only want RTP I believe CDRTool keeps up with RTP traffic but
you have to be sending the RTP streams through Mediaproxy. I've done
both SIP and RTP with Cisco "IP SLA" but t
If you only want RTP I believe CDRTool keeps up with RTP traffic but you
have to be sending the RTP streams through Mediaproxy. I've done both SIP
and RTP with Cisco "IP SLA" but that is once again because the packets are
tagged with QoS and also going through the cisco Router. I would also t
Hi Duane
Yes, You are right.
So, Is it not other method???
If I only want to collect RTP traffic. Do you have any suggest??
Thanks
On 2011年12月02日 04:57, duane.lar...@gmail.com wrote:
So probably the only thing Nick can do if he is wanting to measure how
much bandwidth the SIP packets are usin
Hi folk, i am facing this issue:
I set, just for learing purpose, my opensips.cfg as follow:
loadmodule "siptrace.so"
# - sip_trace --
modparam("siptrace", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "enable_
Hi Bogdan, thanks for reply. It was my mistake. I was creating a user,
whitout compiling all fields. ( i was missing aliases ).
Now it works perfectly.
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Add-User-From-Opensips-Cp-tp7051225p7052539.html
Sent fr
Hi Bogdan,
Any ideas here?
Thanks.
-- James
On Tue, Nov 29, 2011 at 3:57 PM, James Lamanna wrote:
> Realized I got variables from the wrong frame.
>
> (gdb) print p
> $1 = (ua_pres_t *) 0x2aba902bfb00
> (gdb) print *p
> $2 = {hash_index = 305, local_index = 0, id = {s = 0x2aba902bfc44
> "DIALOG
So probably the only thing Nick can do if he is wanting to measure how much
bandwidth the SIP packets are using (if I understand his question
correctly) would be at the router like Cisco's "IP SLA". But in order for
this to work all SIP packets would need to be tagged with QoS.
On , Bogdan
Hi Duane,
No, siptrace cannot provide data about how much bandwidth is used - of
course, for the SIP signaling you can simply sum the size of the
packages, but not really the best method.
But if ti goes to RTP, that is really impossible to say, as opensips
will never know what are the used c
Hi Ryan,
The accounting module catches only calls that are proxy-ed (forwarded).
If the call is locally completed, it is not automatically accounted, but
you can still do it manually via the acc_db_request() function from acc
module:
http://www.opensips.org/html/docs/modules/1.7.x/acc.htm
Hello,
My opensips is not starting. I noticed the /var/log/messages file indicates
it needs a database. I want to use postgresql instead of mysql, so I changed
the DBENGINE parameter to PGSQL in the opensipsctlrc and osipsconsolerc.
Now, when I execute "./opensipsdbctl create", I get the fol
Brett,
Yes you're right, opensips is not starting. I noticed the /var/log/messages
file indicates it needs a database. I want to use postgresql instead of mysql,
so I changed the DBENGINE parameter to PGSQL in the opensipsctlrc and
osipsconsolerc.
Now, when I execute "./opensipsdbctl create
Bogdan there is too little info about this online, can you please help
me a bit more with this, how do I write the if statement, and how do I
set a variable for the first call, and how do I retrieve which server
was used for the first call.
On Wed, Nov 23, 2011 at 9:46 PM, Schneur Rosenberg
wrote:
Bogdan are you saying that siptrace can keep up with how much bandwidth SIP
is using? I thought Nick was wanting to graph the total bandwidth taken up
by SIP.
On Dec 1, 2011 11:03 AM, "Bogdan-Andrei Iancu" wrote:
> Hello Nick,
>
> Take a look at the siptrace module - it is used to dump the SIP t
I am using accounting opensips with cdr accounting and the
failed_transaction_flag set.
I was wondering if anyone knows a way to get calls that we reject from
within our script (e.g. send_reply("503"), etc) to be accounted into
missed_calls? These don't currently seem to be recorded anywhere and
we
Hi,
So, when trying to add a new user, do you get an error or so ? what
about the apache logs ? do you see any errors there ? Are you sure you
populated all fields in the "Add" form ?
Regards,
Bogdan
On 12/01/2011 06:46 PM, spady wrote:
Hi all, I setted up a OpenSIPS-Cp and almost everythin
Hi Jayesh,
If dialog module has restart persistence ( db modes 1, 2 and 3) the CDRs
should survive across a restart. If this is not the case for you, let me
know the exact version of opensips you are running (do "opensips -V").
Regards,
Bogdan
On 12/01/2011 12:28 PM, Jayesh Nambiar wrote:
H
Hello Nick,
Take a look at the siptrace module - it is used to dump the SIP traffic
going through opensips into DB:
http://www.opensips.org/html/docs/modules/1.7.x/siptrace.html
Then you can use OpenSIPs Control Panel to view in a graphical way the
DB content (with call flows).
Regards,
Hi all, I setted up a OpenSIPS-Cp and almost everything it's working good
except for User--->User Managment--> Add New.
If I try to add a new User, when I press OK, nothing happen. User is not
added and I can't see anything into opensips.log.
I created some user by using "opensipsctl add " and it w
Hi,
If you are trying to access the VM from non-host machine then you
definitely need to use Bridged adapter mode in your VM networking option.
That way your VM will have the sam subnet IP as of your LAN.
--
Regards,
Sammy
2011/11/19 Zsolt Pintér
> Hi All!
>
> Im a newbie thus I hope my letter
HI All
I have setup OpenSIPS to act as a B2BUA with "top hiding" scenario.
Additionally, i want to have the ability to dynamically hunt for a
successful route from a list of possible route choices.
Currently i use the following routing script
--SCRIPT BEGINS-
if(is_method("INVITE") &&
Hi All!
Im a newbie thus I hope my letter will be not so disturbing!
I want to configure a SIP server with presence and conferance to use with
android softphones. Thought on doing this with OpenSIPS for presence +
Asterisk for conference.
I found the new OpenSIPS Live VM which I thought will be gr
HI All
I have setup OpenSIPS to act as a B2BUA with "top hiding" scenario.
Additionally, i want to have the ability to dynamically hunt for a
successful route from a list of possible route choices.
Currently i use the following routing script
--SCRIPT BEGINS-
if(is_method("INVITE") &&
Hello All,
I am planning to use CDR accounting in my script starting from version 1.7
and it looks fine and working as expected. Although I had one doubt, how do
I make sure the CDR accounting still happens if the opensips is restarted
and BYE comes after the restart. I have tried db_mode 3 for dia
Hi
I want to monitor SIP Traffic with MRTG.
But, I don't want to monitor network card. In Server, I have other service.
Do everyone have other function with it??
Can snmpstats support this?? Which function can do it??
Thanks for suggest.
Nick
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