The "from" port is not relevant. I need to manipulate the remote port, the
tsrget.
On Tue, Aug 20, 2024, 2:50 AM Pavel Eremin wrote:
> just guess... I suppose you may want to send second leg from 5060 port
> of your server?
>
> пт, 16 авг. 2024 г. в 21:48, Saint Michael :
w.siphub.com
>
> On 8/16/24 7:45 PM, Saint Michael wrote:
> > Using opensips 3.4, I need to change the destination port for an
> > outbound call. The call comes in at port 56000, where I have a socket,
> > and I need to send the second leg to change the destination
Using opensips 3.4, I need to change the destination port for an
outbound call. The call comes in at port 56000, where I have a socket,
and I need to send the second leg to change the destination port to
5060.
Is there a way to to this?
___
Users mailing
INVITE sip:19206661392@38.95.11.250;transport=UDP
when this happens, opensips has a contaminated ru variable
and since it's read-only, I cannot fix it in code
I tried
if ($ru =~ ";transport=UDP") {
$ru = $(ru{s.select,0,-13});
but it makes no difference, ru does not change
How do I get aro
(xlog) NOTICE:ru sip:19206661392@1.1.1.1;transport=UDP rU 19206661392
DST= 1.1.1.1 rd= 1.1.1.1
CRITICAL:core:mk_proxy: could not resolve hostname: ";transport=UDP"
ERROR:tm:uri2proxy: bad host name in URI
ERROR:tm:t_forward_nonack: failure to add branches
ERROR:tm:w_t_relay: t_forward_nonack faile
(xlog) NOTICE:ru sip:19206661392@1.1.1.1;transport=UDP rU 19206661392
DST= 1.1.1.1 rd= 1.1.1.1
CRITICAL:core:mk_proxy: could not resolve hostname: ";transport=UDP"
ERROR:tm:uri2proxy: bad host name in URI
ERROR:tm:t_forward_nonack: failure to add branches
ERROR:tm:w_t_relay: t_forward_nonack faile
ating-Config-Files-3-2
>
> -Brett
>
>
> On Wed, Jul 3, 2024 at 4:09 PM Saint Michael wrote:
>
>> I need something like
>> #include "/etc/opensips/test.cfg", inside opensips.cfg
>> Is this possible using some mechanism?
>> Philip
>>
&
I need something like
#include "/etc/opensips/test.cfg", inside opensips.cfg
Is this possible using some mechanism?
Philip
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
I have an opensips box with 40+ IP addresses. No matter the call's
"received at" address, I need the second leg to use the same IP, so
when the carrier receives it, he sees the same IP as "source" that I
saw as "received at".
How can I do this? is it possible?
Philip
__
I need to save the media address offering to the CDR. I understand
that it comes with the INVITE but it changes upon connection. Any
idea how to do this?
Also, I need to drop the call if the media address matches a list of
blocked addresses.
Philip
___
whatever query I call, db_unixodbc_get_columns returns only db_int for
every field.
DBG:avpops:ops_dbquery_avps: query [CALL get_user_details(1)]
DBG:core:db_new_result: allocate 48 bytes for result set at 0x71c49935ba20
DBG:db_unixodbc:db_unixodbc_get_columns: 3 columns returned from the query
I am.very interested in this case.
On Fri, Dec 8, 2023, 9:20 AM Bogdan-Andrei Iancu
wrote:
> Simon, I do not need the corefile, usually it is huge - as per web
> instructions (see the prev link), extract the backtrace (using gdb) and
> share it with me.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
>
I am using opensips 3.4
my /etc/opensips/opensips.cfg has
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/home/opensips/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
and it does exists:
file /home/opensips/opensips_fifo
/home/opensips/opensips_fifo: fifo (named pipe)
stat /ho
Question:
is there a significant speed difference in servicing events from
version 2.4 to 3.4?
On Wed, Oct 4, 2023 at 1:19 PM Liviu Chircu wrote:
>
> Hi, everyone!
>
> The 3.4.2, 3.3.8 and 3.2.15 OpenSIPS minor versions are scheduled for release
> on Wednesday, Oct 18th.
>
> In preparation for t
I attempted an upgrade to version 3.4 from 3.1 and It failed
ERROR:core:db_check_table_version: invalid version 8 for table dispatcher
found, expected 9
ERROR:dispatcher:mod_init: failed to init database support
ERROR:core:init_mod: failed to initialize module dispatcher
ERROR:core:main: error whil
The issue is: opensips uses
dig -t NAPTR sip.x.com
so you need do that first and get the real terminating hosts and add them
to /etc/hosts
then, it works
On Tue, Sep 5, 2023 at 3:43 AM Karsten Wemheuer wrote:
> Hi,
>
> Am Montag, dem 04.09.2023 um 07:56 -0400 schrieb Sain
Using opensips 3.1, I am sending calls to a carrier whose address is
sip.x.com. I add an entry to /etc/hosts/ that redirects sip.x.com
to a different address, and I test with "resolvectl query sip.x.com" an
it works as I need it. But opensips keeps sending calls to the DNS address,
not
Yes, I am in bash shell...
On Tue, Jul 4, 2023 at 12:33 AM mayamatakeshi
wrote:
> Are you on a bash shell when you execute these commands?
> If not, I would try to switch to it.
>
> On Tue, Jul 4, 2023 at 2:58 AM Saint Michael wrote:
>
>> my script is
>> cd /usr/src
0: Illegal option --
make[1]: --libs: Command not found
/bin/sh: 0: Illegal option --
make[1]: --libs: Command not found
/bin/sh: 0: Illegal option --
make[1]: --libs: Command not found
make[1]: Leaving directory '/usr/src/opensips-3.1/modules/lua'
On Mon, Jul 3, 2023 at 8:43 PM Saint M
My compilation is failing by first time, version 3.1
is there a list of pre-requisites for compiling opensips on Ubuntu 20.04?
make[1]: Entering directory '/usr/src/opensips-3.1/modules/lua'
/bin/sh: 0: Illegal option --
/bin/sh: 0: Illegal option --
/bin/sh: 0: Illegal option --
make[1]: --libs:
the output format of tshark. With
> ngrep you can use "-W byline" to make sure that the SIP messages are
> displayed line by line.
>
> Have a nice weekend,
>
> Karsten
>
> Am Donnerstag, dem 30.03.2023 um 06:21 -0400 schrieb Saint Michael:
> > I am al
;
> Have a nice day
>
> Karsten
>
> Am Donnerstag, dem 30.03.2023 um 01:40 -0400 schrieb Saint Michael:
> > Is there any command line tool for Linux (ubuntu 20.04 or better)
> > that can generate opensips SIP CDR from a group of large *.pcap files
> > that only con
Is there any command line tool for Linux (ubuntu 20.04 or better) that can
generate opensips SIP CDR from a group of large *.pcap files that only
contain SIP information?
I have not found it. I tried with tshark and it would take a lot of coding
to get this done. I am looking for these fields
fi
ing the duplicated invite as a new call.
>
>
> On Fri, Mar 24, 2023 at 11:40 PM Saint Michael wrote:
>
>> I have on a typical day many calls that arrive twice, same SIP CALL ID,
>> almost the same time, maybe the next millisecond. This happens because the
>> dialer does n
I have on a typical day many calls that arrive twice, same SIP CALL ID,
almost the same time, maybe the next millisecond. This happens because the
dialer does not wait for a confirmation, and resends the same call multiple
times. But it confuses my CDR.
Is there a way to check for the CALL ID and
Feb 15, 2023 at 2:14 PM Daniel Zanutti
wrote:
> Virtual Router Redundancy Protocol (VRRP)
>
>
> https://www.techopedia.com/definition/13483/virtual-router-redundancy-protocol-vrrp
>
>
> On Wed, Feb 15, 2023 at 3:21 PM Saint Michael wrote:
> >
> > what is VRRP ?
> >
> &g
stops replying to OPTIONS
> * it responds 503 to INVITE if other nodes are still up
> * when all early dialogs have ended, the IP moves to another node
>
> I can then restart OpenSIPS on this node without losing anything.
>
> Cheers,
> Kingsley.
>
> On Thu, 2023-01-12 at 23:2
Dear friends
I need to understand how to verify a token using the public key of the
issuer, using Bash and openssl. I guess liviuchircu knows this subject very
well.
So far I have done this, using a real token:
token=“eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM
n
> to the endpoint - he is the one that should respond.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 2/3/23 01:24, Saint Michael wrote:
> > The Customer is unable to keep calls open past 15 minutes.
> >
The Customer is unable to keep calls open past 15 minutes.
If the duration of the call was a multiple of 15 minutes, please make
sure that you can properly respond to the keep-alive RE-INVITE that
the carrier sends every 15 minutes.
How do I make sure that Opensips responds to any REINVITES?
_
This is the nth times that it happens. Opensips 3.1 It stops writing
cdr to the hard drive. The calls keep flowing, and I keep losing
records, money.
Is there a command that I may run without restarting opensips that
restarts the process, internally?
if I restart opensips I lose all the records.
Th
I have a lot of calls that should work with RTPPROXY and instead I get dead air.
Maybe this is the issue.
On Fri, Dec 23, 2022 at 2:45 PM Wadii ELMAJDI | Evenmedia
wrote:
>
> hello , i do have a question related to rtpproxy module documentation.
>
> The doc describes that rewriting sdp body shoul
Opensips+ RTPProxy only works fine with plain LXC containers,
privileged, which basically have access to all the resources of the
box.
That is the model I use with great success.
On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote:
>
> Hello Terrance,
> I wouldn't really recommend this. RTPProxy
LOOKING FOR A CONSULTANT
My usual engineer Vlad is on vacation util Dec 5th,
Today I lost around $1000, of which the client only agreed to pay 50%
(they never think that they sent so much traffic), because opensips
3.1 decided to stop writing calls to the hard drive.
My question is: what may cause
The greatest question I've read so dar.
On Sun, Oct 30, 2022 at 10:48 AM M S wrote:
>
> Hi list,
> When a dialog is created with initial INVITE, it seems like it is not
> available to dlg_end_dlg until the call is answered (OK received) and the
> caller also acknowledges (ACK) to the OK message
I want to thank everybody. I finished my project with the help of Vlad Paiu.
Saint Michael
On Thu, Oct 27, 2022 at 12:52 PM Saint Michael wrote:
>
> Thanks Bogdan-Andrei
>
> however, if I try to set the whole flag to 0 (from an initial value of
> 15), I get
>
> /usr/local/b
https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/27/22 7:05 PM, Saint Michael wrote:
> > this code does not seem to work: I set the initial value to 15, inside
> > opensips.cfg
t; OpenSIPS Bootcamp 5-16 Dec 2022, online
>https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/27/22 4:15 PM, Saint Michael wrote:
> > YES
> > Maybe we should fix the documentation?
> > now the real question is, what goes inside the bracket if the flag
&
e thing. I apologize.
On Thu, Oct 27, 2022 at 11:14 AM Saint Michael wrote:
>
> I am afraid I am not seeing it clearly:
> from the command line I set
> opensips-cli -x mi set_gflag 1
> "OK"
> what function will test if this is true inside the code?
> this does n
1\r\n");
}
it is counterintuitive that you set Bit 1 from the command line and
then test it inside the code but it fails the test.
How can I print the current gflag inside xlog();
On Thu, Oct 27, 2022 at 11:00 AM Liviu Chircu wrote:
>
> On 27.10.2022 17:55, Saint Michael w
e
code, it cannot be used. The module gflag is of a great importance
when managing Opensips, because it allows for an orderly shutdown
without losing CDR, that is: money.
Many thanks.
Thank you!
On Thu, Oct 27, 2022 at 10:48 AM Liviu Chircu wrote:
>
> On 27.10.2022 17:20, Saint Michael w
Dear friends
I set flag to 1, using
opensips-cli -x mi set_gflag 1
and it works, in fact:
opensips-cli -x mi get_gflags
{
"hex": "0x1",
"dec": "1"
}
but in the code, this test never works:
if (is_gflag(1)) {
xlog("$ci Shutting Down\r\n");
}
else {
xlog("$ci Gflag is not 1\r\n")
gt; Hi,
>
> I am not familiar with the gflags module, but
>
> Am Donnerstag, dem 27.10.2022 um 08:39 -0400 schrieb Saint Michael:
> > for testing I added
> > route{
> > if (is_gflag("1")) {
> > t_relay("udp:10.0.0.1:5060&quo
mp_2022/
>
> On 10/27/22 8:05 AM, mayamatakeshi wrote:
>
> It can be done with:
> https://opensips.org/docs/modules/3.1.x/gflags.html
>
>
> On Thu, Oct 27, 2022 at 1:18 PM Saint Michael wrote:
>>
>> Dear Friends
>> I successfully wrote a script that termi
odules/3.2.x/dialog.html#mi_profile_end_dlgs
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On
s.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/26/22 5:28 AM, Saint Michael wrote:
> > Dear Friends
> > local_route
> > {
> > if (is_method("200"))
> > {
> >
Dear Friends
local_route
{
if (is_method("200"))
{
}
}
I need to intercept the code at the moment when a call connects, so I
may notify that information to the database. What specific code can I
use inside a Local Route so the 200 OK corresponds with a real
connection
I just caught a callID that is 160 characters long.
What is the max length that it should be? I am using 128.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Will that work on version 3.1X
On Tue, Oct 25, 2022, 10:36 AM Social Boh wrote:
> maybe:
>
>
>-
>
>*modparam("tm", "disable_6xx_block", 1)*
>
> to TM module
>
> ---
> I'm SoCIaL, MayBe
>
> El 25/10/2022 a las 9:22 a. m., Richard Robson escribió:
>
> I am currently testing version 3.3 an
Dear friends
is there a command I can send to Opensips 3.1, so all calls are shut
down and BYEs are sent to both the caller and the callee? Then of
course, opensips closes down.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cg
method support: poll, epoll, sigio_rt, select.
git revision: 8cdc3e3f6
main.c compiled on 05:02:26 Oct 14 2022 with gcc 9
On Mon, Oct 24, 2022 at 7:15 PM ideanet help wrote:
>
> Hi Philip,
> Which version do you have?
>
> Regards,
> Jazzi
>
> On Tue, Oct 25, 2022 at 11:30
any idea what may cause this?
[519171] ERROR:core:io_wait_loop_epoll: failed to remove from epoll
Bad file descriptor(9)
Oct 24 22:23:00 proxy opensips[519171]: Oct 24 22:23:00 [519171]
ERROR:core:io_wait_loop_epoll: [UDP_worker] unset/bogus map (idx=367)
triggered for 1 by epoll (fd=-1,type=0,fla
#idp5880336
> Retrieving ->
> https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp5887712
>
> Or work with flags, if just true or false value
> https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp341408
>
> Regards
> On Sun, Sep 25, 2022 at 1:45 PM Saint M
Question:
can you write your own functions with opensips?
On Sun, Sep 25, 2022 at 12:05 PM Saint Michael wrote:
> Dear Daniel
> Can you point me to an example?
> Right now Opensios will get a clogged memory.
> Many thanks.
>
>
> On Sun, Sep 25, 2022, 11:45 AM Daniel Zanut
Dear Daniel
Can you point me to an example?
Right now Opensios will get a clogged memory.
Many thanks.
On Sun, Sep 25, 2022, 11:45 AM Daniel Zanutti
wrote:
> You have to use dialog variable storing.
> Take a look at dialog module.
>
> Em dom., 25 de set. de 2022 10:42, S
I noticed that the variable
$avp(lineid)
set in the section of the code handling the original INVITE, is null when I
need to close the call.
Is there a way to store a variable that will be available throughout the
call, everywhere?
I am trying:
cache_store("local","lineid_$ci","$avp(lineid)",0);
bu
, "OK");
>
> - Jon Abrams
>
> On Thu, Sep 22, 2022 at 10:29 AM Saint Michael wrote:
>
>> Dear friends
>> I have a call extender technology that is very useful, but it has a
>> problem, and my support provider, who wrote it, is missing in action.
>>
Dear friends
I have a call extender technology that is very useful, but it has a
problem, and my support provider, who wrote it, is missing in action.
This is the issue: the call extender technology does not confirm the BYE to
the caller
if ($avp(hold_seconds) > 0) {
");
> }
> }
>
>
> On Wed, Sep 14, 2022 at 5:04 PM Johan De Clercq wrote:
>
>> Xlog(….);
>>
>> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
>> --
>> *Van:* Users namens Saint Michael <
>> vene...
ok voor iOS <https://aka.ms/o0ukef> downloaden
>> --
>> *Van:* Users namens Saint Michael <
>> vene...@gmail.com>
>> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
>> *Aan:* OpenSIPS users mailling list
>> *Onderwer
>
> On Wed, Sep 14, 2022 at 4:40 PM Saint Michael wrote:
>
>> This is a trace showing a BYE from Opensips, but none of the sides did
>> actually hangup.
>>
>>
>> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael wrote:
>>
>>> I use opensips 3.1, an
This is a trace showing a BYE from Opensips, but none of the sides did
actually hangup.
On Wed, Sep 14, 2022 at 3:33 PM Saint Michael wrote:
> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
>> upgraded all calls fail after 20 seconds.
>
> cd /usr/s
>
> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
> upgraded all calls fail after 20 seconds.
cd /usr/src/opensips-3.1/
git pull
make clean;make proper;make all
make modules
make install
clearlog.sh
systemctl restart opensips
opensips -V
How do I go back?
__
tpproxy.html#event_E_RTPPROXY_DTMF
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 9/14/22 10:46 AM, Sain
>
> My goal is to close the call as soon as the callee presses any DTMF
when any DTMF is detected, then I need to access the callID variable, $ci
but it's nowhere to be found
Also I need $avp(start_time) and $avp(duration)
I am at a loss as to how to retrieve this information. The call is connec
n error is detected by RTPProxy,
> and usually the error is that it cannot bind the IP you asked to bind on
> (pub.lic.i.p). You should check the rtpproxy logs for more information.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
&
I keep getting this error:
SCRIPT: Failed to engage rtpproxy for trunk XX.XXX.XX.135 -
104678ZWJjOWU2ZDlkZWQ3MmE0MThjZWEzNTNlMzVhOTVhYTg
ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy
The call comes from another opensips box with rtpproxy enabled.
On connect, I get
Thanks
It's solved.
Federico
On Wed, May 11, 2022, 10:40 AM Daniel Zanutti
wrote:
> https://www.opensips.org/Documentation/Tutorials-Topology-Hiding
>
> On Tue, May 10, 2022 at 2:15 PM Saint Michael wrote:
>
>> Dear friends
>> I am using opensips 3.1.9, with rt
Dear friends
I am using opensips 3.1.9, with rtp proxy, and without topology hiding it
would not talk to any carrier who has a Sonus box. I need to add topology
hiding urgently and my support provider is missing in action. Can somebody
provide instructions and code samples?
Federico
__
Dear Friends
I see a lot of the unixodbc errors below, which I guess are not real
errors, just a reconnect from a dead connection after a few hours. The
question is how to avoid logging them with error_level=-1. Is that even
possible?
ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call
reserved character and not
> allowed on a string:
> https://datatracker.ietf.org/doc/html/rfc3261#section-25.1
>
> Should be something on client of your customer, since you received on 180
> ringing, but i'm not sure if you can just solve it. It's violating RFC.
>
>
> O
Dear friends
Kindly look at the file attached. I am losing 10% of my CDR because some
messages cannot be parsed by Opensips
opensips -V
version: opensips 3.1.9 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAP
My new business is to provide 302 Redirect services and Opensips does not
genrate a CDR for those calls. Other type of calls do generate a record. Is
this by design or is it a bug?
Every call that goes through Opensips should generate a record.
Any idea about what is going on?
_
In my case, Opensips is losing 30% of CDR, but there is no error.
At routing time I create a CDR record based on SIP callID, and Opensips is
supposed to "close" it when the call drops by executing a stored procedure
from the place when the call is finished. It does not in 30% of cases. I
started do
I have a similar issue, which probably started with an update
30% of my calls' CDR do not get written to disk after
OPENSIPSCTL=/usr/local/bin/opensips-cli
/usr/bin/timeout -k 5 5 ${OPENSIPSCTL} -x mi flat_rotate 2>&1 >>
/usr/src/cdr.log
any idea what can be happening?
On Sun, Feb 27, 2022 at
My carrier rejected me for interconnection with this message
"Invalid Content-Length, other from the size of the actual body
In reINVITE from customer did not present Header Content-Length: 231
customer replace Content-Length: 231 on his opensips to X-: 231
Content-Length must be present in case
>
> I needd to send out only compact sip. I use Opensips. 2.4. Can somebody
> indicate how do I change my configuration to achieve 100% compact sip? I
> found some information but I am afraid is not enough for a succesful
> implementation..
Philip
___
Us
I would love to see a SQL engine, a real one, being pulled inside Opensips.
Right now I have to use unixODBC and execute a stored procedure 3000 times
per second, only to achieve a routing decision. It is very inefficient.
Some engines like RocksDB, open-source, should be part of Opensips and run
i
stem Architect II
>
> 900 Main Campus Drive, Suite 100, Raleigh, NC 27606
>
>
>
> m: 919-578-3421 • o: 919-727-4614
>
> e: rrev...@bandwidth.com
>
>
> On Fri, Oct 2, 2020 at 12:53 AM Saint Michael wrote:
>
>> I need to close the CDR files so I can start processi
The naming convention I am using includes Hour and Minute, then all files
are named differently.
On Fri, Oct 2, 2020 at 10:56 AM Vic Jolin wrote:
> If you rotate before moving or renaming you might have an issue.
>
> On Fri, Oct 2, 2020, 10:48 PM Saint Michael, wrote:
>
>> B
But I move the file to another box, via an NFS mount.
That's why my approach is better. I build a list of files, rotate, and then
move the files across the Internet.
On Fri, Oct 2, 2020 at 10:32 AM Liviu Chircu wrote:
> On 02.10.2020 17:23, Saint Michael wrote:
>
> the order of
the order of actions seems to be problematic. what if I issue a rename,
move command, and opensips is actually at the same time writing to the file?
On Fri, Oct 2, 2020 at 9:41 AM Ben Newlin wrote:
> There is no automatic configuration in OpenSIPS to close and/or roll the
> flatstore files. Yo
How often the flatstore log files get closed and a new one is created?
I need to move the file to another directory and it is still open.
Philip
On Thu, Aug 13, 2020 at 11:03 AM Bogdan-Andrei Iancu
wrote:
> I think this is irrelevant for the topic. Again, the flatstore backend
> creates one file
>
> I need to close the CDR files so I can start processing them.
I want a new file generated every 60 seconds, if there are calls.
is there any way to have opensips behave this way?
>
>
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ns.com
> OpenSIPS Summit 2020 online
> https://www.opensips.org/events/Summit-2020Distributed/
>
> On 8/6/20 5:21 PM, Saint Michael wrote:
>
> Maybe somebody has seen this?
> I am using
> opensips -V
> version: opensips 2.4.8 (x86_64/linux)
> flags: STATS: On, DIS
Maybe somebody has seen this?
I am using
opensips -V
version: opensips 2.4.8 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
pol
I call this:
append_hf("Identity:\r\n$avp(signature)\r\n");
and the outbound INVITE is wrong, since the header must be located before
the "Content-Type: application/sdp.
X-: 381."
in fact, the 381 are wrong because my new header itself is way over 500
characters.
Allow: INVITE, ACK, CANCEL, OPTION
dmesg shows this:
[849895.444677] opensips[37151]: segfault at 0 ip 55f471304456 sp
7ffc04f79dc0 error 6 in opensips[55f4711ce000+2ec000]
[849895.444690] Code: 5b 08 48 85 db 75 ef 48 83 c1 10 48 39 d1 75 d3 e9 3d
fe ff ff 4c 8b 4b 10 4c 8b 43 08 49 81 fe 00 40 00 00 0f 87 6a 01 00 00
<4d>
tion abut the environment you're running on?
> Is it debian or redhat based, could you send us the fifo file
> permissions, as well as the user OpenSIPS is running with?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-soluti
/usr/local/sbin/opensipsctl fifo flat_rotate
/usr/local//lib64/opensips/opensipsctl/opensipsctl.fifo: line 121:
/tmp/opensips_fifo: Permission denied
This happened after upgrading my kernel to 5.4.0-37-generic
although I cannot see the connection.
___
Use
Not really, I am root. The fifo es also owned by root.
On Sat, Jun 13, 2020 at 2:59 PM Ovidiu Sas wrote:
> Most likely the user that is running the command doesn't have
> write|execute access to the fifo file.
>
> -ovidiu
>
> On Sat, Jun 13, 2020 at 10:52 AM Saint Micha
WARNING:dialog:register_dlgcb: Cannot register callbacks in DELETED state
(type 2000)!
ERROR:acc:acc_onreply: cannot register callback for context serialization
Any idea what these errors mean?
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/usr/local/sbin/opensipsctl fifo get_statistics
I get
/usr/local//lib64/opensips/opensipsctl/opensipsctl.fifo: line 121:
/tmp/opensips_fifo: Permission denied
My new kernel is
5.4.0-37-generic
Any idea what am I doing wrong?
How can I know how may open channels I have?
I do 3000+ CPS with Opensips and MySQL using unixODBC, no problem. My
query is for routing only. I read a 280 MM table (RocksDB) for every call.
So it's comparable.
It seems to work flawlessly. However, I only use $60.000 servers from Dell,
R920s, with 64 physical cores and 120 threads, plus 1.5
I have been wondering why my billing never matched the invoice from my
carrier and I think I found the issue. I pass a few million calls a day. It
turns out that Opensips is not rounding upwards the ms to calculate
seconds, and that is done at the source code. This issue leads to absurds
like a cal
Calvin, feel free to login to the container and change the config for
MariaDB, it is located in a single file /etc/my.cnf.
I use TokuDB, a new engine that is better than InnoDB. Lately, I shifted to
RocksDB, which is even better, designed by Facebook.
I have not updated that box because: "if it ain
I need to add a new SIP header to the response below.
if ($rm=="INVITE") {
/* add the redirect destinations as branches */
$branch = "sip:bat...@gotham.com";
$branch = $avp(my_custom_uri);
/* sending a 3xx reply will automatically push all
* existing branches as Contact URIs
I have an opensips box, and the programs below are running. I did not
install the box, so I am not sure what that do. Does anybody know?
├─master─┬─cleanup
│├─local
│├─pickup
│├─qmgr
│└─trivial-rewrite
___
I need to identify all private IPs vs public IPs
Right now I am doing
$rd =~ "192.168" || $rd =~ "10." || $rd =~ "172.16."
but in regex there is a faster way, chaining several ORs
like
$rd =~ "192.168|10.|172.16."
what is the correct way to do this in opensips?
_
The ideal platform to run opensips, asterisk, etc. is LXC containers, not
docker. Unless I misunderstand docker, you may only dockerize applications,
while in LXC you conteinarize the equivalent of full virtual machines, all
apps together as a unit. Performance is identical to docker, for both use
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