[OpenSIPS-Users] INVITE brings in extra transport=UDP parameter

[Saint Michael]

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 around this?

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[OpenSIPS-Users] Maybe this is a bug version 3.4 latest

[Saint Michael]

(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 failed
My box is multihomed, and I am using rttpproxy.
this: socket=udp:*:5060 use_workers 80
or this: socket=udp:1.1.1.1:5060 use_workers 80
make no difference
also to use rttpproxy or not makes no difference
The issue seems to be caused by ru
sip:19206661392@1.1.1.1;transport=UDP having now the ;transport=UDP,
but I only use RTP, so the tansport= part is irrelevant

The main problem is that my CDR is useless
because if the call fails, the field dst_ip comes as ";transport=UDP"
when it should have been "1.1.1.1"

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[OpenSIPS-Users] This may be a bug

[Saint Michael]

(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 failed
My box is multihomed, and I am using rttpproxy. Since I added a new IP to my box
socket=udp:*:5060 use_workers 80

then I get these errors. The main problem is that my CDR is useless
because if the call fails, the field dst_ip comes as ";transport=UDP"
when it should have been "1.1.1.1"

How do I get around this?

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Re: [OpenSIPS-Users] Is there any way to include files in the configuration?

[Saint Michael]

Include_file "/etc/opensips/demo.cfg"

On Wed, Jul 3, 2024, 7:10 PM Brett Nemeroff  wrote:

> OpenSIPS depends on external preprocessors for this kind of functionality.
> See if this peg helps you out. If not, let us know:
>
> https://www.opensips.org/Documentation/Templating-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
>>
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[OpenSIPS-Users] Is there any way to include files in the configuration?

[Saint Michael]

I need something like
#include "/etc/opensips/test.cfg", inside opensips.cfg
Is this possible using some mechanism?
Philip

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[OpenSIPS-Users] Originating address in multihomed computer

[Saint Michael]

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

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[OpenSIPS-Users] capturing the media address upon conenction

[Saint Michael]

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

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[OpenSIPS-Users] possible big in unixodbc queries

[Saint Michael]

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
 DBG:core:db_allocate_columns: allocate 84 bytes for result columns at
0x71c49935ba88
 DBG:db_unixodbc:db_unixodbc_get_columns:
RES_NAMES(0x71c49935baa0)[0]=[username]
 DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
 DBG:db_unixodbc:db_unixodbc_get_columns: RES_NAMES(0x71c49935bab0)[1]=[email]
 DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
 DBG:db_unixodbc:db_unixodbc_get_columns: RES_NAMES(0x71c49935bac0)[2]=[phone]
 DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
 DBG:core:db_allocate_rows: allocate 112 bytes for result rows and
values at 0x71c49935bb18
 DBG:db_unixodbc:db_unixodbc_str2val: converting INT [johndoe]
 ERROR:core:db_str2int: Unexpected characters: [johndoe]
 ERROR:db_unixodbc:db_unixodbc_str2val: converting integer value from
string failed
 ERROR:db_unixodbc:db_unixodbc_convert_row: failed to convert value

opensips -V
version: opensips 3.4.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: e50c6736e
main.c compiled on 00:29:53 Mar  2 2024 with gcc 9

dpkg -l | grep unixodbc
ri  unixodbc 2.3.6-0.1build1
amd64Basic ODBC tools
ri  unixodbc-dev:amd64   2.3.6-0.1build1
amd64ODBC libraries for UNIX (development files)

odbc.ini

[opensips]
Description = MySQL ODBC Driver Testing
Driver = maria
Socket = /var/run/mysqld/mysqld.sock
#Server = localhost
User = x
Password = 
Database = asterisk
Option = 3

odbcinst.ini

[maria]
Description=ODBC for MySQL
Driver=/usr/lib64/mariadb/libmaodbc.so
FileUsage=1
Threading=0

a Unixodbc trace shows that everything is normal,so the issue must be
with opensips
Statement = 0x6003034831b0
Column Number = 3
Target Type = 1 SQL_CHAR
Buffer Length = 1024
Target Value = 0x77a24e1ff318
StrLen Or Ind = 0x7ffc233ad5f0
[ODBC][103886][1709340852.301423][SQLGetData.c][545]
Exit:[SQL_SUCCESS]
Buffer = [123-456-7890]
Strlen Or Ind = 0x7ffc233ad5f0 -> 1

as you can see, the target Type is SQL_CHAR

any ideas?

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Re: [OpenSIPS-Users] Fwd: opensips is restarting after call is finished

[Saint Michael]

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
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 08.12.2023 16:13, Simon Gajski wrote:
> > Hi Bogdan
> >
> > you were right
> > we have the core file now
> >
> > Please let me know where to send it (gzipped is approx 2MB)
> >
> >
> > BR
> > Simon
> >
> > Bogdan-Andrei Iancu je 08.12.2023 ob 14:17 napisal:
> >> Hi Simon,
> >>
> >> Maybe you have some apport service running on the machine, moving the
> >> core file.
> >>
> >> Regards,
> >>
> >> Bogdan-Andrei Iancu
> >>
> >> OpenSIPS Founder and Developer
> >>   https://www.opensips-solutions.com
> >>   https://www.siphub.com
> >>
> >> On 08.12.2023 13:35, Neven Grzancic wrote:
> >>> Dear Bogdan,
> >>>
> >>> we tried following
> >>> https://opensips.org/Documentation/TroubleShooting-Crash and trying
> >>> different things, and even though the syslog shows that the core
> >>> file was dumped:
> >>>
> >>> Dec  8 12:25:13 sbc2 /usr/sbin/opensips[4363]:
> >>> CRITICAL:core:sig_usr: segfault in process pid: 4363, id: 7
> >>> Dec  8 12:25:13 sbc2 kernel: [ 3394.269733] opensips[4363]: segfault
> >>> at 0 ip  sp 7ffdc458f6a8 error 14 in
> >>> opensips[55816fbd+1f000]
> >>> Dec  8 12:25:18 sbc2 mariadbd[922]: 2023-12-08 12:25:18 400
> >>> [Warning] Aborted connection 400 to db: 'opensips' user: 'opensips'
> >>> host: 'localhost' (Got an error reading communication packets)
> >>> Dec  8 12:25:18 sbc2 /usr/sbin/opensips[4356]:
> >>> INFO:core:handle_sigs: child process 4363 exited by a signal 11
> >>> Dec  8 12:25:18 sbc2 /usr/sbin/opensips[4356]:
> >>> INFO:core:handle_sigs: core was generated
> >>> Dec  8 12:25:18 sbc2 /usr/sbin/opensips[4356]:
> >>> INFO:core:handle_sigs: terminating due to SIGCHLD
> >>> Dec  8 12:25:18 sbc2 /usr/sbin/opensips[4359]: INFO:core:sig_usr:
> >>> signal 15 received
> >>> Dec  8 12:25:18 sbc2 /usr/sbin/opensips[4360]: INFO:core:sig_usr:
> >>> signal 15 received
> >>>
> >>> the core does not get generated, or at least not in the directory
> >>> provided via options at runtime (-w /tmp). The issue is always
> >>> repeatable (the daemon crashes after every finished call), if there
> >>> are more active calls at the moment of the crash, whose calls remain
> >>> active and proceed normally until termination (which again causes a
> >>> crash). We also tried enabling debug mode, but in that case the
> >>> process does not start successfully and runs into a timeout. In any
> >>> case, any additional help or suggestion on how to successfully
> >>> generate the core dump would be much appreciated, as we seem to have
> >>> ran into a wall here...
> >>>
> >>> On 08.12.23 12:26, Simon Gajski wrote:
> 
> 
> 
>   Posredovano sporočilo 
>  Zadeva: Re: [OpenSIPS-Users] opensips is restarting after call
>  is finished
>  Datum: Thu, 7 Dec 2023 17:52:37 +0200
>  Od: Bogdan-Andrei Iancu 
>  Za: Simon Gajski , OpenSIPS users mailling
>  list 
> 
> 
> 
>  Hi Simon,
> 
>  Well, the log is very self explanatory:
>  Dec 7 15:59:47 sbc2 /usr/sbin/opensips[28488]:
>  INFO:core:handle_sigs: child process 28495 exited by a signal 11
>  Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]:
>  INFO:core:handle_sigs: core was generated
>  Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]:
>  INFO:core:handle_sigs: terminating due to SIGCHLD
> 
>  Or shortly, you opensips just crashed :). So see
>  https://opensips.org/Documentation/TroubleShooting-Crash for how to
>  report further.
> 
>  Regards,
> 
>  Bogdan-Andrei Iancu
> 
>  OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> https://www.siphub.com
> 
>  On 07.12.2023 17:17, Simon Gajski via Users wrote:
> >
> > Hi
> >
> > we are running opensips 3.4.2
> > on Ubuntu 22.04.3 LTS
> > and also use RTPengine Version: 10.5.0.0+0~mr10.5.0.0
> > git-master-74075f63
> >
> > Opensips acts as SBC with RTP engine enabled.
> >
> > Calls are working fine, however after each call is finished,
> > bellow action happens.
> >
> > We had same problem with opensips 3.2 and Ubuntu 20. So we did
> > upgrade to latest stable release, and it is the same.
> >
> > Is this opensips script configuration issue or a bug? And how
> > could I fix it?
> >
> > []
> >
> > Thanks for all tips.
> >
> > BR
> > Simon
> >
> >
> > ___
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> > Users@lists.opensips.org
> > 

[OpenSIPS-Users] fifo not found

[Saint Michael]

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 /home/opensips/opensips_fifo
  File: /home/opensips/opensips_fifo
  Size: 0   Blocks: 0  IO Block: 4096   fifo
Device: fe00h/65024dInode: 21530043Links: 1
Access: (0666/prw-rw-rw-)  Uid: ( 1001/opensips)   Gid: ( 1001/opensips)
Access: 2023-10-24 01:13:01.766009199 -0400
Modify: 2023-10-24 01:13:01.766009199 -0400
Change: 2023-10-24 01:13:01.766009199 -0400

opensips is running
 ps ax | grep opensips
 /usr/local/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg -m 5096 -M


but
/usr/local/bin/opensips-cli -x mi address_reload

ERROR: fifo file /tmp/opensips_fifo does not exist!
ERROR: Is OpenSIPS running?

it is trying to find opensips_fifo at the default place, how do  make
it look in the place indicated by opensips.cfg?

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Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases

[Saint Michael]

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 the releases, starting Monday, Oct 9th, we will impose the 
> usual freeze on any significant fixes (as complexity) on these stable 
> branches, in order to ensure a safe window for testing in the days ahead.
>
> Happy testing,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS eBootcamp, Oct 16-27 | www.opensips.org/training
>
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[OpenSIPS-Users] upgrade failed

[Saint Michael]

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 while initializing modules

how do I upgrade the dispatcher table from 8 version 8 to version 9?
The script is fine.
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Re: [OpenSIPS-Users] DNS question

[Saint Michael]

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 Saint Michael:
> > 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 to my new address.
> > I tried restarting opensips but the issue does not go away.
> > I tried restarting systemd-resolved, and nothing
> > I tried opensipsctl fifo cache_flush and it does hang.
> >
> > So how does Opensips obtains DNS information?
>
> Beside the system resolver also check the settings of the systemd
> resolver (/etc/systemd/resolver.conf) (if systemd is in use). There is
> a setting to ignore the hosts file, if I remember correctly.
>
> HTH,
> Karsten
>
>
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[OpenSIPS-Users] DNS question

[Saint Michael]

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 to my new address.
I tried restarting opensips but the issue does not go away.
I tried restarting systemd-resolved, and nothing
I tried opensipsctl fifo cache_flush and it does hang.

So how does Opensips obtains DNS information?
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Re: [OpenSIPS-Users] compilation fails

[Saint Michael]

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/opensips-3.1/
>> git reset --hard HEAD
>> git pull
>> make clean;make proper;make all
>> make modules
>> make install
>> and is generating errors
>> 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: 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 Michael  wrote:
>>
>>> 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: 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]: Entering directory '/usr/src/opensips-3.1/modules/python'
>>> /bin/sh: 1: python: not found
>>> /bin/sh: 1: python: not found
>>> /bin/sh: 1: python: not found
>>>
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Re: [OpenSIPS-Users] compilation fails

[Saint Michael]

my script is
cd /usr/src/opensips-3.1/
git reset --hard HEAD
git pull
make clean;make proper;make all
make modules
make install
and is generating errors
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: 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 Michael  wrote:

> 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: 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]: Entering directory '/usr/src/opensips-3.1/modules/python'
> /bin/sh: 1: python: not found
> /bin/sh: 1: python: not found
> /bin/sh: 1: python: not found
>
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[OpenSIPS-Users] compilation fails

[Saint Michael]

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: 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]: Entering directory '/usr/src/opensips-3.1/modules/python'
/bin/sh: 1: python: not found
/bin/sh: 1: python: not found
/bin/sh: 1: python: not found
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Re: [OpenSIPS-Users] looking for PCAP to CDR utility

[Saint Michael]

I solved it. You can do all this with tshark
tshark -r "$input_file" -Y "sip.Call-ID == \"$call_id\"" -w
"$output_dir/$call_id.pcap"



On Wed, Apr 5, 2023 at 2:28 AM Karsten Wemheuer  wrote:

> Hi Frederico,
>
> I unfortunately do not know about 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 alradybusing tshark and awk, but I thought that there was some
> > tools written to do this
> >
> > Federico
> >
> > On Thu, Mar 30, 2023, 1:55 AM Karsten Wemheuer  wrote:
> > > Hi,
> > >
> > > maybe You can use ngrep (converting pcap to text) and than a lot of
> > > scripting in the language of Your choice.
> > >
> > > 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 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
> > > >
> > > >
> > > fields="method,from_tag,to_tag,callid,sip_code,sip_reason,time,orig
> > > _
> > > >
> > > fu,orig_ru,out_fu,out_ru,source_ip,dst_ip,lrn,rate,provrate,jurisd,
> > > ro
> > > >
> > > unding,minsec,singlecharge,setuptime,closetime,duration,ms_duration
> > > "
> > > >  one record per call of course
> > > >
> > > >
> > > >
> > > >
> > > > ___
> > > > Users mailing list
> > > > Users@lists.opensips.org
> > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] looking for PCAP to CDR utility

[Saint Michael]

I am alradybusing tshark and awk, but I thought that there was some tools
written to do this

Federico

On Thu, Mar 30, 2023, 1:55 AM Karsten Wemheuer  wrote:

> Hi,
>
> maybe You can use ngrep (converting pcap to text) and than a lot of
> scripting in the language of Your choice.
>
> 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 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
> >
> >  fields="method,from_tag,to_tag,callid,sip_code,sip_reason,time,orig_
> > fu,orig_ru,out_fu,out_ru,source_ip,dst_ip,lrn,rate,provrate,jurisd,ro
> > unding,minsec,singlecharge,setuptime,closetime,duration,ms_duration"
> >  one record per call of course
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] looking for PCAP to CDR utility

[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
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

 
fields="method,from_tag,to_tag,callid,sip_code,sip_reason,time,orig_fu,orig_ru,out_fu,out_ru,source_ip,dst_ip,lrn,rate,provrate,jurisd,rounding,minsec,singlecharge,setuptime,closetime,duration,ms_duration"
 one record per call of course
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Re: [OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

[Saint Michael]

I use 3.1, is that applicable?


On Sat, Mar 25, 2023 at 9:49 AM Daniel Zanutti 
wrote:

> Hi  Federico
>
> Yes it does, need to create the transaction inside your script:
> https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_newtran
>
> This will avoid opensips handling 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 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 discard the INVITE if there
>> is another INVITE already in progress?
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>
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[OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

[Saint Michael]

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 discard the INVITE if there is
another INVITE already in progress?
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Re: [OpenSIPS-Users] Version 3.1 it stops writing cdrs

[Saint Michael]

Amazing, thanks
What I need urgently and cannot figure out, is how to verify a Stir Shaken
token using Bas and openssl
This is a real example:
token="eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9hei50YXgvMSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNjc2MjY1NTE0LCJvcmlnIjp7InRuIjoiMTcyNzQ0MzMwMTkifSwib3Jp
Z2lkIjoiMzU4ZDA0OGItZTAwNy0xMWVhLWFjMzctNDctMjA1LTE3Mi04OSJ9.YD9kpeGaKFa6Gh-TIX7mAJWl9W3EpitTCpHxS7UEmddfT0E3DfRvpwEIsGqC8ouKxwmuYGKq1hvgucKqLLYFhA"

echo "${token}" > token.txt
echo "-BEGIN PUBLIC KEY-
MFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEkCmapE3ygZFy5ZDAjFSGh5rp766W
EZ2+adZvlPeVk5kJzenhDJROfhh7aIKiG/npK8VnYE0WOf2OjyJU6LE9OQ==
-END PUBLIC KEY-" > pk.pem
IFS='.' read -r header payload signature <<< "$token"
echo "${signature}" > signature.bin

#echo -n "$signature"|basenc -d --base64url > signature.bin #ERROR
echo "${header}.${payload}"|openssl dgst -sha256 -binary > payload_hash.bin
openssl pkeyutl -verify -inkey pk.pem -pubin -sigfile signature.bin -in
payload_hash.bin



On Wed, 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 ?
> >
> > On Wed, Feb 15, 2023 at 1:16 PM Kingsley Tart 
> wrote:
> >>
> >> FWIW, I set up OpenSIPS here in a couple of clusters with VRRP managing
> >> the service "floating" IP addresses. If I want to restart a node, I
> >> mark it as disabled, which then means that
> >>
> >> * it 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 23:29 -0500, Saint Michael wrote:
> >> > 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.
> >>
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Version 3.1 it stops writing cdrs

[Saint Michael]

what is VRRP ?

On Wed, Feb 15, 2023 at 1:16 PM Kingsley Tart  wrote:

> FWIW, I set up OpenSIPS here in a couple of clusters with VRRP managing
> the service "floating" IP addresses. If I want to restart a node, I
> mark it as disabled, which then means that
>
> * it 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 23:29 -0500, Saint Michael wrote:
> > 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.
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Looking for help with Stir Shaken

[Saint Michael]

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=“eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9hei50YXgvMSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNjc2MjY1NTE0LCJvcmlnIjp7InRuIjoiMTcyNzQ0MzMwMTkifSwib3JpZ2lkIjoiMzU4ZDA0OGItZTAwNy0xMWVhLWFjMzctNDctMjA1LTE3Mi04OSJ9.YD9kpeGaKFa6Gh-TIX7mAJWl9W3EpitTCpHxS7UEmddfT0E3DfRvpwE
IsGqC8ouKxwmuYGKq1hvgucKqLLYFhA”
echo “${token}” > token.txt
echo “-BEGIN PUBLIC KEY-
MFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEkCmapE3ygZFy5ZDAjFSGh5rp766W
EZ2+adZvlPeVk5kJzenhDJROfhh7aIKiG/npK8VnYE0WOf2OjyJU6LE9OQ==
-END PUBLIC KEY-” > pk.pem
IFS=‘.’ read -r header payload signature <<< “$token”
echo “$signature” > signature.bin
echo “${payload}”|openssl dgst -sha256 -binary > payload_hash.bin
openssl pkeyutl -verify -inkey pk.pem -pubin -sigfile signature.bin -in
payload_hash.bin

But it fails verification. The same token passes verification on
https://jwt.io
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Re: [OpenSIPS-Users] Re-Invites being ignored

[Saint Michael]

And how do that? Isn't this done already, automatically?

On Fri, Feb 3, 2023, 2:42 AM Răzvan Crainea  wrote:

> Hi, Michael!
>
> You can't (and shouldn't, at least not in an easy way) respond to
> re-INVITE from a proxy - your best chance is to route the re-INVITE down
> 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.
> > 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?
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Re-Invites being ignored

[Saint Michael]

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?

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[OpenSIPS-Users] Version 3.1 it stops writing cdrs

[Saint Michael]

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.
This is the script that I rn every 60 seconds

#move .csv to .csx , flat_rotate, move .csx & process them
LOCKFILE="/var/run/opensips_cdr_worker.lock"
OPENSIPSCTL=/usr/local/bin/opensips-cli

source /etc/profile
lockfile -r 0 $LOCKFILE
if [ $? -ne 0 ] ; then
echo "ERROR FROM LOCKFILE ${LOCKFILE}"
rm -f ${LOCKFILE}; exit
fi
trap "rm -f $LOCKFILE; exit" INT TERM EXIT

ip=`hostname -I | cut -d " " -f1`

cd /cdr

if [[ -z "$(ls -1 *.csv 2>/dev/null)" ]] ; then echo "no files to
process";exit;fi
...
/usr/bin/timeout -k 5 5 ${OPENSIPSCTL} -x mi flat_rotate

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Re: [OpenSIPS-Users] RTPPROXY / OPENSIPS

[Saint Michael]

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 should happen during either INVITE 
> , 200 OK or ACK.
> In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer 
> during the invite and rtpproxy_answer during the 200 OK.
>
> "Documentation : Rewrites SDP body to ensure that media is passed through an 
> RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 
> 200 OK and on 200 OK when SDPs are in 200 OK and ACK."
>
> But sometimes opensips receives a 183 Session Progress containing SDP before 
> the 200 which i think is related to earlymedia .
> I think those sdp packets should also be rewritten with the right rtp proxy 
> ip/port. In which case the doc should mention "SDPs are in INVITE and 200 OK 
> /183 Session Progress".
>
> My second question : since we can handle most cases with 
> rtpproxy_offer/answer methods, what is the purpose of rtpproxy_engage ? is it 
> to ease the management i described above ?
>
> Thank you
>
>
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Re: [OpenSIPS-Users] RTPProxy Docker Image

[Saint Michael]

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 is going to use a lot of ports in 
> a very large range. That just doesn't work great in docker, but even worse in 
> K8S.
>
> I personally would put the RTPProxy outside of K8S. While you might be able 
> to get it to work, you are likely going against some basic design concepts in 
> containerization. I feel like the tech should propel the solution and not be 
> a hindrance to it. In this case, I'm not sure that K8S is buying you anything 
> of value, but instead creating architectural challenges.
>
> I'd love to hear feedback or experiences from others. There's always 
> something to learn :)
> -Brett
>
> On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor  wrote:
>>
>> Was it something I said?
>>
>> Terrance
>>
>> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor  wrote:
>>>
>>> Hello Everyone,
>>>
>>> Wow! Blast from the past... I am a long time member of this list, been a 
>>> while.
>>>
>>> Question, anyone successful in deploying RTPProxy to a dockerized 
>>> environment? Preferably to a Kubernetes managed environment.
>>>
>>> Please Help Team :)
>>>
>>> Kind Regards,
>>> Terrance
>>
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[OpenSIPS-Users] CDR stopped writing files

[Saint Michael]

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 this and what can I do to preclude this
from happening again?

I checked the directory "/cdr" where opensips is supposed to write CDR
it remained empty.

I saved the syslog messages from opensips but there was nothing that
would indicate a problem.

Yours
Philip

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Re: [OpenSIPS-Users] dialog behavior when no ACK is received

[Saint Michael]

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.
> So in a scenario where a caller sends an INVITE and then dies, the call goes 
> to callee, gets answered, OK is sent to caller but there is no response, and 
> the dialog stays open until default_timeout and that's when the CDR is 
> generated (if default_timeout is 12 hours then a 12 hour call is in CDR).
> Anybody has any ideas how we can fix this? If I use PpB for dialog, it never 
> PINGs the callee because apparently the dialog is not fully established on 
> the callee side. If I use rtpengine to detect media timeout, it correctly 
> detects it but when it calls dlg_end_dlg opensips says "dialog not found".
>
> Thank you!
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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

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/bin/opensips-cli -x mi set_gflag 0
> ERROR: command 'set_gflag' returned: 400: Bad parameter value
>
> if we somehow get through that issue, then the code would be?
>
> if (is_gflag(0)) {
> xlog("$ci Shutting Down\r\n");
> sl_send_reply(503,"Temporary Failure");
> append_to_reply(503,"Temporary Failure\r\n");
> exit;
> else {
> xlog("$ci Gflag is not set\r\n");
> }
>
> Sorry for being so pushy. I am open to pay consultancy fees. It's just
> that I already lost a lot of money because I killed opensips with 700
> open calls and lost the billing.
>
>
> On Thu, Oct 27, 2022 at 12:21 PM Bogdan-Andrei Iancu
>  wrote:
> >
> > OK, let's do a "back to school" when comes to binary representation and ops
> >
> > Decimal 15 (what you set for initial) is , so all bits (index 0-3)
> > are set to 1, to true. So, when you set (to value 1) the gflag with idx
> > "1" , nothing will be actually changed as it is already 1 :P
> >
> > 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 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
> > > modparam("gflags", "initial", 15)
> > >
> > > Then:
> > > /usr/local/bin/opensips-cli -x mi set_gflag 1
> > > "OK"
> > > /usr/local/bin/opensips-cli -x mi get_gflags
> > > {
> > >  "hex": "0xF",
> > >  "dec": "15"
> > > }
> > >
> > > Maybe I am not following the whole thing. I apologize.
> > > I imagine that we have a 32 bit field, and we set a bit to 1 for bits 
> > > 0..31.
> > >
> > > On Thu, Oct 27, 2022 at 11:32 AM Bogdan-Andrei Iancu
> > >  wrote:
> > >> Ok, docs fixed, thanks Karsten for spotting the error [0]
> > >>
> > >> Michael, that;s right, if the gflag is set, just reject the code with
> > >> 503 Service Unavailable . You also may consider adding a Retry-After
> > >> header [1] using the append_to_reply() function.
> > >>
> > >> [0]
> > >> https://github.com/OpenSIPS/opensips/commit/5945746b09b480dd0b7f8ea4b3584e4521bdac3a
> > >> [1] https://www.ietf.org/rfc/rfc3261.html#section-20.33
> > >> [2]
> > >> https://opensips.org/html/docs/modules/3.2.x/sipmsgops.html#func_append_to_reply
> > >>
> > >> 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 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
> > >>> value matches, to reject the call:
> > >>> I propose
> > >>> {
> > >>> sl_send_reply(503,"Declined");
> > >>>   exit;
> > >>> }
> > >>>
> > >>>
> > >>> On Thu, Oct 27, 2022 at 9:04 AM Karsten Wemheuer  wrote:
> > >>>> 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");
> > >>>>>   } else {
> > >>>>>   t_relay("udp:10.0.0.2:5060");
> > >>>>>   }
> > >>>> I think, You should remove the quotes around the "1".
> > >>>>
> > >>>> I think, the example in the docs may be wrong (Your code was from the
> > >>>> docs, I think). See section 1.4.3, where
> > >>>>   if(is_gflag(4))
> > >>>> is written.
> > >>>>
> > >>>> HTH,
> > >>>>
> > >>>> Karsten
> > >>>>
> > >>>>
> > >>>>
> > >>>> ___
> > >>>> Users mailing list
> > >>>> Users@lists.opensips.org
> > >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >

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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

Thanks Bogdan-Andrei

however, if I try to set the whole flag to 0 (from an initial value of
15), I get

/usr/local/bin/opensips-cli -x mi set_gflag 0
ERROR: command 'set_gflag' returned: 400: Bad parameter value

if we somehow get through that issue, then the code would be?

if (is_gflag(0)) {
xlog("$ci Shutting Down\r\n");
sl_send_reply(503,"Temporary Failure");
append_to_reply(503,"Temporary Failure\r\n");
exit;
else {
xlog("$ci Gflag is not set\r\n");
}

Sorry for being so pushy. I am open to pay consultancy fees. It's just
that I already lost a lot of money because I killed opensips with 700
open calls and lost the billing.


On Thu, Oct 27, 2022 at 12:21 PM Bogdan-Andrei Iancu
 wrote:
>
> OK, let's do a "back to school" when comes to binary representation and ops
>
> Decimal 15 (what you set for initial) is , so all bits (index 0-3)
> are set to 1, to true. So, when you set (to value 1) the gflag with idx
> "1" , nothing will be actually changed as it is already 1 :P
>
> 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 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
> > modparam("gflags", "initial", 15)
> >
> > Then:
> > /usr/local/bin/opensips-cli -x mi set_gflag 1
> > "OK"
> > /usr/local/bin/opensips-cli -x mi get_gflags
> > {
> >  "hex": "0xF",
> >  "dec": "15"
> > }
> >
> > Maybe I am not following the whole thing. I apologize.
> > I imagine that we have a 32 bit field, and we set a bit to 1 for bits 0..31.
> >
> > On Thu, Oct 27, 2022 at 11:32 AM Bogdan-Andrei Iancu
> >  wrote:
> >> Ok, docs fixed, thanks Karsten for spotting the error [0]
> >>
> >> Michael, that;s right, if the gflag is set, just reject the code with
> >> 503 Service Unavailable . You also may consider adding a Retry-After
> >> header [1] using the append_to_reply() function.
> >>
> >> [0]
> >> https://github.com/OpenSIPS/opensips/commit/5945746b09b480dd0b7f8ea4b3584e4521bdac3a
> >> [1] https://www.ietf.org/rfc/rfc3261.html#section-20.33
> >> [2]
> >> https://opensips.org/html/docs/modules/3.2.x/sipmsgops.html#func_append_to_reply
> >>
> >> 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 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
> >>> value matches, to reject the call:
> >>> I propose
> >>> {
> >>> sl_send_reply(503,"Declined");
> >>>   exit;
> >>> }
> >>>
> >>>
> >>> On Thu, Oct 27, 2022 at 9:04 AM Karsten Wemheuer  wrote:
> >>>> 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");
> >>>>>   } else {
> >>>>>   t_relay("udp:10.0.0.2:5060");
> >>>>>   }
> >>>> I think, You should remove the quotes around the "1".
> >>>>
> >>>> I think, the example in the docs may be wrong (Your code was from the
> >>>> docs, I think). See section 1.4.3, where
> >>>>   if(is_gflag(4))
> >>>> is written.
> >>>>
> >>>> HTH,
> >>>>
> >>>> Karsten
> >>>>
> >>>>
> >>>>
> >>>> ___
> >>>> Users mailing list
> >>>> Users@lists.opensips.org
> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

this code does not seem to work: I set the initial value to 15, inside
opensips.cfg
modparam("gflags", "initial", 15)

Then:
/usr/local/bin/opensips-cli -x mi set_gflag 1
"OK"
/usr/local/bin/opensips-cli -x mi get_gflags
{
"hex": "0xF",
"dec": "15"
}

Maybe I am not following the whole thing. I apologize.
I imagine that we have a 32 bit field, and we set a bit to 1 for bits 0..31.

On Thu, Oct 27, 2022 at 11:32 AM Bogdan-Andrei Iancu
 wrote:
>
> Ok, docs fixed, thanks Karsten for spotting the error [0]
>
> Michael, that;s right, if the gflag is set, just reject the code with
> 503 Service Unavailable . You also may consider adding a Retry-After
> header [1] using the append_to_reply() function.
>
> [0]
> https://github.com/OpenSIPS/opensips/commit/5945746b09b480dd0b7f8ea4b3584e4521bdac3a
> [1] https://www.ietf.org/rfc/rfc3261.html#section-20.33
> [2]
> https://opensips.org/html/docs/modules/3.2.x/sipmsgops.html#func_append_to_reply
>
> 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 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
> > value matches, to reject the call:
> > I propose
> > {
> > sl_send_reply(503,"Declined");
> >  exit;
> > }
> >
> >
> > On Thu, Oct 27, 2022 at 9:04 AM Karsten Wemheuer  wrote:
> >> 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");
> >>>  } else {
> >>>  t_relay("udp:10.0.0.2:5060");
> >>>  }
> >> I think, You should remove the quotes around the "1".
> >>
> >> I think, the example in the docs may be wrong (Your code was from the
> >> docs, I think). See section 1.4.3, where
> >>  if(is_gflag(4))
> >> is written.
> >>
> >> HTH,
> >>
> >> Karsten
> >>
> >>
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] Problem with gflags

[Saint Michael]

this code does not seem to work: the initial value is 15, inside opensips.cfg

/usr/local/bin/opensips-cli -x mi set_gflag 1
"OK"
/usr/local/bin/opensips-cli -x mi get_gflags
{
"hex": "0xF",
"dec": "15"
}

Maybe I am not following the whole 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 not work
> if (is_gflag(1)) {
> xlog("$ci Shutting Down\r\n");
> }
>  else {
> xlog("$ci Gflag is not 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 wrote:
> > > I just need to know how to check for the gflag inside the code
> > > this:
> > >   if (is_gflag("1")) {..}
> > > does not work, even if the flag is (1), as I wrote this test from the
> > > command line works
> > >
> > > opensips-cli -x mi is_gflag 1
> > > true
> > >
> > > As you can see, unless you can test for the gflag value inside the
> > > 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.
> >
> > Script: is_gflag(0) (is bit "0" set?)
> > MI: is_gflag(1) (are all bits in bitmask "1" set?)
> >
> > --
> > Liviu Chircu
> > www.twitter.com/liviuchircu | www.opensips-solutions.com
> >

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Re: [OpenSIPS-Users] Problem with gflags

[Saint Michael]

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 not work
if (is_gflag(1)) {
xlog("$ci Shutting Down\r\n");
}
 else {
xlog("$ci Gflag is not 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 wrote:
> > I just need to know how to check for the gflag inside the code
> > this:
> >   if (is_gflag("1")) {..}
> > does not work, even if the flag is (1), as I wrote this test from the
> > command line works
> >
> > opensips-cli -x mi is_gflag 1
> > true
> >
> > As you can see, unless you can test for the gflag value inside the
> > 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.
>
> Script: is_gflag(0) (is bit "0" set?)
> MI: is_gflag(1) (are all bits in bitmask "1" set?)
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>

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Re: [OpenSIPS-Users] Problem with gflags

[Saint Michael]

I just need to know how to check for the gflag inside the code
this:
 if (is_gflag("1")) {..}
does not work, even if the flag is (1), as I wrote this test from the
command line works

opensips-cli -x mi is_gflag 1
true

As you can see, unless you can test for the gflag value inside the
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 wrote:
> > I only get the second xlog()
> >
> > The documentation may be wrong, it asks us to use
> > if (is_gflag("1")) {..}
> > But opensips fails to start.
> >
> > How do I test for gflag=1 inside my
> > route
>
> Hi SM,
>
> The set of functions in "gflags" module is a bit strange, however they
> are correct.  The script functions work with *bits* (e.g. bit 0, 1, 2,
> ...), while the MI functions work with *bitmasks* (e.g. 1, 2, 4, ...).
> Be careful not to mix the two, and you should get the correct results.
>
> I believe the docs are somewhat accurate about the above - if not, let
> me know and we can make improvements together.
>
> Best regards,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>

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[OpenSIPS-Users] Problem with gflags

[Saint Michael]

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");
}

I only get the second xlog()

The documentation may be wrong, it asks us to use
if (is_gflag("1")) {..}
But opensips fails to start.

How do I test for gflag=1 inside my
route
{

}

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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

YES
Maybe we should fix the documentation?
now the real question is, what goes inside the bracket if the flag
value matches, to reject the call:
I propose
{
sl_send_reply(503,"Declined");
exit;
}


On Thu, Oct 27, 2022 at 9:04 AM Karsten Wemheuer  wrote:
>
> 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");
> > } else {
> > t_relay("udp:10.0.0.2:5060");
> > }
>
> I think, You should remove the quotes around the "1".
>
> I think, the example in the docs may be wrong (Your code was from the
> docs, I think). See section 1.4.3, where
> if(is_gflag(4))
> is written.
>
> HTH,
>
> Karsten
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

for testing I added
route{
if (is_gflag("1")) {
t_relay("udp:10.0.0.1:5060");
} else {
t_relay("udp:10.0.0.2:5060");
}
but I see these errors

Oct 27 12:34:33 proxy opensips[140294]: Aliases:
Oct 27 12:34:33 proxy opensips[140296]: Oct 27 12:34:33 [140296]
ERROR:core:fix_cmd: Param [1] expected to be an integer or variable
Oct 27 12:34:33 proxy opensips[140296]: Oct 27 12:34:33 [140296]
ERROR:core:fix_actions: Failed to fix command 
Oct 27 12:34:33 proxy opensips[140296]: Oct 27 12:34:33 [140296]
ERROR:core:fix_actions: fixing failed (code=-6) at
/etc/opensips/opensips.cfg:126
Oct 27 12:34:33 proxy opensips[140296]: Oct 27 12:34:33 [140296]
CRITICAL:core:fix_expr: fix_actions error
Oct 27 12:34:33 proxy opensips[140296]: Oct 27 12:34:33 [140296]
ERROR:core:main: failed to fix configuration with err code -6
Oct 27 12:34:33 proxy systemd[1]: opensips.service: Control process
exited, code=exited, status=255/EXCEPTION
Oct 27 12:34:33 proxy systemd[1]: opensips.service: Failed with result
'exit-code'.

at the beginning of the code I added
loadmodule "gflags.so"
modparam("gflags", "initial", 0)

it does load with those two lines, but as soon as I try to use the
module, it blows up.



On Thu, Oct 27, 2022 at 2:55 AM Bogdan-Andrei Iancu  wrote:
>
> To be more specific here, use the suggested gflags to test for initial INVITE 
> and reject any new call if the gflag is set - and you can toggle the flag via 
> MI cmds.
>
> 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 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 terminates all open calls in an
>> orderly fashion.
>> The question is, before I execute my script, is there any similar
>> command (like opensips-cli -x mi dlg_end_dlg "${callid}") that would
>> stop taking new calls?  Like "reject any call attempts with 503 or
>> 404", because, in a high-pressure environment, 2000 new calls may
>> arrive before I close all calls and shut down Opensips.
>> I use version:
>> opensips -V
>> version: opensips 3.1.12 (x86_64/linux)
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll, sigio_rt, select.
>> git revision: 15ef14a89
>> main.c compiled on 03:16:12 Oct 25 2022 with gcc 9
>>
>> On Wed, Oct 26, 2022 at 10:49 AM Bogdan-Andrei Iancu
>>  wrote:
>> >
>> > Hi,
>> >
>> > Sending a sig TERM to the opensips processes will trigger a controlled
>> > shutdown of opensips (meaning with flushing data to DB, if needed,
>> > cleanup, etc). Nevertheless, the opensips shutdown does not ends the
>> > call, as the calls may be continued after restarting opensips.
>> >
>> > Still, if you want to terminate all ongoing calls before shutting down,
>> > better use dialog module and check these MI cmds:
>> > https://opensips.org/html/docs/modules/3.2.x/dialog.html#mi_dlg_end_dlg
>> > https://opensips.org/html/docs/modules/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 10/25/22 2:15 PM, Saint Michael wrote:
>> > > 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/cgi-bin/mailman/listinfo/users
>> >
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

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Re: [OpenSIPS-Users] Orderly shutdown

[Saint Michael]

Dear Friends
I successfully wrote a script that terminates all open calls in an
orderly fashion.
The question is, before I execute my script, is there any similar
command (like opensips-cli -x mi dlg_end_dlg "${callid}") that would
stop taking new calls?  Like "reject any call attempts with 503 or
404", because, in a high-pressure environment, 2000 new calls may
arrive before I close all calls and shut down Opensips.
I use version:
opensips -V
version: opensips 3.1.12 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 15ef14a89
main.c compiled on 03:16:12 Oct 25 2022 with gcc 9

On Wed, Oct 26, 2022 at 10:49 AM Bogdan-Andrei Iancu
 wrote:
>
> Hi,
>
> Sending a sig TERM to the opensips processes will trigger a controlled
> shutdown of opensips (meaning with flushing data to DB, if needed,
> cleanup, etc). Nevertheless, the opensips shutdown does not ends the
> call, as the calls may be continued after restarting opensips.
>
> Still, if you want to terminate all ongoing calls before shutting down,
> better use dialog module and check these MI cmds:
> https://opensips.org/html/docs/modules/3.2.x/dialog.html#mi_dlg_end_dlg
> https://opensips.org/html/docs/modules/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 10/25/22 2:15 PM, Saint Michael wrote:
> > 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/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] Local Route question

[Saint Michael]

Thank you for the clarification.
But then how do you intercept the Connect?
Where in the code?
I need to fire a db query when the call connects.
Is this even possible?



On Wed, Oct 26, 2022, 10:51 AM Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> please see
> https://www.opensips.org/Documentation/Script-Routes-3-2#local_route -
> the local route is triggered only for internally generated (UAC)
> requests, so you will never have replies over there.
>
> 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 10/26/22 5:28 AM, Saint Michael wrote:
> > 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 and not any other kind of 200?
> > Is this doable?
> >
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Local Route question

[Saint Michael]

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 and not any other kind of 200?
Is this doable?

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[OpenSIPS-Users] How long can the callID be in SIP

[Saint Michael]

I just caught a callID that is 160 characters long.
What is the max length that it should be? I am using 128.

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Re: [OpenSIPS-Users] 603 not treated as a failure

[Saint Michael]

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 and I have a scenario where we will
> receive a 603 to an invite. this needs to be failed over to the next
> destination. we are using dispatcher for this.
>
> While testing I am finding that opensips is treating the 603 a a relpy and
> not as a failure and its being handled by the reply route and not by the
> failure route. passing the message back to the UAC..
>
> Is this by design or is there some setting where the 603 can be treated as
> a failure and be handled by the specified failure route and failed over to
> the next destination.?
>
>
> Regards,
>
>
> Richard
>
>
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[OpenSIPS-Users] Orderly shutdown

[Saint Michael]

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.

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Re: [OpenSIPS-Users] Error

[Saint Michael]

opensips -V
version: opensips 3.1.11 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll 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 AM Saint Michael  wrote:
>>
>> 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,flags=0,data=(nil)) -> removing
>> from epoll
>>
>>
>> Philip
>>
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[OpenSIPS-Users] Error

[Saint Michael]

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,flags=0,data=(nil)) -> removing
from epoll


Philip

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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Saint Michael]

I use opensips 3.1, does it matter?


On Mon, Sep 26, 2022 at 10:20 AM Daniel Zanutti 
wrote:

> can you write your own functions with opensips?
> Yes -> using routes
>
> Can you point me to an example?
> Storing->
> https://opensips.org/html/docs/modules/2.2.x/dialog.html#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 Michael  wrote:
>
>> 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 Zanutti 
>>> wrote:
>>>
>>>> You have to use dialog variable storing.
>>>> Take a look at dialog module.
>>>>
>>>> Em dom., 25 de set. de 2022 10:42, Saint Michael 
>>>> escreveu:
>>>>
>>>>> 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);
>>>>> but I need this value to disappear when this call is closed. I cannot
>>>>> set an expiration because the call may last for 2 hours or 2 seconds.
>>>>>
>>>>> many thanks for your help and guidance
>>>>>
>>>>> Philip
>>>>>
>>>>>
>>>>> ___
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>>>>>
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>>>>
>>> ___
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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Saint Michael]

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 Zanutti 
> wrote:
>
>> You have to use dialog variable storing.
>> Take a look at dialog module.
>>
>> Em dom., 25 de set. de 2022 10:42, Saint Michael 
>> escreveu:
>>
>>> 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);
>>> but I need this value to disappear when this call is closed. I cannot
>>> set an expiration because the call may last for 2 hours or 2 seconds.
>>>
>>> many thanks for your help and guidance
>>>
>>> Philip
>>>
>>>
>>> ___
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>>
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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Saint Michael]

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, Saint Michael 
> escreveu:
>
>> 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);
>> but I need this value to disappear when this call is closed. I cannot set
>> an expiration because the call may last for 2 hours or 2 seconds.
>>
>> many thanks for your help and guidance
>>
>> Philip
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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[OpenSIPS-Users] Question about cache_store and lifetime of variables

[Saint Michael]

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);
but I need this value to disappear when this call is closed. I cannot set
an expiration because the call may last for 2 hours or 2 seconds.

many thanks for your help and guidance

Philip
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Re: [OpenSIPS-Users] Call extender technology failing to work

[Saint Michael]

It worked fine, many thanks.
I am very grateful for this list.

On Thu, Sep 22, 2022 at 11:44 AM Jonathan Abrams  wrote:

> Hi Michael,
>
> It looks like you may need to send an OK to the caller side before
> invoking the asynchronous sleep. Something like:
>
> sl_send_reply(200, "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.
>>
>> This is the issue: the call extender technology does not confirm the BYE
>> to the caller
>>  if ($avp(hold_seconds) > 0) {
>> $avp(total_duration) =
>> $DLG_lifetime + $avp(hold_seconds);
>>
>> cache_store("local","start_$ci","$dlg_val(start_time)",180);
>>
>> cache_store("local","duration_$ci","$avp(total_duration)",180);
>> async( sleep($avp(hold_seconds)),
>> bye_resume );
>> exit;
>> }
>> if that code executes, because hold_seconds > 0, then we don't confim the
>> BYE and we keep
>> getting more BYEs.
>> The call extender technology should confirm the BYE to the caller as
>> normally, and it does not. It should hold the call open towards the callee,
>> not the caller. Can you suggest a change in the code?
>> I guess the new line if code should  be called before
>> $avp(total_duration) = $DLG_lifetime + $avp(hold_seconds);
>>
>> Many thanks for your help
>>
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[OpenSIPS-Users] Call extender technology failing to work

[Saint Michael]

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) {
$avp(total_duration) =
$DLG_lifetime + $avp(hold_seconds);

cache_store("local","start_$ci","$dlg_val(start_time)",180);

cache_store("local","duration_$ci","$avp(total_duration)",180);
async( sleep($avp(hold_seconds)),
bye_resume );
exit;
}
if that code executes, because hold_seconds > 0, then we don't confim the
BYE and we keep
getting more BYEs.
The call extender technology should confirm the BYE to the caller as
normally, and it does not. It should hold the call open towards the callee,
not the caller. Can you suggest a change in the code?
I guess the new line if code should  be called before
$avp(total_duration) = $DLG_lifetime + $avp(hold_seconds);

Many thanks for your help
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

I added this code to the end of my script
local_route
{
  if (is_method("BYE"))
  {
  xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
  }
}
and now my system is filed with these errors:
Sep 15 14:25:36 node5 opensips[238750]: Sep 15 14:25:35 [238750]
LOCAL_ROUTE - BYE - RTPProxy Timeout - sip:7867830513@208.73.234.96:5060 -
0e87838610bfe6e1670023aa6bbe9590@208.73.234.96Sep 15 14:25:35 [238750]
LOCAL_ROUTE - BYE - RTPProxy Timeout - sip:192.69.217.154;did=8fc1.e2444961
- 0e87838610bfe6e1670023aa6bbe9590@208.73.234.96
but my calls do not use the rtpproxy, so why is this affecting my traffic
and killing my calls?
I have 10 rttpproxy services:
grep rtpproxy
  rtpproxy1.serviceloaded active running
RTPProxy1
  rtpproxy10.service   loaded active running
RTPProxy10
  rtpproxy2.serviceloaded active running
RTPProxy2
  rtpproxy3.serviceloaded active running
RTPProxy3
  rtpproxy4.serviceloaded active running
RTPProxy4
  rtpproxy5.serviceloaded active running
RTPProxy5
  rtpproxy6.serviceloaded active running
RTPProxy6
  rtpproxy7.serviceloaded active running
RTPProxy7
  rtpproxy8.serviceloaded active running
RTPProxy8
  rtpproxy9.serviceloaded active running
RTPProxy9

here is number 1
[Unit]
Description=RTPProxy1
After=network.target
Requires=network.target

[Service]
Type=forking
PIDFile=/var/run/rtpproxy1.pid
#Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d
INFO:LOG_LOCAL5'

Restart=on-failure
RestartSec=5


ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy1.pid -l
Public.Ip.Address \
   -s udp:127.0.0.1:7890 -F -L 10240 -m 1 -M 15000 -T 20 -d
WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
ExecStop=/usr/bin/pkill -F /var/run/rtpproxy1.pid


StandardOutput=syslog
StandardError=syslog
SyslogIdentifier=rtpproxy1
SyslogFacility=local5

TimeoutStartSec=10
TimeoutStopSec=10

[Install]
WantedBy=multi-user.target


Here is number 2

[Unit]
Description=RTPProxy2
After=network.target
Requires=network.target

[Service]
Type=forking
PIDFile=/var/run/rtpproxy2.pid
#Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d
INFO:LOG_LOCAL5'

Restart=on-failure
RestartSec=5


ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy2.pid -l
Public.IP.address \
   -s udp:127.0.0.1:7891 -F -L 10240 -m 15000 -M 2 -T 20 -d
WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
ExecStop=/usr/bin/pkill -F /var/run/rtpproxy2.pid


StandardOutput=syslog
StandardError=syslog
SyslogIdentifier=rtpproxy2
SyslogFacility=local5

TimeoutStartSec=10
TimeoutStopSec=10

[Install]
WantedBy=multi-user.target


what am I doing wrong?

Federico





On Wed, Sep 14, 2022 at 4:55 PM Daniel Zanutti 
wrote:

> Hi
>
> Everytime opensips sends the BYE, it's generated inside local_route:
> https://www.opensips.org/Documentation/Script-Routes-3-1#toc6
>
> So put a xlog there to see why. Something like this:
> local_route
> {
>   if (is_method("BYE"))
>   {
>   xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
>   }
> }
>
>
> 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...@gmail.com>
>> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
>> *Aan:* OpenSIPS users mailling list 
>> *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all
>> calls fail after 20 seconds, how do I go back?
>>
>> how do I do this:
>> " put some log on local_route"
>> Sorry I am learning
>>
>>
>> On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
>> wrote:
>>
>> So your Opensips is hanging up the call.
>>
>> Do you see any log on it? Try put some log on local_route if you don't
>> see anything.
>>
>>
>>
>> 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, 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?
>>
>>
>>
>> __

Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

I will, thanks.


On Wed, Sep 14, 2022, 4:55 PM Daniel Zanutti 
wrote:

> Hi
>
> Everytime opensips sends the BYE, it's generated inside local_route:
> https://www.opensips.org/Documentation/Script-Routes-3-1#toc6
>
> So put a xlog there to see why. Something like this:
> local_route
> {
>   if (is_method("BYE"))
>   {
>   xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
>   }
> }
>
>
> 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...@gmail.com>
>> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
>> *Aan:* OpenSIPS users mailling list 
>> *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all
>> calls fail after 20 seconds, how do I go back?
>>
>> how do I do this:
>> " put some log on local_route"
>> Sorry I am learning
>>
>>
>> On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
>> wrote:
>>
>> So your Opensips is hanging up the call.
>>
>> Do you see any log on it? Try put some log on local_route if you don't
>> see anything.
>>
>>
>>
>> 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, 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?
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

how do I do this:
" put some log on local_route"
Sorry I am learning


On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
wrote:

> So your Opensips is hanging up the call.
>
> Do you see any log on it? Try put some log on local_route if you don't see
> anything.
>
>
>
> 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, 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?
>>>
>>>
>>>
>> ___
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>>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

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/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?
>
>
>
INVITE sip:1957408@208.78.161.135:5060 SIP/2.0
Via: SIP/2.0/UDP 
38.95.11.9:5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f;rport
Max-Forwards: 70
Contact: 
To: 
From: "7869440397";tag=fa388212
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, 
INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 282

v=0
o=3cxPS 9251273505243136 8956523271159809 IN IP4 38.95.11.9
s=3cxPS Audio call
c=IN IP4 38.95.11.9
t=0 0
m=audio 9092 RTP/AVP 0 8 9 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 100 Giving it a try
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;branch=z9hG4bK-524287-1---8d06285b2fbb097f;rport=5060
To: 
From: "7869440397";tag=fa388212
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Server: OpenSIPS (3.1.11 (x86_64/linux))
Content-Length: 0

INVITE sip:1957408@216.82.236.252:5060 SIP/2.0
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
t: 1957408 
f: 15617596254;tag=fa388212
CSeq: 1 INVITE
i: KNCLJbZ-sV2tfAQT5Qv7Ig..
m: 
Max-Forwards: 69
c: application/sdp
l: 195
k: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, 
INFO, MESSAGE, UPDATE
Identity: 
eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9hei50YXgvMSJ9.eyJhdHRlc3QiOiJCIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNjYzMTgzMzU0LCJvcmlnIjp7InRuIjoiMTU2MTc1OTYyNTQifSwib3JpZ2lkIjoiMzU4ZDA0OGItZTAwNy0xMWVhLWFjMzctMzgtOTUtMTEtOSJ9.E5pg-DMkY06sGwFTK5JniLaEBsRhEMS192Zb1OIbdAzN0tWopRxWNRWvBrLxECb5juZWtPSNu3m_VgCDvUYi9A;info=<https://az.tax/1>;alg=ES256;ppt=shaken

v=0
o=3cxPS 9251273505243136 8956523271159809 IN IP4 38.95.11.9
s=3cxPS Audio call
c=IN IP4 38.95.11.9
t=0 0
m=audio 9092 RTP/AVP 0 8 9 3 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 100 Trying
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Content-Length: 0

SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length:   238
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 216.82.236.252
s=SIP Media Capabilities
c=IN IP4 216.82.236.27
t=0 0
m=audio 30648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;rport=5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f
From: "7869440397";tag=fa388212
To: ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 257
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 208.78.161.135
s=SIP Media Capabilities
c=IN IP4 208.78.161.135
t=0 0
m=audio 39112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=nortpproxy:yes
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length:   238
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 216.82.236.252
s=SIP Media Capabilities
c=IN IP4 216.82.236.27
t=0 0
m=audio 30648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;rport=5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f
From: "7869440397";tag=fa388212
To: ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 257
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 208.78.161.135
s=SIP Media Capabilities
c=IN IP4 

[OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

>
> 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?
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Re: [OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Saint Michael]

Is there way to obtain the SIP Call ID from the "id" or when is_callid=0?
if I save the callID on a global variable, will it be visible around all
the code?
Sorry I am still learning the ropes.




On Wed, Sep 14, 2022 at 9:38 AM Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> As per docs [1], you have in event_route :
>
>  id - represents the identifier of the call for which that event was
> received.
>
>  is_callid - is 0 if the id parameter represents the Dialog ID, or 1 if it
> is a callid.
>
> [1]
> https://opensips.org/html/docs/modules/3.2.x/rtpproxy.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, Saint Michael wrote:
>
> 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
> connected
> I need to fire a database query to close the call.
> I imagine that when the call got connected, some variable stored that
> unix-time .
> But it does not seem to be available in this part of the code.
> Philip
>
>
>
>
>
>
>>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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[OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Saint Michael]

>
> 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 connected
I need to fire a database query to close the call.
I imagine that when the call got connected, some variable stored that
unix-time .
But it does not seem to be available in this part of the code.
Philip






>
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Re: [OpenSIPS-Users] Failed to engage rtpproxy for trunk

[Saint Michael]

The public IP is the only public IP of the box.
I have 10 processes, does this configuration look normal?
grep udp /etc/systemd/system/rtpproxy*.service
rtpproxy1.service:   -s udp:127.0.0.1:7890 -F -L 10240 -m 1 -M
15000 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy2.service:   -s udp:127.0.0.1:7891 -F -L 10240 -m 15000 -M
2 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy3.service:   -s udp:127.0.0.1:7892 -F -L 10240 -m 2 -M
25000 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy4.service:   -s udp:127.0.0.1:7893 -F -L 10240 -m 25000 -M
3 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy5.service:   -s udp:127.0.0.1:7894 -F -L 10240 -m 3 -M
35000 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy6.service:   -s udp:127.0.0.1:7895 -F -L 10240 -m 35000 -M
4 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy7.service:   -s udp:127.0.0.1:7896 -F -L 10240 -m 4 -M
45000 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy8.service:   -s udp:127.0.0.1:7897 -F -L 10240 -m 45000 -M
5 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy9.service:   -s udp:127.0.0.1:7898 -F -L 10240 -m 5 -M
55000 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
rtpproxy10.service:   -s udp:127.0.0.1:7899 -F -L 10240 -m 55000 -M
6 -T 20 -d WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889


On Tue, Jul 5, 2022 at 3:20 AM Răzvan Crainea  wrote:
>
> Hi, Michael!
>
> "port 0" is returned by RTPProxy when an 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
>
> On 7/4/22 15:57, Saint Michael wrote:
> > 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 this SDP from the carrier
> >
> > Content-Length: 209
> >
> >  v=0
> >  o=- 655206240 655206240 IN IP4 XXX.XX.XX.XX
> >  s=ENSResip
> >  c=IN IP4 XX.XX.XX.XX
> >  t=0 0
> >  m=audio 18634 RTP/AVP 0 101
> >  a=rtpmap:0 PCMU/8000
> >  a=rtpmap:101 telephone-event/8000
> >  a=fmtp:101 0-15
> >  a=ptime:20
> >
> > My proxy's configuration is
> >
> >   cat rtpproxy1.service
> > [Unit]
> > Description=RTPProxy1
> > After=network.target
> > Requires=network.target
> >
> > [Service]
> > Type=forking
> > PIDFile=/var/run/rtpproxy1.pid
> > #Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d 
> > INFO:LOG_LOCAL5'
> >
> > Restart=on-failure
> > RestartSec=5
> >
> >
> > ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy1.pid -l pub.lic.i.p \
> > -s udp:127.0.0.1:7890 -F -L 10240 -m 1 -M 15000 -T 20 -d
> > WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
> > ExecStop=/usr/bin/pkill -F /var/run/rtpproxy1.pid
> >
> >
> > StandardOutput=syslog
> > StandardError=syslog
> > SyslogIdentifier=rtpproxy1
> > SyslogFacility=local5
> >
> > TimeoutStartSec=10
> > TimeoutStopSec=10
> >
> > [Install]
> > WantedBy=multi-user.target
> > --
> > version:
> > version: opensips 3.1.10 (x86_64/linux)
> > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> > Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> > MAX_URI_SIZE 1024, BUF_SIZE 65535
> > poll method support: poll, epoll, sigio_rt, select.
> > git revision: 45c8875d5
> > main.c compiled on 12:08:55 Jul  4 2022 with gcc 9
> > 
> > there is a single call open, no traffic, for this is a development box
> >
> >
> > Any idea?
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Failed to engage rtpproxy for trunk

[Saint Michael]

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 this SDP from the carrier

Content-Length: 209

v=0
o=- 655206240 655206240 IN IP4 XXX.XX.XX.XX
s=ENSResip
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 18634 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

My proxy's configuration is

 cat rtpproxy1.service
[Unit]
Description=RTPProxy1
After=network.target
Requires=network.target

[Service]
Type=forking
PIDFile=/var/run/rtpproxy1.pid
#Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d INFO:LOG_LOCAL5'

Restart=on-failure
RestartSec=5


ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy1.pid -l pub.lic.i.p \
   -s udp:127.0.0.1:7890 -F -L 10240 -m 1 -M 15000 -T 20 -d
WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
ExecStop=/usr/bin/pkill -F /var/run/rtpproxy1.pid


StandardOutput=syslog
StandardError=syslog
SyslogIdentifier=rtpproxy1
SyslogFacility=local5

TimeoutStartSec=10
TimeoutStopSec=10

[Install]
WantedBy=multi-user.target
--
version:
version: opensips 3.1.10 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 45c8875d5
main.c compiled on 12:08:55 Jul  4 2022 with gcc 9

there is a single call open, no traffic, for this is a development box


Any idea?

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Re: [OpenSIPS-Users] Topology Hiding

[Saint Michael]

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 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
>>
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[OpenSIPS-Users] Topology Hiding

[Saint Michael]

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
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[OpenSIPS-Users] Errors in the machine's log

[Saint Michael]

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
asterisk.gatekeeper('48c77ca0077fe1de555db2f40c5a4...@zz.zz.zz.zz
','AAA7830513','AAA4536538','ZZ.ZZ.ZZ.ZZ')
ERROR:db_unixodbc:db_unixodbc_extract_error:
unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal]
ERROR:db_unixodbc:reconnect: Attempting DB reconnect
ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call
asterisk.gatekeeper('336f8bfc3d3e9eab7a624b507e7c7...@zz.zz.zz.zz
','AAA7830513','AAA3012713','ZZ.ZZ.ZZ.ZZ')
ERROR:db_unixodbc:db_unixodbc_extract_error:
unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal]
ERROR:db_unixodbc:reconnect: Attempting DB reconnect
ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call
asterisk.gatekeeper('06867d7363616c6c044c5...@zz.zz.zz.zz
','AAA5599174','AAA7284460','ZZ.ZZ.ZZ.ZZ')
ERROR:db_unixodbc:db_unixodbc_extract_error:
unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal]
ERROR:db_unixodbc:reconnect: Attempting DB reconnect
ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog

The next question is: I see a lot of
ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp
proxy
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -

But none of those inbound trunks are using media-proxy, I mean, they are
just going through the normal process for a call, where RTP goes straight
between the parties. Why do I get these errors? The box has sometimes 1
call attempts at the same time, and 10 RTP systemd services.

many thanks
Federico

SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
6b2197a60ed901941be16a5a10855...@zz.zz.zz.zz
ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
07744b7b64f5805e2a4cadda40dfe...@zz.zz.zz.zz
ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
3740afb80c04f1122b470b044e1cc...@zz.zz.zz.zz
ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp
proxy
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
584e46cd63616c6c044c7...@zz.zz.zz.zz
ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp
proxy
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
058056e963616c6c044c9...@zz.zz.zz.zz
ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call
asterisk.gatekeeper('7173b32375fb54d41bb96e60041bc...@zz.zz.zz.zz
','AAA7830513','1AAA4991471','ZZ.ZZ.ZZ.ZZ')
ERROR:db_unixodbc:db_unixodbc_extract_error:
unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal]
ERROR:db_unixodbc:reconnect: Attempting DB reconnect
ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp
proxy
SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ -
27f9b17b63616c6c044cb...@zz.zz.zz.zz
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Re: [OpenSIPS-Users] incompatibility leads to massive CDR loss

[Saint Michael]

Thanks.
It's my carrier. I have already opened a ticket.
Yours
Federico

On Fri, May 6, 2022 at 11:17 PM Daniel Zanutti 
wrote:

> I think this is your problem: branch=z9hG4bK-524287-1---b8aced18b4075aa3
> *=49972*
>
> You have char "=" inside a string, which is a 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.
>
>
> On Fri, May 6, 2022 at 4:42 PM Saint Michael  wrote:
>
>> 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-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll, sigio_rt, select.
>> git revision: 1a71fded7
>> main.c compiled on 13:37:30 May  2 2022 with gcc 9
>>
>> I need some paid help generating a patch or fixing this somehow.
>> I normally have a consultant but he may be tied up with the war and is
>> not responding.
>>
>> Yours
>> Federico
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[OpenSIPS-Users] incompatibility leads to massive CDR loss

[Saint Michael]

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-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 1a71fded7
main.c compiled on 13:37:30 May  2 2022 with gcc 9

I need some paid help generating a patch or fixing this somehow.
I normally have a consultant but he may be tied up with the war and is not
responding.

Yours
Federico
May 06 17:26:11 federico opensips[358436]: May  6 17:26:11 [358436] 
ERROR:core:parse_via: parsed so far:
May 06 17:26:11 federico opensips[358436]: May  6 17:26:11 [358436] 
ERROR:core:get_hdr_field: bad via
May 06 17:26:11 federico opensips[358436]: May  6 17:26:11 [358436] 
ERROR:core:parse_msg: message=;tag=29749fd7
May 06 17:26:11 federico opensips[358436]: t:9990118135950761 
;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02
May 06 17:26:11 federico opensips[358436]: 
i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY
May 06 17:26:11 federico opensips[358436]: m:
May 06 17:26:11 federico opensips[358436]: CSeq: 1 INVITE
May 06 17:26:11 federico opensips[358436]: Allow: 
ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
May 06 17:26:11 federico opensips[358436]: Server: sbc_5
May 06 17:26:11 federico opensips[358436]: l:0
May 06 17:26:11 federico opensips[358430]: May  6 17:26:11 [358430] 
ERROR:core:receive_msg: Unable to parse msg received from [ZZ.ZZ.ZZ.ZZ:5060]
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:parse_via_param: invalid char <=> in state 202
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:parse_via_param: parse_via_param
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:parse_via:  ;tag=29749fd7
May 06 17:26:42 federico opensips[358430]: t:9990118135950761 
;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02
May 06 17:26:42 federico opensips[358430]: 
i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY
May 06 17:26:42 federico opensips[358430]: Record-Route: 

May 06 17:26:42 federico opensips[358430]: m:
May 06 17:26:42 federico opensips[358430]: CSeq: 1 INVITE
May 06 17:26:42 federico opensips[358430]: Allow: 
ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
May 06 17:26:42 federico opensips[358430]: Server: sbc_5
May 06 17:26:42 federico opensips[358430]: c:application/sdp
May 06 17:26:42 federico opensips[358430]: l:361
May 06 17:26:42 federico opensips[358430]: 
May 06 17:26:42 federico opensips[358430]: v=0
May 06 17:26:42 federico opensips[358430]: o=SBCSIPUAS 1993104899 1 IN IP4 
206.20.217.23
May 06 17:26:42 federico opensips[358430]: s=SBCSIPUAS SIP STACK v1.0
May 06 17:26:42 federico opensips[358430]: c=IN IP4 206.20.217.23
May 06 17:26:42 federico opensips[358430]: t=0 0
May 06 17:26:42 federico opensips[358430]: m=audio 20556 RTP/AVP 0 101
May 06 17:26:42 federico opensips[358430]: a=rtpmap:0 PCMU/8000
May 06 17:26:42 federico opensips[358430]: a=rtpmap:101 telephone-event/8000
May 06 17:26:42 federico opensips[358430]: 
a=msi:mavodi-0-15b-13f-4--7a3c-5dc6608fbb33e-1168--@127.0.0.1-127.0.0.1&
May 06 17:26:42 federico opensips[358430]: a=fmtp:101 0-15
May 06 17:26:42 federico opensips[358430]: a=sendrecv
May 06 17:26:42 federico opensips[358430]: a=rtcp:36607
May 06 17:26:42 federico opensips[358430]: >
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:parse_via: parsed so far:
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:get_hdr_field: bad via
May 06 17:26:42 federico opensips[358430]: May  6 17:26:42 [358430] 
ERROR:core:parse_msg: message=;tag=29749fd7
May 06 17:26:42 federico opensips[358430]: t:9990118135950761 
;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02
May 06 17:26:42 federico opensips[358430]: 
i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY
May 06 17:26:42 federico opensips[358430]: Record-Route: 

May 06 17:26:42 federico opensips[358430]: m:
May 06 17:26:42 federico opensips[358430]: CSeq: 1 INVITE
May 06 17:26:42 federico opensips[358430]: Allow: 
ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
May 06 17:26:42 federico opensips[358430]: Server: sbc_5
May 06 17:26:42 federico opensips[358430]: c:application/sdp
May 06 17:26:42 federico opensips[358430]: l:361
May 06 17:26:42 federico opensips[358430]: 
May 06 17:26:42 federico opensips[358430]: v=0
May 06 17:26:42 federico opensips[358430]: o=SBCSIPUAS 1993104899 1 IN IP4 
206.20.217.23
May 06 17:26:42 federico opensips[358430]: s=SBCSIPUAS SIP STACK v1.0
May 06 17:26:42 federico opensips[358430]: c=IN IP4 206.20.217.23
May 06 17:26:42 federico opensips[358430]: t=0 0
May 06 17:26:42 federico opensips[358430]: 

[OpenSIPS-Users] CDR not generated on 302 redirect

[Saint Michael]

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?
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Re: [OpenSIPS-Users] OpenSIPS 3.2 + MySQL - do_accounting()

[Saint Michael]

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 doing this because my carrier was showing a lot more than I did in
minutes. The CDR written to disk the traditional way also shows no record
of so many calls. In 541000 calls I have no accounting for 80.000.
nightmare.
Any idea what can it be?
Vlad is my advisor, please copy him on the information.


On Tue, Mar 1, 2022 at 9:31 AM Bogdan-Andrei Iancu 
wrote:

> Hi Eugen,
>
> What is the capture of the call that produced that CDR ? Can you
> reproduce it ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS eBootcamp
>https://www.opensips.org/Training/Bootcamp
>
> On 2/23/22 11:21 PM, Eugen Prieb via Users wrote:
> > Hello,
> >
> > I will collect all CDRs, also failled, in DB.
> >
> > I see in log follow message:
> > Feb 23 21:58:33 opensips-32 /sbin/opensips[2623292]:
> > CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error
> > (1048): Column 'to_tag' cannot be null
> >
> > As MySQL is MariaDB 10.5.12 installed. Have you any idee?
> >
> >
>
>
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Re: [OpenSIPS-Users] acc not writing INVITE to db

[Saint Michael]

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 4:44 PM Marcin Groszek  wrote:

> You may disregard. I have located the issue.
>
> --
> Best Regards:
> Marcin Groszek
> Business Voip Resource.
> http://www.voipplus.net
>
>
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[OpenSIPS-Users] Bug found

[Saint Michael]

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 when SDP present"

The reinvite shows:
 Message Header
Via: SIP/2.0/UDP xxx.xx.xxx.x:5060;branch=z9hG4bKf611.5be13266.0
To: ;tag=as7f8965f0
From: ;tag=gK06f2e8a9
CSeq: 16844 INVITE
Call-ID:
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-
[Generated Call-ID:
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-]
Max-Forwards: 67
X-: 231
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, PRACK, BYE, CANCEL, OPTIONS, UPDATE
User-Agent: SIPMAV

It should be "Content-Length: 231"
I am attaching the full trace. In the file, the issue is at line 636.
I am using these special modules:
loadmodule "topology_hiding.so"
loadmodule "compression.so"
No. Time   SourceDestination   Protocol 
Length Info
  1 0.00   yy.yyy.yyy.yy xxx.xx.xxx.x  SIP/SDP  
1132   Request: INVITE sip:8802#+17198886...@xxx.xx.xxx.x;user=phone | 

Frame 1: 1132 bytes on wire (9056 bits), 1132 bytes captured (9056 bits)
Linux cooked capture
Internet Protocol Version 4, Src: yy.yyy.yyy.yy, Dst: xxx.xx.xxx.x
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:8802#+17198886...@xxx.xx.xxx.x;user=phone SIP/2.0
Method: INVITE
Request-URI: sip:8802#+17198886...@xxx.xx.xxx.x;user=phone
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP yy.yyy.yyy.yy:5060;branch=z9hG4bK4c59.4cd41e02.0;cid=5
Max-Forwards: 65
From:  ;tag=as7f8965f0
To: 
Contact: 

Call-ID: 
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-
[Generated Call-ID: 
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-]
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 280
P-Asserted-Identity: 
Remote-Party-ID: ;privacy=off;screen=yes
Allow: INVITE, ACK, PRACK, BYE, CANCEL, OPTIONS
X-HCV-Session: 
XlRdWwcQW1JfUhBQU1NWQkJbV1pcWgAEQghSWV1fU1dyCFFUR15aT0tZSA4=
User-Agent: SIPMAV
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 1114254434 1114254434 IN IP4 
zzz.zz.zzz.zzz
Session Name (s): Asterisk PBX 1.6.0.26-FONCORE-r78
Connection Information (c): IN IP4 zzz.zz.zzz.zzz
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 18242 RTP/AVP 0 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv
[Generated Call-ID: 
HOWTHISBATMAN_K0FdWDAqZWJiFi0oOB1hER4zNCQCHjwKCjYiKCIsYwkkOCMvEB4nARomJF4ZAQc3PjscDQQBFmMLNgIKBR4oIA4xVyd+]
[Generated Call-ID: 
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-]

No. Time   SourceDestination   Protocol 
Length Info
  2 0.000597   xxx.xx.xxx.x  yy.yyy.yyy.yy SIP  401 
   Status: 100 Giving a try | 

Frame 2: 401 bytes on wire (3208 bits), 401 bytes captured (3208 bits)
Linux cooked capture
Internet Protocol Version 4, Src: xxx.xx.xxx.x, Dst: yy.yyy.yyy.yy
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (100)
Status-Line: SIP/2.0 100 Giving a try
Status-Code: 100
[Resent Packet: False]
[Request Frame: 1]
[Response Time (ms): 0]
Message Header
Via: SIP/2.0/UDP yy.yyy.yyy.yy:5060;branch=z9hG4bK4c59.4cd41e02.0;cid=5
From:  ;tag=as7f8965f0
To: 
Call-ID: 
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-
[Generated Call-ID: 
d186cc51-fHFkT1BQCQJQWlYEFGFqe3ZkHFACWwRUVA0JHWdqKycWHF0DFgdVWxsAA2I-]
CSeq: 102 INVITE
Server: OpenSIPS (2.4.9 (x86_64/linux))
Content-Length: 0

No. Time   SourceDestination   Protocol 
Length Info
  3 0.004691   xxx.xx.xxx.x  38.147.130.145SIP/SDP  777 
   Request: INVITE sip:8802#17198886456@38.147.130.145;user=phone | 

Frame 3: 777 bytes on wire (6216 bits), 777 bytes captured (6216 bits)
Linux cooked capture
Internet Protocol Version 4, Src: xxx.xx.xxx.x, Dst: 38.147.130.145
User Datagram Protocol, Src Port: 5060, Dst Port: 5060

[OpenSIPS-Users] Compact SIP

[Saint Michael]

>
> 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
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Re: [OpenSIPS-Users] Working out OpenSIPS 3.2 roadmap

[Saint Michael]

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
in the same process space. This would be a first for any softswitch
technology.
We need full SQL technology, with tables, views, stored procedures,
functions, timers, etc.
Philip Orleans



On Thu, Dec 3, 2020 at 5:49 AM Bogdan-Andrei Iancu 
wrote:

> Gentle reminder,
>
> We do not want you to miss your favorite feature for 3.2 and the we are
> getting closer and closer to closing this poll.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>https://opensips.org/training/OpenSIPS_eBootcamp_2020/
>
> On 11/19/20 2:00 PM, Bogdan-Andrei Iancu wrote:
> > Hi OpenSIPSers !
> >
> > Is it the time to turn our eyes to the next major release, to OpenSIPS
> > 3.2 . Is it the time to plan it, to decide what should be the main
> > area to focus on when developing OpenSIPS 3.2, to decide what ideas
> > and features just be counted when putting together the roadmap for 3.2.
> >
> > So, we want to take the community pulse on the upcoming 3.2. Any
> > suggestions, ideas, opinions are (as always) more than welcome for us
> > and for the project. So, make your statement and please fill in this
> > short (but relevant) form :
> >
> > https://bit.ly/2ISE9uX
> >
> > Best regards,
> >
>
>
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Re: [OpenSIPS-Users] CDR rotation

[Saint Michael]

it works perfectly, thanks.


On Tue, Oct 6, 2020 at 6:18 AM Richard Revels  wrote:

> https://opensips.org/docs/modules/3.1.x/db_flatstore.html#mi_flat_rotate
>
> 1) move CDR files to different directory (opensips continues to write to
> them)
> 2) issue fifo command to cause opensips to write to new files
> 3) insure an accounting record is written to see that new files in
> original directory are indeed created
> 4) work with old files in different directory
>
>
>
>
>
> [image: BandwidthMaroon.png]
>
>
>
> Richard Revels  •  System 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 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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Re: [OpenSIPS-Users] Flatstore files missing calls

[Saint Michael]

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:
>
>> 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 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?
>>>
>>> :) that is the part which confuses everyone when working with this
>>> module (myself included, when I began using it).
>>>
>>> The answer: it simply works!  OpenSIPS will continue writing to the *moved
>>> *file until you *rotate* it, as the "move" operation simply changes the
>>> inode of the file (its reference, hook, pointer, etc.), without touching
>>> its contents.  This seems to work well on ext filesystems, such as ext4.  I
>>> have a feeling this is not the case for all filesystems, though.
>>>
>>> --
>>> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>>>
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Re: [OpenSIPS-Users] Flatstore files missing calls

[Saint Michael]

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 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?
>
> :) that is the part which confuses everyone when working with this module
> (myself included, when I began using it).
>
> The answer: it simply works!  OpenSIPS will continue writing to the *moved
> *file until you *rotate* it, as the "move" operation simply changes the
> inode of the file (its reference, hook, pointer, etc.), without touching
> its contents.  This seems to work well on ext filesystems, such as ext4.  I
> have a feeling this is not the case for all filesystems, though.
>
> --
> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>
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Re: [OpenSIPS-Users] Flatstore files missing calls

[Saint Michael]

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. You must handle it. It is a 2 step process:
>
> * moves/renames the current file
>
> * calls OpenSIPS MI command to create a new file [1]
>
>
>
> I think most people do it with a scheduled cron job.
>
>
>
> [1]
> https://opensips.org/docs/modules/3.1.x/db_flatstore.html#mi_flat_rotate
>
>
>
> Ben Newlin
>
>
>
> *From: *Users 
> *Date: *Friday, October 2, 2020 at 12:57 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] Flatstore files missing calls
>
> 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 for each OpenSIPS process.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS Summit 2020 online
>
>   https://www.opensips.org/events/Summit-2020Distributed/
>
> On 8/11/20 2:50 PM, Vic Jolin wrote:
>
> What I did was increase the number of process from 4 to 8. Will that
> somehow work?
>
>
>
> On Tue, Aug 11, 2020, 7:44 PM Bogdan-Andrei Iancu, 
> wrote:
>
> Vic,
>
> Note that OpenSIPS will produce one file per process, so maybe your record
> is in a different file
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS Summit 2020 online
>
>   https://www.opensips.org/events/Summit-2020Distributed/
>
> On 8/11/20 11:43 AM, Vic Jolin wrote:
>
> Hi Bogdan-Andrei,
>
>
>
> The issue is just there are a few calls not being written to the
> flatstore. So we were missing those calls in the cdr. I saw it in the logs.
> But it was not written in the flatstore file
>
>
>
> On Tue, Aug 11, 2020, 4:15 PM Bogdan-Andrei Iancu, 
> wrote:
>
> Hi Vic,
>
> The files are created upon first write into them. So be sure you are
> looking for the files into the right directory and be sure there is an
> actual write operation to the file.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS Summit 2020 online
>
>   https://www.opensips.org/events/Summit-2020Distributed/
>
> On 8/5/20 7:45 PM, Vic Jolin wrote:
>
> Hello,
>
>
>
> What are the reasons why flatstore files are not being created?
>
>
>
> Im  seeing this output in a binary journal file, and not from a normal log
> file I have my output logs in /var/log/messages (but we do not see it
> coming here as well)
>
>
>
>
>
>
>
> ACC: call ended:
> created=1596585092;call_start_time=1596585108;duration=5;ms_duration=5268;setuptime=16;method=INVITE;from_tag=13c1b24f27e408db;to_tag=ZtNe611a9391D;call_id=2a2ac4f263616c6c0015c430
>
>
>
> But no flatstore file created or updated
>
>
>
> But there is no flatstore files created. Is this a server issue? A
> resource like HD write speed? or some misconfiguration?
>
>
>
> ___
>
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>
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>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
>
>
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Re: [OpenSIPS-Users] Flatstore files missing calls

[Saint Michael]

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 for each OpenSIPS process.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Summit 2020 online
>   https://www.opensips.org/events/Summit-2020Distributed/
>
> On 8/11/20 2:50 PM, Vic Jolin wrote:
>
> What I did was increase the number of process from 4 to 8. Will that
> somehow work?
>
> On Tue, Aug 11, 2020, 7:44 PM Bogdan-Andrei Iancu, 
> wrote:
>
>> Vic,
>>
>> Note that OpenSIPS will produce one file per process, so maybe your
>> record is in a different file
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Summit 2020 online
>>   https://www.opensips.org/events/Summit-2020Distributed/
>>
>> On 8/11/20 11:43 AM, Vic Jolin wrote:
>>
>> Hi Bogdan-Andrei,
>>
>> The issue is just there are a few calls not being written to the
>> flatstore. So we were missing those calls in the cdr. I saw it in the logs.
>> But it was not written in the flatstore file
>>
>>
>> On Tue, Aug 11, 2020, 4:15 PM Bogdan-Andrei Iancu, 
>> wrote:
>>
>>> Hi Vic,
>>>
>>> The files are created upon first write into them. So be sure you are
>>> looking for the files into the right directory and be sure there is an
>>> actual write operation to the file.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 2020 online
>>>   https://www.opensips.org/events/Summit-2020Distributed/
>>>
>>> On 8/5/20 7:45 PM, Vic Jolin wrote:
>>>
>>> Hello,
>>>
>>> What are the reasons why flatstore files are not being created?
>>>
>>> Im  seeing this output in a binary journal file, and not from a normal
>>> log file I have my output logs in /var/log/messages (but we do not see it
>>> coming here as well)
>>>
>>>
>>>
>>> ACC: call ended:
>>> created=1596585092;call_start_time=1596585108;duration=5;ms_duration=5268;setuptime=16;method=INVITE;from_tag=13c1b24f27e408db;to_tag=ZtNe611a9391D;call_id=2a2ac4f263616c6c0015c430
>>>
>>> But no flatstore file created or updated
>>>
>>> But there is no flatstore files created. Is this a server issue? A
>>> resource like HD write speed? or some misconfiguration?
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
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[OpenSIPS-Users] CDR rotation

[Saint Michael]

>
> 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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Re: [OpenSIPS-Users] Possible bug

[Saint Michael]

No, it does not. Opensips keeps on, and the logs show the same over and
over.
I rebooted the server and it stopped happening. I guess there was
something wrong with the kernel at that point.
Federico

On Tue, Aug 11, 2020 at 7:42 AM Bogdan-Andrei Iancu 
wrote:

> Hi Michael,
>
> is OpenSIPs crashing with a coredump (after these logs) or it carries on ?
> If the latest, do you have a large chunk of logs around the event ? Does it
> reproduce somehow ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.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, 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
> poll method support: poll, epoll, sigio_rt, select.
> git revision: 8a9a396d4
> main.c compiled on 13:38:29 Aug  6 2020 with gcc 7
>
> Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59545]
> CRITICAL:core:io_watch_add: [UDP_worker] check failed after successful fd
> add (fd=177,type=17,data=0x7f9e6012ab48,flags=1) already=0
> Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
> CRITICAL:core:io_watch_add: [UDP_worker] check failed after successful fd
> add (fd=75,type=4,data=(nil),flags=1) already=0
> Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
> CRITICAL:core:io_watch_add:
> Aug 06 14:12:04 brian opensips[59481]: >>> fd_array idx 2 (fd=0) points to
> bogus map (fd=0,type=1,flags=1,data=(nil))
> Aug 06 14:12:04 brian opensips[59481]: It seems you have hit a programming
> bug.
> Aug 06 14:12:04 brian opensips[59481]: Please help us make OpenSIPS better
> by reporting it at https://github.com/OpenSIPS/opensips/issues
> Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
> CRITICAL:core:io_watch_add:
> Aug 06 14:12:04 brian opensips[59481]: >>> used fd map fd=0 has bogus data
> (fd=0,type=1,flags=2001,data=(nil))
> Aug 06 14:12:04 brian opensips[59481]: It seems you have hit a programming
> bug.
> Aug 06 14:12:04 brian opensips[59481]: Please help us make OpenSIPS better
> by reporting it at https://github.com/OpenSIPS/opensips/issues
>
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>
>
>
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[OpenSIPS-Users] Possible bug

[Saint Michael]

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
poll method support: poll, epoll, sigio_rt, select.
git revision: 8a9a396d4
main.c compiled on 13:38:29 Aug  6 2020 with gcc 7

Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59545]
CRITICAL:core:io_watch_add: [UDP_worker] check failed after successful fd
add (fd=177,type=17,data=0x7f9e6012ab48,flags=1) already=0
Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
CRITICAL:core:io_watch_add: [UDP_worker] check failed after successful fd
add (fd=75,type=4,data=(nil),flags=1) already=0
Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
CRITICAL:core:io_watch_add:
Aug 06 14:12:04 brian opensips[59481]: >>> fd_array idx 2 (fd=0) points to
bogus map (fd=0,type=1,flags=1,data=(nil))
Aug 06 14:12:04 brian opensips[59481]: It seems you have hit a programming
bug.
Aug 06 14:12:04 brian opensips[59481]: Please help us make OpenSIPS better
by reporting it at https://github.com/OpenSIPS/opensips/issues
Aug 06 14:12:04 brian opensips[59481]: Aug  6 14:12:04 [59518]
CRITICAL:core:io_watch_add:
Aug 06 14:12:04 brian opensips[59481]: >>> used fd map fd=0 has bogus data
(fd=0,type=1,flags=2001,data=(nil))
Aug 06 14:12:04 brian opensips[59481]: It seems you have hit a programming
bug.
Aug 06 14:12:04 brian opensips[59481]: Please help us make OpenSIPS better
by reporting it at https://github.com/OpenSIPS/opensips/issues
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[OpenSIPS-Users] function append_hf may be wrong

[Saint Michael]

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, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Type: application/sdp.
X-: 381.
Identity:.
eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cHM6Ly9jci5jY2lkLm5ldXN0YXIuYml6L2NjaWQvYXV0aG4vdjIvY2VydHMvMTExMTguMTAwMzIifQ.eyJhdHRlc3QiOiJDIiwiZGVzdCI6eyJ0biI6WyIxNjMxNzkxODM3OCJdfSwiaWF0IjoxNTk0NDQxMTY1LCJvcmlnIjp7InRuIjoiNjM2NTY2NjY2NiJ9LCJvcmlnaWQiOiIxMjNlNDU2Ny1lODliLTEyZDMtYTQ1Ni00MjY2NTU0NDAwMDAifQ.5U7fmN7whQVdmRmU0yOmDFZenUZtX0_mnKvQRB0yUe1i0NUdnTUqa8pHF4VxBdinDTSoOX1_I9mKR3A1x9DXzw;info=<
https://c>;alg="ES256";ppt="shaken".
.
v=0.
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[OpenSIPS-Users] crashes

[Saint Michael]

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> 89 01 48 8b 7b 08 4d 89 f7 49 c1 ef 03 49 c1 e7 04 49 01 ef 49
[849895.690104] opensips[37055]: segfault at 21 ip 55f47130574c sp
7ffc04f7a150 error 4 in opensips[55f4711ce000+2ec000]
[849895.690117] Code: 8d 78 50 45 31 f6 31 ed 45 31 ed 4c 8d 64 24 60 4c 89
64 24 28 45 89 f3 49 8b 17 48 85 d2 0f 84 8b 02 00 00 44 89 eb 49 89 d0
<49> 03 28 4d 8b 40 08 83 c3 01 41 89 dc 45 29 ec 4d 85 c0 75 eb 45

but opensips is not generating the dumps in /home/opensips
I have of course
cho 1 > /proc/sys/kernel/core_uses_pid and echo 1 >
/proc/sys/fs/suid_dumpable
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Re: [OpenSIPS-Users] CDR no longer work for permission denied

[Saint Michael]

Please ask Vlad Paiu. He fixed it. It turns out that in the new kernel
5.4.0-37-generic of Ubuntu 20.04, the fifo has a lock, so he took these
steps:
mkdir /home/opensips
chown -R opensips:opensips /home/opensips/
/etc/opensips/opensips.cfg , changed to modparam("mi_fifo", "fifo_name",
"/home/opensips/opensips_fifo")
/usr/local/etc/opensips/opensipsctlrc , added
OSIPS_FIFO="/home/opensips/opensips_fifo"

I think this should the canonical way this should work.




On Tue, Jun 23, 2020 at 4:12 AM Răzvan Crainea  wrote:

> Can you provide more information 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-solutions.com
>
> On 6/14/20 2:16 PM, Saint Michael wrote:
> > /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.
> >
> >
> >
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
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[OpenSIPS-Users] CDR no longer work for permission denied

[Saint Michael]

/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.
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Re: [OpenSIPS-Users] New kernel, new errror

[Saint Michael]

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 Michael  wrote:
> >
> > /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?
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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[OpenSIPS-Users] New Errors

[Saint Michael]

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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[OpenSIPS-Users] New kernel, new errror

[Saint Michael]

/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?
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Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

[Saint Michael]

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 TB of RAM. My
Opensips is under Vmware ESX 6.X.
Honestly, I paid an Opensips guru to assemble the application for me.
I designed the logic base on my previous switch for Asterisk that could not
handle the pressure. Basically all routing is done in MariaDB. Opensips
just asks a question using a Stored Procedure.
The brain is the database, and opensips executes instructions. It just
works.


On Wed, Jun 10, 2020 at 5:15 PM Jon Abrams  wrote:

> I built a similar functioning platform back in 2015 based on similar
> hardware (Westmere Xeons, hyperthreading enabled)  running bare metal on
> Centos6. At some point we bumped it up to dual X5670s (cheap upgrade
> nowadays), but it was handling 12000 CPS peaks on 1 server with 3000-5000
> CPS sustained for large parts of the day. I don't think you are too far off
> in hardware.
> This was on version 1.9, so there was no Async. IIRC it was either 32 or
> 64 children. Async requires the TM module which adds additional overhead
> and memory allocation.
> The LRN database was stored in mysql with a very simple table (TN, LRN) to
> keep memory usage down so that it could be pinned in memory (server had 48
> or 72GB I think). MySQL was set to dump the innodb buffer cache to disk on
> restart so that the whole database would be back in memory on restart.
> Doing a full table scan would initially populate the MySQL cache.
> Blacklists and other smaller datasets were stored in OpenSIPs using the
> userblacklist module. There are better ways to do that in version 2 and
> onwards. Bigger list were stored in memcached. I prefer redis for this
> purpose now.
>
> I would suggest simplifying testing by using a single MySQL server and
> bypassing the F5 to eliminate that as a source of connection problems or
> additional latencies.
> In the OpenSIPs script, eliminate everything but 1 dip, probably just dip
> the LRN to start.
> Performance test the stripped down scenario with sipp. Based on past
> experience, you should be able to hit or come close to your performance
> goal with only 1 dip in play.
> If you do hit your performance targets, keep adding more dips one by one
> until it breaks.
> If you can't reach your performance target with this stripped down
> scenario, then I'd suggest testing without the async and transactions
> enabled. I wouldn't think transactions would be a necessity in this
> scenario. I ran into CPS problems on that other open source SIP server when
> using async under heavy load. The transaction creation was chewing up CPU
> and memory. I'm not sure how different the implementation is here.
> I seem to start having problems with sipp when I hit a few thousand CPS
> due to it being single threaded. You probably will need to run multiple
> parallel sipp processes for your load test, if not already.
> If using an OS with systemd journald for logging, that will be a big
> bottleneck in of itself with even small amounts of logging.
> In 1.9, I hacked together a module to create a timestamp string with ms
> for logging query latencies for diagnostic purposes. There may be a better
> out of the box way to do it now.
> For children sizing, I would suggest benchmarking with at least 16
> children and then doubling it to compare performance.
> Watch the logs for out of memory or defragment messages and bump up shared
> memory or package memory if necessary. Package memory is probably going to
> be your problem, but it doesn't sound like it is a problem yet.
>
> BR,
> Jon Abrams
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Wed, Jun 10, 2020 at 3:20 PM Calvin Ellison 
> wrote:
>
>> We've checked our F5 BigIP configuration and added a second database
>> server to the pool. Both DBs have been checked for max connections, open
>> files, etc. Memcached has been moved to a dedicated server. Using a SIPp
>> scenario for load testing from a separate host, things seem to fall apart
>> on OpenSIPS around 3,000 CPS with every CPU core at or near 100% and no
>> logs indicating fallback to sync/blocking mode. Both databases barely
>> noticed the few hundred connections. Does this seem reasonable for a dual
>> CPU server with 8 cores and 16 threads?
>>
>>
>> https://ark.intel.com/content/www/us/en/ark/products/47925/intel-xeon-processor-e5620-12m-cache-2-40-ghz-5-86-gt-s-intel-qpi.html
>>
>> What is the OpenSIPS opinion on Hyper-Threading?
>>
>> Is there a way to estimate max CPS based on SPECrate, BogoMIPS, or some
>> other metric?
>>
>> I would love to know if my opensips.cfg has any mistakes, omissions, or
>> inefficiencies. Is there a person or group who does sanity checks?
>>
>> What should I be looking at within OpenSIPS during a load test to
>> identify bottlenecks?
>>
>> I'm still 

[OpenSIPS-Users] Maybe it's a bug

[Saint Michael]

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 call with 100 ms being reported as zero seconds, and a call with
1200 ms being reported as 1 second, etc. In the first case, 100ms, my
carrier will consider this call 1 second, rounded to 6 seconds
minimum (6/6). So there a jus lost 6 seconds that I am not charging to my
customer but I am paying for. If you have millions of calls, 50%, that fall
into the lower half of 1000ms, like 24400 ms, that they are reported as 24
seconds, but my carrier thinks is 25 seconds, and he rounds it to 30, there
I lost another 6 seconds.
I talked to Vlad, who I believe wrote the code, and he does not think it is
a bug and I should use the ms and not the seconds. But thousands of
businessmen will not spot this and thus their billing will never match the
carrier, and they will lose money. If anybody thinks for a second that a
call with a 200 OK will be free, is dreaming. Not in America.
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Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

[Saint Michael]

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't broke, don't fix it".
But I am open to change the engine in the second container, if you
so desire..
MariaDB is better than MySQL because it gives us a pool of threads,
something that MySQL only gives to paying customers. So I think it should
work.


On Thu, Jun 4, 2020 at 3:44 PM Jon Abrams  wrote:

> A) Is the LRN database located locally on the OpenSIPs box or is it remote?
> B) Have you tried only doing sync database queries? Async introduces some
> overhead, and I'm not sure if it causes extra database connections to be
> created. When using sync there is a connection per child process that stays
> up.
> C) Does the database have enough memory to contain the LRN and DNC
> datasets fully in memory? The extra latency for the non-cache hits sent to
> the database may stack up if the database has to hit disk.
> D) How many child processes are you using now? If you are hitting 100% you
> may need to increase them.
> E) Are your memcached processes using heavy cpu? If you are caching
> multiple lists, I've found it helps to use unique memcached instance per
> list.
> F) Look for memory related log messages. If the memory starts getting
> exhausted you will see defrag messages. This will chew up available
> computation cycles.
>
> - Jon Abrams
>
>
> On Thu, Jun 4, 2020 at 2:17 PM Calvin Ellison 
> wrote:
>
>> The scenario is INVITE -> MySQL query -> non-200 final response. No
>> calls are connected here, only dipping things like LRN, Do Not Call,
>> and Wireless/Landline. A similar service runs on a second port,
>> specific to a different kind of traffic and dip. We're using async
>> avp_db_query and memcached, with about 3:1 cache hits.
>>
>> Our target is up to 10,000 CPS across two opensips servers, which are
>> dual-CPU Xeon E5620 with 48G RAM. Both are run memcached, and both
>> servers are using both memcached to share a distributed cache thanks
>> to this:
>>
>> 'modparam("cachedb_memcached","cachedb_url","memcached:lrn://lrn-d,lrn-e/")'.
>> At a glance there are over 200mil total cached items, distributed
>> nearly equally.
>>
>> The issue is that individual child processes start getting suck at
>> 100% CPU. Logs indicate connection failures to the MySQL database
>> causing children to run in sync mode, and there are warnings about
>> delayed timer jobs tm-timer and blcore-expire. Eventually, the service
>> becomes unresponsive. Restarting opensips restores service and the
>> children return to single-digit CPU utilization, but eventually,
>> children get suck again.
>>
>> I'm not certain if the issue is on the database server, or if the
>> opensips servers are overloaded, or if the config is just not right
>> yet.
>>
>> Is there an established method for fine-tuning these things?
>> shared memory
>> process memory
>> children
>> db_max_async_connections
>> listen=... use_children
>> modparam("tm", "timer_partitions", ?)
>>
>> What else is worth considering?
>> Does a child ever return to async mode after running in sync mode?
>> How do I know when my servers have reached their limit?
>> opensips.cfg is available on request.
>>
>> version: opensips 2.4.7 (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
>> poll method support: poll, epoll, sigio_rt, select.
>> git revision: 9e1fcc915
>> main.c compiled on  with gcc 7
>>
>> *re-built using dpkg-buildpackage including the patch to support DB
>> floating point types:
>> https://opensips.org/pipermail/users/2020-March/042528.html
>>
>> $ lsb_release -d
>> Description:Ubuntu 18.04.4 LTS
>>
>> $ uname -a
>> Linux TC-521 4.15.0-91-generic #92-Ubuntu SMP Fri Feb 28 11:09:48 UTC
>> 2020 x86_64 x86_64 x86_64 GNU/Linux
>>
>> $ free -mw
>>   totalusedfree  shared buffers
>>cache   available
>> Mem:  482811085 337  871729
>>45128   46551
>>
>> $ lscpu
>> Architecture:x86_64
>> CPU op-mode(s):  32-bit, 64-bit
>> Byte Order:  Little Endian
>> CPU(s):  16
>> On-line CPU(s) list: 0-15
>> Thread(s) per core:  2
>> Core(s) per socket:  4
>> Socket(s):   2
>> NUMA node(s):2
>> Vendor ID:   GenuineIntel
>> CPU family:  6
>> Model:   44
>> Model name:  Intel(R) Xeon(R) CPU   E5620  @ 2.40GHz
>> Stepping:2
>> CPU MHz: 2527.029
>> BogoMIPS:4788.05
>> Virtualization:  VT-x
>> L1d cache:   32K
>> L1i cache:   32K
>> L2 cache:256K
>> L3 cache:

[OpenSIPS-Users] adding and reading a SIP header to a 302 Redirect

[Saint Michael]

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 */
send_reply("302","Moved Temporarily");
exit;
}
the final result should be:

SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 172.16.7.254:52169
;rport=52169;received=47.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04
Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI
From: "X" ;tag=81a25c36
To: ;tag=9e198dc4-7ce8-433d-ae23-05b9bc14d55a
Identity:eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyI2MzE3OTE4Mzc4Il19LCJpYXQiOjE1OTA4MTEyMzgsIm9yaWciOnsidG4iOiI3Mjc0NDMzMDE5In0sIm9yaWdpZCI6IjEyM2U0NTY3LWU4OWItMTJkMy1hNDU2LTQyNjY1NTQ0MDAwMCJ9.AKViDWA3uonP6tt5cKBh0FUPY5zBuJnwZLQNTrp9LCWJ-vLY1Xx5i3_oXGh1ERL4tnD-KK5wsP3FdByDa_cjGw;info=<
https://cert.example.org/passport.cer>;alg=ES256;ppt=shaken
CSeq: 1 INVITE
Server: Asterisk PBX 16.10.0
Contact: 
Reason: Q.850;cause=0
Content-Length:  0

also if I get that packet at my end, how do I extract the Identity header
and apply it to the next INVITE?
Is this even doable?
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[OpenSIPS-Users] Question

[Saint Michael]

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
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[OpenSIPS-Users] Question about regex

[Saint Michael]

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?
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Re: [OpenSIPS-Users] Dockerize OpenSIPS

[Saint Michael]

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
the same technology at the kernel level.
Please correct me if I am wrong.

On Sat, May 2, 2020 at 8:34 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> They all get their dialplan/config from an API backend. They’re DB
> agnostic.
>
> On Sat, 2 May 2020 at 13:28, H Yavari  wrote:
>
>> Thank you David,
>> Interested to know, you are running a cluster/group of FS with
>> centralized DB, with same functionality?
>>
>>
>> Regards,
>> HYavari
>>
>>
>> On Saturday, May 2, 2020, 4:54:01 PM GMT+4:30, David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>
>> Hello,
>>
>> I have never run into those problems. The people I’ve seen running into
>> them are doing thousands upon thousands of packets por seconds. Hardware is
>> cheap nowadays, I just spread the load.
>>
>> I wouldn’t dockerize rtpengine, although I know people who do.
>>
>> In a normal setup probably you won’t run into them, we run freeswitch on
>> containers for production without a hitch (I must stress I was very
>> reluctant to do that a few years ago, but docker has come a long way since
>> then).
>>
>> David
>>
>> On Sat, 2 May 2020 at 12:42, H Yavari  wrote:
>>
>> Thank you David.
>>
>> What do you think about networking concerns? you mentioned to them but I
>> didn't get your point.
>> RTP restrictions, port proxy, iptables, fail2ban are top ones.
>>
>> PS: No matter which telephony platform (Asterisk/FS/OpenSIPS/Kamailio),
>> these concerns are in general.
>> PS: We have more challenges with projects like Freepbx.
>>
>>
>> Regards,
>> HY
>>
>>
>>
>> On Saturday, May 2, 2020, 3:48:11 PM GMT+4:30, David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>
>> Not sure about OpenSIPS specifically, but I would assume it has been
>> implemented in docker just as much as kamailio and freeSWITCH/Asterisk.
>>
>> This is done all over the world. Docker is not an emulator or a virtual
>> machine host. When you run something on docker, its speed is (almost)
>> exactly the same as running it on the host itself, since there’s no OS
>> overhead, it works by separating processes via Cgroups, no by virtualizing
>> or emulating hardware.
>>
>> Quote:
>>
>> The Docker technology uses the Linux kernel
>>  and
>> features of the kernel, like Cgroups
>> 
>>  and namespaces , to segregate
>> processes so they can run independently. This independence is the intention
>> of containers‐the ability to run multiple processes and apps separately
>> from one another to make better use of your infrastructure while retaining
>> the security you would have
>> with separate systems.
>>
>> So in simple terms, docker simply separates processes.
>>
>> There ARE, nonetheless, some problems with dockerizing everything. I have
>> read issues like If the network traffic is way way way too high, you may
>> encounter issues like dropped packets, etc. but this is a problem on the
>> networking side, I.e: the iptables rules. Also the natting related to using
>> docker can be cumbersome, but once you’re over that, it’s home free.
>>
>> So, as long as you manage your infrastructure well, you shouldn’t have
>> problems.
>>
>> In terms of troubleshooting a failing container. All logging should be
>> sent to some log server, and you can do your troubleshooting there. Also,
>> don’t kill a failing container so you can access it (via ssh or attach or
>> exec) and troubleshoot it.
>>
>> The pros of using docker/k8s greatly outweighs the cons, in my opinion.
>>
>> Hope this help.
>>
>> David
>>
>> On Sat, 2 May 2020 at 11:33, H Yavari via Users 
>> wrote:
>>
>> Thank you Johan,
>>
>> When your infrastructure goes to run with k8s or other same platforms,
>> it's hard to make some exceptions.
>> Also softwares like opensips that are working just with DB, can run very
>> smoothly.
>>
>> Although I haven't seen any problem yet after moving it to containers,
>> but I am interested in hearing from others and developers team.
>>
>>
>> Regards,
>> HY
>>
>>
>> 
>> On Saturday, May 2, 2020, 12:51:51 PM GMT+4:30, johan 
>> wrote:
>>
>>
>> First of all, I am not aware of a production kubernetes cluster.
>>
>> Using containers has advantages : fast install, easy to move.  The
>> annoying thing is that if it goes wrong, it is not easy to troubleshoot.
>> Secondly, you add an extra abstraction layer, abstraction (most of the
>> time) reduces speed and decreases capacity.
>>
>> In short : it all depends on the size of your 

[OpenSIPS-Users] CDR capturing not working

[Saint Michael]

I checked the database and I missed the CDR for many calls.  Later I found
them sitting in files in /var/log/opensips
I run it manually cdr_worker.sh and got this error ./cdr_worker.sh
lockfile: Sorry, giving up on "/var/run/opensips_cdr_worker.lock"which
means is not firing because the lock is being held, but there is no
traffic. I stopped opensips and still the same error , so the script is not
working as it should. The locking mechanism is stuck. I restarted opensips,
placed another call, and odbc did not get executed, so recycling opensips
does not free the lock.
Any idea how can I get my CDR inserted 100% for sure into my database?
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Re: [OpenSIPS-Users] Messages

[Saint Michael]

Let me try 0.

On Tue, Apr 14, 2020, 7:56 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Have you tried with 1 or 0?
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Wed, Apr 15, 2020 at 12:09 AM Saint Michael  wrote:
>
>> what is the lowest debug level that is safe to use?
>>
>>
>> On Tue, Apr 14, 2020 at 6:59 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> Maybe lower the debug?
>>>
>>> On Tue, 14 Apr 2020 at 23:00, Saint Michael  wrote:
>>>
>>>> I see thousands of the messages below. If they are not important, how
>>>> do I hide them
>>>> ARNING:dialog:dlg_onroute: tight matching failed for ACK with
>>>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52,
>>>> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and
>>>> direction=1
>>>> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787]
>>>> WARNING:dialog:dlg_onroute: dialog identification elements are
>>>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52,
>>>> caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13
>>>> ___
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>>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
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Re: [OpenSIPS-Users] Messages

[Saint Michael]

what is the lowest debug level that is safe to use?


On Tue, Apr 14, 2020 at 6:59 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Maybe lower the debug?
>
> On Tue, 14 Apr 2020 at 23:00, Saint Michael  wrote:
>
>> I see thousands of the messages below. If they are not important, how do
>> I hide them
>> ARNING:dialog:dlg_onroute: tight matching failed for ACK with
>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52,
>> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and
>> direction=1
>> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787]
>> WARNING:dialog:dlg_onroute: dialog identification elements are
>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52, caller
>> tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13
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> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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