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gards, Alexey
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for each [1] statement seems to be what I need
[1] https://www.opensips.org/Documentation/Script-Statements-3-2#toc4
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/Tutorials-OpenSIPSAsteriskIntegration-1-8
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Is it possible to cache not exact number of table row
but all values from concrete column from all rows?
The total amount of rows in table is not big, several rows or several dozens.
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customerfullid)
has privileges. Global call-limits not applied.");
...
}
# apply global call-limits here
route[customers_all] {
xlog("L_INFO", "[$ci] This is $route . This $var(customerfullid)
has no privileges. Global call-limits applied.");
... # call-limit logic
}
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o $avp.
Is there any way to extract the whole column
(base_id in my sample) from the cached DB ?
Maybe you can provide right way to make request via
$avp(a) = $sql_cached_value(caching_name:column_name_1:key1); ?
We expect something similar to
avp_db_query("select base_id from test"
Hello,
isn't it what you need?
https://www.opensips.org/Documentation/TipsFAQ#toc7
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which lets us extract the full column (array) from the cached DB ?
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Catch CANCEL requests which are sent by clients,
according to 9.1 section of the RFC [1].
[1] https://datatracker.ietf.org/doc/html/rfc3261#section-9.1
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Hi Alexander,
there's also one important thing when it's about using script variables -
first of all we have to reset its value [1].
1. https://www.opensips.org/Documentation/Script-CoreVar-3-4#toc1
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Hello,
ensure that the m4 package is installed and try again.
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db_text is available as always.
Check please:
[1] https://opensips.org/docs/modules/3.4.x/db_text.html
[2] https://opensips.org/docs/modules/3.4.x/auth_db.html#param_db_url
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This is exactly what you need - the minimal OpenSIPS subscriber db_text
table example:
https://opensips.org/docs/modules/3.4.x/db_text.html#idp107424
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what 'apt search opensips' will show?
i'm sure it will show the package you need, but it is not installed
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rding to the package description:
This package provides the modules that are being used to provide
SIP authentication in OpenSIPS. It consists of both the
authentication interface (the auth module), as well as the UAC
authentication module (uac_auth).
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Hi,
What version do you use?
According to the documentation [1] the flags (string) parameter
denotes a comma-separated list of checks to be performed.
What errors in the log do you have?
[1] https://opensips.org/docs/modules/3.5.x/nathelper.html#func_nat_uac_test
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I think you should start from $socket_in [1] and $socket_out [2] variables.
[1] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc80
[2] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc81
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Hello,
add to /usr/lib/tmpfiles.d/tmp.conf this line:
# Disable auto-remove of /tmp/opensips_fifo
x /tmp/opensips_fifo
[1] https://alexeyka.zantsev.com/?p=1252
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Hello,
I think 'cache_fetch_chunk' [1] MI command is exactly what you need.
But according to the documentation, it appeared only from v.3.5.
[1]
https://opensips.org/docs/modules/3.5.x/cachedb_local.html#mi_cache_fetch_chunk
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According to the documentation, this MI function is available since v.3.5.
Maybe the developers know something more, but as for me -
I'm absolutely sure there's nothing like this for the versions below.
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et variable
$var(x) = 123456;
Otherwise you may catch an OpenSIPS behavior when the variable
sometimes has its value and sometimes does not.
[1] https://www.opensips.org/Documentation/Script-CoreVar-3-5
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_
orked with 12).
If it is so (and if mr12 does not have any backwards compatibility
with older versions
and with iptables usage), you have to start using nftables in your
system instead of iptables.
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What about setport [1] core function?
[1] https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#setport
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] crit
CRITICAL:drouting:dr_reload_cmd: failed to load routing data
Restarting of nginx and php-fpm does not help either.
What may be the reason?
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OpenSIPS 3.2.17
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c_registrant:reg_tm_cback: record [0x7fce2e314018] not found
on hash index [34]
ERROR:uac_registrant:reg_tm_cback: record [0x7fce2e313d60] not found
on hash index [35]
What may be the reason?
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10:51:38 sipgw [2348879] info
INFO:uac_registrant:load_reg_info_from_db: loading [35] records from
db
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ERROR:uac_registrant:reg_tm_cback:
record [0x7f0ccf957e60] not found on hash index [16]
11:49:43 sipgw [2348892] err ERROR:uac_registrant:reg_tm_cback:
record [0x7f0ccfdffbd8] not found on hash index [18]
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Hi Răzvan,
sure, I'll try, if it will be possible to catch this moment,
as it is not always
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Hi Bogdan,
Thank you for the comprehensive answer.
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where X is the group id, which will be different,
but the prefix will be the same.
Thank you one more time!
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checked the documentation for both 2.4 and 3.4 versions.
[1] https://www.opensips.org/Documentation/Install-DBSchema-3-4#GEN-DB-DR-RULES
[2] https://opensips.org/docs/modules/3.4.x/drouting.html#idp5903680
[3] https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc41
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ssigning a value to it.
[1] https://www.opensips.org/Documentation/Script-CoreVar-3-5#varscript
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Hello,
you should always reset the var value first:
...
$var(logkey) = NULL;
$var(logkey) = "condition_" + $avp(user);
...
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Hi Bogdan,
Now it is clear, thank you !
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e_attrs is
$var(rule_attrs)");
}
But I'm not sure if this is the best way.
Maybe there is some more elegant solution.
[1] https://opensips.org/docs/modules/3.2.x/drouting.html
[2] https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-RULES
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http
Hi Dmitriy,
I would also recommend adding sip: to your gateway address,
I mean not just 10.214.0.78 , but sip:10.214.0.78 is correct.
If I'm not mistaken, some OpenSIPS versions returned errors,
when gateways' addresses did not have 'sip:' at the beginning.
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EGISTER-requests
and respectively the same which will send the remote server in its INVITEs.
[1]
https://www.opensips.org/Documentation/Install-DBSchema-3-5#GEN-DB-REGISTRANT
[2] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc77
[3] https://opensips.org/docs/modules/3.5.x/drouting.html
[
ddress is among all those
towards which we are registering to using uac_registrant module'.
[1] https://opensips.org/docs/modules/3.5.x/permissions.html
[2] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc79
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Hi Răzvan, hi Saint Michael,
thank you both for the advice, but nothing works for me :]
alexey@lubutnu-24-10:~$ pipx install opensips-cli
Fatal error from pip prevented installation. Full pip output in file:
/home/alexey
yes, I've installed pipx before trying and writing here the output
with the error
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Hello,
pay attention to the "sort_alg" column [1]
[1]
https://www.opensips.org/Documentation/Install-DBSchema-3-4#GEN-DB-DR-CARRIERS
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Just to compare how the list of gateways looks like when the weight is set
and when it is not set:
Ex:“gw1=10,gw4=10” or “gw1,gw2”
This is from official module documentation.
Setting the same weight to several gateways will turn them on for
simultaneous use, as you want.
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Hi ,
it is in documentation under item 1.1.4.2.
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Excuse me, Michael *
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Hi Dragomir,
Could you show the dump of the outgoing Linphone REGISTER?
Especially the Contact: header.
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Hi Dragomir,
Could you show the dump of the outgoing Linphone REGISTER?
Especially the Contact: header.
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Incoming.
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username
...
Logs:
...
ERROR:core:get_cmd_fixups: Variable in param [2] is not a string
ERROR:core:do_action: Failed to get fixups for command
in /etc/opensips/opensips.cfg.m4, line 276
OpenSIPS v.3.4.9.
dp_translate manual:
https://opensips.org/docs/modules/3.4.x/dialplan.html#func_dp_translate
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Hi Bogdan,
yes, the solution was in one more condition/check,
because the Diversion header is not present in each
incoming INVITE:
if (is_present_hf("Diversion")) {
dp_translate(10224, $di, $rU);
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Oh, sure, this is exactly what the topicstarter needs.
Nice to know that OpenSIPS has such a feature.
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tions.html#func_options_reply
[2]
https://opensips.org/docs/modules/3.5.x/load_balancer.html#param_probing_method
[3] https://opensips.org/docs/modules/3.5.x/dispatcher.html#param_ds_ping_method
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Hi Sasmita,
you should use permissions [1] module.
And here's some examples [2] beyond the official documentation.
[1] https://opensips.org/docs/modules/3.4.x/permissions.html
[2] https://alexeyka.zantsev.com/?p=1469
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F0-8E08A2C6-A7AC8E91@10.1.31.1] fnumber:
info [79c4e65b089ffb570fb480821f454...@u.r46.ru] fnumber: 7471244
info [SD44-XJwMYbQpm2on@10.243.214.142] fnumber: 73022333333
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s:
modparam("dialog", "profiles_with_value", "numbers/b;")
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{
"Profile": {
"name": "numbers",
"value": null,
"count&quo
.5.x/topology_hiding.html#func_topology_hiding
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empt option works).
If such a command existed, I could use it in keepalived config/script
and run it during switching to backup, something like -
/usr/bin/opensips-cli -x mi clusterer_shtag_set_INactive vip/1
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Errors appear not always, only sometimes.
OpenSIPS v. 3.2.18.
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Excuse me, the reason was in unsyncronized time on the virtual machine.
Now it's OK.
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o download metadata for repo 'opensips': Cannot
download repomd.xml: Cannot download repodata/repomd.xml: All mirrors
were tried
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. But again, if Active node restarts and still Active
- we will have a problem.
-
Alexey Vasilyev
> 31 Dec 2018, в 18:04, John Quick написал(а):
>
> Hi Alexei,
>
> Many thanks for your reply to my query about syncing the seed node for
> usrloc registrations.
> I just
2 nodes (or more), which one to
choose for syncing? And if they have different data (they were not synced
between each other), what should it do?
Thanks.
чт, 3 янв. 2019 г. в 11:33, Liviu Chircu :
> Happy New Year John, Alexey and everyone else!
>
> I just finished catching up with this t
exists. But for TLS it is not
enough. We additionally should check, what certificate uses this connection (or
what domain it is related).
If on server side we have SNI, why not to have more control for client TLS side?
-
Alexey Vasilyev
alexei.vasil...@gmail.com
> 25 Mar 2019, в 18
tePort
connection 2: localIP2:localPort2 <--> remoteIP:remotePort
but I think the is very unrealistic */
So I got exactly this scenario.
чт, 28 мар. 2019 г. в 13:47, Bogdan-Andrei Iancu :
> Hi Alexey,
>
> It make sense (logically speaking) to get the TLS domain involved in the
> TCP
Hi Bogdan,
Sorry that I mentioned He-Who-Must-Not-Be-Named. Just to simplify search later:
https://github.com/OpenSIPS/opensips/issues/1651
-
Alexey Vasilyev
alexei.vasil...@gmail.com
> 28 Mar 2019, в 16:45, Bogdan-Andrei Iancu написал(а):
>
> Hi Alexey,
>
> oh, if i
aders. But it seems, that function topology_hiding()
encodes only RR headers from original UAC request, not those ones that were
added by OpenSIPS.
So the question - how to pass this in-dialog data?
Should we add functions add_th_param/get_th_param for this purpose?
Thanks.
-
---
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crypted or unencrypted.
You should choose, what you want. Use only one profile as parameter.
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that they can choose to send or not its'
certificate?
If you like to test it in your environment?
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es from DB, there is parameter for domain:
modparam("tls_mgm", "ca_client_send", "[dom]1")
For this domain OpenSIPS adds DNs to Certificate request.
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uZYcpOhf1g/h+oG
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=audio 60812 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
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Thanks, Giovanni and Bogdan!
If somebody has questions, I think we can discuss here.
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*/
#define DEFAULT_TCP_LISTEN_BACKLOG 10
If we have thousands TLS/TCP connections at the same time, and for example
32 processes, so we'll have only 10*32=320 socket backlog limit for
OpenSIPS? Is this right?
Should it be increased?
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Hi Bogdan,
Yes, this is exactly what I need. I'll change it and will see, how it works
with next switching.
Thanks.
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In the case of UDP, may be it make sense to use simple mirroring mode with
nathelper pings.
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> 27 Sep 2019, в 22:58, Igor Olhovskiy [via OpenSIPS (Open SIP Server)]
Hi Mark,
I'd use uaCSTA, if your phones support it. For example, snom or yealinks
definitely can be controlled over SIP.
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Hi Callum,
I have almost the same problem. But with 2.4.6.
Just today I opened issue: https://github.com/OpenSIPS/opensips/issues/1976
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Hi Jeff,
Transactions are not replicated.
Here
https://blog.opensips.org/2018/03/21/full-anycast-support-in-opensips-2-4/
Razvan explains why. Section "Distributed transactions handling".
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Please forward requests to Asterisk with simple record_route().
Only to MS Teams you should use preset with domain name.
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.
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call "dr_enable_probing 0" after
starting opensips.
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erisk, the only difference was RTPEngine to transcode
SRTP-RTP.
And within test lab I've tested not only calls, but transfers worked fine
too.
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Hi James,
Just to be sure, that this is not a typo:
check your SDP
o= 137.117,136.143
c=IN IP4 137.117,136.143
This will definitely fail.
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Alexey Vasilyev
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answer that
if(is_method("ACK") && has_body("application/sdp")){
xlog("[RTPPROXY] onreply_route[handle_nat] ACK rtpproxy_answer\n");
rtpproxy_answer("co", "137.117,136.143");
}
and change comma to point.
May be this can help?
reset("sbc.ip-sentinel.com:5091;transport=tls");
of course without: add_rr_param(";r2=on");
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e same URI parameter "transport=tls".
Check this please.
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s-sbc/
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Hi Pasan,
This was explained here:
http://opensips-open-sip-server.1449251.n2.nabble.com/TLS-handshake-failure-td7619394.html
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Alexey Vasilyev
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Hi Mark,
try this:
if (nat_uac_test("123")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
}
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Alexey Vasilyev
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But config is for 2.4
A lot of things has changed.
You should go through the whole config and follow first this:
https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
and then this:
https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0
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Alexey Vasilyev
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Hi, yes there is an issue.
Please follow here: https://github.com/OpenSIPS/opensips/issues/2433
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Users
how i can do that? Any help will be appreciated.
Regards, Alexey.
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Hi.
For example, you can try this book... "Building Telephony Systems with
OpenSIPS - Second Edition"
https://www.amazon.com/Building-Telephony-Systems-OpenSIPS-Second/dp/1785280619
Regards, Alexey Borisov.
пн, 17 мая 2021 г. в 09:23, HS :
> Hi all.
>
> I keep writin
Yes... I would say "TRY-AND-CRY"
there is a usual method to work with SIPS (IMHO)
Regards, Alexey.
вт, 18 мая 2021 г. в 14:51, HS :
> Hi Alexey.
>
> Thanks a lot for taking the time to respond. Incidentally, I had read the
> book a few months ago and took a look at
, Alexey
вт, 11 мая 2021 г. в 14:07, Alexey Hamstr :
> Hi All.
> I'am trying to setup RADIUS accounting and i want to fill in the
> Acct-Session-Time by $DLG_lifetime in STOP call as i have found in the
> list archive
> http://lists.opensips.org/pipermail/users/2010-August/014
good day.
Could anybody advise me if it is possible to remove ISUP part from whole
SIP dialogue
Now I have only found that there is no problem to remove it from the common
methods
like
if (is_method("INVITE") ) {
remove_body_part("application/isup");
}
but i see ISUP
ISUP part from the INVITE request, from 10.10.20.20
to the upper trunk but when I receive the 180 ringing signal with the ISUP
part I need to remove it too.
Regards, Alexey.
ср, 28 июл. 2021 г. в 19:46, Vlad Patrascu :
> Hi Alexey,
>
> The remove_body_part() function should wor
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