Hi Gang
I'm having another go with a dedicated kamailio running topos to replace
a B2BUA SBC.
I came across this situation:
Server Topology:
Kamaiolio SBC (topos) <=> Kamailio Registrar <=> Kamailio Router (DLG +
CDRS)
Core IP Address 10.0.0.1
Registrar IP 10.0.0.2
Two different CPE are registered on Registrar via SBC
One CPE is calling the other CPE
So the path the call is taking is:
CPE A => SBC => Registrar => Core => Registrar => SBC => CPE B
Invite A => B goes through, no issue.
180 Ringing, required 100rel B => A requires the route to be recorded
therefore contains 3 record route statements:
Record Route: <sip:10.0.0.2;lr>
Record Route: <sip:10.0.0.1;lr>
Record Route: <sip:10.0.0.2;lr>
To ensure answer will get route back to the registrar => core
=> registrar and then the contact pointing to the SBC with topos to
restore the last hop.
Ringing B => A works as expected and is answered by A with PRACK.
topos would now need set Route: in this PRACK message, according to the
received Record Route Header.
Unfortunately I fear Topos 'removes' the routes with duplicate host ip.
I only get:
Route: <sip:10.0.0.2>
Route: <sip:10.0.0.1>
Causing the Core not to be able to route the PRACK back to the
registrar.
To test, which line is being dropped, I added an mark to each one and
determined the 3rd Record Router is being dropped by topos, probably
considered a duplicate.
I found no setting of the 'do not perform record route header dupe
detection'.
Interestingly, when I look at the database field topos_d.b_rr all
three entries are present in the correct order.
Has anyone come over the same issue and found a solution?
Maybe related to:
https://github.com/kamailio/kamailio/issues/3778 ?
Mit freundlichen Grüssen
-Benoît Panizzon-
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Hello,
Kamailio SIP Server project is organizing another meeting of its
developers and community members during November 19-20, 2024 (Tue-Wed),
hosted again by sipgate.de in Dusseldorf, Germany.
The event is intended to facilitate the interaction between Kamailio
developers and contributors in order to offer a convenient environment
for working together on several topics of high interest for the project,
including writing code for Kamailio and its tools, improving
documentation, or discuss about future development.
Everyone from the community is welcome to join, developer or user
interested in helping the project. Please note we have a limited
capacity of seats in the meeting room, the main policy for accepting
participants being first come first server. Also, very important to be
aware that this is not an event to learn how to use Kamailio.
More details about the event, the venue, how to register, are available at:
* https://www.kamailio.org/w/developers-meeting/
Looking forward to those two intensive hacking Kamailio days in Dusseldorf!
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Hello,
Kamailio SIP Server v5.8.4 stable release is out.
This is a maintenance release of the latest stable branch, 5.8, that
includes fixes since the release of v5.8.3. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.8.x. Deployments running previous v5.8.x
versions are strongly recommended to be upgraded to v5.8.4.
For more details about version 5.8.4 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2024/11/kamailio-v5-8-4-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Hello,
I am considering to release Kamailio v5.8.4 (out of branch 5.8) sometime
next week, most likely on Tuesday, Nov 12, 2024. If anyone is aware of
issues not yet on the bug tracker, report them there asap in order to
have a better chance to be fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
When Kamailio message parser produces this kind of syslog message:
024-11-11T21:16:40.757364+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\340#001>
fail2ban complains:
Consider setting logencoding to appropriate encoding for this jail. Continuing to process line ignoring invalid characters: b'2024-11-11T21:06:15.347318+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\xc0#001>\n': 1 Time(s)
Error decoding line from '/var/log/syslog' with 'UTF-8'.: 1 Time(s)
Unable to retrieve Watch object associated to <_RawEvent cookie=0 mask=0x2 name='' wd=13 >: 3 Time(s)
Would it be possible to somehow ensure that Kamailio syslog messages
only contain valid UTF-8 characters?
-- Juha
Hi all
I noticed, in my config, that if the last line of the reply_route is a
function that does not succeed, then the SIP response is not relayed
and kamailio throws an error. I've searched documentation to indicate
why this might be, but I can't find anything, so this looks to me like
a bug.
I use 5.8.1 for testing.
Here's a sample config that shows what goes wrong.
####################
#!KAMAILIO
listen=udp:203.0.113.57:5060
debug=1
loadmodule "textops"
request_route{exit;}
reply_route{remove_hf_re("^X-Custom");}
####################
Here's a sample SIP message that works. When I send this to
udp:203.0.113.57:5060, kamailio relays it.
####################
SIP/2.0 200 Testing
Via: SIP/2.0/UDP 203.0.113.57:5060;branch=z9hG4bK-d8754z
Via: SIP/2.0/UDP 198.18.0.157:5060;branch=z9hG4bK-d8754z
T: <sip:c@s>;tag=676fbf45
F: <sip:c@s>;tag=676fbf45
X-Custom: lkjdfl
i: testing-XXX
CSeq: 1 INFO
l: 0
####################
If I then remove the X-Custom header field from the SIP reply and try
again, the message does not get relayed and I see this in the log.
5(1356) WARNING: <core> [core/receive.c:605]: receive_msg(): error
while trying onreply script
If I put a log command after the remove_hf_re(), then the message gets
relayed. It looks like the reply_route always fails if the last
function is not successful.
Is this normal or is it a bug?
James
Hi!
We have recently had issues with one of our Production Kamailios.
When those happened, the log was filled with the following message:
CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list
Now I wonder what the best approach is to prevent this.
We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that.
Shared memory is set to 512MB
Can you give me a hint on what to look for?
Thank you very much!
P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉)
FLORIAN FLOIMAIR
Software Development - Symphony Cloud Services
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
[signature_2072127332]
commend.com
LG Salzburg / FN 178618z
When I look at the debian repositories maintained by the project, there are instructions to import the public key of the repository for apt using this command:
wget -O http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add -
which results in:
Warning: apt-key is deprecated. Manage keyring files in trusted.gpg.d instead (see apt-key(8)).
gpg: no valid OpenPGP data found.
Is there an alternate method that can be posted in the repository that will get better results?
Thank you,
TL
Hello,
I want to get a body for content-type "application/isup" but I have the following error:
4(16) DEBUG: textops [textops.c:3394]: ki_get_body_part_helper(): found content type application/isup
4(16) ERROR: textops [textops.c:3397]: ki_get_body_part_helper(): no CRLF found after content type
Here is the INVITE sent:
INVITE sip:+883390000010022@[remote_ip]:[remote_port];user=phone SIP/2.0
From: <sip:+883390000010007@[local_ip];user=phone>;tag=[call_number]
To: <sip:+883390000010022@[remote_ip];user=phone>
Max-Forwards: 70
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: <sip:[local_ip]:[local_port];transport=udp;lr>
Record-Route: <sip:[local_ip]:[local_port];transport=udp;lr>
Call-ID: [call_id]
CSeq: 1 INVITE
P-Asserted-Identity: <sip:+883390000010007@[local_ip]:[local_port];user=phone>
Accept: application/sdp,application/isup,multipart/mixed
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE
P-Charging-Vector: icid-value=IBCF-1-161460836711652742337;icid-generated-at=BC00.QX00.ORANGE-MSS.FR;orig-ioi=ORANGE-MSS.FR
Supported: 100rel
Content-Type: multipart/mixed;boundary=unique-boundary-1
MIME-Version: 1.0
Contact: <sip:[local_ip]:[local_port];transport=udp>
Content-Length:[len]
--unique-boundary-1
Content-Type: application/sdp
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
a=sendrecv
m=audio [media_port] RTP/AVP 8 96
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=maxptime:40
--unique-boundary-1
Content-Type: application/isup;version=spirou;base=itu-t92+
Content-Disposition: signal;handling=required
\x01\x11\x20\x01\x0a\x00\x02\x0c\x0a\x04\x10\x88\x33\x09\x00\x00\x01\x20\xf2\x0a\x0a\x84\x13\x88\x33\x09\x00\x00\x01\x00\x07\x1d\x03\x80\x90\xa3\x3f\x07\x83\x97\x26\x87\x41\x00\x00\x39\x04\x34\xc0\x3f\xc0\x00
--unique-boundary-1--
CRLF is well added at the end of Content-Type header, is anybody could help me to solve this issue?
Regards,
Anthony
Hi Dear Community,
I want to use db_unixodbc to connect to the dm8 database.
Although I use Opensips, I found that this module db_unixodbc, Kamailio and Opensips are the same.
My question is how to specify the database with kamailio? I add "Database = sbc" into odbc.ini, But nothing happened。
######
cat /usr/local/etc/odbc.ini
[DM8]
Description = ODBC DRIVER FOR DM8
Driver = DM8 ODBC DRIVER
SERVER = 172.16.7.131
TCP_PORT = 5230
odbcinst -j
unixODBC 2.3.12
DRIVERS............: /usr/local/etc/odbcinst.ini
SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini
FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources
USER DATA SOURCES..: /root/.odbc.ini
SQLULEN Size.......: 8
SQLLEN Size........: 8
SQLSETPOSIROW Size.: 8
#######
####################
modparam("usrloc", "db_url", "unixodbc://opensips:opensips#$@172.16.7.131:5230/dm8")
####################
DBG:db_unixodbc:db_unixodbc_new_connection: opening connection: unixodbc://xxxx:xxxx@172.16.7.131/dm8
Nov 7 17:05:21 [145] DBG:db_unixodbc:db_unixodbc_new_connection: connection succeeded with reply <DSN=dm8;SERVER=172.16.7.131;UID=opensips;PWD=opensips#$;TCP_PORT=5230;>
DBG:core:db_do_init: connection 0xfffd5bbbaf08 inserted in pool as 0xfffd5bbbb0e0
Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= select table_version from version where table_name='dialog'
Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=42S02:1:-2106:Error in line: 1
####################