Hello,
I am using topos module to hide Record-Route and Route header as per the
requirement from carrier. But the topos module hides, contact, via and
routing headers also. Due to this reason when I am getting bye from the
carrier I am not able to route the bye back to the agent.
Bye from Agent to carrier is working fine.
Bye from carrier to my internal network interface is not routing to client
through public interface. I assume this may be because Via / Contact header
has been hidden and changed by the topos module with the invite
please help me for this
Thanks
Arun
Hello,
Kamailio SIP Server v5.8.2 stable release is out.
This is a maintenance release of the latest stable branch, 5.8, that
includes fixes since the release of v5.8.1. 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 v5.8.2.
For more details about version 5.8.2 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-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 v5.8.2 (out of branch 5.8) on
Wednesday,
June 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
Hi,
In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support
for server (RFC7339)". Where can I find some more details about the feature?
Also, the similar question about the new "influxdbc" module. What are the
possible use cases for this module?
Thank you,
Leonid Fainshtein
Xorcom Ltd
I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3
route[HANDLEREDIRECT] {
xlog("L_INFO", "DBG: HANDLEREDIRECT");
msg_apply_changes();
$var(test) = $hdr(Contact);
xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]");
# send to gateway
$var(new_uri) = "sip:" + $tU + "@" + "gateway";
if ($var(new_uri) != "") {
xlog("DBG: Redirecting to: $var(new_uri)\n");
$ru = $var(new_uri);
$tu = $ru;
record_route();
route(RELAY);
exit;
# i have also tried using
#t_newtran();
# and
#t_relay();
} else {
xlog("DBG: No contact header found in 300 response\n");
}
}
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("DBG: new branch [$T_branch_idx] to $ru\n");
xlog("L_INFO", "DBG: manage branch");
route(NATMANAGE);
return;
}
onreply_route[MANAGE_REPLY] {
xdbg("DBG: incoming reply\n");
xlog("L_INFO", "DBG: manage reply");
if (t_check_status("3[0-9][0-9]")) {
xlog("L_INFO", "DBG: got 3xx redirect");
route(HANDLEREDIRECT);
drop;
}
return;
}
I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params)
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) exit;
if (t_check_status("3[0-9][0-9]")) {
xlog("L_INFO", "DBG: following 3xx redirect");
get_redirects("3:1");
t_relay();
exit;
}
}
Rob Green
Chief Information Officer (CIO)
Tel: 1-352-218-1490
Cell: 1-407-620-0932
Email: rob.green(a)nuwave.com
Web: www.nuwave.com
This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.
Hi everyone,
I am creating a dialogue when I receive INVITE. Is there a way to set
key/value pair like attributes at dialog level, so that, when I receive BYE
message I can retrieve that information and do some processing?
Thank you,
Pavan Kumar
Hi
I wonder if anyone could maybe recommend what to use as signaling B2BUA
to work with Kamailio to handle spiralling / looping calls and to reduce
the number of Route, Via Header and message size, which cause issue
with a large number of CPE.
Requirements:
* New CallID needs to be generated
* Handling Route and Via
* ~ 2000 concurrent calls.
* Preserving custom SIP X- header
* Routing NOTIFY for MWI etc.
Handling RTP is not required!
What I have been looking at more closely:
= FreeSwitch =
Got stuck trying to solve the dependencies to get it to run/build on
ubuntu jammy.
Has anyone experience in running it in a docker container? Will it
perform?
= Sippy =
Very small, very simple, unfortunately removes customer SIP headers and
does not forward NOTIFY. (It's python, so maybe this could be easily
implemented/fixed).
= SEMS =
Does not seem to be actively developed in the last, couple of years.
Looks quite bloated to me for what I need.
= Asterisk =
Each customer header has to be copied, wants to handle RTP and bridge
by issuing Re-Invites. Not sure how to route stuff like NOTIFY.
Also from experience I fear asterisk would translate sip reply codes
(like 608 and 607) it does not know to some 4XX default.
--
Mit freundlichen Grüssen
-Benoît Panizzon- @ HomeOffice und normal erreichbar
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hi all
I'm trying to make kamailio not to insert a row in missed_calls table when
reply code is 407. But even if I set the "failed_filter" modparam the 407 row
is inserted.
The acc mmodparams are:
modparam("acc", "db_table_acc", "kam_acc")
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACC_MISS)
modparam("acc", "db_url",DBURL)
modparam("acc", "db_insert_mode", 1)
modparam("acc", "time_mode", 2)
modparam("acc", "time_attr", "time_hires")
modparam("acc", "failed_filter", "401,407")
modparam("acc", "db_extra", "whatever"
db is mysql. Kamailio version is 5.7.4
When I send a call that receives a 407 and after auth a 480 y get both rows in
missed_calls table.
Log:
DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc
DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401
DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407
DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for
t(0x7f7cec807480) event type 512, reply code 407
DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407
flags: 0x220
DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off
DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for
t(0x7f7cebf59bc0) event type 512, reply code 486
DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486
flags: 0x220
DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off
am I configuring this wrong?
cheers,
Jon
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
Hey everyone,
I have the following setup:
External User/Trunk <===> Kamailio <===> FreeSWITCH
I have configured FreeSWITCH and Kamailio to work only with phone numbers
in E.164 format. Recently, I needed to integrate a new SIP trunk that sends
phone numbers in an 11-digit US format.
My initial idea to solve this is to use Kamailio as a translator that
converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to
11-digit when sending to the trunk.
Before writing the code, I want to know if there is already a solution to
this problem in one of the modules, so that I won't reinvent the wheel.
Thanks,
Pavan Kumar