Hi,
I tried to setup rtpengine to make my sip clients communicate through proxy
following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/
I started sip communication since the service was running and saw that the
RTP packets go to exposed rtp interface instead of other sip client. But i
don't receive packets from the other end.
When analyzed I noticed the rtp-engine daemon gives a few errors.
*modprobe: FATAL: Module xt_RTPENGINE not found in directory
/lib/modules/4.15.0-118-generic*
*iptables: No chain/target/match by that name.ip6tables: No
chain/target/match by that name. * Starting RTP/media proxy
ngcp-rtpengine-daemon
[1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such
file or directory), KERNEL FORWARDING DISABLED*
*(Note: *uname -r gives 4.15.0-118-generic and /lib/modules/ has only
4.15.0-115-generic*)*
Please help me with this issue. Thanks in Advance.
Regards,
Pavithra
Hello,
please keep the list in CC. I can see Kamailio in your log:
Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received
Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open Source SIP Server...
Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received
About tracing/debugging, one good option is sngrep.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: Santanu B <santanu.br(a)gmail.com>
Sent: Wednesday, October 14, 2020 1:41 PM
To: Henning Westerholt <hw(a)skalatan.de>
Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio
Hi,
My syslog file is enclosed for your reference.
Thanks,
Santanu
On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote:
Hello,
have a look to /var/log/messages or /var/log/syslog, depending on the distribution. Kamailio might be logged there.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Santanu B
Sent: Wednesday, October 14, 2020 7:09 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio
Dear David,
In /var/log folder I am unable to see any kamailio log. Only syslog file is available. Please help me to find the Kamailio log.
Also please guide me to find out the trace also as I am very new to SIP Server.
Thanks a lot.
Regards,
Santanu
On Tue, Oct 13, 2020 at 4:05 PM David Villasmil <david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>> wrote:
Hello,
You will need to provide more information. What happens? What errors do you see in kamailio’s log? A trace would also be nice.
On Tue, 13 Oct 2020 at 06:24, Santanu B <santanu.br(a)gmail.com<mailto:santanu.br@gmail.com>> wrote:
Hi,
We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure.
Thanks and regards,
Santanu
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>
phone: +34669448337
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi All,
I have setup a tls connection to a sip provider and it works perfectly
But they are requesting i send srtp
i can find any documentation how i do it with rtpengine etc?
any guidance would be appreciated
Hii,I am trying to use pstn_route module for the first time, and i want to route the invite to specific pstn server based on its prefix.I have loaded the module & setup the DB, and it's giving error saying route name (from DB) is not defined after adding the a prefix into DB table., and there were no errors when DB table is empty.Not sure how to defined it, or i dont know what did i miss. Please help.
#### CONFIG ##########loadmodule "prefix_route.so"modparam("prefix_route", "db_url", "mysql://xxxxx:xxxxx@localhost/kamailioDB")modparam("prefix_route", "db_table", "new_prefix_route")
#PSTN Routing table contains if (!prefix_route("+44")) { xlog("L_INFO", "+44 prefix didnt match with prefix_route DB\n"); }
########## DB####MariaDB [kamailio]> select * from new_prefix_route;Empty set (0.00 sec)
MariaDB [kamailio]> exit;##########
Kamailio looks good untill here, no errors..
# Now i have added a prefix into sql DBINSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route for this prefix");###########Error log after adding a prefix into new_prefix_route table & restart of kamailio
0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route(): route name '10.10.1.1:5060' is not defined 0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load(): Total prefix routes loaded: 1 0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error flushing tree 0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db load failed
I have tried a "name" instead of IP (though i dont know how to convert that "name" into a IP:Port), But i am still getting same error using "name" instead of IP:port @ route field in DB.
Please help.
Hii
Please help me to get the a value from SIP INVITE header reached to kamailio like INVITE sip:+341930203454@sub.domain.com;myid=+34@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a variable, without the domain name.
$var(uri) = $sel(ruri);
xavp_params_explode("$(var(uri){s.unbracket})", "uri");
xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n");
I tried above and it prints +34(a)sub.domain.com But i want to just save +34 into a variable to further check the prefix based routing from the database.
1) Could you please help how to get it or If there is any alternate/single line approach to get this value?
2) $var & $xavp are process-local variables, and they cant be shared with other calls? Right
3) What is the best module to route calls based their prefix ? pstn_route or dynamic routing or any suggestions?
Please help, thanks in advance.
Hello,
I see parsing errors on 302 contact header that seems compliant to RFC :
Contact:<sip:172.16.0.21:5060>;q=0.5,<sip:172.16.0.22:5060>;q=0.25
<core> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in status 27: [;q=0.5]
Is there an error in my contact or is it truly a parsing error ?
Regards,
David
Hello Team,
I am working for a telecom company and want to use Kamailio for my VoLTE
testing.
I had earlier used these tools for my VoLTE testing using a Polaris CNE.
But i have a requirement to perform SRVCC Handover for CS/CS+PS.
will Kamailio support the SRVCC Handover?
If supported, can it be possible to share the details ?
--
Best Regards,
Anwesh Babu T
Hi All,
What is the best way to approach this?
example if a call comes into kamailio with a specific number it example
to:123456789 it must go to the dispatcher module
but if a call comes with any other number it must route to a uac
or can you use uac with dispatcher?
basically i want to route inbound calls to a dispatcher and calls from that
dispatcher out a uac
Hello,
Is it possible write more readable regular expression
if ($tU =~ "(?i)^test$") {
...
}
instead of "^[tT][eE][sS][tT]$"?
On the first option, I get the error:
0(47250) ERROR: <core> [core/rvalue.c:3038]: fix_match_rve(): Bad regular
expression
Regards,
Marat
