Hello,
I am using siptrace module in Kamailio. As you know, it sends a copy of
signalling message for specific destination. Like HOMER project.
Because of i won't run main route block in destination, i just have
changed this block like below:
request_route {
//do some works.
drop();
exit;
}
I know it would make a question in your mind why i do this? No problem,
Let's come back to question. All things works correct except Replying SIP
message. The destination is trying to relay Replying SIP message that it
get from siptrace module in other server.
How can i disable replying SIP mesages?
With Best Regards
--Mojtaba Esfandiari.S
Hi.
When using the attributes feature of the dispatcher module, is it possible to directly access/address an individual attribute directly?
For example, if I use the following dispatcher text file:
--
# line format
# setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt)
1234 sip:10.10.10.10:5060;transport=tcp 0 0 attr1=red;attr2=blue
--
And this config snippet, I was hoping to be able to access the attribute with a variable.
--
modparam("dispatcher", "attrs_pvname", "$vxavp(ds_attrs)")
if(ds_is_from_list(1234,1)) {
x_log(“L_NOTICE”,”attr1: $xavp(ds_attrs=>attr1)\n”);
}
--
Jason Park | Lead Engineer | Voice Services | • Target<http://www.target.com/> | TNC | 816-273-8336
Hi,
I have the following architecture - SIP provider <-> Kamailio <-> Asterisk
servers
Currently I have everything setup and incoming calls from Sip are routed to
my asterisk server. The issue is however that when I answer the call, there
is no media in the call. I have tried connecting with a normal local
extension(not SIP,eg 1001) and there is a normal flow of media.
When i try to sniff my connection via Wireshark on the asterisk server,
there is an outflow of RTP packets but the same RTP traffic does not appear
on the Wireshark of my Kamailio server connection.
I am not sure if this is an RTP engine issue and how to resolve this.
I have -
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
Hello,
I tried to test the rabbitmq module in kamailio with the debian 9 repo
from kamailio.org but there is a segfault when the service is launched.
Is there is people also have this issue?
/etc/init.d/kamailio restart
[....] Restarting kamailio (via systemctl): kamailio.serviceJun 9
13:02:03 siprouter kamailio: DEBUG: <core> [core/cfg.y:1659]: yyparse():
loading module rabbitmq.so
Jun 9 13:02:03 siprouter kamailio: DEBUG: <core>
[core/sr_module.c:575]: load_module(): trying to load
</usr/lib/x86_64-linux-gnu/kamailio/modules/rabbitmq.so>
Jun 9 13:02:03 siprouter kamailio: DEBUG: <core> [core/kemi.c:1295]:
sr_kemi_modules_add(): adding module: rabbitmq
Jun 9 13:02:03 siprouter kamailio: DEBUG: <core> [core/cfg.lex:1737]:
pp_define(): defining id: MOD_rabbitmq
Jun 9 13:02:03 siprouter kernel: [159315.610811] kamailio[13226]:
segfault at 7f03b2296287 ip 00007f03b2080ca3 sp 00007fff133285c0 error 7
in librabbitmq.so.4.2.0[7f03b2074000+13000]
Jun 9 13:02:03 siprouter /usr/sbin/kamailio[13226]: DEBUG: <core>
[core/sr_module.c:988]: init_mod(): rabbitmq
Sylvain
Hi, I would like to know if there is a possibility to see a user's
connection history, or at least their last connection. Thank you
--
José Antonio Gutiérrez Delgado
Responsable técnico
Oficina: (+34) 923040031
<http://www.arsoft-company.com/>
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Hello,
We're having an issue with rtpengine (used by Kamailio) where audio works
initially, but then after an apparently random amount of time stop working.
We see that when audio stops working rtpengine logs this:
Dec 10 09:58:57 hostname rtpengine[376]: WARNING: [Pl1SeGDssOsDNWQdvey4lg..
port 48766]: Discarded invalid SRTP packet: authentication failed
It then logs similar messages until the call hangs up. No such messages
were logged while audio was working.
Searching for this error message suggests that a change in the SSRC can
cause the problem, but we don't see any such change in the PCAP. The source
IP, port, codec, and SSRC all stay the same, and the Sequence increments as
normal.
Does anyone have suggestions on where to look next? We can share the PCAP
privately if that would help anyone.
Thanks for any advice!
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
Yes, The DNS NAPTR records are correct. I have all running on single laptop, P:4060, I:5060, S:6060 and all DNS tests verified.
I don’t have any traffic other than the P talking to itself until the number hops is exceeded.
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Iman Mohammadi
Sent: Wednesday, January 9, 2019 9:49 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: [EXT] Re: [SR-Users] P-CSCF returns SIP error: 483 Too may hops
Hi
Check the Ports in DNS and in configuration files.for example if you set 5060 in configuration of pcscf,set 5060 in dns for pcscf too
For Registration REGISTER goes from pcscf to icscf first then icscf gets address of scscf in UAA from HSS,then icscf forwards REGISTER to scscf and ...
On Wed, Jan 9, 2019, 18:04 Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> wrote:
Hi all,
I’ve stood up IMS 4.3.4. When I try to register with a boghe IMS client the P-CSCF gets in a loop and a SIP error: 483 Too may hops is returned.
I would think the path would b to S-CSCF:6060 which would then check the HSS for user bob.
The P-CSCF log attached.
I believe Im missing some basic configuration setting here.
If there is configuration (test scenario) documentation that I can be pointed to, it’d be appreciated.
R,
martin
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