Hi Guys,
I have a kamailio server 1.3 with clear text Password in Subscriber table.... I wish I can encrypt that password by doing some kind of combinations between username, domain and let kamailio server be reading the encrypted password
Is there a way to do it?
modparam("auth_db", "db_url", DBURL)modparam("auth_db", "calculate_ha1", 0)modparam("auth_db", "password_column", "password")#modparam("auth_db", "password_column", "ha1")modparam("auth_db", "password_column_2", "ha1b")modparam("auth_db", "load_credentials", "")modparam("auth_db", "use_domain", MULTIDOMAIN)
I have made a calling service through kamailio and rtpengine.In this setup
of calling service i have also included webRTC for cross platform(like
android phone to web browser).Same calling service is not working after
addition of freeswitch.Additional integration of Freeswitch in this current
setup is making trouble.
I have followed below link to setup my rtpengine module.
https://github.com/caruizdiaz/kamailio-ws
<https://github.com/caruizdiaz/kamailio-ws>
My requirement is to integrate all these three to make cross platform
calling system
Kamailio<-->Rtpengine<-->Freeswitch
But the calls are not relaying to freeswitch via rtpengine.
Help me out to relay calls from rtpengine to freeswitch in this setup.
Thanks and Regard
Md Safdar Khan
Thanks for reply.
I check one more time, and does not see any non SIP (HEP) packets
received on port 5060 of kamailio.
May bee parser trying to parse (HEP) packet when it going out via
socket to capture server?
When I turn of topoh module I does not see any errors in log.
And in both cases I see HEP packets on kamailio capture server.
--
Best regards,
Sergey Basov e-mail: sergey.v.basov(a)gmail.com
SATB-RIPE
SATB-UANIC
tel: (+38067) 403-62-54
Hello,
What is the simplest way to configure Kamailio to generate periodical
OPTIONS to the caller, for an established dialog?
Can you give me some starting references?
Thanks,
Stefan
Hi all,
I need to implement a WebRTC gateway for an existing conference bridge. The
WebRTC gateway has to support Signaling, ICE and DTLS. The webrtc clients
can be JsSIP or any webrtc client.
The conference bridge is an existing working one for SIP clients, and I am
trying to add webrtc support for that.
The webrtc gateway needs to be implemented in a way like a library because
it needs to be integrated into the existing platform.
There are some init functions and config function from the existing
conference platform, based on which the webrtc gateway has to be
configured.
Also, when a webrtc call come from a webrtc client, it needs to handle the
signaling and the media(RTP) has to go to the conference bridge platform.
It would be really helpful if you suggest whether I can use openSIPS for
this purpose and use it as a library and integrate into the exiting
platform?
Your suggestions will be more helpful.
Thanks.
Hi, All.
I have a strange issue occuring, when I enable the siptrace
functionality within the routing logic, I begin to get a lot of
parser error as show below. Is this normal behavior?
This errors appears only when topoh module is enabled...
As I see with wireshark, there is no duplicates or returned HEP packets...
I enable siptrace as follows:
request_route {
sip_trace();
setflag(22);
...
}
Configuration of the topoh and siptrace modules:
# ----- topoh params -----
modparam("topoh", "mask_key", "123456789")
modparam("topoh", "mask_ip", "127.0.0.8")
#Siptrace
modparam("siptrace", "duplicate_uri", "sip:10.1.23.20:9060")
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "force_send_sock", "sip:10.1.23.23:5060")
modparam("siptrace", "traced_user_avp", "$avp(s:user)")
output from kamailio.log with debug enabled:
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh
[th_msg.c:180]: th_mask_via(): body: 120: [SIP/2.0/UDP 10.10.206.39:5060
;received=10.10.206.39;TH=dcv;branch=z9hG4bK-d8754z-2e9df22b7c5cb6c9-1---d8754z-;rport=5060]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh
[th_msg.c:191]: th_mask_via(): +body: 199: [SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-s7wTDLa0zUfYZXl5zpl0zpl0z.lRD.zok.sEG.lvBJY.euwReuTfzpl0zpl0z.lRD.zokqZaxuZ.P.c3BJX01R5fO.wCZLZ3WSqMkhBqGgCczJsoeV1SzJa21LY.1.e.kW7jDW7ceh52GpZKDpcSBVfSPh7qzh1E]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh
[th_msg.c:322]: th_mask_record_route(): no record route header
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_hooks.c:266]:
run_trans_callbacks_internal(): DBG: trans=0x7fcfacccb7a8, callback type
4194304, id 0 entered
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:766]: sip_trace_store_db(): database connection not initialized
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:1875]: pipport2su(): the port string is 5060
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:1875]: pipport2su(): the port string is 5060
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[proxy.c:265]: mk_proxy(): doing DNS lookup...
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[resolve.c:1208]: srv_sip_resolvehost(): 10.1.23.20:9060 proto=1
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[resolve.c:1329]: srv_sip_resolvehost(): returning 0x9da020 (10.1.23.20:9060
proto=1)
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:1687]: trace_send_hep_duplicate(): setting up the socket_info
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:1694]: trace_send_hep_duplicate(): force_send_sock activated,
grep for the sock_info
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 &&
[10.1.23.23] == [10.56.41.23]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0)
matches port 5060
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 &&
[10.1.23.23] == [10.56.42.23]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0)
matches port 5060
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 &&
[10.1.23.23] == [10.1.23.23]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0)
matches port 5060
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace
[siptrace.c:1701]: trace_send_hep_duplicate(): found socket while grep:
[10.1.23.23] [10.1.23.23]
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: INFO: <core>
[parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line:
method not followed by SP
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core>
[parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad
message (offset: 0)
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core>
[parser/msg_parser.c:602]: parse_msg(): parse_msg: invalid message
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core>
[parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg:
message=<#002#020#002#021#023�#023�#0128*#027#0128*#024zx�V�#016#003>
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh
[topoh_mod.c:214]: th_prepare_msg(): outbuf buffer parsing failed!
Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_funcs.c:362]:
t_relay_to(): SER: new transaction fwd'ed
Is there any way I can disable seeing these messages? What are these
messages?
Any thoughts are appreciated.
Thanks in advance.
--
Best regards,
Sergey Basov e-mail: sergey.v.basov(a)gmail.com
Hi,
We are using attached two configuration file to start icscf node.
But while we run "kamailio start" command getting below errors.
loading modules under config path: /usr/lib/i386-linux-gnu/kamailio/modules/
ERROR: bad config file (1 errors)
Segmentation fault
Can anyone help me here to resolve this issue.
Regards,
-kranti
Hi friends:
I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine.
But there is always prompts like following in rtpengine’s log:
Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30136] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible
Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible
Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible
So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen?
Thx a lot!
------------------------------------
北京邮电大学网络技术研究院
网络与交换技术国家重点实验室
田军
+86 18810315790
mozillafire(a)bupt.edu.cn
------------------------------------
Alright I guess this is one of those duhh… moments, all this time I was forgetting to add the content length to the NOTIFY event. Remark: DO NOT FORGET the content length!
Thanks for the help, very much appreciated!
I see, what i’m trying to do is actually to limit querying the DB by updating the MWI only when there is activity.
In other words to send a NOTIFY when the user receives a new message in his voicemailbox.
I am not sure if it’s possible to send the NOTIFY for example 5-30 mins after the initial subscribe.