Hello, i need your help,
I'm having problems with the reverse response of an INVITE passing through a
rtpproxy (NAT). Does anyone have any idea what might be going on?
UE is informed of the result INVITE, but the UAS does not receive the test
bench detail.
Greetings.
--
Rodrigo M.
(37) 9132-4539
(34) 9889-3069
rodrigo.moreira2007
I am using kamailio with ubuntu, but I can't obtain CDRs.
In my conf. file related to CDR I have:
modparam("acc", "cdr_enable", 1)
modparam("acc", "cdr_start_on_confirmed", 1)
modparam("acc", "cdrs_table", "acc_cdrs")
modparam("acc", "cdr_start_id", "start_time")
modparam("acc", "cdr_end_id", "end_time")
modparam("acc", "cdr_duration_id", "duration")
But I don't know where is the problem.
Regards
Diogenes Marcano
+58 414 117 2011
Hello everybody,
I am having issues with one SIP vendor.
I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk
and Media Gateways.
Calls get established and I have two way audio but when the remote party
hangs up the call, the BYE arrives to the Kamailio and does not move
forward.
I think the problem is SIP vendor rewrite the BYE header and change the
asterisk IP with the public IP of the kamailio.
The IP that appears in the header of the BYE have to be the same that
appears in the contact (UAC that send the call, in my case the Asterisk).
Vendor should not change that IP. ¿Am I correct?
Thank you
-----------------------------------------------------------------------------------------------------
INVITE
----------------------------------------------------------------------------------------------------
2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060
INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0
Record-Route:
<sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U
G9jYWw-;did=09b.9572;nat=yes>
Record-Route:
<sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U
G9jYWw-;did=09b.9572;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1
Via: SIP/2.0/UDP
PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060
Max-Forwards: 69
From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c
To: <sip:DESTINATION-NUMBER@VENDOR-IP>
Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060>
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 102 INVITE
User-Agent: UAC
Date: Mon, 17 Oct 2016 16:53:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP
s=Asterisk PBX
c=IN IP4 PUBLIC-KAMAILIO-IP
t=0 0
m=audio 23456 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
-----------------------------------------------------------------------------------------------------
BYE
-----------------------------------------------------------------------------------------------------
2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060
BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0
Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0
Via: SIP/2.0/UDP
VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206
Max-Forwards: 34
Route:
<sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1>
Route:
<sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1>
To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c
From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 154701 BYE
User-Agent: VENDOR
Content-Length: 0
-----------------------------------------------------------------------------------------------------
Hi all,
I am testing Edge Proxy setup with the example given here :
http://kamailio.org/docs/modules/4.2.x/modules/outbound.html
and I get this error :
U 2016/10/06 09:18:48.331580 10.0.0.4:5060 -> 52.233.25.164:5060
SIP/2.0 420 Bad Extension.
Via: SIP/2.0/UDP
10.1.0.4;branch=z9hG4bK162a.e9f73fe5e52ded1f30801aa00ee2a771.0;
received=52.233.25.164.
Via: SIP/2.0/UDP
10.114.70.81:1228;received=113.193.149.49;rport=1228;branch=z9h
G4bKPj3ec7c87e09d6452b9e68e60052cb2651.
From: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;
tag=ac1e36aec8bf445bbec8e43cf28e9f48.
To: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;ta
g=b27e1a1d33761e85846fc98f5f3a7e58.b172.
Call-ID: a270d285de6145c6b8670b909b104e74.
CSeq: 20149 REGISTER.
Contact: <sip:user1@10.114.70.81:1228;ob>;expires=300.
Unsupported: path.
Path: <sip:10.1.0.4;lr>.
Supported: outbound.
P-Registrar-Error: No support for found Path indicated.
Server: kamailio (4.4.3 (x86_64/linux)).
Content-Length: 0.
Both my proxy and registrar are behind NAT. rtpproxy is installed and started.
I already have modparam("registrar", "use_path", 1) in my registrar.
Any idea what is missing? Thanks.
regards
Ganesh Kumar
Hi,
Can we use sip router as IMS PCSCF to connectivity with IMS.
Does sip router supports the diameter RX interface to connectivity with PCRF .
Regards
Surender Singh
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Hi,
I am trying to register sipml5 webrtc client to my kamailio IMS setup. I
have tried to register the client both with ws and wss, but it seems to be
that the sipml5 doesn't calculate the authentication digest right. The
authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple
wireshark file is attached. .10 being the host, .11 being the kamailio
server.
>From the output of scscf we can see that the digests do not match.
[REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test]
ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test
nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA=
response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=00000001
cnonce=2d4545cf4c935c8094c8b1da3d4a2976
hbody=d41d8cd98f00b204e9800998ecf8427e
ims_auth [authorize.c:872]: authenticate(): UE said:
61cbdbeb47c9880ededfca51c3801800 and we expect
7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77
(REGISTER)
tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a
message for which no T-state has been established
Thanks a lot,
Serhat
How to use the sql transformation?
https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transform…
has the following example:
xlog("$$rm = $rm = $(rm{s.sql})");
But adding this to the request_route and starting kamailio will fail:
ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: sql}/sql/3!
ERROR: <core> [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}]
ERROR: <core> [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm"
ERROR: <core> [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(rm{s.sql})] at (4)
ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = $(rm{s.sql})]
ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:372
line 372 is the above xlog and sqlops.so is loaded (and works). Anybody
got a working example of this? Or an other hint to prevent sql
injections when using user supplied variables in sql queries?
Hi
503 Service unavailable received at kamailio SIP server is forwarded as 500
server internal error to the UAC. Which seems fine based on the fact that
there is no retry-after header.
Would like to know if there is any way to to control the SIP Headers like
adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created
by kamalio stack for 503 service unavailable (without retry-after Header)
It seems that Kamalio does not retain the proprietary sip headers received
in 503 to 500 response forwarded.
Any insight into the issue would be helpful
--
Warm Regards
Dhruvin Desai.
Hello,
I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which IP I should configure RTPengine to send the stats?
We have a Kamailio server configured with the sipcapture module, which works perfectly for SIP traffic, but RTP stats are not visible in Homer.
Do I need to point RTPengine to the sip capture server or directly to Homer?
Running RTPEngine 4.5.2 and Kamailio 4.4.1.
Regards,
Grant
Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if
it is under development?
It would be useful to extract RTCP stats from SRTP streams, the
standalone HOMER captagent can't see inside them when they are encrypted
(SRTCP)
Regards,
Daniel
