Good evening,
I'm quite new in the Kamailio community so at the beginning I just want to say "Hello" to everyone.
I have a following problem with SNMP in Kamailio.
I want to collect data from Kamailio by SNMP. I've installed snmpd on the machine (Linux RHEL) and it's working. I've load snmpstats module in Kamailio and it starts up without error. I've configured Kamailio to use agentXsocket and enable agentx in SNMP. Port 705 is opened and netstat shows connection established between Kamailio and SNMP. Wireshark shows some packets going through. The problem is that there is no data during snmpwalk from Kamailio. Just standard data from SNMP, nothing from Kamailio. I've copied Kamailio MIBs to the proper directory, added it in the snmp.conf. Moreover I've tried to add -m +KAMAILIO-REG-MIB to snmpwalk. Still no data from Kamailio during snmpwalk.
Any ideas?
Best Regards,
Michal Muszalski
------phona A-------kamailio---------asterisk-----
OPTION 1: configure asterisk or kamailio
i used asterisk, and install kamailio for traffic RTP can be send between
end points that behind NAT router, and do not have to go through RTP
proxy,, plz help!
i think to the moment install kamailio, headrs'sdp fix IP private, but no!,
how can fix it plz!!?
regards!
or
OPTION 2: edit sip/sdp
mi sip/sdp is
[code]
<--- SIP read from UDP:152.74.21.12:5060 --->
ACK sip:1001@152.74.21.12:6112 SIP/2.0
Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 190.164.204.227:41553
;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553
Max-Forwards: 16
Contact: <sip:JavierTren@190.164.204.227:41553;transport=UDP>
To: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1
From: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c
Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.
CSeq: 2 ACK
Proxy-Authorization: Digest
username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri="
sip:1001@152.74.21.12
;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5
User-Agent: Zoiper r27147
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:152.74.21.12;lr=on;ftag=2b8fa52c> for
address/port to send to
set_destination: set destination to 152.74.21.12:5060
Audio is at 12064
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 152.74.21.12:5060:
INVITE sip:JavierTren@190.164.204.227:41553;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 152.74.21.12:6112;branch=z9hG4bK3992226d;rport
Route: <sip:152.74.21.12;lr=on;ftag=2b8fa52c>
Max-Forwards: 70
From: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1
To: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c
Contact: <sip:1001@152.74.21.12:6112>
Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.0
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 1842142539 1842142540 IN IP4 192.168.1.8
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.1.8
t=0 0
m=audio 8000 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
[/code]
and the field o and c i need to IP public y no private,,
plz any?
Greetings,
I'm fighting a number of errors in the 'acc' table like so:
Nov 21 01:43:05 landsort postgres[53327]: [4-1] ERROR: column "time" of relation "acc" does not exist at character 78
That is, I believe, the default column name.
Thing is, though, that the DDL that I used to build the table, is the
one in in scripts/ and it does not does not contain that string:
(on master but it is the same in 4.2 which is what I've got in production)
$ for d in mysql oracle postgres ; do egrep '\btime\b' scripts/${d}/*create* ; done
$
How are those files generated?
How do I come about to submitting an useful patch?
What is the desired datatype for the time column?
--
Måns Nilsson primary/secondary/besserwisser/machina
MN-1334-RIPE +46 705 989668
Edwin Meese made me wear CORDOVANS!!
Richard Fuchs wrote:
> The command flags "RTP/AVP" etc are translated into the key
> "transport-protocol=$X" within the control protocol. If you wish to
> force usage of one of the UDP/TLS/... protocol, you should be able to do
> so by spelling it out in your command. I believe this is not documented,
> but should work.
i tried and the offer and audio worked accordingly:
Nov 21 05:37:15 rautu /usr/bin/sip-proxy[11113]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 transport-protocol=UDP/TLS/RTP/SAVP trust-address)
for consistency, it might be a good idea to make the dtls flags known to
rtpengine module.
-- juha
Hello! I'm having this issue and I'm not able to solve it. I have searched
it many times on the Internet but I found nothing, that's why I am asking
here... I'm sorry I do not wanna cause any troubles but I need to get this
done...
This is my scenario, I want to make a call through a Kamailio server that
redirects to another server. I have an User A registered to Server A and a
User B registered to Server B, what I want to do is the following:
User A calls User B using Server A, because Server A doesn't have the User
B the Server A will redirect to Server B and make the call. I'm using Ekiga
and of course Kamailio (and I want to make it work both ways).
This is the code I'm using to do this in kamailio.cfg, I replaced this
lines of code:
*$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) =
$rc; route(TOVOICEMAIL); t_newtran(); switch
($var(rc)) { case -1: case -3:
send_reply("404", "Not Found"); exit; case -2:
send_reply("405", "Method Not Allowed");
exit; } }*
For this ones:
*$avp(oexten) = $rU; if (!lookup("location")) {
sl_send_reply("300",("REDIRECT"));
rewritehostport("10.80.129.177"); forward(uri:host,uri:port);
break; }*
And it didn't work, then I putted the lines above as the following:
*$avp(oexten) = $rU; if (!lookup("location")) { *
* forward(uri:host,uri:port); rewritehostport("10.80.129.177");*
* sl_send_reply("300",("REDIRECT")); break; }*
And it worked but if I do a Wireshark Capture on the Server I get a lot of
REDIRECT packets and my teacher says this is wrong, that I should only get
1. He did not help me and said I must learn and I want but I'm not able to
understand what's going on, I need to do this because I want to pass the
exam and not to fail...
Again I'm sorry for taking your time, I'm asshamed.
Thank you very much
On 11/18/2014 08:29 PM, Juha Heinanen wrote:
> Richard,
>
> Enclosed find syslog that includes rtgpengine log at level 7 and also
> pcap of the call. I made re-invite from baresip to sems once I saw this
> in syslog:
>
> Nov 19 03:22:14 rautu rtpengine[29718]: [5315d797a9a5e7ce port 50761] DTLS-SRTP successfully negotiated
>
> I didn't include the to this message due to the pcap.
Looking at your logs, I think what happened is that you've included the
"trust-address" flag in the original answer, but haven't included it in
the answer to the re-invite. This means that in the second answer,
rtpengine erroneously saw a different endpoint than before (using the
source address of the SIP packet), in response to this it also opened a
new endpoint on its own side, which was then sent to the DTLS-SRTP
client. This client then saw the new endpoint and correctly started a
new DTLS connection, meanwhile rtpengine was still using the old keys
and (I'm guessing) neither expected nor processed the new DTLS
handshake, thus en/decrypted SRTP using the old keys, resulting in
mangled RTP packets.
So the short-term fix would be to include trust-address to avoid
switching endpoints. As for DTLS restarts in rtpengine, this is
something that I was just working on and DTLS restarts on changed
endpoints should be working soon.
cheers
Does setting $shv()s in script require lock()ing, or is it inherently
thread-safe?
Thanks!
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Hi Dears,
I'm trying to configure Kamailio as SBC in multi home mode for Asterisk by
authenticating the inbound SIP registration requests,i'm following this
tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
to achieve this goal. i have modified the necessary changes like the
Asterisk DB URL and the SIP table name and Username and password column and
verified the connection.
My topology like this *Asterisk (192.168.100.10)
<----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP
Phone (192.168.50.2)*
But when trying to register a SIP phone Kamailio does NOT forward the
authentication request to Asterisk and sends 401 Unauthorized error
message.I've attached my config file if any one wants to check it and
thanks in advance.
Best Regards
U 192.168.50.2:37297 -> 192.168.50.1:5060
REGISTER sip:192.168.50.1;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 192.168.50.2:37297
;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport;transport=UDP.
Max-Forwards: 70.
Contact: <sip:1001@192.168.50.2:37297
;rinstance=1d7c44dbcb8a7a2f;transport=UDP>.
To: <sip:1001@192.168.50.1;transport=UDP>.
From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19.
Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
CSeq: 2 REGISTER.
Expires: 70.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
User-Agent: Z 3.2.21357 r21367.
Authorization: Digest
username="1001",realm="192.168.50.1",nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D",uri="sip:192.168.50.1;transport=UDP",response="8bbd01d879250585eafee4f510689f73",algorithm=MD5.
Allow-Events: presence, kpml.
Content-Length: 0.
#
U 192.168.50.1:5060 -> 192.168.50.2:37297
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.50.2:37297
;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport=37297;transport=UDP.
To: <sip:1001@192.168.50.1
;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe8b.
From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19.
Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
CSeq: 2 REGISTER.
WWW-Authenticate: Digest realm="192.168.50.1",
nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D".
Server: kamailio (4.1.6 (i386/linux)).
Content-Length: 0.
Hello,
I saw that rtpengine docs still advertise support for SDES SRTP and I
was wondering if anyone was (is still) using it for
decryption/encryption of this type of SRTP, mainly in scenarios like
SRTP from a classic sip phone (e.g., snom) to another SIP endpoint
wihtout srtp support.
As I cc-ed Richard, another question is about the plans for the future,
if at this moment is properly working/maintained, will it be kept for
the future or the plan is to deprecate/remove/don't invest any effort in
its maintenance?
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
For the life of me I can't find any in depth online resources on setting up and configuring Kamailio. Do anyone know of some good resources? Specifically I want to put this in front of a couple Asterisk servers so I need to learn how to connect a trunk to it, and then based on the call destination route it to the right Asterisk servers. I feel like this shouldn't be complicated I just don't understand where/how Kamailio is configured coming from only using Asterisk previously.
