[SR-Users] Routing calls to Asterisk using dispatch
Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 7...@testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770@1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a6...@testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947.0 To: sip:770@1.1.1.1 From: sip:770@1.1.1.1;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742@1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: sip:770@1.1.1.1:5060 Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Unknown caller gets online user's identity
Well, this
*if (from_uri!=myself uri!=myself)*
Means neither source nor destination is our user. Which implies that if our
domain is A, then call from domain B to C is not possible. However, calls
from B or C to A and A to B or C are possible. That is way an
unauthorized user gets passed and reaches asterisk. Asterisk accepts it
since call is coming from kamailio and tries to route it back to kamailio,
where kamailio finds user online and thus it goes through.
You should really break down this,
*if (from_uri!=myself uri!=myself)*
into something like this for clarity,
*if (from_uri!=myself) { *
* if (uri!=myself) {*
* # neither source nor destination is our user*
* } else {*
* # source is not our user but destination is our user*
* };*
*} else {*
* if (uri!=myself) {*
* # source is our user but destination is not our user*
* } else {*
* # both source and destination are our users*
* };*
*};*
Hope this helps.
Thank you.
On Fri, Jul 11, 2014 at 5:36 PM, g.aloi...@gmail.com wrote:
Hello,
I'm using Kamailio version 4.1.4+precise (amd64).
I have followed Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration
using Asterisk Database (http://kb.asipto.com/
asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main
difference in my setup compared to that one is that I continued use of
Kamailio's database.
The problem is as follows:
I decided to put Kamailio and through it Asterisk reachable from internet.
I have tried to configure Asterisk so that only calls of registered users
would be possible, and they could only call to other registered users or
conference rooms and echo test number.
Then I took the following steps:
I ensured that there was no online users with kamctl online. Then I
launched MicroSIP (www.microsip.org), but I did not defined account, I
simply set the protocol to tls and media encryption to mandatory, because
I'm using these.
I called to extension with x...@my.public.ip.address (where xxx is
extension) getting unauthorized. And that was what I wanted.
But if there is online users, calls go through, and incoming call is
coming from Asterisk (in syslog I can find out that src_user=asterisk).
Kamailio and Asterisk are listening the same IP address, but different
port. I have refused connections to the Asterisk's port with iptables.
I have defined my public IP address as domain in sip.conf. There is also
other domain defined which corresponds to users' domain I am using in
Kamailio's database.
In kamailio.cfg there is if statement which prevents Kamailio not to be
open relay:
if (from_uri!=myself uri!=myself)
...
If I change this for example:
if (from_uri!=myself || uri!=myself)
I get what I want this time: no calls from outside, but I somewhat think
that this is not a final solution.
I have not found from log files such information which would have helped
me. I have not yet investigated this problem so much that I could tell the
logic behind the selection of online user's identity which is used.
However, if I make a call to conference room I notice that Asterisk is
thinking that one of online users has joined the conference.
If I can recall correctly, I started with Kamailio version 3.2, and
integrated it with Asterisk 11 (currently 11.10.2). Is there something
which has changed in Kamailio, but what I have not changed in my setup
which could explain this.
Best,
Teijo
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[SR-Users] Groups in Kamailio vs Freeswitch
Hi guys, I'm designing a new service for a client and I was wondering what your opinion is of the 2 options I'm considering to separate users on the server. Basically I want to create closed groups where each user can only call and receive calls from members of the same group (single domain install). Initially I was thinking of just forwarding all the invites to FS and using the dialplan to enforce this segregation, but now I realize there is a groups module in kamailio too. Which 2 courses of action do people recommend? I'm more familiar with FS than Kamailio, so it would be easier for me to just forward everything to FS, although that will be more expensive and probably unnecessary if I could do it all in Kamailio with Siremis. Would kamailio's group module do the trick? Are there any tutorials out there I can follow? Cheers, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Rtpengine vs. TURN?
Hi, On my server, I have the option of using either Rtpengine for NAT traversal or pure TURN without rtpengine. Rtpengine has the obvious plus that it only needs 1 public IP, while TURN (with STUN) will need 2 public IPs, although that's not a problem in my case. Having said that, I'd like to take advantage of the huge experience that users of this list have in real world deployments. in your experience, which option is more reliable in a real world deployment? Cheers, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
