Hi,
I have a trunk trying to connect to Kamailio but when it sends me OPTIONS I
am not replying with 200 OK.
This trunk is no authenticated in any way.
Any ideas?
Keith
I have kamailio 4.1.0 running on a server on a real ip and behind a
firewall.
NAT detection is enabled on kamailio because many remote clients are behind
NAT, so NAT is working fine along with rtpproxy.
Everything else (incoming, outgoing) is working fine except the following:
Users who are connected to our openvpn server (bridged mode) which is on the
same subnet with kamailio, fail to register.
I suspect that kamailio detects NAT on these clients as all of them are
behind NAT, but they also have obtained a real ip from our openvpn server on
their tap interface and as a result, REGISTER fails.
One solution, but not the best is to exclude kamailio from our openvpn
routes but I would not prefer this because I wil not be able to manage the
server remotely
My config: http://pastebin.com/JSxzgmKH
Any suggestions?
Hello everyone:
I want to know what is the parameter for the RPC function "tm.cancel" in the TM module ?
I input the parameters such as :
tm.cancel d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ 2
but it can accept the only parameter,and inform me "error 400-Callid and CSeq expected as parameters " .
The context of the Callid.s is "d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ".
But the context of the CSeq.s is "if you get this string, you don't check rpc_scan return code!!!(very bad)".
Could you give me an example ?
Message: 3
Date: Mon, 13 Jan 2014 09:00:53 +0100
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] The RPC Function in The TM Module
Message-ID: <52D39D35.5000205(a)gmail.com>
Content-Type: text/plain; charset="utf-8"; Format="flowed"
Hello,
can you try to put callid in between double quotes. Try with the number
only and if it doesn't work, try also with number and method. Like:
kamcmd tm.cancel "CALLID" "NNN"
kamcmd tm.cancel "CALLID" "NNN INVITE"
With kamcmd should be an option to force string input parameter to rpc
command, iirc, like:
kamcmd tm.cancel "CALLID" s:NNN
Let us know if it works.
Cheers,
Daniel
hi,
I try to put callid between double quotes.Like:
tm.cancel "CALLID" "NNN"
but it inform : error: 400-Transactions not found.
the context of the callid is " "CALLID" "
the context of the Cseq is " "NNN" " .
note:
Thus,the double quotes is treated as a part of the string.
Hi Jason,
Please find below my response inline,
I have some questions for you as we have used suspend/continue quite a lot
in the IMS code and don't have any leaks.
Firstly, why are you using pkg_mem for your hash_id and label? Remember that
you will be in 2 different processes in the suspend and continue portions of
the code... so pkg_mem will not work - you should use shm_mem instead.
[Shankar] We use pkg_mem because we are invoking t_continue from the same
process ( using thread ).
Secondly, how are you using top to tell that you have a leak? Kamailio's
memory is internally managed.
[Shankar] After running for say 20minutes, we get out of shared memory
error. Also in top output we observed incremental increase in the shared
usage of shared memory for the process.
Cheers
Jason
On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote:
> Re-sending without the attachment.
>
>
>
> *From:* Shankar [mailto:shankar.rk@plintron.com]
> *Sent:* Monday, January 13, 2014 4:57 PM
> *To:* 'sr-users(a)lists.sip-router.org'
> *Subject:* Regd. t_suspend() and t_continue()
>
>
>
> Hi,
>
>
>
> We are trying out the t_suspend() and t_continue() in our test setup.
> We are facing memory leak ( both shm and pkg as per top command results).
>
>
>
> Please find below the scenario,
>
>
>
> 1) Do a t_newtran()
>
> 2) Allocate pkg memory for hashid and label.
>
> 3) Call t_suspend()
>
> 4) Do t_continue() when async result is available
>
> 5) De-allocate pkg memory reserved for hashid and label
>
> 6) Do a t_relay() which forwards the sip message to another sip node.
>
>
>
> In the step (6) above, we see t_newtran() allocates one more time
> shared memory for the same transaction.
>
>
>
> We tried t_release() after step (4) to release the transaction as
> t_relay() anyways allocates new shared memory. Nothing helped.
>
>
>
> Please let me know what are the logs you would require to debug the same.
> I am attaching syslog for this run.
>
>
>
> Regards,
>
> Shankar
So Im looking at a way of implementing IP Network ACL's in kamailio..
block all except specific subnets etc..
it seems I can do what I need with the IPOps module and is_in_subnet but
is_in_subnet
will only take 1 CIDR notation subnet, and I want to be able to put ( For
example )
192.168.1.1/24,172.16.10.1/24 in there ... ( or any number of subnets
really )
is there a benevolent kamailio developer on the list who is able to add
this simple feature for me ?
--
Sincerely
Jay
Hello,
No one has an idea? I was thinking that each request goes to RELAY but even
if I try to modify the R-URI in this route, it fails. I still had the
contact URI taken from location instead of my modification.
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com]
Envoyé : lundi 13 janvier 2014 17:15
À : sr-users(a)lists.sip-router.org
Objet : Update R-URI after lookup("location")
Hello,
I do an association between X aliases with 1 contact. This contact is
connected one time in the location.
But sometimes, for some scenario, this contact can be connected 2 or 3 times
and the INVITE are sent with parallel forking.
When an INVITE is received, after lookup(aliases), I set the R-URI with
the original SIP TO with the following instruction:
avp_pushto("$ruri/username", "$tU"); before relay.
The issue occurred with the next contacts. They dont passed through this
instruction. How can I update all available R-URI in location before relay?
Regards,
Igor.
---
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Hi All,
I just installed Kamailio in one server and Asterisk in another.
Asterisk loads it sipusers info from database which is in Kamailio server.
I don't know how to go further. How can I authenticate Asterisk users
through Kamailio. I am trying to authenticate using a sipphone. But no
luck.
I am missing alot here. I know we can add users using the following command
>> kamctl add username password
But I already have users in asterisk realtime db. Is there any
difference? Awaiting your reply,
Thanks in advance,
--
Thanks & Regards,
Kasinath K P
System Engineer
Admin-Ahead Server Technologies
email:kasinathk@admin-ahead.com
Hi All,
I am using Kamailio 4.1 as SIP proxy and registrar with websocket and msrp modules. As a client I am using the JsSIP stack. The client can open websocket for SIP messages by calling "new WebSocket('ws://1.2.3.4:8000','sip')" with sip as a subprotocol, this works fine. But when the client tries to open a websocket for MSRP sessions using new WebSocket('ws://1.2.3.4:8000','msrp') in the browser log I see the error 'WebSocket connection to 'ws://1.2.3.4:8000/' failed: Unexpected response code: 400' . For MSRP file transfer I am using the crocodile msrp lib.
The HTTP GET request of sent by the client is:
GET / HTTP/1.1
Via: SIP/2.0/TCP 10.147.66.197:54818
Upgrade: websocket
Connection: Upgrade
Host: 192.168.144.48:8000
Origin: http://localhost
Sec-WebSocket-Protocol: msrp
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: gccXstTCf+egnPtufU2xng==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/31.0.1650.63 Safari/537.36
Cookie: PHPSESSID=pdlbnc3h24i5r8rdg1979tino2
In the log of Kamailio I see these lines:
11(2114) WARNING: websocket [ws_handshake.c:318]: ws_handle_handshake(): required headers not present
11(2114) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl)
11(2114) DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(10.147.66.197, 10.147.66.197, 0)
11(2114) DEBUG: <core> [tcp_main.c:2320]: tcpconn_send_put(): tcp_send: send from reader (2114 (11)), reusing fd
11(2114) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending...
11(2114) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f37ab162bd0 n=182 fd=8
11(2114) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=
HTTP/1.1 400 Bad Request
Via: SIP/2.0/TCP 10.147.66.197:54744
Sec-WebSocket-Protocol: sip
Sec-WebSocket-Version: 13
Server: kamailio (4.1.1 (x86_64/linux))
Content-Length: 0
So, anyone has an idea whats the problem is?
Thanks,
Medo
Hello,
I do an association between X aliases with 1 contact. This contact is
connected one time in the location.
But sometimes, for some scenario, this contact can be connected 2 or 3 times
and the INVITE are sent with parallel forking.
When an INVITE is received, after lookup("aliases"), I set the R-URI with
the original SIP TO with the following instruction:
avp_pushto("$ruri/username", "$tU"); before relay.
The issue occurred with the next contacts. They don't passed through this
instruction. How can I update all available R-URI in location before relay?
Regards,
Igor.
---
Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active.
http://www.avast.com
