sr-users January 2014

sr-users@lists.kamailio.org

93 participants
141 discussions

SIP OPTIONS
by Keith 15 Jan '14

15 Jan '14

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

3 2

15 Jan '14

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?

2 1

The RPC function in the TM module
by 无愿无悔 15 Jan '14

15 Jan '14

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.

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 15 Jan '14

15 Jan '14

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

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is_in_subnet & Multiple CIDR subnets
by jay binks 15 Jan '14

15 Jan '14

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

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Re: [SR-Users] Update R-URI after lookup("location")
by Igor Potjevlesch 14 Jan '14

14 Jan '14

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. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

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Registration to Kamailio World 2014 is open
by Daniel-Constantin Mierla 14 Jan '14

14 Jan '14

Hello, I would like to announce that the registration to Kamailio World 2014 is now open. This year there is also a pre-conference event for half a day on April 2, with technical tutorials for those looking on more practical and hands-on content. You can see more details on the web site at: - http://conference.kamailio.com/k02/registration/ - http://conference.kamailio.com/k02/2014/01/registration-is-open/ Looking forward to meeting many of you in Berlin during April 2-4, 2014. Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Help with authenticating using Kamailio
by Kasinath 14 Jan '14

14 Jan '14

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

2 1

WebSocket with MSRP
by mohammed alyaseen 14 Jan '14

14 Jan '14

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

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Update R-URI after lookup("location")
by Igor Potjevlesch 13 Jan '14

13 Jan '14

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

1 0

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sr-users January 2014