- sr-users - kamailio.org

Debugging and syslog
by Daniel W. Graham 26 Mar '13

26 Mar '13

Hello, Reaching out for some assistance with debugging, I am not seeing any xlog messages in my log file. If there is any other info I didn't include that would be helpful please let me know. Any ideas are appreciated. Using Kamailio 3.3.2 & Cent OS 6 Rsyslog.conf -------------------- *.info;mail.none;authpriv.none;cron.none;local0.none /var/log/messages # Kamailio local0.* -/var/log/kamailio Kamailio CFG ------------------------------ #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif log_facility=LOG_LOCAL0 loadmodule "xlog.so" Example xlog- xlog("L_INFO", "No more gateways \n"); Thanks, Dan-

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KAMAILIO Config Hot/Live reload
by Omar 26 Mar '13

26 Mar '13

Kamailio Team. Was this feature added already or we still need to restart kamailio after we modify the Kamailio.cfg. O

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str2ip6 bug is also in 3.3.3
by Juha Heinanen 26 Mar '13

26 Mar '13

i tried to compile latest 3.3 and found that it failed to compile due to same str2ip6 bug that was fixed in 4.0: dns_cache.c: In function ��dns_cache_do_request��: dns_cache.c:1894: warning: implicit declaration of function ��str2ip6�� -- juha

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Help improving the modules readmes & docs
by Daniel-Constantin Mierla 25 Mar '13

25 Mar '13

Hello, I thought some people may have spare cycles and are willing to help with the documentation. Here is some updates I would like to get for modules: 1) adding ids for parameters, functions, and other sections of the docbook xml files. Now we generate the alphabetic indexes pointing to the online html readmes, would be better to point directly to the exact section. Practically, for example, sections such as next from async module: <section> <title><varname>workers</varname> (int)</title> should become: <section id="async.p.workers"> <title><varname>workers</varname> (int)</title> To get unique ids over all files, I suggest using following pattern: [module name] [dot] [type] [dot] [title] The [type] should be: - p - parameters - f - functions - m - mi commands - r - rpc commands - s - statistics 2) Add RPC commands to readmes -- some of modules already do, but many don't. There is a perl script that should get the list of RPC commands from source code, but it requires some patch to a perl lib, not working on all systems. Moreover, the output is quite minimalistic, not easy to improve the content or add examples. Last generated index is: - http://www.kamailio.org/docs/docbooks/3.4.x/rpc_list/rpc_list.html 3) Document selects -- although their name are quite suggestive, a bit of text around won't harm. - http://www.kamailio.org/wiki/cookbooks/devel/selects Because 1) and 2) requires git access, the easiest way to contribute is to provide patches to the xml files. If anyone commits to do an extensive work on these tasks in short term, we may eventually arrange for git access. The 3) is in the wiki, everyone can register an account and then update the pages. I would recommend these guidelines to be followed from now on by developers when adding new elements to the module documentation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Acknowlidging BYE in advance using t_reply
by Efelin Novak 25 Mar '13

25 Mar '13

Hi, I'd like to ask you whether it is a good idea to acknowledge a BYE message in advance using t_reply("200", "OK") on my border proxy. Not acknowledged BYE messages by clients cause me problems in the middle of my network. Is there any problem with this behavior from Kamailio side, from script side or from SIP side? Do you have any experience with such logic manipulation? I am aware that this is not a proxy function, but it would solve many problems. My script is as follows: if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... t_newtran(); t_reply("200", "OK"); } route(RELAY); } My test shows that 200 OK from client are absorbed. Also if there is no reply from a client, Kamailio does not generate 408 Request Timeout. I'm using older version kamailio 3.1.0 (i386/linux) 21a375. Thanks for an advice Efelin

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Does Kamailio work with ICE ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, 1. Does Kamailio work with ICE? What does Kamailio do with ICE candidates in SDP ? 2. Does rtpproxy acts as TURN server ? Many thanks. Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Store GeoIP in Accounting
by Ali Jawad 25 Mar '13

25 Mar '13

Hi I am storing accounting entries in the db, and I am also inserting some custom AVP values, however I also need to store the GeoIP value that I am extracting using gip. I did try the below, but it is not working, any other ideas how I could achieve this ? modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;CountryCode=$gip(src=>cc)") Regards

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How to enable TLS in Kamailio ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, I want to enable TLS in Kamailio, as in here http://kamailio.org/docs/modules/stable/modules/tls.html But how can I get the certificate and private key ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Problems with Mar portion of registration
by mrichardson 22 Mar '13

22 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } }

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Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 22 Mar '13

22 Mar '13

Hi Barry, Thanks for trying to help me out. Unfortunately your suggestion didn't work out. Not sure why. I tried a few other functions like t_relay_to. Guess I still have a lot to learn about this kamailio stuff in more complex setups. Any other ideas, anyone? Ttyl, Dave -----Original Message----- From: Barry Flanagan <barry(a)flanagan.ie> Date: Thu, 21 Mar 2013 12:01:18 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name On 20 March 2013 03:05, David Thomson <jdavidthomson(a)hotmail.com <mailto:jdavidthomson@hotmail.com> > wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com <http://siptest.testdomain.com> with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Hi I think you can use the t_replicate function within your register route so that each server will replicate the registration request to the other. I did this many years ago on OpenSER using the following just before the consume_credentials() line. On each of the servers you want to check that the src_ip is not that of the OTHER sever so they don't keep sending back and forth forever :-) So in the example below 10.0.1.1 is the IP of the other server. --------8<----------------------------------------- if (!src_ip==10.0.1.1) { t_replicate("udp:10.0.1.1:5060 <http://10.0.1.1:5060> "); xlog("L_INFO", "Replicating $ct, URI = $ru"); }; consume_credentials(); --------8<----------------------------------------- Have a look at http://www.kamailio.org/docs/modules/3.3.x/modules/tm.html#t_replicate Hope this helps. -Barry

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GeoIP Issue
by Ali Jawad 21 Mar '13

21 Mar '13

Hi I did setup geoIP module and got the latest geoipcity free dat file and csv file and did setup all parameters in kamailio, when doing a test the instead of showing the country name the code is showing the region name, while region name is empty "I only need the country name", see : Config : if(geoip_match("$si", "src")) xlog("L_ERR","SIP message from: $gip(src=>cc) $gip(src=>regn) $gip(src=>regc)\n"); Log : SIP message from : A2 <null> <null> >From the CSV file of geoipcity just for cross reference : locId,country,region,city,postalCode,latitude,longitude,metroCode,areaCode 190132,"FR","A2","Lézardrieux","",48.7851,-3.1059,, As you can see above A2 is the region it is the 3rd column, src=>CC should show FR instead of A2. Working on Kamailio 3.3.4. Is there anything I missed here ? Regards

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2 kamailio servers sharing one database and one dns name
by David Thomson 21 Mar '13

21 Mar '13

hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows:User 1 registers to Server1User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Any ideas? ttyl,Dave

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New developer: Hugh Waite
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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New developer: Carlos Ruiz Díaz
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Need help with port fowards.
by Mini UN 21 Mar '13

21 Mar '13

Hi, was on the chat channel, so emailed as no activity. I have a new Cisco/Linksys EA-2700 and port forwarding is the same as my old Linksys WRT-610N. However my VoIP and website don't work. I tried DMZ aswell. Confused? Unfortunately, this router isn't compatibel for DD-WRT firmware. Thanks if anyone answers.

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kamailio 4.0.0 and websockets
by Kelvin Chua 20 Mar '13

20 Mar '13

Hi guys, i have yet to finish my readings on the websocket standards but just wanted to fire away with this question. is the behavior of protocol conversions between UDP and TCP the same as if you include websockets? i have this twinkle issue (an old SIP stack) Received from: udp:192.168.122.100:5060 INVITE sip:kelvin@192.168.122.1 SIP/2.0 Record-Route: <sip:192.168.122.100;r2=on;lr=on> Record-Route: <sip:192.168.122.100:8080;transport=ws;r2=on;lr=on> Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0 Via: SIP/2.0/WS z173czhz21tk.invalid;rport=54765;received=192.168.122.1;branch=z9hG4bK3818745 Max-Forwards: 16 To: sip:kelvin@192.168.122.100 From: sip:kelvin2@192.168.122.100;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Contact: <sip:kelvin2@z173czhz21tk.invalid ;alias=192.168.122.1~54765~5;transport=ws;ob> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.2.1 Content-Length: 2103 v=0 o=- 3148117784 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio m=audio 55736 RTP/SAVPF 103 104 111 0 8 106 105 13 126 c=IN IP4 192.168.122.1 a=rtcp:55736 IN IP4 192.168.122.1 a=candidate:2625852906 1 udp 2113937151 <cut off> --- +++ 16-3-2013 10:41:25.584732 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100> From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Content-Length: 0 --- +++ 16-3-2013 10:41:25.589231 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100>;tag=pxtmo From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Warning: 302 X340precise "Incompatible transport protocol" Content-Length: 0 Kelvin Chua

2 1

SIP debug.
by Victor V. Kustov 20 Mar '13

20 Mar '13

Good day. How could I debug SIP dialogs? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.2-calculate GNU/Linux

4 5

NAT & Firewall Issues
by Amr Elsharqawi 20 Mar '13

20 Mar '13

Hello, I am facing NAT & Firewall problems how I can solve this issue I've tried RTPProxy but voice quality is bad. any tutorial, please Amr

4 8

Re: [SR-Users] dialog ka_timer and ka_interval problem
by dolphin 20 Mar '13

20 Mar '13

Hello, Thank you for your reply I follow your suggestions to modify SHM_MEMORY ,PKG_MEMORY value and the system only two uac register. See the program after I start kamailio. shm memory 1G pkg memory 10M sh-3.00# ps -ef | grep kamailio root 23627 1 0 12:57 ? 00:00:00 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio root 23635 23627 0 12:57 ? 00:00:03 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio This problem is still the same last crazy the sending options message for uac But the /var/log/syslog not the same as follows Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime 2013/3/20 Rinor Hoxha <rinorhoxha(a)gmail.com> > Hi, > I had this "issue" some times ago. It turned out that I needed to increase > the value of SHM_MEMORY, since by mistake I forgot a 0 on the value (1024 > vs. 10240). However this happened only when high loads on system. > I needed also to tune PKG_MEMORY to my needs. ( both on > /etc/default/kamailio ) > > Check your settings and see if this helps. > > Br, Rinor > > > > On Tue, Mar 19, 2013 at 4:07 PM, dolphin <dolphinctk(a)gmail.com> wrote: > >> hi , >> >> i use kamailio 3.3.4 version >> >> I set and load dialog module >> >> modparam("dialog", "ka_timer", 10) >> modparam("dialog", "ka_interval", 20) >> >> use dlg_set_property("ka-src"); >> >> When uac A call uac B connection success , After 20 seconds, sip proxy >> send the OPTIONS message uac A >> >> Crazy sent to fill my bandwidth , I use ngrep tools see sip server >> >> U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 >> OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. >> To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. >> From: sip:xxxxxxx@xxx.xxx.xxx.xxx >> :5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. >> CSeq: 1 OPTIONS. >> Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. >> Content-Length: 0. >> User-Agent: SIP Server. >> >> and /var/log/syslog >> >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> >> Please help me >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >

1 0

Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 20 Mar '13

20 Mar '13

Thanks for the idea. After sending the original email I got user2 to call through to user1. Zrtp even was negotiated. So I think usrloc is sort of sharing the registration information but sometimes the call doesn't get sent through under certain circumstances. Maybe it's because user1 is behind a nat and rtpproxy is being used to keep its connection alive while user2 is not nat'd. I think it is almost working. -----Original Message----- From: David | StyleFlare <david(a)styleflare.com> Date: Wed, 20 Mar 2013 03:27:06 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name I dont think it works like that, obviously the server 1 does not have a connection if the user is registered on server 2. On 3/19/13 11:05 PM, David Thomson wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060> <udp:xxx.xxx.xxx.xxx:5060> ...ignoring. Any ideas? ttyl, Dave _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

sip-router mailing list
by Nikita Kozlov 19 Mar '13

19 Mar '13

Hello, I don't know if it's something wanted, but the email header "List-Id" has changed for both sr-dev and sr-users. The modifications : List-Id: "SIP Router - Kamailio" -> List-Id: "Kamailio $SER$ - Users Mailing List" List-Id: "Development mailing list of the sip-router project" -> List-Id: "Kamailio $SER$ - Development Mailing List" Regards, -- Nikita

2 1

dialog ka_timer and ka_interval problem
by dolphin 19 Mar '13

19 Mar '13

hi , i use kamailio 3.3.4 version I set and load dialog module modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 20) use dlg_set_property("ka-src"); When uac A call uac B connection success , After 20 seconds, sip proxy send the OPTIONS message uac A Crazy sent to fill my bandwidth , I use ngrep tools see sip server U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. From: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. CSeq: 1 OPTIONS. Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. Content-Length: 0. User-Agent: SIP Server. and /var/log/syslog Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Please help me

2 1

Kamailio not increasing cseq
by Camila Troncoso 19 Mar '13

19 Mar '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * * * When I make a call from one client connected to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing though a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , because the second gateway is also in the cloud, but when this happens, FreeSWITCH answer with : “482 Request merged” because it detects that is the same call. This is because the second INVITE has the same Call-ID and same Cseq. Kamailio *is not* increasing CSeq. Is there a way to resolve this? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 5

CentOS 6 crash
by Krishna Kurapati 19 Mar '13

19 Mar '13

Daniel, Thanks for the help. YES after disabling TLS for Kamailio, it is working fine. Is there a solution to have both TLS for Kamailio and SSL for LUA? Thanks Krish Kura

2 1

Multiple calls uac_replace_from()
by Alex Balashov 19 Mar '13

19 Mar '13

Forgive me if I failed to follow a recent discussion to its conclusion, but what is the bottom line on calling uac_replace_from() consecutively, in the course of processing one request? Is it possible, while using restore_mode 'auto'? Is there a correct way to do it, such as calling msg_apply_changes() after the first call but not the second one? If not, what is the alternative? -- Alex -- 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/

4 7

Kamailio 4.0 crashing in LUA on CentOS 6 - What could be wrong?
by Krishna Kurapati 19 Mar '13

19 Mar '13

36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd474542320 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): sqlops 9(8567) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 36(8606) DEBUG: <core> [sr_module.c:674]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/lib64/kamailio/modules/db_mysql.so] 36(8606) DEBUG: <core> [db.c:213]: using db bind api for db_mysql 36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd4745424f0 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): app_lua 36(8606) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 39(8615) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 34(8604) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 40(8616) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 41(8617) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 41(8617) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 6, pid 8558 (shutting down?) 0(8543) ALERT: <core> [main.c:788]: child process 8558 exited by a signal 11 0(8543) ALERT: <core> [main.c:791]: core was generated And here is backtrace done. Loaded symbols for /usr/lib64/lua/5.1/ssl.so Core was generated by `/usr/local/sbin/kamailio -L /usr/local/lib64/kamailio/modules/:/usr/local/lib64'. Program terminated with signal 11, Segmentation fault. #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 269 FRAG_END(next)->prev_free=prev; Missing separate debuginfos, use: debuginfo-install lua-5.1.4-4.1.el6.x86_64 (gdb) backtrace #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 #1 qm_malloc (qm=0x7fd370304000, size=32) at mem/q_malloc.c:386 #2 0x00007fd47052ae71 in shm_malloc (size=24) at ../../mem/shm_mem.h:262 #3 ser_malloc (size=24) at tls_init.c:285 #4 0x0000003928c5d95e in CRYPTO_malloc (num=24, file=0x3928d24a84 "o_names.c", line=195) at mem.c:306 #5 0x0000003928c6085f in OBJ_NAME_add (name=0x3928d27c52 "RSA-SHA512", type=1, data=0x3928d27c56 "SHA512") at o_names.c:195 #6 0x0000003928cc076a in EVP_add_digest (md=0x3928f84440) at names.c:99 #7 0x000000392b442f18 in SSL_library_init () at ssl_algs.c:165 #8 0x00007fd36ea1d3c9 in luaopen_ssl_core () from /usr/lib64/lua/5.1/ssl.so #9 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #10 0x00007fd36f678a54 in ?? () from /usr/lib64/liblua-5.1.so #11 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #12 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #13 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #14 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #15 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so #16 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #17 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #18 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #19 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #20 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so ---Type <return> to continue, or q <return> to quit--- #21 0x00007fd36f678137 in ?? () from /usr/lib64/liblua-5.1.so #22 0x00007fd36f6781b2 in ?? () from /usr/lib64/liblua-5.1.so #23 0x00007fd36f673b61 in lua_pcall () from /usr/lib64/liblua-5.1.so #24 0x00007fd36f89e69d in lua_sr_init_child () at app_lua_api.c:227 #25 0x00000000004e4aa3 in init_mod_child (m=0x7fd474485060, rank=-1) at sr_module.c:893 #26 0x00000000004e4a24 in init_mod_child (m=0x7fd474485980, rank=-1) at sr_module.c:890 #27 0x00000000004e4a24 in init_mod_child (m=0x7fd474485bf0, rank=-1) at sr_module.c:890 #28 0x0000000000489d15 in fork_process (child_id=-1, desc=<value optimized out>, make_sock=<value optimized out>) at pt.c:347 #29 0x0000000000466dec in main_loop () at main.c:1700 #30 0x0000000000468afe in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566

2 1

Survey on SIP overload control
by Yang Hong 18 Mar '13

18 Mar '13

Dear all, SIP overload may block too many calls unnecessarily. "Requirements for Management of Overload in the Session Initiation Protocol. IETF RFC 5390." http://www.ietf.org/rfc/rfc5390.txt Build-in overload mechanism cannot prevent overload effectively. Therefore, IETF SIP Overload Control (soc) Working Group works on IETF RFC "SIP Overload Control" currently. http://tools.ietf.org/html/draft-ietf-soc-overload-control-12 IETF SIP Overload Control discussion archives http://www.ietf.org/mail-archive/web/sip-overload/current/threads.html#00874 We have published a survey on SIP overload control algorithms (including IETF RFC "SIP Overload Control"). http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI… http://arxiv.org/abs/1210.1505 http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67… We have also used control theorectic approaches to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - redundant retransmission ratio control and round trip delay control (IEEE Globecom 2010 and ICC 2011). "Mitigating SIP Overload Using a Control-Theoretic Approach" IEEE Globecom 2010 http://www.researchgate.net/publication/221284946_Mitigating_SIP_Overload_U… http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5683124&url=http%3A%2… Implementation of implicit SIP overload control in the real system "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach" http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15# "Design Of A PI Rate Controller for Mitigating SIP Overload" IEEE ICC 2011 http://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Contr… http://features.rr.com/article/01vBgtc8TR17z?q=Skype http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5963029&url=http%3A%2… Best regards, Winston Hong Software Engineer InBay Technologies Inc. Ottawa, Canada K2K 1Y3 http://www.inbaytech.com/

1 0

Kamailio, NAT and RTPProxy
by Amr Elsharqawi 18 Mar '13

18 Mar '13

Dears, Finally I've got Kamailio 4 running on FreeBSD 9 with MySQL. in the beginning it was working fine with NAT enabled, but after I've configured NAT & RTPProxy users can register, ring but voice is very bad. here is my kamailio.cfg #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: #!define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql #!define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql #!define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: #!define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _._._._ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql #!define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://sip:s510Sg4lrw@localhost/sip" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:XXXXXXXXX:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $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; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif }

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Runaway TCP Log
by Krishna Kurapati 18 Mar '13

18 Mar '13

Hi, My Log filled to 20G overnight with these Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files^C I looked at the code segment it is here where it is logging. new_sock=accept(si->socket, &(su.s), &su_len); if (unlikely(new_sock==-1)){ if ((errno==EAGAIN)||(errno==EWOULDBLOCK)) return 0; LOG(L_ERR, "WARNING: handle_new_connect: error while accepting" " connection(%d): %s\n", errno, strerror(errno)); return -1; } I am using 3.3.x Is this an issue with Kamailio or something else? Appreciate any help regarding this matter Krish Kura

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Kamailio strange behavior : bad Via, bad port
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. The INVITE goes through Kamailio. In the Kamalio 's log file, I see it say "bad Via", bad port. And it doesnot reply A with 100 Trying not forwarding INVITE to B. The INVITE message (espcially the Via header) is OK. This problem sometimes happens. Do you know why ? Please help. Thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Kamailio not send 100/180 for INVITE !
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. Kamailio will handle A's INVITE message. But I see that B doesnot receive the INVITE. And A doesnot receive any 100/180 privisional message. Why does Kamailio not send 100/180 ? Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

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How does Kamailio handle INVITE message by default ?
by Khoa Pham 17 Mar '13

17 Mar '13

Hi, After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP According to rtpproxy flow http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info. When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf). 1. Add an SDP command direction:active to the SDP content 2. Change the c= line to a.b.c.d 3. Force RTP to go through a proxy by changing the c-line to c=IN IP4 address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101. When will SER do 2, 3 ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Kamailio only allowing traffic with machines in the dispatcher.cfg?
by Benjamin Henrion 16 Mar '13

16 Mar '13

Hi, Does someone know/uses a simple rule so that Kamailio only exchanges traffic with machines in the dispatcher? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Re: [SR-Users] sr-users Digest, Vol 94, Issue 51 - Problem with Mar portion of registration with Kamailio ims modules
by mrichardson 15 Mar '13

15 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } ============================================ On 03/15/2013 03:06 AM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. FreeBSD build from git: i use that patch (Victor V. Kustov) > 2. Re: Problems with Mar portion of registration [IMS] > (Daniel-Constantin Mierla) > 3. Re: UAC module and SRV records (Daniel-Constantin Mierla) > 4. Re: registrar and check registered (Daniel-Constantin Mierla) > 5. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > 6. Re: FreeBSD build from git: i use that patch (Victor V. Kustov) > 7. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 15 Mar 2013 11:43:55 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <20130315114355.2409d851@bkstv> > Content-Type: text/plain; charset=US-ASCII > > diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c > index faabd85..24686e7 100644 > --- a/modules/ims_icscf/cxdx_lir.c > +++ b/modules/ims_icscf/cxdx_lir.c > @@ -51,6 +51,11 @@ > #include "mod.h" > #include "location.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable > int create_lia_return_code(int result) { > int rc; > diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c > index bd5216c..8b6b79c 100644 > --- a/modules/ims_icscf/scscf_list.c > +++ b/modules/ims_icscf/scscf_list.c > @@ -47,6 +47,11 @@ > #include "db.h" > #include "../../lib/ims/useful_defs.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list > > extern struct tm_binds tmb; //Structure with pointers to tm funcs > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.3-calculate GNU/Linux > > > > ------------------------------ > > Message: 2 > Date: Fri, 15 Mar 2013 08:44:13 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems with Mar portion of registration > [IMS] > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D14D.7080808(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > I haven't played with IMS modules, but you said is causing scscf to die > -- is that a kamailio instance? What means 'to die' -- does it crash? If > yes, can you get the syslog messages and the gdb backtrace? > > Cheers, > Daniel > > On 3/13/13 3:01 PM, mrichardson wrote: >> The MAR portion of registration is either not happening or failing >> causing scscf to die. Using Kamailio 4.0 with the ims_auth module. >> We're trying to determine what >> the REGISTER block should look like and what ims_auth parameters are >> necessary for a successful MAR challenge with auth vector delivery. We >> currently only get the UAR by ICSCF then the SAR by SCSCF. >> >> Module parameters: >> # -- ims_auth params -- >> modparam("ims_auth", "name", URI) >> modparam("ims_auth", "registration_default_algorithm", >> REG_AUTH_DEFAULT_ALG) >> modparam("ims_auth","ignore_failed_auth",0) >> #!ifdef CXDX_FORCED_PEER >> modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) >> #!endif >> modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) >> modparam("ims_auth", "av_request_at_once", 1) >> modparam("ims_auth", "registration_qop", "auth") >> >> route[REGISTER] { >> xlog("L_ERR", "Enter register block"); >> t_newtran(); >> >> ims_www_authenticate(NETWORKNAME); >> #check to see if user is authenticated - ie sip header has auth >> information - (already challenged) >> if ($avp(maa_return_code) == 1) { >> # user has not been authenticated. Lets send a challenge >> via 401 Unauthorized >> ims_www_challenge("$td"); >> exit; >> } else { >> # We need to check if this user is registered or not >> if (!impu_registered("location")) { >> save("location"); >> if ($avp(saa_return_code) == 1) { >> isc_match_filter_reg("0","location"); >> exit; >> } >> } else { >> save("location"); >> if($avp(saa_return_code) == 1) { >> isc_match_filter_reg("1","location"); >> exit; >> } >> } >> } >> } >> >> >> 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer >> hss.ims.somecompany.com:3868 connected >> 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: >> ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : >> [saa_return_code] - [-2] >> 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: >> cxdx_get_experimental_result_code: Failed finding avp >> 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP >> successfully : [saa_return_code] - [1] >> 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't >> release transaction >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 3 > Date: Fri, 15 Mar 2013 08:48:53 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] UAC module and SRV records > To: "Olle E. Johansson" <oej(a)edvina.net> > Cc: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D265.3060104(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 3/13/13 10:00 AM, Olle E. Johansson wrote: >> 13 mar 2013 kl. 09:35 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> On 3/13/13 9:25 AM, Olle E. Johansson wrote: >>>> 13 mar 2013 kl. 09:08 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> do you set the port in r-uri/dst-uri? >>>> No. That would invalidate SRV record lookup, right? >>> yes. >>> >>> Have you watched the dns traffic? Is directly A/AAAA query? Afaik, this function should use just the usual logic to determine the next hop based on dst-uri/r-uri. >> I will debug in more detail. Had to set an outbound URI with a hostname and a port number to get it to work. > There could be the possibility that some of the functions in the chain > of uac_req_send() set port to default 5060 if not set explicitly, but > from sending point of view, it should the same as for the rest of > messages. Thus a look over the code should be done as well. I'll do it > soon if nobody gets the time meanwhile. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 4 > Date: Fri, 15 Mar 2013 08:53:29 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] registrar and check registered > To: "Victor V. Kustov" <coyote(a)bks.tv>, sr-users(a)lists.sip-router.org > Message-ID: <5142D379.20402(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > On 3/13/13 10:00 AM, Victor V. Kustov wrote: >> Hello, Daniel. Thanks for reply. >> >>> do you want to check if caller (origin) is registered or if callee >>> (destination) is registered? Like one can call to pstn only if has a >>> phone registered for itself? >> In our scheme all calls forwards to PSTN GW, if that call from our to >> our users, GW turn call back and kamailio receive it as incoming. >> So, no "local calls" in this scheme. >> >> For outgoing calls (to PSTN) i want check caller (origin) is registered. >> For incoming calls (from PSTN) i want callee is registered. >> >> Need cache registrations (DB, memcached or something). I try that way: >> 1. REGISTER - if ok, than we save("location"). >> 2. Other methods - check !registered("location") or !lookup("location"). >> But in INVITE seems both !registered("location") >> and !lookup("location") return TRUE. I dont understand why. >> >> In topicstart letter i citate part of my cfg: in INVITE i check >> registration again and it work - registered users may call to PSTN, >> unregged may not. But no cache for registration and look as overhead. >> Must be more "beauty" solution. > when you need to check if caller is registered use: > > - registered("location", "$fu") > See more details at: > - http://kamailio.org/docs/modules/stable/modules/registrar.html#id2512620 > > If used without the second parameter is checking if R-URI ($ru) is > registered. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 5 > Date: Fri, 15 Mar 2013 08:55:28 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D3F0.4090408(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > thanks for checking on bsd, can you please sent the patch as attachment? > Because it can saved locally and applied easier. > > Cheers, > Daniel > > On 3/15/13 8:43 AM, Victor V. Kustov wrote: >> diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c >> index faabd85..24686e7 100644 >> --- a/modules/ims_icscf/cxdx_lir.c >> +++ b/modules/ims_icscf/cxdx_lir.c >> @@ -51,6 +51,11 @@ >> #include "mod.h" >> #include "location.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable >> int create_lia_return_code(int result) { >> int rc; >> diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c >> index bd5216c..8b6b79c 100644 >> --- a/modules/ims_icscf/scscf_list.c >> +++ b/modules/ims_icscf/scscf_list.c >> @@ -47,6 +47,11 @@ >> #include "db.h" >> #include "../../lib/ims/useful_defs.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list >> >> extern struct tm_binds tmb; //Structure with pointers to tm funcs >> >> > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 6 > Date: Fri, 15 Mar 2013 12:01:37 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: sr-users(a)lists.sip-router.org > Message-ID: <20130315120137.72b404c4@bkstv> > Content-Type: text/plain; charset="utf-8" > > ? Fri, 15 Mar 2013 08:55:28 +0100 > Daniel-Constantin Mierla <miconda(a)gmail.com> ?????: > >> Hello, >> >> thanks for checking on bsd, can you please sent the patch as >> attachment? Because it can saved locally and applied easier. >> > > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.2-calculate GNU/Linux >

1 0

FreeBSD build from git: i use that patch
by Victor V. Kustov 15 Mar '13

15 Mar '13

diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c index faabd85..24686e7 100644 --- a/modules/ims_icscf/cxdx_lir.c +++ b/modules/ims_icscf/cxdx_lir.c @@ -51,6 +51,11 @@ #include "mod.h" #include "location.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable int create_lia_return_code(int result) { int rc; diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c index bd5216c..8b6b79c 100644 --- a/modules/ims_icscf/scscf_list.c +++ b/modules/ims_icscf/scscf_list.c @@ -47,6 +47,11 @@ #include "db.h" #include "../../lib/ims/useful_defs.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list extern struct tm_binds tmb; //Structure with pointers to tm funcs -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.3-calculate GNU/Linux

3 4

registrar and check registered
by Victor V. Kustov 15 Mar '13

15 Mar '13

Hi. I try to use Kamailio 4.1 git version as registrar and PSTN gw. Need accept register from users. Its OK, I made it and use save() function - from keep registrations in case of kamailio restart. Need forward calls to gw, but only for registered users. If I understand correctly in INVITE enough check registered("location"), isnt it? but it's not work, i need verify auth again. Routing part of my config: route { # initial sanity checks - skip in letter # We are registrar! if (is_method("REGISTER")) { sl_send_reply("100", "Trying"); # send "Trying, dont threat on me if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } # we must check auth right now # WORK: if (!www_authorize("$fd", "subscriber")) if (!auth_check("$fd", "subscriber", "1")) { # WORK: www_challenge("$fd", "0"); auth_challenge("$fd", "0"); # send me requisites plz exit; } else { # ok. auth accepted, save in location if (save("location")) #if (save("location","0x04","$fd")) { xlog("L_WARN", "Save OK in location sip:$au@$fd"); } else { xlog("L_WARN", "Save FAIL in location sip:$au@$fd"); } } xlog("L_WARN", "REGISTERED method done"); exit; } if (is_method("INVITE")) { if ($si == $sel(cfg_get.pstn.gw_ip)) { # incoming calls xlog("L_WARN", "Incoming call"); } else { # outgoing calls xlog("L_WARN", "Outgoing call"); # check auth # we must check auth right now if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } consume_credentials(); ### THAT FUNCTIONS ALWAYS WRITE IN LOGS # "!registered" "!lookup" for both registered and unregged users # if (!registered("location")) xlog("L_WARN","!registered"); # if (!lookup("location")) xlog("L_WARN", "!lookup"); # number change - FIX late # check may be used dialplan automatically? # route $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_WARN", "Rewrite to $ru, send to PSTN."); if (!t_relay()) { sl_reply_error(); } } } else { # reply 2 keepalive and drop all # if (!t_relay()) { sl_reply_error(); } } } -- SY, Victor V. Kustov +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.7.7-calculate GNU/Linux

2 3

UAC module and SRV records
by Olle E. Johansson 15 Mar '13

15 Mar '13

Hi! When using the UAC module to send a request to a domain that only exists in SRV records, I get an error message that no A or AAAA records can be found for the domain. Is this by design or possibly a bug? Anyone else that have seen this? /O

2 5

Problems with Mar portion of registration
by mrichardson 15 Mar '13

15 Mar '13

The MAR portion of registration is either not happening or failing causing scscf to die. Using Kamailio 4.0 with the ims_auth module. We're trying to determine what the REGISTER block should look like and what ims_auth parameters are necessary for a successful MAR challenge with auth vector delivery. We currently only get the UAR by ICSCF then the SAR by SCSCF. Module parameters: # -- ims_auth params -- modparam("ims_auth", "name", URI) modparam("ims_auth", "registration_default_algorithm", REG_AUTH_DEFAULT_ALG) modparam("ims_auth","ignore_failed_auth",0) #!ifdef CXDX_FORCED_PEER modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) #!endif modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) modparam("ims_auth", "av_request_at_once", 1) modparam("ims_auth", "registration_qop", "auth") route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); ims_www_authenticate(NETWORKNAME); #check to see if user is authenticated - ie sip header has auth information - (already challenged) if ($avp(maa_return_code) == 1) { # user has not been authenticated. Lets send a challenge via 401 Unauthorized ims_www_challenge("$td"); exit; } else { # We need to check if this user is registered or not if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer hss.ims.somecompany.com:3868 connected 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [-2] 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_experimental_result_code: Failed finding avp 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [1] 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't release transaction

2 1

A weird message looping scenario
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, We have an ugly message looping scenario where a SIP request message received at Kamailio is forwarded to the switch/router and back to Kamailio in a loop until the Max-forwards header value becomes 0 or the transaction times out. After receiving an incoming INVITE request from a local user, Kamailio forwards the INVITE request to another proxy located in a different network. The call is established successfully. During the call setup the external proxy sends an OPTIONS request to Kamailio (actually to the external public IP address from where it received INVITE request from Kamailio). The OPTIONS request arrives at Kamailio. But the request uri contains the mapped public address instead of the local ip address where Kamailio is listening. So Kamailio forwards the request to the public IP address (i.e. the switch) which again sends it back to Kamailio again and it keeps looping. Here is the message flow. 192.168.1.3 is the local user registered to Kamailio which is running at 192.168.1.5. IP address of the external proxy server is 10.139.90.161 and 10.139.90.137 is the public IP address of the switch/router. NAT isn't enabled at Kamailio. Internet Protocol Version 4, Src: 192.168.1.3 (192.168.1.3), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@video-conf SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Server: kamailio (3.3.3 (i386/linux)) Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.161 (10.139.90.161) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@vc-vcs-control.air2gnd.net SIP/2.0 Message Header Record-Route: <sip:192.168.1.5;lr=on> Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@ video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1002@10.139.90.161:5060> Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: sip:1002@10.139.90.161:5060 Content-Length: 0 P-Received: 10.139.90.137 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.139.90.161:5060;rport;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.161>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "uuuu" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 This message forwarding goes on for a long time in a loop, each time with an extra Via header and Max-Forwards value decremented by one. After sometime it eventually times out and Kamailio starts sending 408 response for the OPTIONS request which again keeps looping. Probably adding an alias for the public IP address i.e., 10.139.90.137 in the Kamailio config file would solve the problem. But this IP address is not fixed and may change and I guess there is no way Kamailio can learn the public IP address on its own. Any help or reference to previous posts if it is already discussed and solved will be much helpful. Regards, Ajay

2 1

What factor determine NATed or not ?
by Khoa Pham 15 Mar '13

15 Mar '13

Hi, When we send REGISTER or INVITE to Kamailio, which part of the request message does Kamailio use to determine if client is NATed or not ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 2

search_hf returns always true
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, I want to check whether a parameter exists in a header and I am using the search_hf() function defined in textops module. I am surprised that is always returning true whether the searched pattern exists or not. To be specific, I need to check if the "received" parameter exists in the top most via header and this is how I do it. The statements inside the if block get executed all the time irrespective of whether the received parameter exists in the top most via or not. Any help what am I doing wrong here? if (search_hf("Via", "received=", "f")) { xdbg("received param $sel(via.received) in top most via header \n"); # perform necessary action } Case 1: no received parameter in via Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK39bc.c1076d93.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK56870034 And the corresponding debug message from Kamailio is: ./sbin/kamailio[22600]: DEBUG: <script>: received param <null> in top most via header Case 2: received parameter exists Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK006e.18417697.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK960971659 And the corresponding debug message: ./sbin/kamailio[22600]: DEBUG: <script>: received param 10.139.90.137 in top most via header Thanks, Ajay

2 1

testing Supported and Require keywords
by Juha Heinanen 14 Mar '13

14 Mar '13

is there currently a proper way to test if a particular keyword appears in Supported or Require header or is a regex test on $hdr(X)[*] the only possibility? -- juha

3 3

Kamailio World Updates - March 14
by Daniel-Constantin Mierla 14 Mar '13

14 Mar '13

Hello, the groups of speakers at Kamailio World Conference got new members, among them Prof. Dr. Thomas Magedantz from Technische Universitat Berlin/FhG Fokus and Alan Duric, a pioneer in VoIP, among other positions being co-founder of Telio, more details at: - http://conference.kamailio.com/k01/speakers/ It is about one month till the start of the event, agenda is pretty much done, with possibility of few updates in the near future. Regarding Kamailio project, Monday we released v4.0.0, with really good feedback so far. Thank you again to everyone making it possible. New contributions started to show up on master branch, once you are done upgrading to v4.0.0, don't forget to give a try to development version and check the list of new additions. If you haven't registered yet for the Kamailio World conference, you should hurry up, the speakers and the other participants form an excellent group of professionals working with rich communication services and products, it is your chance to meet with them and get updated on the field: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ We are glad to announce the first group of participating companies with all day availability for demos and presentations of their products and services: Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. The list will be updated soon with more participants. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

I added the append_hf() inside the if condition in oureply_route[] block as given below, and it is working. I still want to learn how to store data in a variable to use it later in another block. Thanks, Ajay Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio with PostgreSQL as persistence
by Hoai-Anh Ngo-Vi 13 Mar '13

13 Mar '13

Dears, I have tried out Kamailio 3.3.4 with PostgreSQL as persistence but it does not work. make FLAVOUR=kamailio include_modules="db_postgres" cfg was OK. I found /usr/local/lib/kamailio/modules/db_postgres.so after installation. I have successfully set up database openser using /usr/local/sbin/kamdbctl create with DBENGINE=PGSQL in /usr/local/etc/kamailio/kamctlrc. I have edited /usr/local/etc/kamailio/kamailio.cfg with: #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB ... #!ifdef WITH_PGSQL #!define DBURL "postgres://openser/openserrw@localhost:5432/openser" #!endif ... #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif Run /etc/init.d/kamailio start|debug I got Listening on .... kamailio error, failed to start. Does someone know some hints or have a working example? When I comment out the first 3 define kamailio starts fine. Thank you for your help. -- Ngô Vi Hoài Anh Biedenkopfer Weg 13 60489 Frankfurt am Main 069 74 22 36 63 0172 53 97 0 82

2 2

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Need help understanding/separating signaling from media
by Grant Bagdasarian 13 Mar '13

13 Mar '13

Hello, I want to separate signaling and media in our architecture but I'm not entirely sure if the following will work (note that it is still in a design phase and these kinds of constructs are still new to me): User Agent 1-------------- Kamailio -------------- User Agent 2 | | | | | | -------------- Media Server Cluster -------------- The signaling will go through Kamailio but the media stream should pass through one of the media servers in the cluster for each call. Is it possible to achieve this by having Kamailio instruct the media server which ports to open and alter sdp information? I think that the RTP Proxy module does precisely this, but what I still don't understand is how the rtp stream is passed between these two agents for each call. Also, is RTP proxy the best way to do this or are there any other third party media servers which I could use? I don't need any fancy functionality, just basic media handling/pass-through. Regards, Grant

2 8

Send Bye to all calls in one profile
by Camila Troncoso 13 Mar '13

13 Mar '13

Hi, I have a kamailio proxy server that´s controlling the limit of calls that clients can make with profile grouping. In other side I also limit clients call duration base on users credit ( DATABASE) and *timeout *feature . Before the calls get answer I calculate how much time the client has left for this call and set the timeout variable. When the time is reached the proxy sends BYEs two both sides of this call. I want when the call timeout, as before explain, all other calls in the same profile end too, because the client has expired his credit. Is there a way to achieve this? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

4 6

java code to interact with kamailio
by Rupayan Dutta 13 Mar '13

13 Mar '13

Dear Sir, I want to authenticate users as well as getting buddy list from kamailio using java .Is there any existing library already in place. Rupayan Dutta

3 2

how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi, I want to extract the IP address in the received parameter of the top most via header of a response and use that in a custom header while sending response back to the caller. Can somebody please tell how to implement that in the Kamailio configuration file? I need to handle a NAT traversal scenario without enabling NAT in Kamailio which is used as the proxy server. The caller which is in the same network as Kamailio first sends a OPTIONS request to the callee before sending the INVITE. It uses the public IP address returned in the custom header of the OPTIONS response to build the contact header and SDP of the INVITE request. In fact, it first looks for the IP address in the received parameter of the via header to do that. But Kamailio removes the top most via with received parameter and then sends the OPTIONS response back to the caller. So we are using a custom header for that purpose. I would appreciate other suggestions to achieve this in Kamailio, may be by enabling NAT or some other alternative. Thanks, Ajay

2 1

Malformed PUBLISH from pua_dialoginfo (4.0.0)
by Charles Chance 13 Mar '13

13 Mar '13

Hi all, We are testing 4.0.0 in the lab and since upgrading, PUBLISHes generated by pua_dialoginfo appear to be malformed (content-length mismatch). Please see attached output from Wireshark for both the PUBLISH and the 400 response from Kamailio. PUBLISH is being sent from one Kamailio to another - nothing has changed other than upgrading from 3.3.3 to 4.0.0. Has anyone else noticed the same? Regards, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

2 5

Failover on carrierroute
by Ali Jawad 13 Mar '13

13 Mar '13

Hi I did setup carrierroute with failover using t_on_failure, I do have 4 ".25 ratio" primary gateways for one carrier and 4 ".25 ratio"secondary gateways for a secondary carrier. However, given the current setup, if one gateway is down carrieroute does use the secondary carrier if one gateway of the primary carrier is down. What I need is to only go to the secondary carrier if all the gateways of the primary carrier are down. I thought of setting up 4 t_on_failure branches with carrierroute but that will be troublesome to maintain. Is there another method for this ? Regards

2 2

High availability
by Moacir Ferreira 12 Mar '13

12 Mar '13

Hi, What would be the community's architecture advice for deploying Kamailio in a high availability environment? I am looking for something simple, still efficient. Thanks,

3 3

Porting kamailio4 to OpenWrt
by Jiri Slachta 12 Mar '13

12 Mar '13

Hello everyone, I am new to this list and I am a newbie when it comes to Kamailio details, so sorry for any of my misunderstandings. I am the author of kamailio3 package in OpenWrt and currently I am trying to port recent major release of kamailio to OpenWrt. Let's jump directly to my question. The current state of my Kamailio4 package is that all modules which does not depend on any external libraries, are succesfully built. Any module that depends on external library is not built at all. But if I pass LD="$(TARGET_CC)" to the linker via: make -C/path/to/module LD="$(TARGET_CC)" then the module is succesfully compiled. I build kamailio4 via macro Build/Compile in following Makefile - http://liptel.vsb.cz/svn/besip/Trunk/packages-trunk/net/kamailio4/Makefile My question is - has anything changed in build procedures or variables significantly? It seems that LD variable is not passed into Makefiles of specific modules. Am I doing anything wrong? I am open to any suggestions. Thank you! ~ Jiri Slachta

2 1

Fwd: Re: memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

---------- Forwarded Message ---------- Subject: Re: memcached module problems Date: Dienstag, 12. März 2013, 14:50:41 From: Dragos Oancea <droancea(a)yahoo.com> To: Bis: Henning Westerholt <henning.westerholt(a)1und1.de> Thank you Henning, that was a good hint. I've noticed that kamailio and my perl script (which is using Cache::Memcached) set the value in memcached somehow differently. kamailio: set 888_HA1 1 10800 32. 6fa22552ea04d443ab5864a7b3c2168c. perl script: set 888_HA1 0 0 32. 6fa22552ea04d443ab5864a7b3c2168c. VAR_VAL_STR is 1. I will investigate further. Regards, Dragos ps: I will also subscribe to sr-users . From: Henning Westerholt <henning.westerholt(a)1und1.de> To: Dragos Oancea <droancea(a)yahoo.com> Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 12, 2013 9:35 AM Subject: RE: memcached module problems Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt -------------------------------------------------------------

1 0

File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

2 2

Kamailio MySQL DB Creation
by Amr Elsharqawi 12 Mar '13

12 Mar '13

Hello Kamailio Community!! I am newbie to Kamailio & SIP. And I am interested in Kamailio. I've used asterisk for some time then I heard about FreeSWITCH unfortunately FreeSWITCH was not good enough for me like Asterisk. I've found NAT problem with Asterisk Finally I've decided to work with Kamailio as SIP server & Asterisk as PBX. to be fair I need to learn more about FreeSWICH to test it well. My Question is about Kamailio installation; I am using FreeBSD 9 attached my uname -a FreeBSD PBX.XXXXX 9.1-RELEASE FreeBSD 9.1-RELEASE #0 r243826: Tue Dec 4 06:55:39 UTC 2012 root@obrian.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 I've used gmake for building, compiling and installation. Everything goes smooth EXCEPT mysql db installation when I use sh /usr/local/sbin/kamdbctl create I am getting the following error: /usr/local/lib/kamailio//kamctl/kamdbctl.base: 36: Syntax error: Missing '}' Thanks a lot for your valuable time and that amazing project (Kamailio). Amr

3 6

Strange TLS issue
by Jirka Jirout 12 Mar '13

12 Mar '13

Hello, I have a very strange issue - we are running kamailio 3.3.2 as a part of Sip Provider CE and some of our clients have issues with TLS connectivity. 1. some clients such as Jitsi on Mac OS X and Bria or idoubs on iOS can always connect 2. imsdroid on Android can sometimes connect and works (about 10 % of attempts) When I run tcpudmp on the server during the unsuccessfull attempts, I see something like this: > 1 1 0.0347 (0.0347) C>S SSLv2 compatible client hello > Version 3.1 > cipher suites > TLS_RSA_WITH_AES_128_CBC_SHA > 1 2 0.0360 (0.0013) S>C Handshake > ServerHello > Version 3.1 > session_id[0]= > cipherSuite TLS_RSA_WITH_AES_128_CBC_SHA > compressionMethod NULL > 1 3 0.0361 (0.0001) S>C Handshake > Certificate > 1 4 0.0361 (0.0000) S>C Handshake ServerHelloDone > 1 5 0.0782 (0.0420) C>S Alert > level fatal > value unexpected_message > 1 0.0785 (0.0002) S>C TCP FIN > 1 6 0.1092 (0.0307) C>S Handshake > ClientKeyExchange > 1 7 0.1092 (0.0000) C>S ChangeCipherSpec > 1 8 0.1092 (0.0000) C>S Handshake What I find particularly strange is that the client sends fatal alert and then attempts to continue with handshake. What exactly does this "unexpected_message" mean? Is the client letting the server know the last message was unexpected? Or is this telling me the server had not expected the message? It cannot be a certificate problem, can it? Considering other clients were fine, I blamed our code at first, but I am no longer sure that is the (only) reason for this behavior. I decided to start from scratch and wrote an extremely simple application in plain C, that connects to the TLS port using openssl (which imsdroid also uses), sends some query and then prints response. It works fine on https sites, even on the same server kamailio is running and with the same certificates (our certificates come from Verisign CA G3). When I run ssldump, I can see handshake is fine, socket opened, the client is sending application data. I can see data coming in on the server using tcpdump. However, the data never reaches Kamailio - no sign of the packets in the log and no answer whatsoever (I use a fake REGISTER message so I would expect to get something like "Unauthorized" back). The same happens, when I try to use openssl s_client to connect to the site and send "register" message (because R at the start of packet triggers renegotiation in s_client, uff). My TLS settings in Kamailio are pretty much default, only using my own key and certificate. regards, jjj

4 5

Re: [SR-Users] memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt

1 0

RPMs for v4.0.0
by Daniel-Constantin Mierla 12 Mar '13

12 Mar '13

Many RPM packages are are now available on OpenSUSE build server (for Centos 5.5 and 6, RedHat 5 and 6, Fedora 16, 17 and 18, OpenSuse 11.4, 12.1, 12.2 and 12.3, on i586 and x86_64 architectures). Links to repositories are at: - http://www.kamailio.org/wiki/packages/rpms There is listed the link to Peter's RPM repository as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Routing calls
by mark＠brightvoip.co.uk 12 Mar '13

12 Mar '13

Hi, I'm looking for pointers on the following scenario. I have two sites (A and B) and sip traffic is sent from A to B. At each site there are a number of SIP (Asterisk) servers and a Kamailio server. The two sites have various network routes (X, Y & Z) between them. I have been asked to load balance the inter-site SIP traffic across the available links with a 20/20/60 traffic split (dispatcher, carrierroute). Each Kamailio server has multiple IP addresses and these are used to direct traffic to specific routes (Kamailio at site A IP 111.111.111.1 to Kamailio site B 111.111.111.2 defines route X for example). For any given call, the complete call (SIP and both RTP streams) should travel wholly over one route. I would welcome any suggestions about how best to route the whole call (sip & rtp) over a specific route, I was working on a solution with multiple RTPProxy instances, but was wondering if this is the best solution. Thanks in advance. Mark

1 0

Kamailio v4.0.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Kamailio v4.0.0 is out – a special major release, shifting up first number of versioning string! It is the landmark for the end of Kamailio-SER integration, bringing out as well a very large set of new features and improvements, like websocket support or dozens of IMS extensions. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.0.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

releasing v4.0.0 tomorrow
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Hello, the plan is to release final Kamailio v4.0.0 tomorrow, Monday, March 11, 2013. If any developer/packager has something to add to the release, please commit by 12:00GMT to GIT branch 4.0. After that, announce first on sr-dev and wait for an ack, to avoid packaging inconsistent content. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

kamailio is returning 500
by chanea 11 Mar '13

11 Mar '13

Dear all While lots of request come to my Kamailio 3.1.x in the same time , server is returning "500, Server Internal Error" to some of the request. What should i do if i want to reduce the number of "500" response? chanea

3 7

SIP and XMPP server integration
by Vinay V 11 Mar '13

11 Mar '13

Hi All, I am in process of integrating a sip server (Lync) with XMPP server (kamailio). Can a kamailio server be used as a sip to xmpp gateway and vice versa. If i am using any other xmpp server and want to integrate with lync will kamailio can be used as a gateway. This integration is maily for IM and presence. If it can be done what configuration should be done to kamailio.conf file. Please provide the inputs. I am very much need of it. If anybody has done already the above testing please provide DOCS. Thanks in advance, Vinay

2 2

Problem with frequent use of FIFO commands
by Mino Haluz 11 Mar '13

11 Mar '13

Hi, I have some nagios scripts binded to kamctl fifo commands, as well as some web scripts that are directly calling kamctl fifo statements. I do not know why, but it sometimes returns ** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] => ** ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config [2] => ** ERROR: and also have loaded the mi_fifo module. when it is called with web script with www-data account. From shell it works everytime, but with web 20% of results are errors. Other 80% are sucessful, so no problem with permissions. Does mi_fifo support multiple access, are there any other limits in this regard ? Within OS level? kamailio-3.3.4 Thanks, Mino

2 1

Kamailio 4.0 install guide
by Jesús Pérez Rubio 11 Mar '13

11 Mar '13

Hi, I think there is a little error on this guide about 4.0 version install: http://www.kamailio.org/wiki/install/4.0.x/git First it is said this: "8. Edit configuration file To fit your requirements for the VoIP platform, you have to edit the configuration file. /usr/local/etc/kamailio/kamailio.cfg" At the end it is said that only this line of init.d script have to be changed: DAEMON=/usr/local/sbin/kamailio I would add this one too: CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Another option would be to work directly with "/etc/kamailio/kamailio.cfg" Anyway, I take this oportunity to thank you for your great job and to notice that this version 4.0 seems to work stil better with our QoffeSIP Javascript SIP stack. :) -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

2 1

no destination sets in dispatcher module
by Ali Jawad 11 Mar '13

11 Mar '13

Hi I am using carrieroute to distribute traffic, I am not sure if there is a way for carrieroute to detect if a destination is down or not, but I read that the dispatcher module does that so I did compile it into the existing kamailio installation and added the following settings : loadmodule "dispatcher.so" modparam("dispatcher","db_url", "mysql://connectionstring") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) and under ROUTE if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; xlog("L_ERR","************************AFTER TOO MANY HOPS******************************************* cr\n"); ds_select_dst("1", "0"); forward(); # t_relay(); All I get is too many hops in ngrep and kamailio reports : (26467) ERROR: dispatcher [dispatch.c:1564]: no destination sets In the DB I do have a setid 1 and sip:ip.of.carrer:5060 as destination. Regards

2 1

why use received as RURI for nathelper OPTIONS ping?
by Andrew Pogrebennyk 11 Mar '13

11 Mar '13

Some Fritzbox devices don't accept the OPTIONS ping generated by kamailio with RURI: sip:11.22.3.4:5060. In the location table we have: received: sip:11.22.3.4:5060 contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967 path: <sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'> Kamailio version is 3.3.2 and the nathelper config is nothing fancy: modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sipwise.local") modparam("nathelper", "received_avp", "$avp(s:received)") What I don't understand is why kamailio sets RURI of the OPTIONS message to value of "received" instead of the contact. I suspect a bug in the parser somewhere along these lines: > rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), > ((unsigned int)(unsigned long)timer_idx)*natping_interval+iteration, > natping_processes*natping_interval); > if (rval != 0) { > pkg_free(buf); > goto done; > } > } > > if (buf == NULL) > goto done; > > cp = buf; > while (1) { > memcpy(&(c.len), cp, sizeof(c.len)); > if (c.len == 0) > break; > c.s = (char*)cp + sizeof(c.len); > cp = (char*)cp + sizeof(c.len) + c.len; > memcpy( &send_sock, cp, sizeof(send_sock)); > cp = (char*)cp + sizeof(send_sock); > memcpy( &flags, cp, sizeof(flags)); > cp = (char*)cp + sizeof(flags); > memcpy( &(path.len), cp, sizeof(path.len)); > path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ; > cp = (char*)cp + sizeof(path.len) + path.len; > > /* determin the destination */ > if ( path.len && (flags&sipping_flag)!=0 ) { > /* send to first URI in path */ > if (get_path_dst_uri( &path, &opt) < 0) { > LM_ERR("failed to get dst_uri for Path\n"); > continue; > } > /* send to the contact/received */ > if (parse_uri(opt.s, opt.len, &curi) < 0) { > LM_ERR("can't parse contact dst_uri\n"); > continue; > } > } else {

4 5

scheduling inside SER
by irma zujovic 11 Mar '13

11 Mar '13

Hello, I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. My question is : where inside SER configuration scripts, the *scheduling *and *queueing *of SIP messages is done? I am interested in a special function or part of the written program. I hope that engineers that worked on developing of SER could help me with this issue. Thank you in advance Irma

3 2

INVITE messages not authenticated (default configuration)?
by Paul Belanger 08 Mar '13

08 Mar '13

Greeting, Hopefully, I'm understanding the following default kamailio.cfg[1] file. Over the weekend, I was attached by SipVicious. Following along with the example Daniel[2] create with kamailio and asterisk, I have almost the same setup. Rather then storing my SIP profiles in Asterisk database, I have then in Kamailio. To my point, the attacker was actually able to by pass any sort of authentication, but simply sending an INIVTE message: ./svmap.py -e 18885551234 kamailio.example.org -m INVITE Which kamailio, forwarded to Asterisk and because there is no additional auth within asterisk, was able to hit the asterisk context for getting processed (they did not get out to the real world). However, my question is.... why do we not authenticate INVITE messages? If my understanding is correct, if would require something like the following: if (is_method("INVITE")) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } } If so, why not also do it in the default configuration file? [1] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=et… [2] http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

6 8

Update existing module or create new?
by Charles Chance 08 Mar '13

08 Mar '13

All, We are working on a memcached module which uses the libmemcached library, as opposed to the (no longer under active development) libmemcache that the existing module depends on. Am I correct in thinking that in this instance, it would be better to create a brand new module (with a different name) rather than modify the existing one? In addition to changing the dependency, we will also be exporting new commands and potentially changing the format of the exported PVs. Regards, Charles -- www.sipcentric.com <http://www.sipcentric.com/> Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

4 8

Asterisk realtime with kamailio Load balancing issue for sip user
by Prakash N 07 Mar '13

07 Mar '13

Hi All, We have finished the Kamailio & Asterisk real time integration and load balancing also done using dispatcher module. Queue and voice mails are load balancing as well.When we are calling extension to extension it is showing in all the servers.It seems extension are not load balancing as per our knowledge. I have attached the kamailio.cfg for your reference,Find my coding below as mentioned. *# -- dispatcher params for DB support --* *modparam("dispatcher","db_url", "mysql:// openser:openserrw@192.168.1.170/openser")* *modparam("dispatcher", "table_name", "dispatcher")* *modparam("dispatcher", "setid_col", "setid")* *modparam("dispatcher", "destination_col", "destination")* *modparam("dispatcher", "flags_col", "flags")* *modparam("dispatcher", "priority_col", "priority")* * * * ----------------------------------------------------------------------------------------- * *# Dispatch requests* *route[DISPATCH] {* *if ( method=="INVITE" ) {* *# dst_select( "GROUP", "HASH METHOD")* * ds_select_dst("1","4");* * sl_send_reply("100","Trying");* * forward();#uri:host, uri:port);* * exit();* *}}* Kindly suggest the solution for the same. Thanks in advance. Regards, N.Prakash

3 10

Re: [SR-Users] scheduling inside SER
by Jiri Kuthan 07 Mar '13

07 Mar '13

not sure if there is a point in doing that, at least for incoming traffic since it already happens in kernel, unless you want to do it in kernel. jiri On 3/7/13 2:25 PM, irma zujovic wrote: > Thank you for your answer. > > So is there any way to produce some kind of algorithm for prioritization of SIP messages inisde FOKUS IMS Core platform by configuring SER? That was my first > goal, when asking the previous question. I tought that scheduling and buffering of messages inside FOKUS IMS is done inside SER, and tought about priorituzation > inside SER. > > If you can help me in any way i would be very greatful. > > Thank you in advance > > Irma > > On Tue, Mar 5, 2013 at 4:13 PM, Jiri Kuthan <jiri(a)iptel.org <mailto:jiri@iptel.org>> wrote: > > On 3/5/13 2:36 PM, irma zujovic wrote: > > Hello, > > I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. > > My question is : where inside SER configuration scripts, the *scheduling * and *queueing *of SIP messages is done? I am interested in a special function > or part > of the written program. > > > this is not done in SER, it is done in kernel. kernel dispatches incoming traffic to > one of multiple processes listening on sockets. Same for outgoing traffic -- SER leaves > it to kernel's decision. > > jiri > > > I hope that engineers that worked on developing of SER could help me with this issue. > > Thank you in advance > > Irma > > > _________________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

1 0

Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 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. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 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=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 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. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 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=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 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. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

2 1

installing v4.0.x from git
by Daniel-Constantin Mierla 07 Mar '13

07 Mar '13

Hello, I created the wiki page with the how-to install kamailio v4.0.x from git. The tutorial is available at: - http://www.kamailio.org/wiki/install/4.0.x/git It should make life easier for those willing to give a try to v4.0.0 these days. Anyhow, a review of the content is very appreciated. Feel free to enhance the content by adding to the wiki page or reply here with your ideas to improve. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

4 7

Fwd: kamailio rls mod
by Aleksandrs Semenenko 06 Mar '13

06 Mar '13

Hello, I'm looking for assistance in RLS module setup for kamailio 3.3.2 The problem is that I need to handle resource-list updates on my kamalio server. For testing purposes I use Jitsi client. Expected: 1) When a new contact is added to Jitsi or an old one is removed, Jitsi sends updated resource-list XML to caps. - This works fine, I can see PUT request in Wireshark and I can see following log from kamailio.cfg: xlog("===== xhttp put: refreshing resource-list for $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); which hopefully means that rls_update_subs function has been called. 2) After this function was called, I expect SUBSCRIBE messages in backend, not in Jitsi, but I still see Jitsi sending SUBSCRIBE for each contact. What am I doing wrong? Any help would be useful! Regards, Alex.

4 3

New developer: Hugh Waite
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Hugh Waite - he is quite old contributor by now, with many patches and activity on mailing lists. He is part of the Crocodile RCS team, which developed very important features for Kamailio so far, like websocket and outbound extensions, Lua API and presence/rls improvements. His git commit id is: hpw My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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early rpm and deb packages from branch 4.0
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, if anyone is preferring to use packages for pre-release testing of v4.0.0, I created rpms for centos, redhat, opensuse and fedors, plus packages for debian squeeze. You can find them on opensuse build service servers: - https://build.opensuse.org/package/show?package=kamailio40&project=home%3Ak… - https://build.opensuse.org/package/show?package=kamailio40-d6.0&project=hom… I made them mainly to be sure there are no issue compiling on common linux OSes. Debian packages should be available at the usual repository for more versions (including ubuntu), courtesy of Jon Bonilla. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Kamailio Geoip
by Ian French 06 Mar '13

06 Mar '13

Hey List, I'm looking at an article by Daniel, http://by-miconda.blogspot.ie/2010/10/best-of-new-in-kamailio-310-5-geoip-a…. In it he's uses the below to block source IP addresses by country, How could this be used to block multiple country sources?? I've tried a couple of things aleardy but have failed geoip_match("$si", "src"); if($gip(src=>cc)=="CA") { xlog("SIP message from Canada (ip: $si) - drop it\n"); send_reply("403", "Forbidden"); exit; } Thanks in advance Ian

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Kamailio World Updates - March 06
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, the schedule for Kamailio World Conference has been updated, among topics of the recently added talks: North American VoIP market, Jitsi softphone, usage in embedded systems and enterprises. Check the current schedule at: - http://conference.kamailio.com/k01/schedule/ The group of speakers for the two days is filled up at this moment, as soon as we received the profiles and presentation summaries, the schedule will get final. Regarding Kamailio project, these are the last days before the next major release, version 4.0.0 is planned to be out next week. This week is CeBIT in Hanover, many community members will visit the event in various days (to my knowledge, Thursday and Friday will be the busiest). If you want face to face meetings, announce yourself on our forums. I plan to be there tomorrow. It is just a bit over one month till the event starts, if you haven't registered yet, you should hurry up, there is no similar event in Europe this year that gives the chance to meet and learn from many professionals working with rich communication services and products: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Simple look table
by Gareth Rylance 05 Mar '13

05 Mar '13

Is it possible to have a custom look-up table to map ids to other ids in the configuration file? E.G. TestLookValues key | value 1 |100 2 | 100 3 | 600 lookup("TestLookValues", "3") gives $var(lookupval) equal to 600 It is similar to MTree functionality only I am looking for an exact match not just a prefix TestLookValues prefix | value 1 |100 2 | 100 3 | 600 mt_match("TestLookValues", "3003)", "0") gives $var(mtval) = 600 I would need $var(mtval) = -1 (not found) Thanks Gareth

2 1

Troubles with call between 2 users
by TF Julian Plata 05 Mar '13

05 Mar '13

Hello, I'm Julian Plata and I have some troubles to do a call between 2 users: julianplata(a)iptel.org since Media5 in my phone, and marinastereo(a)iptel.org in a Comrex box. When I try to connect, Media5 shows a message "The caller can not be found". Also, in the Iptel User Management the users julianplata@iptel & marinastereo@iptel are shown as online. Could you help me to fix the issue ? Could you verify if the accounts are OK for establish a connection?? Thanks, Best Regards, Julian Plata -- ------------------------------------------------------------------------------------ La informacion contenida en este E-mail es confidencial y solo puede ser utilizada por el individuo o la compania a la cual esta dirigido. Si no es el receptor autorizado, cualquier retencion, difusion, distribucion o copia de este mensaje es prohibida y sera sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo al remitente y borrar el mensaje recibido inmediatamente. Cualquier punto de vista expresado en este mensaje es unicamente de quien lo envia excepto donde especificamente se indique que es la posicion de la Armada Nacional de Colombia (ARC). ------------------------------------------------------------------------------------ Apaga la pantalla de su computador cuando salgas a almorzar. No malgastemos la Energía. Cuidemos lo que es de Todos. ------------------------------------------------------------------------------------ Antes de imprimir este e-mail piensa bien si es necesario hacerlo: 1 hoja de papel tamaño carta = 2 litros de agua + 10 gramos de materia prima (decolorantes, fijadores, pigmentos) + 1 bombillo de 40 Vatios prendido por 1 hora + 14 gramos de madera pura.

2 1

Kamailio problem with both clients in the same NAT
by mohavere36 04 Mar '13

04 Mar '13

4 4

releasing v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, from my point of view, the last to sort out for v4.0.0 is packaging (I just sent a dedicated email to decide on that). Personally I have been testing v4.0.0 thoroughly in various cases, no critical issue being on my list at this moment. Also, v4.0 is running for quite some time on voipuser.org (our usual life testing platform). I propose releasing v4.0.0 next Monday, on March 11, 2013. The second option, if more people prefer it, would be Thursday, March 14, 2013. Opinions? If anybody has other dates in mine, let us know here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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packaging v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, (cross-posting as this decision may be done properly with feedback from both developers and users community). Now that we have some band new modules but also several that were not packaged by Kamailio, just by ser (due to single modules directory), I was thinking we may want to reconsider a bit the packages we do for v4.0.0. 1) A new package is related to the bunch of IMS modules, that is pretty much obvious. 2) There are few modules targeting development samples or troubleshooting: print, print_lib and malloc_test - I think they should not be packaged. 3) I would create a special package for the modules using uid-based database schema: db2_ops uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db uri_db 4) Maybe we should merge some packages by dependencies: - a) some depend on libxml2: cpl-c xhttp_pi xmlrpc xmlops - b) some depend on libssl: tls auth_identity outbound 5) Some experimental modules with external deps will stay unpackaged like so far: db_oracle Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

When to rewrite o= in SDP ?
by Khoa Pham 04 Mar '13

04 Mar '13

Hi, I read in http://kamailio.org/docs/modules/1.4.x/nathelper.html that Kamailio might rewrite ip address in o= in SDP to facilitate NAT traversal. It may decide to use to original IP or use IP of RTP server. What factors affect its decision ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

UnixODBC cant insert sip_capture record on sql server
by Grant Bagdasarian 03 Mar '13

03 Mar '13

Hello, We are using SQL Server 2008 as our database. When the sip_capture module tries to insert a row, the following error is given: db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL Server]Incorrect syntax near the keyword 'authorization'. Using Kamailio 3.3.4. Is there a way to force brackets around the column values? Regards, Grant

2 3

uac_replace_from / New URI shorter than old URI
by Ronald Voermans 03 Mar '13

03 Mar '13

Hi, this last weekend we went live from the old 3.0 version to the latest stable release of Kamailio (3,3,0). All went well, except for one issue. We make use of uac_replace_from in our config, in case a client needs CLIR: uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); Since the upgrade we receive the following error: uac [replace.c:521]: new URI shorter than old URI Which causes the call to be disconnected after about 20/30 seconds. This, because the ACK is received by Kamailio, but cannot be 'linked' to the transaction i guess. What is exactly causing this issue and, more importantly, is there any way to resolve this? Thanks in advance, Regards, Ronald Voermans

2 4

Which is first, t_on_reply or callbacks in C ?
by David 03 Mar '13

03 Mar '13

Hello, Can someone tell me which is executed first ? the on reply route setup with t_on_reply() or the callbacks in the code that allows the module dialog to insert the confirmed dialog into the dialog table ? Thanks, David

2 1

not able to install Kamailio
by Julián Quintero 03 Mar '13

03 Mar '13

Hello thereI hope that you can help me, I have an issue installing Kamailio, when I issue th command "make all" I receive the following error resolve.h:471:4: error: #error neither gethostbyname2 or getipnodebyname present make: *** [action.o] Error 1 I am using debian squeeze, please let me know if there is something missing in my system or what is not allowing to install the SIP Server THanks Julián

3 3

t_lookup.c: WARNING: script writer didn't release transaction - problem
by Efelin Novak 03 Mar '13

03 Mar '13

Hi folks, we have a serious issue with tm module. In Fail route we would like to send a reply 404 when a cr_route function fails(no destination found). We use t_reply("404", "No route to destination"). However after the call is disconnected with a 404 reply the tm module writes a following warning WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Very soon after this error message is shown a segfault occurs. Log is as follows: Mar 3 15:26:46 hostname kamailio[791]: ERROR: carrierroute [cr_func.c:594]: desired routing domain doesn't exist, prefix 0039333333333, carrier 14, domain 2 Mar 3 15:26:46 hostname kamailio[791]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction //ACK is received Mar 3 15:26:52 hostname kernel: [771134.093250] kamailio[796]: segfault at 4 ip 0815c182 sp bf8ace00 error 6 in kamailio[8048000+1ea000] Mar 3 15:26:53 hostname kamailio[803]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:785]: child process 796 exited by a signal 11 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:788]: core was generated Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Mar 3 15:26:53 hostname kamailio[802]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[799]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[801]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[798]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[800]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[797]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[792]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[803]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[795]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[791]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[794]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[793]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x886b3ad8/0x886b3ae0) - ignore Mar 3 15:26:53 hostname kernel: [771135.681720] kamailio[790]: segfault at 7dd193e0 ip 0816f5cd sp bf8acc20 error 4 in kamailio[8048000+1ea000] Currently we have fixed it using a custom route route(T_REPLY); . . . route[T_REPLY] { sl_send_reply("404","No route to destination"); } This works correctly with no error or a warning message or a segfault. We are using kamilio version: kamailio 3.3.4 (i386/linux) f8c8f2 Is this behavior a bug or should I dig deeper into the tm documentation? Kind regards Efelin

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Karl Smith
by Karl Smith 02 Mar '13

02 Mar '13

witwcvrf ictniydtoz.vsd/http://www.loudounwalkin.com/sq/f7nax0z0n2ouq77adgkgejjvnz.g… yklxx.vsd/knc hj.vsd/

1 0

RPM packages for v3.3.4
by Daniel-Constantin Mierla 01 Mar '13

01 Mar '13

Hello, the RPM packages for latest minor release v3.3.4 were created on openSuse build service. More details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Karl Smith
by Karl Smith 01 Mar '13

01 Mar '13

http://www.louiselemelin.com/sinkzm/hhir5ew9drwu4s54d429jdnj.d?a6dfinu09fqc… Smith3/1/2013 3:00:01 PM

1 0

How to let Kamailio to always use RTP proxy
by Khoa Pham 01 Mar '13

01 Mar '13

Hi, Supposed A call B in a non-NAT environment. How to let Kamailio to always force A, B to use RTP proxy ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

4 4

Help with Asterisk RT integration
by Jon Morby 28 Feb '13

28 Feb '13

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

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Re: [SR-Users] SBC
by Keith 28 Feb '13

28 Feb '13

Hi All, So following a guide I have successfully setup Kamailio and FreeSwitch together working as an SBC. SIP registration with Kamailio and using FreeSwitch as the media server which in turn authenticates with users in Kamailio. I have managed to pass calls between users etc. So the next challenge is bolting 2 Asterisks servers behind this SBC and load balancing the media. As the SBC will be public facing I also have to consider NAT which I believe will be configured in FreeSwitch. I have had some fantastic advice in the past so just interested to get peoples views again. Thanks. -- Keith

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Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 28 Feb '13

28 Feb '13

Hi Govoiper Free Memory on my server 2309 MB total used free shared buffers cached Mem: 2795 486 2309 0 70 180 -/+ buffers/cache: 234 2561 Swap: 2927 0 2927 *Kamailio error:* * * ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *tail -f /var/log/messages:* Feb 12 23:41:28 localhost /usr/local/sbin/kamailio[4280]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: usrloc [ul_mod.c:407]: child(1): failed to connect to database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module usrloc (/usr/local/lib64/kamailio/modules_k/usrloc.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [pt.c:350]: ERROR: fork_process(): init_child failed for process 3, pid 4283, "udp receiver child=2 sock=127.0.0.1:5060" Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: : <core> [main.c:1625]: main_loop: Cannot fork With Regards N.Prakash ---------- Forwarded message ---------- > From: SamyGo <govoiper(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:49 PM > Subject: Re: [SR-Users] [sr-dev] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: Muhammad Shahzad <shaheryarkh(a)gmail.com>, sr-dev(a)lists.sip-router.org, > Prabhakaran B <prabhakaran(a)tevatel.com>, Indu Priya <indu(a)tevatel.com> > > > Hi Prakash, > > Please paste All ERROR appearing and Warnings when you start your > kamailio. What is the free memory situation on your server ? > > Regards, > Sammy > > > > > On Tue, Feb 12, 2013 at 5:12 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Hi Muhammad, >> >> Yes, I have define Mysql on top in Kamailio.cfg >> >> DBURL "mysql://openser:openserrw@192.168.1.170/openser" >> >> >> I have attached Kamailio.cfg ,please advice >> >> With Regards >> >> N.Prakash >> >> >> On Tue, Feb 12, 2013 at 5:37 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com>wrote: >> >>> What is your connection string in kamailio i.e. DBURL? Also make sure >>> you have defined WITH_MYSQL on the top of kamailio.cfg to load necessary >>> drivers. >>> >>> #!define WITH_MYSQL >>> >>> Thank you. >>> >>> >>> On Tue, Feb 12, 2013 at 12:56 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>> >>>> Hi Shahzad , >>>> >>>> It is not ,We can able to connect remote mysql ( please find >>>> the detail below),moreover we can able create openser db from Kamailio >>>> server to remote mysql server >>>> >>>> *Mysql server : 192.168.1.170* >>>> *Kamailio server : 192.168.1.101* >>>> >>>> >>>> *Mysql output :* >>>> * >>>> * >>>> *mysql -u openser -h 192.168.1.170 -p* >>>> Enter password: >>>> Welcome to the MySQL monitor. Commands end with ; or \g. >>>> Your MySQL connection id is 68 >>>> Server version: 5.1.67 Source distribution >>>> >>>> Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights >>>> reserved. >>>> >>>> Oracle is a registered trademark of Oracle Corporation and/or its >>>> affiliates. Other names may be trademarks of their respective >>>> owners. >>>> >>>> Type 'help;' or '\h' for help. Type '\c' to clear the current input >>>> statement. >>>> >>>> mysql> >>>> >>>> With Regards >>>> >>>> N.Prakash >>>> >>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Muhammad Shahzad <shaheryarkh(a)gmail.com> >>>>> Date: Tue, Feb 12, 2013 at 5:16 PM >>>>> Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server >>>>> To: Development mailing list of the sip-router project < >>>>> sr-dev(a)lists.sip-router.org> >>>>> Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, >>>>> sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, >>>>> Prabhakaran B <prabhakaran(a)tevatel.com> >>>>> >>>>> >>>>> Its not a Kamailio error, it is problem with connecting to mysql >>>>> server. Error 4 means No MySQL Server Running On Specified Host. >>>>> >>>>> May be MySQL server is down or there is some firewall running or MySQL >>>>> not listening on IP you specified. >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>>>> >>>>>> Dear Daniel, >>>>>> >>>>>> Thanks for your quick response >>>>>> >>>>>> We can able to create operseb db on remote mysql server ,but >>>>>> Kamailio not getting start >>>>>> >>>>>> Error detail: >>>>>> >>>>>> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to >>>>>> MySQL server on '192.168.1.170' (4) >>>>>> ERROR: <core> [db.c:32 0]: could not add connection to the pool >>>>>> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >>>>>> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >>>>>> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >>>>>> >>>>>> >>>>>> I have attached my kamailo.cfg ( we followed Kamailio and >>>>>> Asterisk real time installation) ,please advice >>>>>> >>>>>> With Regards >>>>>> >>>>>> N.Prakash >>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>>>> Date: Tue, Feb 12, 2013 at 2:41 PM >>>>>>> Subject: Re: [SR-Users] openser db on an external mysql server >>>>>>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>>>>>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>>>>>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>>>>>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> kamailio database is created with: >>>>>>> >>>>>>> kamdbctl create >>>>>>> >>>>>>> the host of mysql server has to be specified in kamctlrc file via >>>>>>> DBHOST variable - kamctlrc is located in same directory as kamailio.cfg. >>>>>>> >>>>>>> In kamailio.cfg, update the db_url value (or DBURL define) to >>>>>>> reflect the mysql server host. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> On 2/12/13 4:16 AM, Prakash N wrote: >>>>>>> >>>>>>> Hi All, >>>>>>> >>>>>>> I am new this Kamailo ,ther is any step by step document to set >>>>>>> up openser db on an external mysql server ? >>>>>>> >>>>>>> With Regards >>>>>>> >>>>>>> N.Prakash >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Kamailio World Conference, April 16-17, 2013, Berlin >>>>>>> - http://conference.kamailio.com - >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sr-dev mailing list >>>>>> sr-dev(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Muhammad Shahzad >>>>> ----------------------------------- >>>>> CISCO Rich Media Communication Specialist (CRMCS) >>>>> CISCO Certified Network Associate (CCNA) >>>>> Cell: +49 176 99 83 10 85 >>>>> MSN: shari_786pk(a)hotmail.com >>>>> Email: shaheryarkh(a)googlemail.com >>>>> >>>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >

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GIT master branch is open for development of v4.1.x
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT master branch is again open for new development. Common testing period of releasing v4.0.0 has ended, if you have pending commits or personal branches to merge, you can go ahead. v4.0.0 will be released out of newly created branch 4.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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