- sr-users - kamailio.org

Kamailio 3.3 presence with resource lists and integrated XCAP
by Sangeeta Shah 28 Sep '12

28 Sep '12

Hello All, I am trying to configure Kamailio v 3.3 to serve as a presence server with resource lists and integrated XCAP. I followed the respective module documentation and am trying to test with the Boghe IMS client. I just can't get it to work! I think I still don't have everything I need in the config file to handle the xcap documents. Attached is the debug log from the server. If someone has had luck setting up the presence server with integrated xcap and resource lists please take a few minutes to take a look at the log file and see if you can help me out. I am also posting the xcap portion of the config file: #!ifdef WITH_XCAPSRV ##!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); #!ifdef WITH_XHTTPAUTH if (!www_authorize("10.50.251.12", "subscriber")) { www_challenge("10.50.251.12", "0"); exit; } #!endif if($hu=~"^/xcap-root/") { set_reply_close(); set_reply_no_connect(); # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": #xcaps_put("$var(uri)", "$hu", "$rb"); #if($xcapuri(u=>auid)=~"pres-rules") #{ # xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); # pres_update_watchers("$var(uri)", "presence"); # pres_refresh_watchers("$var(uri)", "presence", 1); #} xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { rls_update_subs("$var(uri)", "presence"); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } -- Sangeeta Shah

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DB accounting missing if t_newtran() is called explicitly
by Andrew Pogrebennyk 28 Sep '12

28 Sep '12

Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply("100", "Trying"); xlog("L_INFO", "New request - $ci\n"); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life.

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SRV record in via header not resolving.
by Jijo 28 Sep '12

28 Sep '12

Hello, I'm observing that Via Header domain name of type SRV is resolving as DNS A. I looked at the code and it does support only A. It is basically setting to default SIP port check_via_address which is causing DNS A query. Is there any specific reason to do so? I changed the code to support SRV as below and working fine. static int check_via_address(struct ip_addr* ip, str *name, unsigned short port, *short proto*, int resolver) { : : //if (port==0) port=SIP_PORT; if (resolver&DO_DNS){ DBG("check_via_address: doing dns lookup\n"); /* try all names ips */ he=sip_resolvehost(name, &port, *(char *) &proto*); /* don't use naptr */ if (he && ip->af==he->h_addrtype){ for(i=0;he && he->h_addr_list[i];i++){ if ( memcmp(&he->h_addr_list[i], ip->u.addr, ip->len)==0) return 0; } } } : : : } /* check if IP address in Via != source IP address of signaling */ int received_test( struct sip_msg *msg ) { int rcvd; rcvd=msg->via1->received || check_via_address(&msg->rcv.src_ip, &msg->via1->host, msg->via1->port,* msg->via1->proto*, received_dns); return rcvd; } Thanks Jijo

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Kamailio sip domain
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

Hi, What is the sip domain in kamctlrc file? Can I change it whatever I want? Is it must be a valid sip domain? Can I give a name like "mysipdomain.com", but there is no server with this name?

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invalid version 6 for table starting from git rev 78dae896127ce6762e3fa7c2541e1b5f9b8a9023
by siklub 28 Sep '12

28 Sep '12

Hi All, This is what i have done. * get kamailio from git git clone git://git.sip-router.org/sip-router * Installing kamailio from working rev A * destroy the database * update kamailio code from git * installing kamailio from not working rev B * create the database again (kamdbctl create) * starting kamailio This is the error that i got. ep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: <core> [db.c:440]: invalid version 6 for table location found, expected 5 (check table structure and table "version") Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: registrar [reg_mod.c:484]: failed to register domain I try to find last working rev Last working rev is (rev A) : ca551f7cb3770a08832758e543587415b3c6d80d (2012-09-11) First not working rev is (rev B) : 78dae896127ce6762e3fa7c2541e1b5f9b8a9023 There is no special reason for destroying the database, i'm just playing with kamailio from https://www.kamailio.org/wiki/install/devel/git, it seems that i have done right procedure. Am i missed something? Thanks

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dispatching xcap requests
by Juha Heinanen 28 Sep '12

28 Sep '12

is it possible to use kamailio based dispatcher to dispatch also xcap requests to one or more kamailio instances that serve them, i.e., is it possible to call t_relay in event_route [xhttp:request]? -- juha

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Kamailio sip user authentication
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

I installed kamailio from this link: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And I used Linphone and/or CSipSimple android apps as sip clients. And if I give the correct sip server name (usually server ip address), kamailio does not check the sip username and password and sip clients can register. In short, kamailio does not have authentication machanism for sip clients. Is it true or I'm doing something wrong? I searched the web and saw an authentication module called auth_identity. Is it a module to make authentication and if yes, how to activate it? Thanks Regards

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non-local socket info message
by Juha Heinanen 27 Sep '12

27 Sep '12

when i do registrar save in multiple registrar setup, i get this kind of info message to syslog when socket field value in location table does not match local socket of the registrar: INFO: usrloc [udomain.c:321]: non-local socket <tcp:192.98.103.2:5070>...ignoring looks like location table entries are updated no matter of the message. what does "ignoring" mean? in my case i don't use socket field value for anything, because requests to contacts are always routed to dispatcher. would it make sense to have a module param that tells usrloc to skip handling of socket info? -- juha

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help with kamailio crash
by Vicente Hernando 26 Sep '12

26 Sep '12

Hello, we are using a kamailio 3.3.1 version here and suddenly it crashed with a core. Any point about how to debug this will be appreciated! # /usr/local/kamailio/sbin/kamailio -V version: kamailio 3.3.1 (x86_64/linux) 91e8cb flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 91e8cb compiled on 04:59:57 Sep 13 2012 with gcc 4.6.3 (gdb) bt full #0 0x00007f242514e445 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f2425151bab in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x00000000004740b2 in sig_alarm_abort (signo=<optimized out>) at main.c:695 No locals. #3 <signal handler called> No symbol table info available. #4 0x00007f24252066d7 in syscall () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000542c58 in futex_get (lock=0x7f23e1ceb7a8) at mem/../futexlock.h:110 i = <optimized out> #6 timer_del_safe (tl=0x7f23e1d378e0) at timer.c:729 ret = -1 #7 0x0000000000447ab6 in destroy_dns_cache () at dns_cache.c:245 No locals. #8 0x0000000000474f95 in cleanup (show_status=1) at main.c:572 memlog = <optimized out> #9 0x0000000000475969 in shutdown_children (show_status=1, sig=15) at main.c:712 No locals. #10 0x00000000004767ac in handle_sigs () at main.c:803 chld = <optimized out> chld_status = 139 memlog = <optimized out> #11 0x0000000000478ed5 in main_loop () at main.c:1762 i = <optimized out> pid = <optimized out> si = <optimized out> si_desc = "udp receiver child=5 sock=79.170.68.187:5060\000\177", '\000' <repeats 18 times>, "\001\000\000\000\000\000\000\000\000\000\300>\000\000\000\000\001\000\000\000#\177\000\000Q\251U\000\000\000\000\000\020p\321$$\177\000\000\b\000\000\000\000\000\000\000x\264\316\341#\177\000\000\000gE\345\322\024P\027" nrprocs = <optimized out> #12 0x000000000041add2 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2546 cfg_stream = <optimized out> c = <optimized out> r = <optimized out> tmp = 0x7fff551d8f24 "" ---Type <return> to continue, or q <return> to quit--- tmp_len = 1427999204 options = 0x5df520 ":f:cm:M:dVIhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 603487729 rfd = 0 debug_save = <optimized out> debug_flag = <optimized out> dont_fork_cnt = <optimized out> n_lst = 0x0 p = <optimized out> Thanks in advance, Vicente.

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Question about CPU
by Ricardo Martinez 26 Sep '12

26 Sep '12

Hello. Is normal this output from the “top” command?! I have a kamailio with 16 children receiving calls at rate of 30cps and using the dialog module to timeout the calls : This is the top command : Tasks: 168 total, 2 running, 166 sleeping, 0 stopped, 0 zombie Cpu0 : 56.0%us, 0.7%sy, 0.0%ni, 43.4%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu1 : 6.3%us, 3.0%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.7%hi, 2.3%si, 0.0%st Cpu2 : 0.3%us, 0.3%sy, 0.0%ni, 99.3%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu3 : 0.7%us, 0.3%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3924912k total, 535700k used, 3389212k free, 20236k buffers Swap: 6291448k total, 0k used, 6291448k free, 226120k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND *2566 root 20 0 1163m 27m 23m R 56.2 0.7 2:08.32 kamailio * 1103 root 20 0 249m 1656 924 S 2.7 0.0 1:32.89 rsyslogd 2525 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.93 kamailio 2527 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.11 kamailio 2528 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.89 kamailio 2529 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.13 kamailio 2535 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.86 kamailio 2537 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.85 kamailio 2538 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.17 kamailio 2541 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.91 kamailio 2542 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.07 kamailio 2543 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.12 kamailio 2544 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.64 kamailio 2549 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.10 kamailio 2551 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.14 kamailio 2552 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.00 kamailio 2495 root 20 0 15088 1240 900 R 0.3 0.0 0:04.10 top 2526 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.83 kamailio 2532 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.87 kamailio 1 root 20 0 19396 1500 1192 S 0.0 0.0 0:02.86 init 2 root 20 0 0 0 0 S 0.0 0.0 0:00.01 kthreadd 3 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 4 root 20 0 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/0 5 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 6 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 The question is : I always see the process PID : 2566 using more CPU than the rest of the childrens. Is this normal? Regards, Ricardo Martinez-.

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LDAP user authentication
by martian＠centrum.sk 26 Sep '12

26 Sep '12

Hi. I have been looking into LDAP way of authenticating users. Reading this guide http://www.kamailio.org/dokuwiki/doku.php/tutorials:kamailio31-auth-ldap and reading the AUTH and LDAP module documentations, it seems to me that currently you can bind to LDAP (using some service account for example) and perform the SEARCH operation for data only. Therefore you need to retrieve user login and password from the LDAP db and than authenticate the user in Kamailio. My question is: What is you can't simply retrieve the password from LDAP db?? Is it possible to do a BIND operation to LDAP, using login name and password provided by user in REGISTER message? (this means not using the ones specified in the external ldap config file). BIND operation kind of authenticates the user. So theoreticaly, if LDAP binding authentication succeeds, the user is trusted and can be replied with 200 OK. This in fact means: using bind operation instead of search operation when a REGISTER message (with Authorization header) arrives. Any opinions on corectness of this approach are welcome, along with clarifying the possibility to do this. Thanks in advance. Martin

1 0

Audiocide
by Cong Le Kha 26 Sep '12

26 Sep '12

Dear Kamailio I have got a problem as you with Audiocode SIP/ SECurity/ MGmt/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow />NOTIC:( sip_stack)(185 ) AcSIPParser: Problem in SIP Message Headers RECOV:( sip_stack)(186 ) !! [ERROR] AcSIPParser: Parse Error. No appropriate delimiter found in multipart body RECOV:( sip_stack)(187 ) !! [ERROR] Message type: INVITE RECOV:( sip_stack)(188 ) !! [ERROR] Source header: RECOV:( sip_stack)(189 ) !! [ERROR] Line: 41. Column: 1 Can you help me?

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[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

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Kamailio and SylkServer
by Gary Shergill 25 Sep '12

25 Sep '12

Hi Kamailio Community, I was wondering how I would integrate Kamailio into SylkServer please? The SIP IM/presence part of Kamailio that is. When I was attempting to get the Presence and IM working I received the following reply; =================== > If you use sylkserver, then you can just forward the >INVITE to sylkserver, which has an MSRP relay integrated, =================== I'm not entirely sure what that means, but I found the following which I believe relates to it; http://sylkserver.ag-projects.com/projects/sylkserver/wiki/Applications I'm not entirely sure if; a) this is the right thing I'm looking at, b) where I would add these lines to Kamailio. Thanks for any help you can provide. Kind Regards, Gary Shergill

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Asterisk and Dispatcher Module
by Ian French 25 Sep '12

25 Sep '12

Hi, I've been working my way throught this tutorial ( http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) and all it working well thanks. In the other benifits section you mention that this configuration can be used with multiple instances of Asterisk. Could you elaborate a little on this?? Is this achieved using the Dispatcher module? I'm trying to get this working with two instances of Asterisk but I can figure out how to specifc multiple asterisk servers as oppossed to the single. asterisk.bindip = "192.168.178.23" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" Thanks in advance Ian

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Re: [SR-Users] is kamilio support RFC5626 now ?
by Peter Dunkley 25 Sep '12

25 Sep '12

Hi, I have been working on adding full RFC5626 support to Kamailio in my spare time - but it is taking longer than I hoped because I haven't been able to spend as much time as I wanted working on it. There is a branch in git (pd/outbound) where I have put some of the module skeleton code, and I still hope to get this done and tested well before the end of the year (and certainly in time for Kamailo 3.4). Regards, Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >_______________________________________________ >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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Starting kamailio. Error on topoh module
by Miguel Baptista 25 Sep '12

25 Sep '12

Hello, I am testing the "topology hiding" module (topoh) on kamailio-3.2.0 and every-time I start kamailio I get these error messages related with the topoh module: Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:79]: +++ ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.- Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:80]: +++ y0GxFtSUVeTOMWzHPam.jKR93gCoNpkicrJBqIEhvYbnd175AfLl4u6wZs-X8Q2D I am only getting those errors when kamailio is starting. Kamailio runs fine and it hides my internal topology, on the SIP Headers, without any error message. I tried to add/modify/remove several topoh parameters but I am always getting those errors starting kamailio. Any idea why I getting it? How can I get rid of it? I don't feel comfortable to apply that configuration on my production servers knowing that kamailio outputs an error when starting. Thanks, - Miguel Baptista

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Dialog Module : "inconsitent dlg timer data"
by Ricardo Martinez 25 Sep '12

25 Sep '12

Hello. Can someone explain to me why i’m getting this error : WARNING: dialog [dlg_req_within.c:173]: inconsitent dlg timer data on dlg 0xb565c500 [2166:2082643875] with clid ' 5ed68b60-fdbe-4aa7-819e-0002a400e27c(a)64.76.155.86' and tags '5e604a9ea4' 'ed505c1da4' This is the output for dlg_list_ctx kamctl fifo dlg_list_ctx dialog:: hash=2995:55475986 state:: 4 ref_count:: 2 timestart:: 1348177242 timeout:: 15084496 callid:: 33d68b60-a09a-b1dc-8198-0002a400e27c(a)64.76.155.86 from_uri:: sip:557100050994@64.76.155.86 from_tag:: 3360b198a4 caller_contact:: sip:557100050994@64.76.155.86 caller_cseq:: 226 caller_route_set:: caller_bind_addr:: udp:64.76.155.82:5060 callee_bind_addr:: udp:64.76.155.82:5060 to_uri:: sip:0267005622408596@64.76.155.82 to_tag:: ba507815a4 callee_contact:: sip:0267005622408596@64.76.155.130 callee_cseq:: 226 callee_route_set:: context:: I can’t find the problem. I’m using this dialog params : # ---- dialog params ------ modparam("dialog", "dlg_flag", 31) modparam("dialog", "profiles_with_value","gwid" ) modparam("dialog", "default_timeout",7200) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) $avp(i:10)=10 $dlg_ctx(timeout_bye) = 4; dlg_manage(); Regards, Ricardo

2 3

kamailio workers and private memory
by Uri Shacked 25 Sep '12

25 Sep '12

Hi, I use "kamctl sercmd pkg.stats" to check the private memory status. I notice that the workers for the proccess "slow timer", "timer", "MI FIFO", use a lot of memory. My system configuration does "mt_reload" for around 5 million lines every hour, it looks like in time, the pkg memory of the workers above is growing...... I set the private memory to 16M instead of 4M. Now the used size (after 3 weeks) is 4M. I will upgrade to 3.3.0 soon and will use mem_join=1, will it help the private memory defrag as well? How can I increase the number of workers for these processes? Will it help increasing it if possible? BR, Uri

2 1

Issue Starting up Kamalio.
by Jay chary 24 Sep '12

24 Sep '12

Hi We are trying to install Kamailio v3.3.1 on CENTOS, I followed the below urls for installation http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Howeever when I try to start Kamailio by issuing kamctl start , I see the below response : INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed and I see the below in the messages log : Sep 20 10:08:49 kamailio kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: usrloc [hslot.c:53]: locks array size 512 Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: auth_db [authdb_mod.c:244]: empty parameter 1 not allowed Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:543 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:552 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:585 Sep 20 10:09:30 kamailio avahi-daemon[2760]: Invalid query packet. Can u please help us ? Regards Chary

3 2

Kamailio presentations
by Pavel Segeč 24 Sep '12

24 Sep '12

Hello Everybody, I'm an university teacher and founder of the nil.uniza.sk portal with some SIP/Kamailio tutorials. I'd like to use the Kamailio server as the basement for my class lecturers about the IP multimedia communication systems. May anyone help me and provide a link to usable presentation about the kamailio, which should I use and probably extend in my classes? I search the web but nothing appropriate have not found thanks palo73

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Re: [SR-Users] xcap server support for org.openmobilealliance.xcap-directory
by Peter Dunkley 23 Sep '12

23 Sep '12

Hi, org.openmobilealliance.xcap-directory isn't supported by the xcap-server module. Adding support for this shouldn't be difficult though. Regards, Peter Juha Heinanen <jh(a)tutpro.com> wrote: >in xcap module readme there is an example xcap server config file and it >does not include in xcap-caps aiud >org.openmobilealliance.xcap-directory. does it mean that xcap-directory >application is not currently supported or could the aiud just be added >there? > >-- juha > >_______________________________________________ >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

2 3

xcap server support for org.openmobilealliance.xcap-directory
by Juha Heinanen 22 Sep '12

22 Sep '12

in xcap module readme there is an example xcap server config file and it does not include in xcap-caps aiud org.openmobilealliance.xcap-directory. does it mean that xcap-directory application is not currently supported or could the aiud just be added there? -- juha

1 0

"Command not found" error during "make install"
by Youngjin Park 22 Sep '12

22 Sep '12

Hi, I faced the following error during installing "sip-router" by command "make install" after successful compilation by command "make all". I will appreciate you so much if you can shed light on. My system is 32 bit Fedora. Thanks in advance. Youngjin install -m 644 README /usr/local/share/doc/ser/ docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth.7] Error 127 make: [modules-man] Error 1 (ignored) docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth_db.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth_db.7] Error 127 make: [modules_s-man] Error 1 (ignored) make[1]: `sercmd' is up to date.

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no socket found for match second RR
by Juha Heinanen 21 Sep '12

21 Sep '12

when proxy gets ack to invite, it spits out this kind of error message: Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: INFO: processing ACK <sip:jh@192.98.103.10:5074> Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: WARNING: rr [loose.c:744]: no socket found for match second RR route headers in the request are: Route: <sip:192.98.103.10:5060;transport=tcp;r2=on;ftag=43884fd989cff6dc;lr> Route: <sip:192.98.103.10:5060;r2=on;ftag=43884fd989cff6dc;lr> and when i start the proxy it tells this: Listening on udp: 192.98.103.2 [192.98.103.2]:5070 udp: 127.0.0.1:5072 tcp: 192.98.103.2 [192.98.103.2]:5070 tcp: 127.0.0.1:5072 tcp: 127.0.0.1:6060 Aliases: *: 192.98.103.10:5060 *: sip.test.fi:5060 why does loose_route() not find the first alias? -- juha

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any easy way to access record-route URIs?
by Juha Heinanen 21 Sep '12

21 Sep '12

is there any easy way to access URIs in one or more record-route headers? there is @via[index], but no corresponding select for record- route URIs. -- juha

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Re: [SR-Users] [sr-dev] kamailio core at qm_status ( patch required important)
by Jijo 21 Sep '12

21 Sep '12

Hi Daniel, This patch needs to be applied to avoid core. Thanks Jijo On Wed, Sep 19, 2012 at 10:54 AM, Jijo <realjijo(a)gmail.com> wrote: > > Hi All, > > Finally i found the issue, > > Here is one of the bad trace for SUBSCRIBE(722bytes) and NOTIFY(1282bytes) > which corrupted the memory. The messages came in the order NOTIFY and > SUBSCRIBE. The core is generated in a different place but I believe this > could be the reason for memory corruption. > > Here is the trace UDP Process 27294 processing NOTIFY and Process 27303 processing > SUBSCRIBE . > > > > The explanation and implementation is below > > > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R39 - Entry R-URI=1234 FD=10.233.20.152 > SI=10.233.20.152 > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27294] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:2357]: tcp_conn_send_put : calling wbufq_add > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:730]: ERROR: wbufq_add(722 bytes): buf:SUBSCRIBE > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;tran > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:747]: ERROR: wbufq_add(722 bytes): first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:774]: ERROR: wbufq_add(2 last free crt_size:722): > first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:796]: ERROR: wbufq_insert(1282 bytes): buf:NOTIFY > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;transpo > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:801]: ERROR: wbufq_insert(2 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:820]: ERROR: wbufq_insert(22 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27359] ERROR: <core> > [tcp_main.c:887]: ERROR: wbufq_run(3 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:755]: child process 27359 exited by a signal 11 > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:758]: core was generated > > 2012-09-19T02:06:18+01:00 [info] sipserver: [27265] INFO: <core> > [main.c:770]: INFO: terminating due to SIGCHLD > > > > Process 27294(NOTIFY) created the TCP connection structure for destination > IP and just before calling wbufq_insert(), context switch happened and > process 27303(SUBSCRIBE) got the cpu. Since the connection structure is > already available process 27303 add the SUBSCRIBE message(722 bytes) to > wbufq. Afterwards process 27294 got the CPU and invoked wbufq_insert() > which basically corrupted the memory due to an overflow with the existing > implementation. > > > > > > inline static int _wbufq_insert(struct tcp_connection* c, const char* > data, > > > unsigned int size) > > { > > struct tcp_wbuffer_queue* q; > > struct tcp_wbuffer* wb; > > > > q=&c->wbuf_q; > > if (likely(q->first==0)) /* if empty, use wbufq_add */ > > return _wbufq_add(c, data, size); > > : > > : > > : > > : > > if ((q->first==q->last) && > ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for > size bytes */ > > > > memmove(q->first->buf+size, > q->first->buf, size); > > memcpy(q->first->buf, data, size); > > q->last_used+=size; > > } > > > > The above condition shall be true in this case and memmove was moving the > pointer which was causing the overflow. > > memmove(void *dest, const void *src, size_t n); > > > > As per the memmove man page, the src shall be copied with size ‘n’ to a > temporary buffer and then temporary buffer to dest. > > > > dest is q->first->buf+size: which is basically (q->first->buf + NOTIFY > MSG SIZE). so the dst will move my 1282 bytes, so we have remaining space > of only 818 bytes. > > > > src is q->first->buf: which is basically copied to temp buffer with NOTIFY > SIZE(1282bytes). > > > > Finally we are moving the buffer from temporary buffer of size 1282 bytes > to buffer which we left with 818 bytes, This basically corrupt the memory > and on wbufq_run we see the memory corruption > > > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > > > I think we don’t need memove, so we can change the code as below > > > > if ((q->first==q->last) && ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for size bytes */ > > //memmove(q->first->buf+size, q->first->buf, size); > > memcpy(q->first->buf+q->last_used, data, size); > > q->last_used+=size; > > } > > OR > we need only the else part as it always add the block to the first. > > Thanks > Jijo > > On Wed, Jul 18, 2012 at 12:42 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> sorry, I just read the last messages in the thread and I didn't noticed >> it is about a crash, but thought it is about an annoying log message. >> >> The backtrace is no longer matching the latest sources of the 3.1 branch, >> but I expect it is due to a double free issue, so you have to update to >> latest 3.1.6, as suggested by other users here. There were many fixes from >> 3.1.0 to 3.1.6. >> >> Cheers, >> Daniel >> >> >> On 7/17/12 6:03 PM, Jijo wrote: >> >> Hi, >> >> This is not happening at shutdown or status check. Its aborting when >> the system is active. >> >> Thanks >> Jijo >> On Tue, Jul 17, 2012 at 3:06 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> does it keep being or it is one time and that's it? >>> >>> That is printed at shut down or if pkg_status() or shm_status() is >>> executed from some part of code or in config via cfgutils module functions. >>> >>> You can get rid of them by setting memdbg and memlog to a value higher >>> than debug global parameter. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 7/16/12 5:28 PM, Jijo wrote: >>> >>> Thanks.. It is not easy to upgrade as it is happening at customer >>> system. >>> Is there any change occurred for this issue.I looked at it, but didn't >>> see anything in q_malloc.c/qm_status() >>> >>> >>> On Mon, Jul 16, 2012 at 11:12 AM, Jon Bonilla <manwe(a)aholab.ehu.es>wrote: >>> >>>> El Mon, 16 Jul 2012 10:27:42 -0400 >>>> Jijo <realjijo(a)gmail.com> escribió: >>>> >>>> > Hi All, >>>> > >>>> > I'm observing a core intermittently at "qm_status (qm=0x786cd000) at >>>> > mem/q_malloc.c:763" for kamailio version 3.1.0 >>>> > >>>> >>>> I'd say that you're using a very old version. You should update your >>>> branch to >>>> 3.1.6 or upgrade to a newer branch. >>>> >>>> >>>> _______________________________________________ >>>> sr-dev mailing list >>>> sr-dev(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>> >>> >>> >>> _______________________________________________ >>> sr-dev mailing listsr-dev@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >> >> >

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Kamailio OS Sysctl Tweaks
by Cenk İlker İzanlı 20 Sep '12

20 Sep '12

Hi, Is there any sysctl tweaks for OS to get the most of Kamailio as performance meaning? Thx

2 2

Kamailio Presence module
by Gary Shergill 20 Sep '12

20 Sep '12

Hi Kamailio Community, Recently installed Kamailio, plan to integrate with SylkServer. Tried the SIP IM feature, but can't seem to get it working. Logged on to Bria and Blink, two seperate accounts. They can call each other, but can't add each other for presence. So I installed the presence module with apt-get and edited the kamailio.cfg file to add the line; #!define WITH_PRESENCE Restarting the server comes up with the following error; ---------------------- root@opensep:~# vi /etc/kamailio/kamailio.cfg root@opensep:~# service kamailio restart Not starting Kamailio: invalid configuration file! -e 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: syntax error 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: Invalid arguments 0(5114) : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 37: ERROR: bad config file (3 errors) ---------------------- Those lines are as follows; ---------------------- #!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 ---------------------- The exact place is where it says DBURL on both params. I am using Ubuntu 12.04 and Kamailio 3.3.x. Thank you for your time. Kind Regards, Gary Shergill

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authentication for client applications
by David Thomson 20 Sep '12

20 Sep '12

Hi, I am working on a project where a custom sip client will be integrated into a suite of applications to provide voip. The sip client will be working with Kamailio. The goal is to ensure that the client is authorized for communication with kamailio before allowing any calls to be made. Conventional username/password authentication for individual users will also be used once the client has been authenticated. Currently other applications in the suite use a digital signature in the http headers when communicating with server processes. If the signature is validated by the server process then the applications identity is validated and communication with the server process is allowed. Is it possible to include a public key and digital signature in the register events and have kamailio perform the transformation to validate the client's identity? If so which module provides such functionality? Has something like this been implemented in the past? Thanks for any input. ttyl,Dave

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call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

In LOCATION_BRANCH, after lookup location, I've added: if ($du==$null) $du=$ru; And now everything works! In summary, either of the following would fix the problem: 1. call fix_nated_contact for all clients, so that 'received' column in location table is populated (so that $du will be set), or 2. set $du after lookup if it is empty Wonder if it is worth adding fixes in the source code. Thanks very much Klaus for pointing me to the right direction! Yufei Date: Wed, 19 Sep 2012 15:13:29 +0100 From: Yufei Tao <yufei.tao(a)redembedded.com> Subject: Re: [SR-Users] call forking using dbaliases not working for un-NATed clients To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5059D309.6030500(a)redembedded.com> Content-Type: text/plain; charset="ISO-8859-1" I should have made it clearer: $du is null both before and after lookup location in LOCATION_BRANCH when no fix_nated_register was done (thus 'received' column in location table was null). When fix_nated_register was done, $du for each branch was null before lookup location, but set to 'received' after lookup location as you said. But in this case (fix_nated_register done), everything works fine. So seems now the problem happens when: 'received' in location table is null, causing $du not to be set. I used to think $du is only set by record-route and thanks for clarifying this Even when without fix_nated_register (thus no 'received'), the lookup location for each branch, i.e. y2 and y3, set $ru to the 'contact' field successfully, but $du isn't set. But Kamailio did relay to the first branch (trunk?) and not the others. So the first branch is handled differently? So maybe when $du is null, should set it to $ru on the branches? Yufei On 19/09/12 11:53, Klaus Darilion wrote: > On 19.09.2012 12:24, Yufei Tao wrote: >> Hi Klaus >> >> Thanks for the reply! >> >> I check the $du, it is always null before and after the lookup. Is it >> only set when relaying to a proxy (from record-route), and not to a >> client? > That's strange. For NATed clients, $du must contain the 'received' > URI. Otherwise they can not be contacted as $ru contains the private > IP address. > >> When no fix_nated_register is called, the lookup location for both >> clients y2 and y3 is successful from the log, when printing out $ru >> after lookup. But seems Kamailio only relays to one client's IP while >> not others. I think there must be some differences when branch route is >> executed first time and second time as you said. As it feels like for >> the first branch (trunk?) it used the 'contact' column from the location >> table, and for the other branches, it tries to use 'received'? > It seems db_lookup() creates multiple branches. Is lookup() only > finding 1 contact in table or multiple contacts? > > regards > Klaus > >> Yufei >> >> On 18/09/12 18:49, Klaus Darilion wrote: >>> I suspect that the branch route is first executed for the NATed >>> client. Then the 'received' column is used as destination URI. When >>> executing the branch route again, the destination URI is still the >>> value from the previous branch, and lookup() will not overwrite is as >>> 'received' is not available. Then Kamailio sends the INVITE again to >>> the first client. >>> >>> You can try to set $du to Null before lookup(). ($du=null or >>> $du=$null, not sure what the correct syntax is). >>> >>> Another workaround is to use fix_nated_register() for every client >>> (the pragmatic and more secure approach). >>> >>> regards >>> Klaus >>> >>> On 18.09.2012 15:16, Yufei Tao wrote: >>>> Hi >>>> >>>> I have a strange problem on forking calls to a group of users. For >>>> example I have two users y2 and y3 in dbaliases, both with >>>> alias_username 'group'. And y2 and y3 both registers with Kamailio >>>> fine. >>>> When I make a call to 'group' from a third client y1, what my >>>> kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to >>>> BRANCH_ALIASDB, where a lookup location will be done for each of the >>>> username resulting from lookup of dbaliases, something like this: >>>> >>>> y1--INVITE 'group'-->lookup >>>> dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup >>>> location---? >>>> >>>> | |-->relay >>>> >>>> ---->[BRANCH_ALIASDB]--->'y3'-->lookup >>>> location---- >>>> >>>> The all works well as long as all clients are NAT'ed. However when >>>> they >>>> are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only >>>> goes to one of the group members, e.g. y2 only. When checking the log, >>>> it seemed to have done the dbaliases lookup fine, and each location >>>> lookup successfully. But Kamailio only relayed y2's IP, e.g. to the >>>> client, while y3's to itself. >>>> >>>> When comparing the location table when clients are NAT'ed or not, I >>>> find >>>> that the 'received' column is only populated when I do >>>> fix_nated_register. And group calls only works when 'received' >>>> column is >>>> populated. That explains why when clients are NAT'ed group calls work, >>>> as I only do fix_nated_register if nat_uac_test returns true. >>>> >>>> But if this is the only reason, if two clients register using the same >>>> username, e.g. both as y3, and when 'received' column of location >>>> table >>>> is empty (no fix_nated_register done), I would expect a call to y3 >>>> should also only make 1 client ring. But in fact both of them rang! >>>> The >>>> flow is like: >>>> >>>> y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client >>>> registered as 'y3' >>>> | >>>> ---> IP of 2nd client >>>> registered as 'y3' >>>> >>>> While a call to 'group' (thus dbaliases lookup took place) under such >>>> un-NAT'ed set up made only 1 client ring. >>>> >>>> So I can make it work by always doing fix_nated_register. But I'm not >>>> clear about these things: >>>> - why does a lookup of dbaliases before lookup of location make such >>>> difference? >>>> - does lookup location work differently depending on whether it is >>>> called from trunk or from a route called from a branch route? >>>> >>>> >>>> Following is relevant parts from my config file: >>>> >>>> ############################################################# >>>> route[LOCATION] >>>> { >>>> if ( alias_db_lookup("dbaliases") ) >>>> { >>>> t_on_branch("BRANCH_ALIASDB"); # in >>>> branch_route[BRANCH_ALIASDB], >>>> # call another route that >>>> looks up >>>> location, >>>> # if not existent, call drop() >>>> >>>> } >>>> else >>>> { >>>> xlog("L_DBG","LOCATION: not alias - go to lookup location >>>> trunk\n"); >>>> route(LOCATION_TRUNK); # normal look up location and sending of >>>> 404 etc >>>> } >>>> >>>> ... ... >>>> } >>>> ############################################################# >>>> branch_route[BRANCH_ALIASDB] >>>> { >>>> xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; >>>> Method:$rm\n"); >>>> route(LOCATION_BRANCH); >>>> } >>>> >>>> route[LOCATION_BRANCH] >>>> { >>>> if (!lookup("location")) >>>> { >>>> # Drop this branch - it's going nowhere >>>> drop(); >>>> } >>>> } >>>> ############################################################# >>>> route[RELAY] { >>>> xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> >>>> #!ifdef WITH_NAT >>>> if (check_route_param("nat=yes")) { >>>> setbflag(FLB_NATB); >>>> } >>>> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { >>>> xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); >>>> route(RTPPROXY); >>>> } >>>> #!endif >>>> >>>> /* example how to enable some additional event routes */ >>>> if (is_method("INVITE")) { >>>> t_on_reply("REPLY_ONE"); >>>> t_on_failure("FAIL_ONE"); >>>> } >>>> >>>> if (!t_relay()) { >>>> sl_reply_error(); >>>> } >>>> >>>> exit; >>>> } >>>> >>>> ############################################################ >>>> route[NAT] { >>>> #!ifdef WITH_NAT >>>> xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> force_rport(); >>>> if (nat_uac_test("2")) { >>>> if (method=="REGISTER") { >>>> fix_nated_register(); >>>> xlog("L_DBG","NAT: Just done fix_nated_register in >>>> REGISTER >>>> message in NAT route\n"); >>>> } else { >>>> xlog("L_DBG","NAT: fix_nated_contact\n"); >>>> fix_nated_contact(); >>>> } >>>> setflag(FLT_NATS); >>>> } >>>> >>>> # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force >>>> media relay >>>> #!endif >>>> return; >>>> } >>>> >>>> ############################################################# >>>> And in the main route, route LOCATION and RELAY are the last two >>>> routes: >>>> route { >>>> ... >>>> route(NAT); >>>> >>>> ... ... >>>> >>>> # user location service >>>> route(LOCATION); >>>> >>>> route(RELAY); >>>> } >>>> >>>> Hope I have made it clear. Thanks very much! >>>> >>>> Yufei >>>> -- >>>> Yufei Tao >>>> Red Embedded -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 0

IM on Kamailio
by Gary Shergill 19 Sep '12

19 Sep '12

Hi Kamailio Community, I've been configuring the presence module on Kamailio and trying to get it working, using the following as a guideline; http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-ka… >From that I assumed that enabling Presence would mean that IM would be enabled too. However, that doesn't seem to be the case. May I ask please how to configure IM on Kamailio? Note that I am testing this with one computer connected by Bria and another computer connected via Blink. I am able to log on to a user on each (test1 and test2) and they are able to call each other. The issue is, with presence enabled, they are unable to IM each other (or add each other as contacts and see online status). Thank you. Kind Regards, Gary Shergill

4 4

Kamailio-Redirect based LCR routing
by Fatima Chahrour~Vanrise Support 19 Sep '12

19 Sep '12

Dear Daniel\List, Following up to a previous post where I was able in my testing lab to redirect call with 1st destination of lcr_gw table to another Kamailio(acting as any SBC) with Daniels assist, I need a hand in redirecting the call with multiple contacts , basically based on the lcr routing rule configured in kamailio LCR module for the rerouted call. Is this possible using code 300 with multiple choices message? Or you advise me with another method? If 300 redirect message works, what should be the variable set as $ru to get all lcr_gw destinations? Example for clarification: My current redirect script used after lcr functions is : .. $ru= $ru; sl_send_reply("302","Moved Temporary"); .. This script is successfully sending the 302 message with Contact: sip:1119613454646@192.168.111.195. (1119613454646 is the dialed number with prefix, 192.168.111.195 is first lcr_gw ip). Now if the call to 192.168.111.195 is failed and I need to redirect the second destination, second lcr_gw ip, how can I achieve this? Thank you for your usual response. Kind Regards, F Chahrour

2 4

AUTOREPLY call forking using dbaliases not working for...
by strobel＠hfo-telecom.de 18 Sep '12

18 Sep '12

******************* AUTOREPLY VON yufei.tao(a)redembedded.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 17.09.2012 bis einschließlich 21.09.2012 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Künzel oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _______________________________________________ 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

record_route_advertised_address question
by Juha Heinanen 18 Sep '12

18 Sep '12

rr readme on record_route_advertised_address function tells: Note: A second Record-Route will be inserted if the transport used on the inbound and outbound interfaces changes. that is not the case. i made call record_route_advertised_address("192.98.103.10:5060"); when request came over udp and went out over tcp and only one rr header was inserted: incoming: U 2012/09/17 17:30:17.467079 192.98.103.10:5060 -> 192.98.103.2:5070 INVITE sip:test@test.fi SIP/2.0. ... outgoing: T 2012/09/17 17:30:17.504631 192.98.103.10:5060 -> 192.98.103.10:38078 [AP] INVITE sip:0x20b7580@192.98.103.10:5050;transport=tcp SIP/2.0. Record-Route: <sip:192.98.103.10:5060;ftag=wumod;lr>. ... i have not set enable_double_rr param and default is 1. is this a bug that i should report on the tracker? -- juha

3 5

Port change 5060
by Vijay Thakur 18 Sep '12

18 Sep '12

All Experts, I want to change my SIP port from 5060 for other one. Before making any change in my live server (Kamailio 3.2.1, i want to be sure. Kindly suggest me that where should i make changes to implement it. This is a security measure for kamailio from port scanning. With thanks in advance. Vijay Thakur

5 6

pidf-lo extraction
by Cenk İlker İzanlı 17 Sep '12

17 Sep '12

Hi, Is there any basic document showing how to extract info from pidf-lo data on kamailio part? Thx.

2 1

rsyslog settings for good kamailio performance
by Albert Petit 17 Sep '12

17 Sep '12

Hi, For load testing I would like to setup rsyslog so Kamailio logs are written asynchronously Would the following configuration be enough for that purpose in * rsyslog.conf*? # # Write log file in asynchronous mode and use an IO Buffer of 64 Kb # IO buffer config http://www.rsyslog.com/doc/rsyslog_conf_global.html # # $OMFileAsyncWriting on $OMFileFlushOnTXEnd off $OMFileFlushInterval 60 $OMFileIOBufferSize 128k #---------------------------------------------------------------------- ## kamailio ##---------------------------------------------------------------------- $outchannel kamailio_log_rotation,/var/log/pocserverplatform/kamailio.log,524288000,/usr/local/bin/kamailio_log_rotation_script :programname, contains, "kamailio" $kamailio_log_rotation local0.* $kamailio_log_rotation Apart of that is there a way to completely disable completely any logs of Kamailio to discard they're causing any performance issue? Thanks Albert

3 2

kamailio prepaid services
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, Is there any guide to install/configure kamailio as prepaid service infrastructure? Thx

1 0

Re: [SR-Users] Limiting concurrent calls
by Alex Balashov 15 Sep '12

15 Sep '12

Htable is another popular method for those disposed less favourably to the dialog module. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Uri Shacked <ushacked(a)gmail.com> wrote:in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

Limiting concurrent calls
by Uri Shacked 15 Sep '12

15 Sep '12

in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri > Hi, > I need to limit the number of concurrent calls go through kamailio. > Its a simple set up(Not per user basis, a global limit will suffice). > > I know i need to employ dialog module. But i'm a little confused about > profiles. I thought for simple application like this, i may be able to > use profile altogether. > > Any code snippet to limit number of concurrent calls will be very helpful. > > Thanks in advance. > -- > -aft

1 0

I get no calls?
by gooly 15 Sep '12

15 Sep '12

Hi, I have registered my (Fritz!Box) ATA successfully registered on my account 8305234(a)iptl.org but what is tried I never received a call? Here is what I can see on 'my' web-page with you: > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:58 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:56 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:20 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:19 500 Server Internal Error Please can you tell me what I have to do to be able to receive Voip-Calls? Regards, Carl

2 1

Limiting concurrent calls
by Aft nix 15 Sep '12

15 Sep '12

Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

cfg. parser: failed to find command force_rtp_proxy
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, I'm installing kamailio and rtpproxy with this guide. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour Versions of the programs: # aptitude versions rtpproxy i 1.2.1-1 stable 500 root@li304-234:~# kamailio -V version: kamailio 3.3.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:18:19 Jun 18 2012 with gcc 4.4.5 When I start kamailio, I get this error, Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule?#012 Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule?#012 Related lines throwing error are below. 724 route[RTPPROXY] { 725 #!ifdef WITH_NAT 726 if (is_method("BYE")) { 727 unforce_rtp_proxy(); 728 } else if (is_method("INVITE")){ 729 force_rtp_proxy(); 730 } 731 if (!has_totag()) add_rr_param(";nat=yes"); 732 #!endif 733 return; 734 } 735 803 #!ifdef WITH_NAT 804 if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) 805 && status=~"(183)|(2[0-9][0-9])") { 806 force_rtp_proxy(); 807 } 808 if (isbflagset("6")) { 809 fix_nated_contact(); 810 } 811 #!endif Does anyone know what's the problem with cfg file? Thx. Cenk

2 2

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

8 37

Howto add peers and routing rules
by Joel Smith | VOZELIA 15 Sep '12

15 Sep '12

Hi, I'm getting started with kamailio and siremis. I've just finished getting the server ready and the software installed. Once in the siremis admin menu, i've added a domian and a subscriber and I have my softphone registered with kamailio. My doubt is, where do I add peers so I can set up the routing rules? Are there any tutorials? Thanks in advanced. Best regards. -- Joel Smith Cell: +34 639 03 13 53 E-Mail: joel(a)vozelia.com <joel(a)vozelia.com> <http://www.vozelia.com> <http://twitter.com/vozelia> <http://www.facebook.com/pages/Vozelia-Operador-de-telefonia-IP-para-empresa…>

2 1

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

Sorry, his name is Daniel Goepp. -----Original Message----- From: jdavidthomson(a)hotmail.com Date: Sat, 15 Sep 2012 03:53:51 To: Daniel-Constantin Mierla<miconda(a)gmail.com> Reply-To: jdavidthomson(a)hotmail.com Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

avp_radius load time high
by Ricardo Martinez 14 Sep '12

14 Sep '12

> Hello List. > > I’m having the next issue. I have a kamailio v3.2 working with a Radius > server as a backend. All the INVITE’s coming to the kamailio server are > challenged against the Radius server and then avp_load is called from > Radius server too. I’m having excessive delay times when I have near 10 > call per second. So I find it very weird. I have made a benchmark for the > avp_load : > > bm_start_timer("timer-tranum"); > > if( !radius_load_caller_avps("$fU@$fd")) { > > sl_send_reply("403","Forbidden - Estamos experimentando problemas"); > > exit; > > } else { > > xlog("L_INFO", "[$ci]:[AUTH_REQUEST]: ok"); > > }; > > bm_log_timer("timer-tranum"); > > > > and this is the output : > > > > Sep 12 18:26:07 benchmark (timer timer-tranum [1]): 112441 [ > msgs/total/min/max/avg - LR: > 100/18446744073695545751/23297/18446744073709236247/184467440736955456.000000 > | GB: > 2100/18446744073708471097/10746/18446744073709542058/8784163844623081.000000] > > Sep 12 18:26:55 benchmark (timer timer-tranum [1]): 280543 [ > msgs/total/min/max/avg - LR: > 100/10238598/21317/18446744073708596212/102385.980000 | GB: > 2200/9158079/10746/18446744073709542058/4162.763182] > > Sep 12 18:28:20 benchmark (timer timer-tranum [1]): 312902 [ > msgs/total/min/max/avg - LR: > 100/5048656/15491/18446744073709546022/50486.560000 | GB: > 2300/14206735/10746/18446744073709546022/6176.841304] > > Sep 12 18:28:31 benchmark (timer timer-tranum [1]): 301818 [ > msgs/total/min/max/avg - LR: > 100/5622716/43785/18446744073708810522/56227.160000 | GB: > 2400/19829451/10746/18446744073709546022/8262.271250] > > Sep 12 18:28:44 benchmark (timer timer-tranum [1]): 18446744073708689124 > [ msgs/total/min/max/avg - LR: > 100/1329199/59661/18446744073709167366/13291.990000 | GB: > 2500/21158650/10746/18446744073709546022/8463.460000] > > Sep 12 18:29:26 benchmark (timer timer-tranum [1]): 44160 [ > msgs/total/min/max/avg - LR: > 100/18446744073701320457/19576/18446744073708922050/184467440737013216.000000 > | GB: 2600/12927491/10746/18446744073709546022/4972.111923] > > Sep 12 18:30:18 benchmark (timer timer-tranum [1]): 218573 [ > msgs/total/min/max/avg - LR: > 100/18446744073707306935/17459/18446744073708876520/184467440737073056.000000 > | GB: 2700/10682810/10746/18446744073709546022/3956.596296] > > Sep 12 18:31:06 benchmark (timer timer-tranum [1]): 132751 [ > msgs/total/min/max/avg - LR: > 100/9611955/17057/18446744073708726431/96119.550000 | GB: > 2800/20294765/10746/18446744073709546022/7248.130357] > > Sep 12 18:31:55 benchmark (timer timer-tranum [1]): 18859 [ > msgs/total/min/max/avg - LR: > 100/18446744073705270953/18859/18446744073708747515/184467440737052704.000000 > | GB: 2900/16014102/10746/18446744073709546022/5522.104138] > > Sep 12 18:32:48 benchmark (timer timer-tranum [1]): 61954 [ > msgs/total/min/max/avg - LR: > 100/7124234/18036/18446744073708596768/71242.340000 | GB: > 3000/23138336/10746/18446744073709546022/7712.778667] > > Sep 12 18:33:39 benchmark (timer timer-tranum [1]): 53489 [ > msgs/total/min/max/avg - LR: > 100/18446744073707180114/18177/18446744073708790040/184467440737071808.000000 > | GB: 3100/20766834/10746/18446744073709546022/6698.978710] > > Sep 12 18:34:35 benchmark (timer timer-tranum [1]): 60604 [ > msgs/total/min/max/avg - LR: > 100/18446744073701108277/33983/18446744073708839351/184467440737011072.000000 > | GB: 3200/12323495/10746/18446744073709546022/3851.092187] > > Sep 12 18:35:24 benchmark (timer timer-tranum [1]): 54705 [ > msgs/total/min/max/avg - LR: > 100/7185862/15880/18446744073708608536/71858.620000 | GB: > 3300/19509357/10746/18446744073709546022/5911.926364] > > Sep 12 18:36:21 benchmark (timer timer-tranum [1]): 60356 [ > msgs/total/min/max/avg - LR: > 100/18446744073705699677/19466/18446744073708844524/184467440737056992.000000 > | GB: 3400/15657418/10746/18446744073709546022/4605.122941] > > Sep 12 18:37:14 benchmark (timer timer-tranum [1]): 18446744073708679441 > [ msgs/total/min/max/avg - LR: > 100/3506212/16102/18446744073708791471/35062.120000 | GB: > 3500/19163630/10746/18446744073709546022/5475.322857] > > Sep 12 18:38:07 benchmark (timer timer-tranum [1]): 93933 [ > msgs/total/min/max/avg - LR: > 100/18446744073704680760/17569/18446744073708741284/184467440737046816.000000 > | GB: 3600/14292774/10746/18446744073709546022/3970.215000] > > Sep 12 18:38:58 benchmark (timer timer-tranum [1]): 142777 [ > msgs/total/min/max/avg - LR: > 100/681953/15933/18446744073708708694/6819.530000 | GB: > 3700/14974727/10746/18446744073709546022/4047.223514] > > Sep 12 18:39:44 benchmark (timer timer-tranum [1]): 165921 [ > msgs/total/min/max/avg - LR: > 100/5153736/15749/18446744073708657622/51537.360000 | GB: > 3800/20128463/10746/18446744073709546022/5296.963947] > > Sep 12 18:40:31 benchmark (timer timer-tranum [1]): 127615 [ > msgs/total/min/max/avg - LR: > 100/18446744073709161716/20690/18446744073708793559/184467440737091616.000000 > | GB: 3900/19738563/10746/18446744073709546022/5061.170000] > > Sep 12 18:41:21 benchmark (timer timer-tranum [1]): 47762 [ > msgs/total/min/max/avg - LR: > 100/9001175/18909/18446744073708632234/90011.750000 | GB: > 4000/28739738/10746/18446744073709546022/7184.934500] > > Sep 12 18:42:11 benchmark (timer timer-tranum [1]): 62944 [ > msgs/total/min/max/avg - LR: > 100/18446744073708163674/17934/18446744073708825788/184467440737081632.000000 > | GB: 4100/27351796/10746/18446744073709546022/6671.169756] > > > > > > If the output is in microseconds and I have from time to time (for the > last 100 messages) values like 184467440737081632.000000 microsecs > > IS THIS NORMAL? > > Then it backs to a more rational value : 90011.750000 microsecs. > > I also made measures in my Radius Server and is answering with average > time around 0.01 secs. > > So… what could it be the problem?.. Could be the RadiusClient-ng ? > > In my attempt to solve the issue I increased the childrens for the > kamailio server from 16 to 60, but the problem persists. > > Can someone help me here? > > > > Best Regards, > > Ricardo Martinez.- > > > > > > >

3 7

collecting stats per subscriber
by Krishna Kurapati 14 Sep '12

14 Sep '12

Hi, Is there a module that can be used to collect number of MESSAGEs sent or received by a subscriber; number of registrations by a subscriber? These will be per subscriber as oppose to statistics collected for entire system. Thanks Krish Kura

2 2

How to do RTPProxy load balancing?
by Mino Haluz 14 Sep '12

14 Sep '12

Hi, mediaproxy has dispatcher and relay, so it load balances the traffic automatically. How it is done with rtpproxy ? Thanks, Mino

3 4

FAQ page on the wiki
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, I just started a FAQ page on the wiki site. Hopefully people will collect there useful content, based on their experience, questions that repeat on the mailing lists or other sources. Added just a few content for the moment, to show the syntax -- the page is available at: * http://www.kamailio.org/wiki/tutorials/faq/main It is using the QNA plugin for dokuwiki: * https://www.dokuwiki.org/plugin:qna Anyone can make an own account to edit the wiki, the registration form is available at: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Re: [SR-Users] Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

thank you for helps cenk and pavel. I see that I had wrong information about kamailio.

2 1

Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

juha writes: > this kind of questions have been asked many times on this list. search > archive for responses. you are right maybe but it is hard to search the archieve. And if it is easier to say "serach" than please show me way to search the arcieve quickly. In this link I m trying to search archieve month by month: http://lists.sip-router.org/pipermail/sr-users/ I m new and thanks your kind help...

3 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 14 Sep '12

14 Sep '12

Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own. None of this has to do with dialog state, though. Just rtpproxy control. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: > >> Peter: Thanks for the tip! Really interesting. But I do not >> understand, why also this list contains the calls that were ended by >> sipp... Should I search for some mistake in my kamaillio config ? > > Perhaps you don't close them with unforce_rtp_proxy: > > if(method=="BYE" || method=="CANCEL"){ > unforce_rtp_proxy(); > } > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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 _______________________________________________ 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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Re: [SR-Users] Max-Forwards: and loops. Testing purpose.
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, do not write private emails from mailing list discussions, if not sensitive data was asked explicitly. If there is no rule to breaking the loop, after some time the message gets very big. Cheers, Daniel On 9/14/12 7:03 AM, Anton Kvashenkin wrote: > Thanks for respond. I just have used forward() based on RURI. > > got > > 0(2558) ERROR: <core> [msg_translator.c:1835]: ERROR: > build_req_buf_from_sip_req: out of memory > 0(2558) ERROR: <core> [forward.c:601]: ERROR: forward_request: > building failed > 0(2558) WARNING: <core> [receive.c:211]: WARNING: receive_msg: error > while trying script > > With maxfwd it works like a charm. http://paste.ubuntu.com/1204093/ > > 2012/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > > On 9/11/12 1:12 PM, Anton Kvashenkin wrote: > > Hi, List. > > I'm using sipsak to fire some predefined sip message with > Max-Forwards: header equal 0 to test mf_process_maxfwd_header > function. I'm doing this exclusively for learning purpose. As > far as we know, this exported function from maxfwd module > services only for adding Max-Forwards header, decrement or > detect Max-Forwards: 0 header to prevent loops. So, my > question guys, how can I create this loop in my lab, with two > proxies for example? What message should I craft? > > what kind of message are you looking to craft? > > Looping from one proxy to another can be achieved with > t_relay("_the other proxy address_") function from tm module. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - > http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - > http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

Does kamailio supports g729 codec? Or which codecs does kamailio supports? Where or how to learn the supported or unsupported codecs for kamailio? Thanks in advance...

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

That's a rather dated study. But it's better than a nonexistent reference point, true. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:According to this (http://transnexus.com/index.php/performance-test-results-for-openser-and-rt…) "For a server hosting both OpenSER and RTPproxy, each 1 GHz of CPU processing capacity can manage a maximum of 325 simultaneous calls." I have 2.4GHz for rtpproxy, but CPU/Mem/network is ok, so the bottleneck should be somewhere else probably.. On Thu, Sep 13, 2012 at 5:56 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > I'm not sure what a single instance of rtpproxy can handle, but most people > squeezing thousand of concurrent calls per box are probably doing it on > multicore boxes by binding multiple instances of rtpproxy with different > core affinities, and round-robining among them. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Mino Haluz <mino.haluz(a)gmail.com> wrote: > The results: > > - rtpproxy calls count 280 > - sipp calls count 2000 > - iptraf on proxy 4.8MB/s > - G711a codec > > So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), > rtpproxy calls count is really the right value. CPU usage is ok on > every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy > cannot serve more than 270-280 calls ? > > On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: >> Ok, so I put there unforce_rtp_proxy even though I'm using >> rtpproxy_manage. The tip with nc now really shows the calls count. >> >> But the dialog count is still higher and higher, so I have bug >> somewhere in the configuration. I'll check it. >> >> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov >> <abalashov(a)evaristesys.com> wrote: >>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE >>> or >>> CANCEL. It'll just figure out what to do on its own. >>> >>> None of this has to do with dialog state, though. Just rtpproxy control. >>> >>> >>> >>> >>> -- Alex >>> >>> -- >>> Sent from my Samsung mobile, and thus lacking in the refinement one might >>> expect from a proper keyboard. >>> >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Decatur, GA 30030 >>> Tel: +1-678-954-0670 >>> Web: http://www.evaristesys.com/ >>> >>> Mino Haluz <mino.haluz(a)gmail.com> wrote: >>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >>> >>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> >>> wrote: >>>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>>> >>>>> Peter: Thanks for the tip! Really interesting. But I do not >>>>> understand, why also this list contains the calls that were ended by >>>>> sipp... Should I search for some mistake in my kamaillio config ? >>>> >>>> Perhaps you don't close them with unforce_rtp_proxy: >>>> >>>> if(method=="BYE" || method=="CANCEL"){ >>>> unforce_rtp_proxy(); >>>> } >>>> >>>> -- >>>> With best regards, Peter Lemenkov. >>>> >>>> _______________________________________________ >>>> 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 >>> >>> _______________________________________________ >>> 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 >>> >>> _______________________________________________ >>> 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 >>> _______________________________________________ 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

I'm not sure what a single instance of rtpproxy can handle, but most people squeezing thousand of concurrent calls per box are probably doing it on multicore boxes by binding multiple instances of rtpproxy with different core affinities, and round-robining among them. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:The results: - rtpproxy calls count 280 - sipp calls count 2000 - iptraf on proxy 4.8MB/s - G711a codec So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ? On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: > Ok, so I put there unforce_rtp_proxy even though I'm using > rtpproxy_manage. The tip with nc now really shows the calls count. > > But the dialog count is still higher and higher, so I have bug > somewhere in the configuration. I'll check it. > > On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or >> CANCEL. It'll just figure out what to do on its own. >> >> None of this has to do with dialog state, though. Just rtpproxy control. >> >> >> >> >> -- Alex >> >> -- >> Sent from my Samsung mobile, and thus lacking in the refinement one might >> expect from a proper keyboard. >> >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Decatur, GA 30030 >> Tel: +1-678-954-0670 >> Web: http://www.evaristesys.com/ >> >> Mino Haluz <mino.haluz(a)gmail.com> wrote: >> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >> >> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: >>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>> >>>> Peter: Thanks for the tip! Really interesting. But I do not >>>> understand, why also this list contains the calls that were ended by >>>> sipp... Should I search for some mistake in my kamaillio config ? >>> >>> Perhaps you don't close them with unforce_rtp_proxy: >>> >>> if(method=="BYE" || method=="CANCEL"){ >>> unforce_rtp_proxy(); >>> } >>> >>> -- >>> With best regards, Peter Lemenkov. >>> >>> _______________________________________________ >>> 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 >> >> _______________________________________________ >> 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 >> >> _______________________________________________ >> 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Really? Interesting, I had no idea. I thought the rtpproxy control protocol was binary and did not lend itself easily to interaction in this manner. Thanks for the tip. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > You can't get it from rtpproxy. You'd really have to use something like the > dialog or htable modules to keep call state and get that from Kamailio. On the contrary it's possible (using raw UDP reads/writes): work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 sessions created: 0 active sessions: 0 active streams: 0 work ~: Where * h1u203u03 is randomly chosen token, * 127.0.0.1 is the rtpproxy's control IP, * 22222 is the rtpproxy's control port, * "-u" means that we're using UDP * -w 1 is the timeout in seconds to wait before closing nc. I can't imagine that someone will use nc in performance testing but I think it looks like a good start. -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Thanks for this info. Netcat is a really underappreciated tool. I can't count how many times I've unnecessarily written utilities to send stuff on UDP sockets, forgetting that nc can do it. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. Yep, this command protocol allows us to do tricks like this which is quite helpful for debugging. Unfortunately this particular command ( 'I' and its counterpart 'Ib' ) seems to be broken - it doesn't prepend received random cookie at the beginning of the answer. Just FYI - I collected some examples of control commands here (just grep for ser_proto:decode): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_proto_test.erl and here (which could be even more interesting since it contains an example of a real session - create session, lookup, destroy, get stats, etc): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_test.erl > Thanks for the tip. Anytime :) -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

If the dialog module is working correctly, and the SIP flows are standards-compliant, the dialog module should automatically track all subsequent state changes and remove calls from tracking after a BYE is processed. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends, it still keeps info about this dialog in list "kamctl fifo dlg_list". Should I somehow close the dialog when the BYE transaction is ended ? Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ? On Thu, Sep 13, 2012 at 3:57 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. > Thanks for the tip. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: >> You can't get it from rtpproxy. You'd really have to use something like >> the >> dialog or htable modules to keep call state and get that from Kamailio. > > On the contrary it's possible (using raw UDP reads/writes): > > work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 > sessions created: 0 > active sessions: 0 > active streams: 0 > work ~: > > Where > > * h1u203u03 is randomly chosen token, > * 127.0.0.1 is the rtpproxy's control IP, > * 22222 is the rtpproxy's control port, > * "-u" means that we're using UDP > * -w 1 is the timeout in seconds to wait before closing nc. > > I can't imagine that someone will use nc in performance testing but I > think it looks like a good start. > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

You can't get it from rtpproxy. You'd really have to use something like the dialog or htable modules to keep call state and get that from Kamailio. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino _______________________________________________ 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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RTPProxy with kamailio : How to get calls count?
by Mino Haluz 13 Sep '12

13 Sep '12

Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino

1 0

script stops processing when relaying e2e ack?
by Andrew Mortensen 12 Sep '12

12 Sep '12

Hi all. I'm working on a custom module and would like to be able to send UPDATEs to the call legs after the script forwards the e2e ACK from the caller. I had hoped to export a function to call in my ACK handling route after the script forwards the ACK. However, the script stops processing after calling t_relay in my ACK handling route. I tracked this back to the fact that t_relay_to invokes forward_request when handling e2e ACKs, and returns the return code of forward_request to the script. Since forward_request returns 0 on success, the t_relay call in the script returns 0, which stops script processing. Is this intentional? My module currently hooks the TMCB_E2EACK_IN, and I can UPDATE the caller here, but the callee seems to be ignoring the UPDATE if it's received prior to the ACK. Thanks for any advice you can offer. andrew

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Kamailio 3.2, NAT, rtpproxy bridging between interfaces
by Russell McConnachie 12 Sep '12

12 Sep '12

Hello, I'm having some issues getting media bridging working in kamailio and rtpproxy. I've attempted to attach as much relevant information as possible. Thanks in advance for taking a look; I'm completely stuck right now. Actual results:No audio is being passed through the rtpproxy from caller "A" to caller "B" even though both media server and UAC are sending media to the rtpproxy. Expected results:Audio that is sent to rtpproxy is bridged/proxied across from caller "A" to caller "B" and vice-versa. ------------------------------------------------------------------------------- Signaling trace: ------------------------------------------------------------------------------- 08:34:57.543224 IP (tos 0x0, ttl 49, id 3237, offset 0, flags [none], proto: UDP (17), length: 1456) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 1428 INVITE sip:9991@r18.test-bwks.local:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Route: <sip:x.x.x.102:5060;lr> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@192.168.2.250:51620;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 619 v=0 o=MxSIP 0 0 IN IP4 192.168.2.250 s=SIP Call c=IN IP4 192.168.2.250 t=0 0 m=audio 51720 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv 08:34:57.545766 IP (tos 0x10, ttl 64, id 58136, offset 0, flags [none], proto: UDP (17), length: 439) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 411 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280;received=x.x.x.199 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Server: kamailio (3.2.4 (x86_64/linux)) Content-Length: 0 08:34:57.547619 IP (tos 0x10, ttl 64, id 47936, offset 0, flags [+], proto: UDP (17), length: 1500) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 1472 INVITE sip:9991@10.0.0.29 SIP/2.0 Record-Route: <sip:10.0.0.38;r2=on;lr=on;nat=yes> Record-Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes> Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0 Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@x.x.x.199:31280;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 629 v=0 o=MxSIP 0 0 IN IP4 10.0.0.38 s=SIP Call c=IN IP4 10.0.0.38 t=0 0 m=audio 29682 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event 08:34:57.551126 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 409) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 381 SIP/2.0 100 Trying Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060> Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Content-Length:0 08:34:57.561084 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 723) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 695 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.562404 IP (tos 0x10, ttl 64, id 58137, offset 0, flags [none], proto: UDP (17), length: 672) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 644 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.609771 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 953) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 925 SIP/2.0 200 OK Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:133 v=0 o=BroadWorks 196 1 IN IP4 10.0.0.24 s=- c=IN IP4 10.0.0.24 t=0 0 m=audio 10498 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 08:34:57.612047 IP (tos 0x10, ttl 64, id 58138, offset 0, flags [none], proto: UDP (17), length: 930) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 902 SIP/2.0 200 OK Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:161 v=0 o=BroadWorks 196 1 IN IP4 x.x.x.102 s=- c=IN IP4 x.x.x.102 t=0 0 m=audio 29774 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sdpmangled:yes 08:34:57.844645 IP (tos 0x0, ttl 49, id 3239, offset 0, flags [none], proto: UDP (17), length: 506) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 478 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes>, <sip:10.0.0.38;r2=on;lr=on;nat=yes> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 08:34:57.845182 IP (tos 0x10, ttl 64, id 47937, offset 0, flags [none], proto: UDP (17), length: 509) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 481 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 ------------------------------------------------------------------------------- The call session is up at this point. CPE Router, Source NAT IP x.x.x.199, Internal Network: 192.168.2.0/24 192.168.2.250 is an Aastra 6757i IP Phone, behind the x.x.x.102 is the external interface on our SIP proxy allowing traffic on: Inbound: port 5060/udp Inbound: range 16384 - 32768/udp 10.0.0.38 is the internal interface facing our BroadWorks lab cluster 10.0.0.29 is the internal interface of our BroadWorks Application Server 10.0.0.24 is the internal interface of our BroadWorks Media Server RTP Proxy has been spawned with the following command line arguments: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy \ -u root \ -s udp:localhost:7722 \ -l x.x.x.102/10.0.0.38 \ -d DBUG LOG_LOCAL5 \ -m 16384 -M 32768 We've changed the /etc/kamailio/kamailio.cfg and added the #!define's at the top of the configuration file: #!define WITH_NAT #!define WITH_DEBUG In the route[NATMANAGE] { } block we've added a few additional lines to handle the setup of the rtpproxy_manage() function which handles the rtpproxy_offer(), rtpproxy_answer() and unforce_rtp_proxy() convientenly in one function. The logic we've got in place currently is: if ((src_ip =~ "10\.0\.0\.*") && (dst_ip =~ "10\.0\.0\.*")) { xlog("L_INFO", "Media session for ($ci) with source address ($si): Internal > External: SDP contact/origin will be x.x.x.102"); rtpproxy_manage("FAIEOC", "x.x.x.102"); } else if (!(src_ip =~ "10\.0\.0\.*") && (dst_ip == x.x.x.102)) { xlog("L_INFO", "Media session for ($ci) with source address ($si): External > Internal: SDP contact/origin will be 10.0.0.38"); rtpproxy_manage("FAEIOC", "10.0.0.38"); } else { xlog("L_INFO", "!!! Uncaught condition for media session for ($ci) with source address ($si) !!!"); } This rtpproxy log is not from the call above, but the same source #, destination # were used: Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session 84bf2a7cb9d4ab24, tag dcfa6a860e;1 requested, type strong Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session on a port 28282 created, tag dcfa6a860e;1 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling caller's address with 192.168.2.250:51720 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: lookup on ports 28282/22500, session timer restarted Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling callee's address with 10.0.0.24:10502 ... Call is currently active and then hung-up ... Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:process_rtp: session timeout Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: session on ports 28282/22500 is cleaned up Example of the RTP: 08:53:13.555834 IP (tos 0x0, ttl 48, id 510, offset 0, flags [none], proto: UDP (17), length: 200) x.x.x.199.32052 > x.x.x.102.16552: UDP, length 172 x.x.x.199:32052 > x.x.x.102:16552 Why is this packet not being forwarded to the media proxy (rtpproxy) to 10.0.0.24 (specified in the 200 OK from the BroadWorks Application Server) 08:53:13.562206 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) 10.0.0.24.10504 > 10.0.0.38.17968: UDP, length 172 10.0.0.24:10504 > 10.0.0.38:17968 Why is this packet not being forwarded from rtpproxy to the UAC at x.x.x.199? ------------------------------------------------------------------------------- System details: ------------------------------------------------------------------------------- OS: CentOS 5.8 x86-64, Hypervisor is qemu-kvm Memory: 2GB Disk: 40G VCPU: 1 /etc/sysctl.conf: # Controls IP packet forwarding net.ipv4.ip_forward = 1 ------------------------------------------------------------------------------- Current software versions: ------------------------------------------------------------------------------- [root@sip-proxy1 kamailio]# rpm -qa | grep -E 'kamailio|rtpproxy' | xargs rpm -qi Name : rtpproxy Relocations: (not relocatable) Version : 1.2.1 Vendor: Fedora Project Release : 2.el5 Build Date: Sun 19 Sep 2010 08:12:23 AM EDT Install Date: Mon 10 Sep 2012 09:23:28 PM EDT Build Host: x86-10.phx2.fedoraproject.org Group : Applications/Internet Source RPM: rtpproxy-1.2.1-2.el5.src.rpm Size : 105682 License: BSD Signature : DSA/SHA1, Mon 20 Sep 2010 12:40:46 PM EDT, Key ID 119cc036217521f6 Packager : Fedora Project URL : http://www.rtpproxy.org Summary : A symmetric RTP proxy Description : This is symmetric RTP proxy designed to be used in conjunction with the SIP Express Router (SER) or any other SIP proxy capable of rewriting SDP bodies in SIP messages that it processes. Name : kamailio Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:32 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 10200784 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : Kamailio, very fast, reliable and flexible SIP Server Description : Kamailio is a very fast, reliable and flexible SIP (RFC3261) proxy server. Written entirely in C, kamailio can handle thousands calls per second even on low-budget hardware. A C Shell like scripting language provides full control over the server's behaviour. It's modular architecture allows only required functionality to be loaded. Among available features: IPv4, IPv6, digest authentication, accounting, CPL scripts, instant messaging, MySQL and UNIXODBC support, SIMPLE presence agent, radius authentication, record routing, SMS gateway, ENUM, UDP, TCP, TLS and SCTP, transaction and dialog module, OSP module, statistics support, registrar and user location, SNMP, SIMPLE Presence and Perl programming interface. Name : kamailio-cpl Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:33 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 296947 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : CPL module (CPL interpreter engine) for Kamailio Description : The kamailio-cpl package provides a CPL interpreter engine for Kamailio

2 1

Max-Forwards: and loops. Testing purpose.
by Anton Kvashenkin 12 Sep '12

12 Sep '12

Hi, List. I'm using sipsak to fire some predefined sip message with Max-Forwards: header equal 0 to test mf_process_maxfwd_header function. I'm doing this exclusively for learning purpose. As far as we know, this exported function from maxfwd module services only for adding Max-Forwards header, decrement or detect Max-Forwards: 0 header to prevent loops. So, my question guys, how can I create this loop in my lab, with two proxies for example? What message should I craft?

2 1

using kamailio with clients in a nat environment
by David Thomson 12 Sep '12

12 Sep '12

hi, I was wondering if in clients in a nat environment need to be in the dmz or need to have ports forwarded for kamailio to be used to connect the clients? I have setup kamailio with the nat settings but am still only getting one way audio. chances are it has not been setup correctly but just checking. If i try to connect the two clients from the same network i can get zrtp and 2 way communication working correctly. if one of the clients is on a different network the call initiates and answers but zrtp and 2way audio does not work. kamailio 3.3 running on ec2 ttyl,Dave

2 4

Re: [SR-Users] Jurisdictional (inter vs intra calling) LCR on Kamailio
by Alex Balashov 11 Sep '12

11 Sep '12

William, This is a business-layer function that is far outside the scope of what built-in Kamailio modules can do. In fact, even the 'lcr' module is something of a misnomer: it does not do LCR in a truly financialised sense, to say nothing of jurisdictionally specific regulatory awareness. It's just a basic weighted routing table. The appropriate place to find such functionality, unless you want to write it yourself, is in a variety of ITSP-oriented commercial platforms built on top of Kamailio. They exist precisely to fill these gaps and bring the somewhat general capabilities of Kamailio into an applied dimension that is useful in specific business and product contexts. Many members of this list represent software or consulting companies that offer such solutions. However, as it is a noncommercial list, I'll refrain from plugging them directly. You may want to ask the question on Kamailio-Business. You can also email me privately and I would be happy to give you some unbiased pointers about the commercial ecosystem. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/William Fulton <wfulton(a)thirdhatch.com> wrote:I can’t find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don’t know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

Jurisdictional (inter vs intra calling) LCR on Kamailio
by William Fulton 11 Sep '12

11 Sep '12

I can't find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don't know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

rls notify delay
by Juha Heinanen 11 Sep '12

11 Sep '12

one rls thing that stilll wonders me is the about 2-5 sec delay that occurs when backend notifies have been received and notify to subscriber ua is sent: Sep 11 17:27:41 siika /usr/sbin/sip-proxy[4495]: INFO: Routing PUBLISH <sip:foo@test.fi> to presence server Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Handling PUBLISH <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Routing locally generated NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> to <sip:jh@test.fi> from <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: presence [notify.c:1575]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:foo@test.fi for event presence Sep 11 17:27:41 siika /usr/sbin/pres-serv[19271]: INFO: Handling in-dialog NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> from <sip:foo@test.fi> to <sip:jh@test.fi> Sep 11 17:27:46 siika /usr/sbin/pres-serv[19265]: INFO: Routing locally generated NOTIFY <sip:cyvrphbz@192.98.103.1:60122;alias=192.98.103.1~47419~2;transport=tcp> to <sip:jh@test.fi> from <sip:jh-buddies@test.fi> Sep 11 17:27:46 siika /usr/sbin/sip-proxy[4482]: INFO: Routing in-dialog NOTIFY <sip:cyvrphbz@192.98.103.1:60122;transport=tcp> from <sip:jh-buddies@test.fi> to <sip:192.98.103.1:47419;transport=tcp> this does not depend on the pua patch nor which db_mode i have in rls module. -- juha

2 2

rls does not generally work
by Juha Heinanen 11 Sep '12

11 Sep '12

it turned out that presence rls module does not work unless pua module is configured to use database mode and database is postgre. the issue is serious, since besides not working, it makes presence/rls server not to accept any new requests. since there is no easy cure, should i go and write a warning in rls readme file so that others don't stumble on this? -- juha

2 6

siptrace and homer
by Krishna Kurapati 11 Sep '12

11 Sep '12

Hi, We have been using homer for some time. The logging is stopping once the disk is full, which make sense. However, it would nice if either kamailio or homer has an option to auto-truncate the siptrace database tables once they grow large. Is there such a feature? Regards. Krishn Kura

2 1

Продукция из Украины
by andrei＠iptel.org 11 Sep '12

11 Sep '12

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1 0

Re: [SR-Users] MTREE reload question
by Daniel-Constantin Mierla 11 Sep '12

11 Sep '12

Hello, the used size is pretty much the same after the reload, so doesn't look as leak. If you do 2-3 reloads is shmem:used_size staying around same value? It is clear boost in the fragments, I would say that after start the number of fragments is quite low, because it should contain the good records in memory as well. Btw, the version 3.3.x has options for defragmentation, see the core cookbook, you can enable it. Cheers, Daniel On 9/9/12 10:18 AM, Uri Shacked wrote: > Hi, > here is the statistics after kamailio starts: > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > here it is after reload of the num table: > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > here is the number of raws from the DB: > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > and attached is the memory log. > thanks, > Uri > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > is this taken only after startup? Get one at startup and another > one after reload, so it can be compared. > > Cheers, > Daniel > > > On 9/6/12 9:22 AM, Uri Shacked wrote: >> Hi, >> here: >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> >> On 9/5/12 3:06 PM, Uri Shacked wrote: >> >> Hi, >> I use MTREE to load 5 million rows from the database. it >> takes about 30 sec to start kamailio and it is running great. >> Whern I check the shmem usage I see the data take about >> 0.8G out of 4G i set on shmem. >> When i reload the data while kamailio is running, the >> memory usage rises to 1.3G and stays there (the second >> and next reloads stays on 1.3G as well). >> Why doesn't it return to 0.8G after the reload is completed? >> >> >> can you send the statistics related to shmem? >> >> kamctl fifo get_statistics shmem: >> >> There are different values there, some usage come from >> overhead of memory chunks management. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda >> <http://twitter.com/#%21/miconda> - >> http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >> http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> >> > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 8

Memory allocation
by David 11 Sep '12

11 Sep '12

Hello, I originally asked this on IRC, but realized that this is probably better suited for the mailing list : If I set memory usage to 100mb ( -m 100 ) and set 100 children ( children=100 in kamailio.cfg ). Will Kamailio use 100 mb or 10, 000 mb ( 10 gb ) of memory ? Is the memory limit per child or global for kamailio ? Is there an MI command I can send to check what kamailio's health ? ( load, free memory, etc... ) ? Thanks, David

2 3

Rtpproxy re-INVITE handling
by Spencer Thomason 11 Sep '12

11 Sep '12

Hello, I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script. Everything works great until the far end (on a public ip) sends a t.38 re-INVITE. The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail. Any help handling these re-INVITEs would be greatly appreciated. I'm happy to post traces if that helps describe the situation. The network topology looks like this: NATed UAC -> Kamailio on a public IP -> UAS on a public IP Thanks in advance, Spencer Connected by DROID on Verizon Wireless

2 12

pua 'Could not convert temporary dialog into a dialog' error
by Juha Heinanen 10 Sep '12

10 Sep '12

when testing presence/rls, i'm getting a pua error when presence server handles subscribe: Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Handling SUBSCRIBE <sip:test@test.fi> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Routing locally generated NOTIFY to <sip:rls@127.0.0.1:5082;transport=tcp> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: presence [notify.c:1581]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:test@test.fi for event presence Sep 8 20:36:12 siika /usr/sbin/pres-serv[22931]: INFO: <core> [tcp_main.c:2787]: quick connect for 0x7f7114cfa490 Sep 8 20:36:12 siika /usr/sbin/pres-serv[22933]: ERROR: pua [send_subscribe.c:689]: Could not convert temporary dialog into a dialog what does pua error message 'Could not convert temporary dialog into a dialog' mean? when does it happen? -- juha

2 9

Example required for XMLRPC module.
by SamyGo 10 Sep '12

10 Sep '12

Hello, I recently posted email relating to using MI_XMLRPC for kamailio monitoring. As Daniel suggested and as mentioned in module docs Mi-xmlrpc is getting obsolete and don't compile on newer OSs. So I need to use XMLRPC module for pulling statistics from kamailio like get_statistics all: Please can anyone point me to some help other than module doc. I need a simple example script to pull data from kamailio and print it. Any help or pointers will be greatly appreciated. Thanks, Sammy G.

1 0

event_route[dialog:failed] and Warning "script writer didn't realese transaction"
by Uri Shacked 10 Sep '12

10 Sep '12

Hi, I started using testing the event_route[dialog:failed]. On my scenario, I receive an INVITE, use t_newtran(), check some stuff and send a failure reply like 3XX or 4XX. I use send_reply() and can see that the TM sends the reply. Now, if I use event_route[dialog:failed], I keep on getting the warning "script writer didn't release transaction". If I delete the event_route[dialog:failed] from the script, the warning does not appear. My solution was to work stateless, which is OK for this scenario. But, still, What does this event_route has that shows this warning? Why doesn't the transaction being released? (I tried to use t_reply as well and still the same). Thanks, Uri

2 1

Re: [SR-Users] MTREE reload question
by Uri Shacked 09 Sep '12

09 Sep '12

> > Hi, > > here is the statistics after kamailio starts: > > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > > here it is after reload of the num table: > > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > here is the number of raws from the DB: > > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > > i have the memory log in a file if needed > > thanks, > Uri > > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> is this taken only after startup? Get one at startup and another one >> after reload, so it can be compared. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 9:22 AM, Uri Shacked wrote: >> >> Hi, >> >> here: >> >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 9/5/12 3:06 PM, Uri Shacked wrote: >>> >>>> Hi, >>>> I use MTREE to load 5 million rows from the database. it takes about 30 >>>> sec to start kamailio and it is running great. >>>> Whern I check the shmem usage I see the data take about 0.8G out of 4G >>>> i set on shmem. >>>> When i reload the data while kamailio is running, the memory usage >>>> rises to 1.3G and stays there (the second and next reloads stays on 1.3G as >>>> well). >>>> Why doesn't it return to 0.8G after the reload is completed? >>>> >>> >>> can you send the statistics related to shmem? >>> >>> kamctl fifo get_statistics shmem: >>> >>> There are different values there, some usage come from overhead of >>> memory chunks management. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.com >>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >>> http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >>> http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> > >

1 0

Possible bug in authentication
by martian＠centrum.sk 07 Sep '12

07 Sep '12

Hello to everybody. I am currently working with Kamailio 3.3.1 on RedHat. The "loose_route" function was not working correctly and I observed some very strange behaviour (not as one described in the documentation of the function). I have found that there needs to be a port included in the "alias" variable for the loose_route function to work correctly. However, upon adding the port to alias, the INVITE messages were no longer authenticated (Kamailio just accepted them and didn't send proxy-auth header in 407 message). My alias: alias="domain.ch:5060" Examining default routing logic, I found the problem here: if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ... } The "from_uri==myself" was no longer evaluated as true, because there was a port at the end of the alias. The FROM Header of the INVITE messages looks like: From: "acc1" ;tag=12345 ..so .. no port number there. Btw, I have fixed this with replacing the "myself" list with my own defined variable MY_DOMAIN. #!define MY_DOMAIN ".*(a)domain.ch" So now the condition looks like this: if (is_method("REGISTER") || from_uri=~MY_DOMAIN) { ... } I am not sure if this is a bug that needs to be fixed or not. I am just pointing my finger at it and I hope it will contribute to the development. Also, a valid description of this behavior (when using port in alias) would be appreciated. Cheers, Martin

3 8

Wrong index on expires column in pua module
by Juha Heinanen 07 Sep '12

07 Sep '12

looks like unique constraint on expires_idx has not been removed yet, because i still see Duplicate entry for key 'expires_idx' errors. i'll go and make the changes. -- juha

1 0

FW: Kamailio as sipRedirect
by Fatima Chahrour~Vanrise Support 07 Sep '12

07 Sep '12

Dears, I have an installed 3CX where I want to generate a call from it to the Kamailo to get a list of suppliers to terminate the call to. Thus, the Kamailio should act as a sip redirect to the 3CX (for routing decision). Have you tried such configuration before? And how do I configure the 3CX on the Kamailio? Regards, F Charour

3 12

kamctlrc SQL definitions and db/schema/entities.xml
by Vieri 07 Sep '12

07 Sep '12

Hi, What is the purpose of the "SQL definitions" settings found in kamctlrc? It is also stated that one should edit db/schema/entities.xml but I can't seem to find this file. kamctlrc contains: # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2020-05-28 21:32:15" # DEFAULT_ALIASES_EXPIRES=$FOREVER # DEFAULT_Q="1.0" # DEFAULT_CALLID="Default-Call-ID" # DEFAULT_CSEQ="13" # DEFAULT_LOCATION_EXPIRES=$FOREVER Thanks, Vieri

2 5

siptrace module with Kamailio 3.3.0 and Siremis 3.2
by Chris Telford 07 Sep '12

07 Sep '12

Hi, Were attempting to get siptrace working on Kamailio 3.3.0 with Siremis 3.2... We've enabled the module, and configured the basic parameters, however were having problems with where exactly in Kamailio's config to call siptrace(). We currently have it at the very end of FAILURE_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE as recommended at the end, and it is logging traces sucessfully. The problem were having, is it is only producing traces with inbound sip messages only. It is also duplicating almost every message (so were clearly call siptrace() too many times per request). Is there a guide on this? or any sample config files to help us speed this process up a little? Regards, Chris Telford

1 0

Re: [SR-Users] sr-users Digest, Vol 88, Issue 19
by Vijay Thakur 07 Sep '12

07 Sep '12

On Friday 07 September 2012 02:34 PM, 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. Re: Monitoring Kamailio using mi_xmlrpc good or bad ? > (Daniel-Constantin Mierla) > 2. Re: Video Call Quality (Tayeb Meftah) > 3. Re: FW: Kamailio as sipRedirect (Fatima Chahrour~Vanrise Support) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 07 Sep 2012 10:36:50 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Monitoring Kamailio using mi_xmlrpc good or > bad ? > To: SamyGo <govoiper(a)gmail.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5049B222.6080405(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 9/7/12 10:21 AM, SamyGo wrote: >> Thanks alot for your time and valuable feedback on the topic. I just >> wanted to be sure that when if comes to collecting stats like 150+ values > did you defined custom stats? IIRC, we don't have so many in the code. > >> from Kamailio per 30 seconds will it impact on the actual performance >> of Kamailio as sip server. I will definitely look for XMLRPC only as >> well if thats scalable. > Stats are just counters, long values, access to them should be very > fast. mi_xmlrpc is a dedicated process, not handling sip at all. > > Cheers, > Daniel >> Thanks >> Sammy >> >> >> >> On Fri, Sep 7, 2012 at 12:35 PM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> if you fetch the statistics, should be ok. >> >> xmlrpc module should be more scalable that mi_xmlrpc, but uses a >> different command format and output. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 3:24 PM, SamyGo wrote: >>> Hello, >>> >>> I've just finished monitoring all the core modules of kamailio >>> via XML-RPC. I use a script placed on kamailio server which sends >>> command to Kamailio over MI_XMLRPC and fetch details of each >>> requested module. The results fetched are then sent back to the >>> monitoring tool for graphing using SNMP OIDs. >>> >>> *My question* here is that if I query kamailio like this for like >>> 7~10 module over xmlrpc per 30/45 seconds how much overhead will >>> it cause to the Kamailio server !? Is this a good approach or bad ? >>> >>> >>> Thanks, >>> Sammy >>> >>> >>> >>> _______________________________________________ >>> 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 >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu >> >> Thanks, Tayeb for your reply. What can be checked at client end. I am using Kamailio 3.2. With thanks in advance. Vijay Thakur

1 0

Video Call Quality
by Vijay Thakur 07 Sep '12

07 Sep '12

Dear All Experts, Recently, i have configure my first kamailio server. After some technical hurdles, it is working very fine. There is a issue related to video call. When i make a video call from jitsi to jitsi, iphone to iphone or android, the image is blur, choppy. After googling for this issue, i have found no satisfactory answer. Kindly suggest me the reasons for this issue and what action can be taken. Thanks in advance , Vijay Tha

2 1

Monitoring Kamailio using mi_xmlrpc good or bad ?
by SamyGo 07 Sep '12

07 Sep '12

Hello, I've just finished monitoring all the core modules of kamailio via XML-RPC. I use a script placed on kamailio server which sends command to Kamailio over MI_XMLRPC and fetch details of each requested module. The results fetched are then sent back to the monitoring tool for graphing using SNMP OIDs. *My question* here is that if I query kamailio like this for like 7~10 module over xmlrpc per 30/45 seconds how much overhead will it cause to the Kamailio server !? Is this a good approach or bad ? Thanks, Sammy

2 3

[ ... ] operator
by Vitaliy Aleksandrov 07 Sep '12

07 Sep '12

"Migrating kamailio v1.5.x to v.3.0.0" says that "test" operator no longer exists. But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone. Can somebody remove it from the documentation ?

2 3

MTREE reload question
by Uri Shacked 07 Sep '12

07 Sep '12

Hi, I use MTREE to load 5 million rows from the database. it takes about 30 sec to start kamailio and it is running great. Whern I check the shmem usage I see the data take about 0.8G out of 4G i set on shmem. When i reload the data while kamailio is running, the memory usage rises to 1.3G and stays there (the second and next reloads stays on 1.3G as well). Why doesn't it return to 0.8G after the reload is completed? BR, Uri

2 3

User accounting
by Ali Jawad 06 Sep '12

06 Sep '12

Hi Is it possible to store user logon/logout dates + user agent used. I know that the info is in the location table but it is not persistent, and I am using the accounting table to store calls. But I am not sure there is a way to store logon/logout/user agent. Any input please ? Regards

2 8

Siremis v3.3.0 Released
by Elena-Ramona Modroiu 06 Sep '12

06 Sep '12

Hi, Siremis v3.3.0 was released - the web management interface compatible with Kamailio v3.3.x. This version is an update of v3.2.1 to be able to work with the database schema of Kamailio v3.3.x. The news of the release, including links to installation and screenshots, is available at: - http://siremis.asipto.com/2012/09/06/siremis-v3-3-0-released/ Regards, Ramona

1 0

Meeting in London
by Carsten Bock 06 Sep '12

06 Sep '12

Hi, anyone from the Kamailio-Community in for a beer tomorrow evening in London? I have several meetings during the day (last one at 5PM) and i'll be travelling back to Germany on Saturday, so i'm free for some chat about Kamailio, SIP, IMS or IPv6 during the evening... :-) If yes, just drop me a mail... Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

Информация для бухгалтера
by jan＠iptel.org 06 Sep '12

06 Sep '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: òÁÂÏÔÁÅÔÅ ÂÕÈÇÁÌÔÅÒÏÍ? ðÏÍÎÉÔÅ, ËÁË ÓÌÏÖÎÏ ×ÁÍ ÒÁÎØÛÅ ÂÙÌÏ ÎÁÊÔÉ ÏÂÕÞÅÎÉÅ ÐÏ ÎÕÖÎÏÊ ÔÅÍÅ ôÅÐÅÒØ, ÎÁ ÓÁÊÔÅ ÎÁÛÅÇÏ ÕÞÅÂÎÏÇÏ ÃÅÎÔÒÁ ÎÁ ÜÔÏ ÕÊÄÅÔ ×ÓÅÇÏ ÎÅÓËÏÌØËÏ ÍÉÎÕÔ! é ÎÅ ×ÁÖÎÏ, × ÞÅÍ ×ÁÍ ÎÕÖÎÏ ÒÁÚÏÂÒÁÔØÓÑ - × îäó, ÒÁÓÞÅÔÅ ÏÔÐÕÓËÏ×, ÏÐÌÁÔÅ ÔÒÕÄÁ, ÎÁÌÏÇÏÏÂÌÏÖÅÎÉÉ, ÓÔÒÁÈÏ×ÙÈ ×ÚÎÏÓÁÈ, ÐÏÄÇÏÔÏ×ÉÔØÓÑ Ë ×ÏÚÍÏÖÎÙÍ ÐÒÏ×ÅÒËÁÍ ÉÌÉ ÒÁÚÏÂÒÁÔØÓÑ Ó ÄÏËÕÍÅÎÔÏÏÂÏÒÏÔÏÍ. úÁÈÏÄÉÔÅ ÎÁ ÎÁÛ ÓÁÊÔ - ÕÞÉÔØÓÑ ÓÔÁÌÏ ÇÏÒÁÚÄÏ ÐÒÏÝÅ. é ÍÙ ×ÓÅÇÄÁ ÐÏÍÏÖÅÍ, ÒÁÓÓËÁÖÅÍ, ÏÂßÑÓÎÉÍ É ÐÏÄÂÅÒ£Í ÔÏ, ÞÔÏ ×ÁÍ ÐÏÄÏÊÄ£Ô! [...] Content analysis details: (9.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 0.9981] -0.0 SPF_PASS SPF: sender matches SPF record 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [2.146.222.10 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [2.146.222.10 listed in bb.barracudacentral.org] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

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Problem with dumping offline messages
by Stoyan Mihaylov 06 Sep '12

06 Sep '12

Following examples, I tried to add processing of offline messages. Structure of my system is: SIP client <->Kamailio<->Asterisk<->Kamailio<->SIP client Where Kamailio in above is same server, but SIP clients are different. Now I have no problems with audio, video calls and messages, if both clients are online. All messages received from client are sent to Asterisk, where I can or not process them and then I forward back to Kamailio. When messages came from Asterisk server, then they are sent directly to end user. Sending messages to end user - I call this function, only if message come from allow_source_address(): route[MSILO_MESSAGE]{ if (lookup("location")){ xlog(" Relying"); t_relay(); }else{ xlog("Storing MSILO_MESSAGE $ru-$tu-$ou"); m_store("$ru"); t_relay(); t_reply("202", "Accepted"); }; exit(); } When client register: 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",0x04)) sl_reply_error(); # MSILO - dumping user's offline messages if (m_dump("$fu")){ xlog("MSILO: offline messages for $fu dumped - if they were \n"); }else{ xlog("MSILO: no offline messages dumped\n"); }; exit; } } I am sending message to offline contact, and I see record in silo table. Record looks perfect. Then offline contact become online, I see in debug MSILO: offline messages for {offlined contact} dumped - if they were But client do not receive nothing. In case all contacts are online, then message go to contact, and is not stored in silo. Setup of msilio is: loadmodule "msilo.so" modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MY DOMAIN") modparam("msilo","contact_hdr","Contact: registrar@MYDOMAIN:5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","***$rU***$rb***") By the way - I receive in Asterisk offline message. Obviously, I am doing something wrong, but I cant understand what. Any help is welcome. Version of Kamailio is: #~> kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 14:43:47 Jul 19 2012 with gcc 4.5.2

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frequency of dispatcher probing
by Juha Heinanen 05 Sep '12

05 Sep '12

i have defined: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 3) modparam("dispatcher", "ds_probing_mode", 2) and still i see that dispatcher is pinging inactive node at least once per second: U 2012/09/05 13:47:04.143977 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:04.644185 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:05.143991 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.143975 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.144056 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... what is that i'm missing in order to make ping interval 10 seconds? -- juha

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Re: [SR-Users] memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

> The 20% initialy loaded is because I load many data with MTREE, > HTABLE,DIALPLAN etc....(5 million rows with MTREE..) > Regarding the PKG, how do i set it? I read a few months ago that the > shared and private memory on version 3.2.2 are the same. > Thanks for the help :-) > > On 09/05/2012 11:44 AM, Uri Shacked wrote: > >* Hi, *>* Thanks, *>* How do i find memory leaks? *>* when my > application starts the it shows the following values: *>* shmem:fragments > = 143898 *>* shmem:freesize = 3446570952 *>* shmem:max_used_size = > 861854768 *>* shmem:real_used_size = 848396344 *>* shmem:total_size = > 4294967296 *>* shmem:used_size = 319676976 *>* I divided the > "real_used_size" with the "total_size" and notice that *>* it is 20%. So, > this is my initial value of the application. *>* After 2 months with > 40,000 calls per day, the relations between these *>* values is 39%. > (doing a reload of the data that occupies 20% failed *>* due to "no > memory") *>* So, 19% occupation of memory is normal for this amount of > calls? or is *>* it a memory leak? if it is a memeory leak, how do i see > where? *Hello, > > Too little information to go on here. A reload might fail mostly because > data from the DB (e.g. with carrierroute case) is first loaded into > private memory (the sql result). Depending of the module there are other > tricks you could do (decrease the number of fetch rows the query returns > etc.). The out of memory condition affects shared or private mem. Try to > use at least 15 MB for PKG mem. > > If the 39% is with traffic, then it's normal, if you don't have any > traffic I still would not worry. The thing that worries me a little is > the value, you are using from the start around 0.8 GB of used shared > memory... this is a lot. From the percentages you provided it looks ok.... > > Marius > >* Uri *>* On 09/05/2012 10:19 AM, Uri Shacked wrote: *>* >/Hi, />/When I > start kamailio it loads some data into memory. />/Now *>* calls are > starting to gather up. How can I check how mach memory *>* />/does my > avp's take? />/BR, />/Uri /Hello Uri, *>* *>* avp's are associated with > the transactions, so they free'd when the *>* transaction stops(and calls > are composed of transactions). This is true *>* for dialog information as > well (dlg_var's etc.). Continuous growth in *>* memory at a constant call > rate (incoming sip requests) might mean a *>* leak. If you need > statistics about the total memory usage you can use *>* `kamctl fifo > get_statistics all` and check for the shmem real_used_size *>* and > max_used_size fields. *>* *>* Cheers, *>* Marius *>* *>* -- *>* Zbihlei > Marius *>* *>* Head of *>* Linux Development Services Romania *>* * > > -- > Zbihlei Marius > > Head of > Linux Development Services Romania > > 1&1 Internet Development srl Tel KA: 754-9152 > Str Mircea Eliade 18 Tel RO: +40-31-223-9152 > Sect 1, Bucuresti mailto: marius.zbihlei at 1and1.ro<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > 71295, Romania > >

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memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, Thanks, How do i find memory leaks? when my application starts the it shows the following values: shmem:fragments = 143898 shmem:freesize = 3446570952 shmem:max_used_size = 861854768 shmem:real_used_size = 848396344 shmem:total_size = 4294967296 shmem:used_size = 319676976 I divided the "real_used_size" with the "total_size" and notice that it is 20%. So, this is my initial value of the application. After 2 months with 40,000 calls per day, the relations between these values is 39%. (doing a reload of the data that occupies 20% failed due to "no memory") So, 19% occupation of memory is normal for this amount of calls? or is it a memory leak? if it is a memeory leak, how do i see where? Uri On 09/05/2012 10:19 AM, Uri Shacked wrote: >* Hi, *>* When I start kamailio it loads some data into memory. *>* Now calls are starting to gather up. How can I check how mach memory *>* does my avp's take? *>* BR, *>* Uri *Hello Uri, avp's are associated with the transactions, so they free'd when the transaction stops(and calls are composed of transactions). This is true for dialog information as well (dlg_var's etc.). Continuous growth in memory at a constant call rate (incoming sip requests) might mean a leak. If you need statistics about the total memory usage you can use `kamctl fifo get_statistics all` and check for the shmem real_used_size and max_used_size fields. Cheers, Marius -- Zbihlei Marius Head of Linux Development Services Romania

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