- sr-users - kamailio.org

username of Contact header
by Al S 27 May '16

27 May '16

Hi, I am trying to read username portion of Contact header: My Contact header content has a URI value such as: sip:8131110000@10.10.10.10.:5060 and I am trying to read : 8131110000 I tried the following perl similar RE and it didn't work: ($var(main_number2)) = $ct =~ /sip:(.*)@/; Thanks,AS

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4.4: Kamailio crashes at start - sometimes
by Sebastian Damm 27 May '16

27 May '16

Hi, we have a machine running 16 Kamailio instances, and while upgrading to 4.4.1 (from 4.3.5), 8 of them wouldn't start. When downgrading to 4.3.5, they all start again. All of them have pretty identical configuration files, except IPs, ports and some code enabled or disabled via defines. After comparing a working and non-working configuration and adjusting setting by setting, we finally ended up with a working configuration. The difference is, that it won't start when a part of the code DOES NOT get included. If it gets included, it will start. This is the mentioned part of our main route: #!ifdef ENABLE_INV_RATELIMIT # Check for INVITE limit if (is_method("INVITE") && $au == $null && !($ua =~ "sipgate") ) { $var(invcount) = $shtcn(invcount=>%~$fU); xlog("L_INFO", "INVITE Requests from $fU in last 30 seconds: $var(invcount)\n"); if ($var(invcount) < 12) { $var(uniqcid) = $ci + $Ts + $ft; $var(tkey) = $fU + '-' + $(var(uniqcid){s.md5}{s.substr,0,10}); $sht(invcount=>$(var(tkey))) = 1; $var(uniqcid) = $null; $var(tkey) = $null; } if ($var(invcount) > 10) { if ($var(invcount) == 11 ) { xlog("L_NOTICE", "User $fU ($var(domain2use)) over ratelimit for new calls, rejecting.\n"); } # Enable this only after evaluating the impact! append_to_reply("Retry-After: 30\r\n"); sl_send_reply("503", "Call Rate Limit Exceeded"); exit; } } #!endif If we put this line at the top of the configuration file, everything works: #!define ENABLE_INV_RATELIMIT If we delete this line, startup does not work. It just sits in ps for one minute without forking, and then gets terminated. We enabled a bit of debugging, and this is apparently the error causing Kamailio to shutdown: May 25 14:50:15 kammel /usr/sbin/kamailio[24989]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 53: nathelper May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0x9f0428 and 0xa34428 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9f0240, 82, 1, (nil)), fd_no=0 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Bad file descriptor [9] May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1747]: tcp_receive_loop(): failed to add tcp main socket to the fd list May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1815]: tcp_receive_loop(): exiting... I have no idea, how this part of the code could lead to this error, but it is reproducable, that at least on this system setting or disabling this define fixes or breaks the startup. Does anybody have an idea, what's happening there? Best Regards, Sebastian

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Multihomed kamailio - one IP for each account
by Attila Megyeri 27 May '16

27 May '16

Hi Folks, I am looking for a solution to do the following, basically an SBC-like solution. I have an internal asterisk system that should make outbound calls towards a provider, but that provider authenticates the customer based on IP address. And I have many such customers. The provider is not willing to make changes to his practice... Asterisk is bound to a single IP, so there is not much I can do. I was thinking about creating several outbound accounts for asterisk, and then a multi-homed Kamailio would do the following: If request comes from user1, Kamailio would send the request to the provider from IP1. If request comes from user2, Kamailio would send the request to the provider from IP2. And so on, for as many IPs as many customers I want to handle. Is there a more elegant way to do this? If not, how should I implement this? Thanks in advance, Attila

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Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is not so much due to the number of entered Via: records but, by the fact that they contain a branch parameter extremely long. Here the call flow example from CARRIER A —> Kamailio1 —> Kamailio2 —> CARRIER B Extract from SIP message sent from CARRIER A to K1 INVITE sip:999912341234@192.168.158.42 SIP/2.0. Via: SIP/2.0/UDP 192.168.220.141:5060;branch=z9hG4bK49e4301c;rport. Extract from SIP message sent from K1 to K2 INVITE sip:999912341234@192.168.127.244 SIP/2.0. Record-Route: <sip:192.168.158.42;lr;did=194.a0f1;nat=yes>. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bK0635.7c1e97c2d5c01ea98e8d0e7fa23a3822.0. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.cuOBV4OBVLMGZfWxF-W.y6Mx1LgRWIC9gIgx4fzxj7MBP7MBVAOBF4M.1jmxuqC93X3heroEWvH9vsCFN43qd4zRj4Mlyf3l1LNSQLpx4uMx3A. Extract from SIP message sent from K2 to CARRIER B INVITE sip:999912341234@192.168.65.248 SIP/2.0. Record-Route: <sip:192.168.127.244;lr;did=194.5122;nat=yes>. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bK0635.f843472880b380acde3a33732975fde9.0. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1LNSQLpx4uMx3AzuaVHRaYpB1JNEt736cQkBIvalaJmly6Mlj7W6Mfg.qw3leqWRMAMRKrz.rIzSPAg.pE3.Vl3.MZMBV7My**. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1JNEt736cQkBIvalaJmwWLORv4mK0Hot3w.jpam6t4kRWWOEWu.4eRWFQGKqfWauYEKwrSOKN7k.gWkxtMgue9mjMsg4IhkxaEkhrbW4uGjxpWPIg0.4eSWF47MRY1kUKfPlqlRxWvg9e5aKp6mxI6N4WS.BPlNRP4kIebWFudkR3loxtM.IWa.U0ZWUKWkxW0. Is there anyone here with the idea how to solve this problem ? Best regards

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rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp udp flow between the phones isn't routed properly by the rtpproxies, ending in the communication drop (all the SIP nego is working well, and the SDP are correctly patched with the rtp proxy address and port). Analyzing the RTP proxy packets, we have found that the Kamailio registrar gives the Kamailio proxy ip address in the RTP proxy create session command, but keeps the original sdp port. command looks like this: Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet> PZU5OITCW;1 We are using rtpproxy_manage() without any flags. It seems to us that this ip and port are used as default forward route as long as the callee hasnt connected to the rtpproxy. *Is it correct? * If its true, Its seems to us that this cant work as we are mixing the proxy sip address with a udp port open on the phone ? *Is our analysis correct ?* Can we use some option in rtpproxy_manage to replace the proxy ip by the phone ip as seen in the via route ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.h… Sent from the Users mailing list archive at Nabble.com.

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Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC>

2 2

webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen.

2 3

SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/sdp Content-Disposition: session Content-Length: 1393 v=0 o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP> s=SBC c=IN IP4 <MyPublicIP> t=0 0 m=audio 30454 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30455 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:iglHZqry a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host m=audio 30484 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30485 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:gM1viWqu a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host My setup : incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy (Kamailio + Rtpengine) -> webRTC clients Is there anybody who could help me with this issue or explain me how to remove the second line of m=audio RTP/SAVPF ? -- Jan

1 0

Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

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nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/rtpengine_conf" loadmodule "db_text.so" modparam("rtpengine", "db_url", DBURLRTP) modparam("rtpengine", "table_name", "rtpproxy") and the database is like root@test:/var/log/rtpengine_conf# ls rtpproxy version root@test:/var/log/rtpengine_conf# cat version table_name(str) table_version(int) rtpproxy:1 root@test:/var/log/rtpengine_conf# cat rtpproxy set_id(int) url(str) weight(int) disable(int) 1:udp\:x.x.x.x\:x:100:0 2:udp\:x.x.x.x\:x:100:0 is there something i missed here? do i must use mysql or postgres drivers for this? (i use a test machine to old to install a driver of those with kam 4.4 maybe?) best regards david

1 1

doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i have a quick question about using module ratelimit on kamailio 4.2 > or 4.4. > on tests made before running it on production, i set the RED > algorithm with a limit of 10 calls per second, but i saw the call > limitation was not done until we got more than 20 calls per second or > so on the queue. we didnt get any INVITE discarded. > after surpassing around 20 INV/sec, it's true the algorithm starts to > work fine and only 10INV/sec are passed more or less > > on production i have no easy way to check this behaviour, besides it's > a feature barely used. > > so i only want to know if this behaviour is known, or if i'm missing > soemthing > > my config is like this > > modparam("ratelimit", "timer_interval", 5) > modparam("ratelimit", "queue", "4:INVITE") > modparam("ratelimit", "pipe", "4:RED:10") > > route[RATE_LIMIT] { > $var(limitation)="4"; > if (!rl_check("$var(limitation)")) { > send_reply("505","Limiting"); > xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, > rejecting with 505\n"); > route(CLEAR); > exit; > } > } > > > thanks a lot and regards > david escartin > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - | CB1 |-----------------| SIPPHONE | ---------- | -------------- | -------- ---------------- | | ---------- | | -------- | SBC2 | - - - - - | | - - - - - | CB2 | ---------- | | -------- | -------------- | ---------- | - - - -| kamailio2 |- - - - -| -------- | SBCn |- - - - - | -------------- | - - - - - | CBn | ---------- | | -------- Kamailio high availabality will be done with loopback address and BGP in two datacenters. SIP carriers will be connected to SBC and customer to one CB. I have made some tests with kamailio, and i use LCR to route call from CB to SBC, using wheigt and prorities to send call to carriers. Then, when a call arrived from carriers, i use DNS NATPER lookup to find the right CB box. CB HA is done using DNS SRV. So user is registred at one location, and only one. No location sharing beetwen boxes. I have some question concerning my architecture, and technologies used to routing call. Is LCR is the best option for routing call between multiple carriers ? DNS NAPTER is a modern solution to find an endpoint on this architecture ? Existing a better way to route call with kamailio, for exemple rt-json and curl ? In my point of view, kamailio should be a routing engine, using internal technologies, to maximise performances. Less work with kamailio and external app, best performances in sip routing. Is it the good vision ? Thanks in advance.

2 4

Kamailio routing to different carriers based on prefix
by José Seabra 25 May '16

25 May '16

Hi there, I'm looking for a solution to route calls based on prefix, my routing table might reach a long list of prefixs. I looked to carrierroute and it seems to be nice for this purpose but i have some doubts to how to handle correctly with carrierroute. IfI understood correctly, if I set the parameter avoid_failed_destinations to 1, can i use the same carrier id and domain on failure route to select another destination that matches the prefix? Can I have the same carrier ID with diferent domains sharing the same prefixs but mapping it to different destinations? Thank you for your help. Best regards -- José Seabra

2 1

call count / and work with despatcher
by jaflong jaflong 24 May '16

24 May '16

Hi Everyone, What is a accurate way to get current call count. I am only relaying calls. Initialy I tried the code below. It seemed to work, it reported the correct number of calls I had dialed. I get the result through kamcmd htable get command. ------------------------------------------ under route[RELAY] # # COUNT # if (is_method("BYE")) { $var(n) = $shtdec(callcount=>active); sht_rm_name_re("callcount=>$ft"); } -------------------------------------------- # Manage incoming replies onreply_route[MANAGE_REPLY] { # # COUNT # if (status=="200") { if (is_method("INVITE")) { # On start if($sht(callcount=>active) == $null) { $sht(callcount=>active) = 0; } # New call if ($sht(callcount=>$ft::from_tag)== $null) { $sht(callcount=>$ft::from_tag) = $ft; $var(n) = $shtinc(callcount=>active); } } xlog("L_INFO", "This is a 200 OK reply\n"); } xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } ------------------------------------------------------- However I wanted to distribute calls to several servers I further added the despatcher code below # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "-----RELAYING going to <$ru> via <$du> time [$Tf] -----\n"); This makes my counting code above give incorrect call count. How can I make call count work with dispatcher ? Do advise how I can do this Regards

2 1

Create "Contac Groups" on the server Kamailio
by ALFONSO MARTINEZ GOMEZ 24 May '16

24 May '16

Hi, I'm working on a VoIP communication application with Kamailio as SIP server and using the SIREMIS interface. As an administrator, I can add "Subscribers" which later will be registered on the server so that they can make the necessary negotiations when they want to make a call. My question is, how you can make groups of "contacts" so that such calls can be made only with each other "contacts" in the same group? that is, we can have many "subscribers" on the server and belong to two different groups and that they can only talk to each other contact each group. Thank you so much and greetings!

1 0

doubt using rate limit
by david escartin 24 May '16

24 May '16

Hello all i have a quick question about using module ratelimit on kamailio 4.2 or 4.4. on tests made before running it on production, i set the RED algorithm with a limit of 10 calls per second, but i saw the call limitation was not done until we got more than 20 calls per second or so on the queue. we didnt get any INVITE discarded. after surpassing around 20 INV/sec, it's true the algorithm starts to work fine and only 10INV/sec are passed more or less on production i have no easy way to check this behaviour, besides it's a feature barely used. so i only want to know if this behaviour is known, or if i'm missing soemthing my config is like this modparam("ratelimit", "timer_interval", 5) modparam("ratelimit", "queue", "4:INVITE") modparam("ratelimit", "pipe", "4:RED:10") route[RATE_LIMIT] { $var(limitation)="4"; if (!rl_check("$var(limitation)")) { send_reply("505","Limiting"); xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, rejecting with 505\n"); route(CLEAR); exit; } } thanks a lot and regards david escartin

2 1

Please help me setting up Kamailio for android application
by Kunal Dhanda 24 May '16

24 May '16

I wanted to create my own voip application using kamailio . I am not able to find any rest integration with kamailio for my server to use for user registration and other task.Please help in developing my appplication. -- Thanks & Regards, Kunal Dhanda, Software Developer E-mail : < "kunaliitkgp09(a)gmail.com" >, Ph. No. (M) : +91 - 8884818805

2 1

Found bug at ACC
by Pavel Eremin 24 May '16

24 May '16

Hi all, when we set syslog facility for ACC to LOCAL3, log_write_cdr() will write at LOG_DAEMON anyway. You have to recompile module acc to user another default syslog facility to save cdr with another syslog facility.

2 1

kamctl stats shmem error changing shared memory
by Mr.Gab 23 May '16

23 May '16

Hi, i have a problem with my kamailio installation. Actually i set the shared memory to 128MB and it's all fine. When i increase the shared memory to 256Mb and more i get the following error while checking kamctl commands (like kamctl stats shmem) ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. If i revert to 128Mb kamailio is able to create the kamailio_fifo file and all works fine. Can somebody help me to solve? Thanks

3 3

Re: [SR-Users] Change CANCEL reason when fr_timer expires
by Bruno Emer 22 May '16

22 May '16

Hello, Alex! Fisrt of all, thanks for you help with this problem! I don't think that this will help me. I can see that this parameter is enabled by default and in fact it is working in some cases. One case where this works is when the INVITE is sended for more than one endpoint (when the same user is registered in many devices). When someone answers the INVITE, Kamailio sends a CANCEL message to the other endpoints and it has the Reason header showing that the call was answered elsewhere. I want to add the CANCEL Reason header when the timer expires, so kamilio will not receive a final replay to that transaction. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-20 15:42 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a situation here where when I send a INVITE to some specific > user-agents in my environment I must change the fr_timer to a value lower > than the default. > > I can acomplish this using the "t_set_fr" commands in my cfg file, and it > works like a charm. > > The point is that when the timer reaches its limit, the kamailio server > sends a CANCEL message to the user-agent. I know that this is the normal > behavior, but I want to add the Reason field on it, so the user-agent > application can know that the call wasn't answered on time. > > I was looking the TM module documentation to understand how to do this, > but I couldn't figure out a way to get this working. > > Is there a way to add the Reason in CANCEL messages generated by the > fr_timer? > > > Thanks. > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >

2 1

Change CANCEL reason when fr_timer expires
by Bruno Emer 21 May '16

21 May '16

Hello all! I have a situation here where when I send a INVITE to some specific user-agents in my environment I must change the fr_timer to a value lower than the default. I can acomplish this using the "t_set_fr" commands in my cfg file, and it works like a charm. The point is that when the timer reaches its limit, the kamailio server sends a CANCEL message to the user-agent. I know that this is the normal behavior, but I want to add the Reason field on it, so the user-agent application can know that the call wasn't answered on time. I was looking the TM module documentation to understand how to do this, but I couldn't figure out a way to get this working. Is there a way to add the Reason in CANCEL messages generated by the fr_timer? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 1

extra characters in re-invite "TO" field
by Al S 20 May '16

20 May '16

Hi, Kamailio adds extra characters in the re-invite "TO" field during session timer. There is no header manipulation in our route[WITHINDLG] block. Re-INVITE received by KamailioTo: <sip:8131111111@?.?.?.?;otg=1>;tag=1c1048632956 Re-INVITE forwarded by KamailioTo: <sip:8132222222@?.?.?.?;T\033\023Z\f1>;tag=1c1048632956 Your help would be greatly appreciated. Thanks,AS

2 2

Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to make such setup that calls without the RTPENGINE are impossible.

1 0

WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

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Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 pejman(a)italkip.com

2 1

Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

2 3

[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne

4 3

Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 <mailto:pejman@italkip.com> pejman(a)italkip.com

1 0

invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

2 2

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> >

1 0

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config file, same results. Thanks in advance. Regards.

2 3

MySQL database for kamailio as S-CSCF
by Christoph Valentin 17 May '16

17 May '16

Hi Community, I'm doing very first steps to use kamailio as an S-CSCF. Now I'm confused about some defines in the examples/scscf/scscf.cfg file. #!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" Tutorials/ims/Installation-howto tell me only I-CSCF needs a database, I do not find anything about database for S-CSCF. However, when I comment out the defines ##!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" then the kamctl start results in a segmentation fault. I have installed kamailio-4.4.0 from RPMs. Is there any hint about database for S-CSCF? Thanks in advance Christoph

1 1

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by M R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Debian, dnssec.so, and missing dnsval.conf
by David Holl 17 May '16

17 May '16

In the interest of sharing and promoting DNSSEC, I was recently stumped on a problem where I noticed I could not call sip:test.echo@sip5060.net without disabling the dnssec.so module in my kamailio.cfg. The last time I had tried the sip5060 echo test was several months ago, and in the intervening time, I had applied the numerous debian package updates. Thus, I could not isolate the cause of the problem to one of the many package updates such as kamailio 4.4.x, libval, my own DNSSEC dns server, etc... My initial only clue was kamailio's syslog messages: May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve "sip5060.net" :unresolvable A or AAAA request (-7) May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) To summarize the problem, I found that kamailio's dnssec.so was failing queries to domains that were not DNSSEC protected such as sip5060.net. This behavior is a bug, since sip5060.net is in this case, provably not participating in DNSSEC from the toplevel "net." zone [1]. The fix was to provide a default configuration file to libval, which was somehow not included in Debian's libval15 package. I created the following configuration from the default dnsval.conf in the source distribution: https://dnssec-tools.org/download/dnsval-2.2.tar.gz https://dnssec-tools.org/download/dnsval-2.2.tar.gz.asc as well as their wiki: https://www.dnssec-tools.org/wiki/index.php/Dnsval.conf cat << 'EOF' > /etc/dnssec-tools/dnsval.conf ################################## # Global Options ################################## global-options trust-oob-answers yes # I'm running a DNSSEC nameserver locally, # so 4 KiB queries to localhost should be OK: edns0-size 4096 env-policy disable app-policy disable rec-fallback no ; : trust-anchor . DS 19036 8 2 49AAC11D7B6F6446702E54A1607371607A1A41855200FD2CE1CDDE32F24E8FB5 ; : zone-security-expectation . validate ; : provably-insecure-status . trusted ; : clock-skew . 0 ; EOF In summary: I find it odd that a default dnsval.conf was not included with my system's normal installation of the "libval15:amd64" package, version 2.2-4. After providing dnsval.conf, kamailio's dnssec.so was happy again. Side notes: * To identify the directory for dnsval.conf, run libval-config --dnsval-config On Debian, libval-config is in the libval-dev pacakge, and this package includes a man page for dnsval.conf as well. * To test your libval and dnsval.conf, try dt-validate -s On Debian, dt-validate is in the libval-bin package. - David [1] -- Authenticated denial of existence: https://tools.ietf.org/html/rfc5155 https://tools.ietf.org/html/rfc7129 And for sip5060.net, there's an NSEC3 record in the top level net. zone proving non-existance of sip5060.net DS records: http://dnsviz.net/d/sip5060.net/dnssec/ Translation: sip5060.net provably not enabled DNSEC.

2 1

kamailio not authenticate a subscribed username
by Al S 16 May '16

16 May '16

Hi,1) A generic subscriber is added to kamailio database:kamctl add pbx password2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call:UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it)The correct password is being sent to kamailio.Your assistance is greatly appreciatedThanks,AR

1 0

kamailio not authenticate a call with subscribed username
by Al S 16 May '16

16 May '16

Hi, 1) A generic subscriber is added to kamailio database:kamctl add pbx password 2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call: UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it) The correct password is being sent to kamailio. Your assistance is greatly appreciated Thanks,AR

1 0

Re: [SR-Users] Another question about RTPengine module
by Dmitry 16 May '16

16 May '16

the problem is: rtpengine was compiled with USE_SCTP,but kamailio that was installed via RPM is without this flag. How to tell compiler NOT TO " USE USE_SCTP" flag? +++++++++++++++++++++++ Makefile.defs:SCTP ?= 0 - and it compiles without sctp. Now kamailio.log:ERROR: rtpengine [rtpengine.c:1113]: child_init(): Name or service not knownERROR: <core> [sr_module.c:929]: init_mod_child(): Error while initializing module rtpengine (/usr/lib64/kamailio/modules/rtpengine.so) View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users On Sunday, May 15, 2016 6:21 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: yes, I wrote that modparam are recognized.MODPARAM must be placed i the sae place where the RTPPROXY resides in the config.+++++++++++++++++++++++++++++++ Another question:kazoo kamailio KAMAILIO-V :4.3.5:kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, 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_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 To add the RTPENGINE - I found Kamailio 4.3.5 sources https://www.kamailio.org/pub/kamailio/4.3.5/src/ When I start kamailio I get the following error:ERROR: <core> [sr_module.c:431]: version_control(): module compile flags mismatch for /usr/lib64/kamailio/modules/rtpengine.so I did the following to add missing modules: #make include_modules="rtpengine topoh" cfg #make all cp /usr/src/kamailio-4.3.5/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so How to fix it? The sources version seems to be the same On Saturday, May 14, 2016 1:21 AM, ycaner <yasin.caner(a)netgsm.com.tr> wrote: Hello; i am not sure. i downloaded from git kamailio. add rtpengine instead of rtpproxy . it works fine. it is hard to solve problem without log. Cheers. try to run with kamailio -E -dd , maybe it gives information. check kamailio.cfg is that rtpengine is loaded like loadmodule "rtpengine.so" 2016-05-13 11:35 tarihinde Dmitry [via SIP Router] yazdı: one more question about rtpengine. I installed Kamailio (centos 6.7) from source as You recommended make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg no rtpengine daemon (it is not installed) but the picture is the same: modparam ("rtpengine,....) kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(19523) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found it means that it does not depend upon rtpengine daemon. # single rtproxy modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") I copied it from http://www.kamailio.net/docs/modules/4.2.x/modules/rtpengine.html STRACE kamctl start -says nothing anout rtpengine What else is it possible to do to find the cause of the problem? On Friday, May 13, 2016 11:45 AM, ycaner <[hidden email]> wrote: Hello; rtpengine module isnt default. you need to add with "make include_modules". after make && make install , automatically it will be copied to lib/modules. Cheers. 2016-05-13 09:42 tarihinde Dmitry [via SIP Router] yazdı: And one more question on it: If the Kamailio was installed from RPM and rtpengine.so is missing - May I add rtpengine.so the following way: make include_modules="rtpengine" cfg make all and then copy rtpengine.so to /modules/ directory Is this way (adding missing modules) is correct? On Friday, May 13, 2016 11:25 AM, Dmitry <[hidden email]> wrote: hello I tested on Centos 6.7 as well as Ubuntu server - the problem is the same. RTPENGINE was installed both from dpkg and from Source (manual compilation) So I need a clarification on it. RTPENGINE daemon is started. The Kamailio module itself is taken from /usr/src/kamailio, where the source was compiled. mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so sudo /etc/init.d/kamailio restart | grep rtp ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <rtpengine> found version: kamailio 5.0.0-dev4 (x86_64/linux) Any help is appreciated. On Thursday, May 12, 2016 6:04 PM, Dmitry <[hidden email]> wrote: Hi, I installed Kamailio from GIT. I did make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg Actually loadmodule "rtpengine.so" does not affect kamailio start/restart. modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") causes this failure: [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found On Thursday, May 12, 2016 12:26 PM, ycaner <[hidden email]> wrote: Hello; on bash , kamailio -V gives Version , kamailio -c gives debug result. you can ask from here https://groups.google.com/forum/#!forum/2600hz-dev for kazoo rpm. if there is something wrong , they will fix it. you can install from source code from here => https://www.kamailio.org/w/ and "make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg " in kamailio folder before 'make all' , helps you add modules. if there is a dependecies is missing , gives error. Cheers. 2016-05-11 19:51 tarihinde Dmitry [via SIP Router] yazdı: I installed Kamailio : yum install -y kazoo-kamailio https://2600hz.atlassian.net/wiki/display/Dedicated/via+RPM git clone https://github.com/sipwise/rtpengine.git the latest version of rtpengine and I do not know how to view the version of Kazoo-Kamailio On Wednesday, May 11, 2016 6:31 PM, ycaner <[hidden email]> wrote: hello ; maybe some devel are missing. which version using?i will try. cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML --

1 4

Another question about RTPengine module
by Dmitry 15 May '16

15 May '16

I will clarify the question in my previous post.I installed Kamailio from 2600hz repo via yum (it is for kazoo). But when cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so the kamailio does not see this moduleNo module matching <rtpengine> found Is there something like modules.list (it is used when I compile Kamailio manually) when the kamailio was installed via RPM?

2 12

WebRTC integration
by Moacir Ferreira 15 May '16

15 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

1 0

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 13 May '16

13 May '16

Hello all! After some time I was able to get this working here, and now I am coming back to share my configuration with you guys (in case someone needs to get this working). Here is what we have done: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } xlog("L_INFO", "checking $tu registered with User-Agent:$hdr(User-Agent)\n"); if(reg_fetch_contacts("location", "$tu", "caller")) { xlog("L_INFO", "$(ulc(caller=>count)) records found for $tu\n"); $var(i) = 0; while($var(i) < $(ulc(caller=>count))) { xlog("L_INFO", "record $var(i) with ruid:$(ulc(caller=>ruid)[$var(i)]) - User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); if ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) { $var(ruid) = $(ulc(caller=>ruid)[$var(i)]); xlog("L_INFO", "unregistering $tu ruid:$var(ruid)"); unregister("location", "$tu", "$var(ruid)"); } $var(i) = $var(i) + 1; } } if (!save("location", "0x00")) sl_reply_error(); exit; } } There are some logs over there because we used it to see what was going on while testing the configurations. With this we can get just one register per User-Agent. And again, thanks Olle for your help in this problem *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Sure it is! > > This is exactly what I was looking for! Actually I wasn't able to see that > there was a way to perform this action! > > Thanks for your help, Olle! > > I will try to create this, and when I have it done, I will post my script > here, so if someone else need the same problem I can help. > > Now, I have just one more question about my scenario: when I register a > user, is there a way to create something like a "custom field" on the > location? To explain better: If possible, I can add something like a custom > header that says if the user is registered from a phone device or the web > interface, then I can perform the loop and replace only the oldest contact > using that device... > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > > 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> This is my problem... I want to have 2 contacts for AOR, and not only >> one. >> >> When the 3rd arrives, it must remove the oldest and continue with 2... >> >> *If I am using the 0x04 parameter, it will save just one contact, and the >> user will not be able to get the Invite both, in the mobile device and web >> interface.* >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello, *Olle!* >>> >>> *Thanks for your help in this case, and don't worry about the time >>> (actually I wasn't even expecting to get an answer today).* >>> >>> >>> *I know the max_contacts parameter. But the problem is that if I set the >>> max_contacts to 2, when the user tries to register again, he gets an error >>> message 503. The point is that I want to get the user registered, and >>> replace the oldest location entry for the newest one.* >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello all. >>>> >>>> I have problem here and I tried to find a solution and search over >>>> internet, but without success. >>>> >>>> My scenario is the following: I have an application that must be >>>> registered in Kamailio when a user logs in the web interface, so he can get >>>> calls (something like a web softphone using webrtc). At this point, we are >>>> OK, and everything is working fine. >>>> >>>> To get these register functions I am using the parameter >>>> "save("location", "0x04")" as described in the REGISTRAR module >>>> documentation, so if a user logs in another web browser or computer, only >>>> the last one will continue registered and all calls will be forwarded to >>>> him. >>>> >>>> The point is that now we are creating a phone app that will do almost >>>> the same thing as the web interface, allowing users to receive calls using >>>> the mobile device, and here is my problem: I want to allow my users to be >>>> registered on two devices at the same time, but if a user logs into another >>>> device, I don't want to reply with a 503. I want to allow the user to >>>> register again, deregistering the oldest contact. >>>> >>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>> description it says "implemented overwrite oldest contact behaviour", >>>> but I couldn't find any documentation about it. >>>> >>>> So, is there a way to get this working today? >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>> >>> >> >

2 1

Changing the SDP connection-address to a value in a pseudovariable
by Kevin McAllister 13 May '16

13 May '16

counter to the docs for nathelper fix_nated_sdp the config parser rejects any pseudovariables in the second parameter claiming they need to be constant. Does anyone know another way to change the "c=" line in an SDP? Or do I have to patch the nathelper module to include a free_fixup_function for the fix_nated_sdp stuff? This is on kamailio 4.2 Thanks, Kevin

1 1

Kamailio Crash
by Sunil More 13 May '16

13 May '16

Hello All, Kamailio instance just crashed with the following logs May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12675]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12674]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12682]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12684]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12667]: INFO: <core> [mem/f_malloc.c:598]: fm_free(): freeing a free fragment (0x7f23ff39cb88/0x7f23ff39cbd0) - ignore to add i have collected a gdb backtrace as below, root@IP:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/corefiles/core.kamailio.sig11.12671 The version of Kamilio used by me is as below;. version: *kamailio 4.4.1 (x86_64/linux) 90be8b* flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b Thanking You, Sunil More Ph : 9503338275

2 2

Blacklists and DB_Cassandra
by SamyGo 13 May '16

13 May '16

Hi, I am tasked to make use of blacklist module for about 4 to 6 million numbers. I am thinking of using Cassandra for the purpose but reading through the documentatiom of module and recent mailing list discussion made me a bit hesitant. I am looking for advise on this whether this is going to perform as expected or is not even going to work with the BlackListing module ! I am using Cassandra 33x and Kamailio 4.4 over Ubuntu 14.04. Looking for suggestions here. Regards, Sammy

2 4

DB_Cassandra + UserBlacklist = ERROR
by SamyGo 12 May '16

12 May '16

Hi List, I'm trying to hook up userblacklist module with db_cassandra. I've kamailio keyspace configured with tables for userblacklist , globalblacklist, and version are created with some data in there. Here is setup info: version: kamailio 4.4.1 *kamailio.cfg* ... loadmodule "db_cassandra.so" modparam("db_cassandra", "schema_path","/etc/kamailio/kamailio") loadmodule "userblacklist.so" modparam("userblacklist", "db_url", "cassandra://:@127.0.0.1:9160/kamailio") modparam("userblacklist", "userblacklist_table", "userblacklist") modparam("userblacklist", "globalblacklist_table", "globalblacklist") *SCHEMA PATH:* root@whit-list:/etc/kamailio/kamailio# ls userblacklist version root@whit-list:/etc/kamailio/kamailio# cat version/version table_name(string) table_version(int) table_name root@whit-list:/etc/kamailio/kamailio# cat userblacklist/userblacklist id(int) username(string) domain(string) prefix(string) whitelist(int) id username *CASSANDRA DB* root@whit-list:/etc/kamailio/kamailio# cqlsh Connected to Test Cluster at 127.0.0.1:9042. [cqlsh 5.0.1 | Cassandra 3.5 | CQL spec 3.4.0 | Native protocol v4] Use HELP for help. cqlsh> use kamailio ... ; cqlsh:kamailio> describe tables; globalblacklist version userblacklist cqlsh:kamailio> select * from version; table_name | table_version -----------------+--------------- uacreg | 2 version | 1 globalblacklist | 1 userblacklist | 1 (4 rows) Every time I start up Kamailio I see the following debug logs: 0(13877) DEBUG: db_cassandra [dbcassa_table.c:572]: dbcassa_read_table_schemas(): Full name= /etc/kamailio/kamailio/ 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/version/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/version/version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1463068672 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_name] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[0] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[0] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [table_name] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/userblacklist/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/userblacklist/userblacklist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1462985556 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [id] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[0] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [username] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [domain] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[2] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[2] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [prefix] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[3] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[3] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [whitelist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[4] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [id] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [username] in primary key ..... 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:149]: db_cassa_new_connection(): opening connection: cassa:// xxxx:xxxx@127.0.0.1:9160/kamailio 0(12780) DEBUG: <core> [mem/f_malloc.c:439]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) called from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:514]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) returns address 0x7f4e361bb5f0 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:161]: db_cassa_new_connection(): 0x7f4e361bb5f0=pkg_malloc(48) 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:117]: dbcassa_open(): Opened connection to Cassandra cluster 127.0.0.1:9160 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:955]: db_cassa_query(): query table=version 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:449]: *cassa_translate_query(): table version does not exist!* 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster 0(12780) ERROR: <core> [db.c:397]: db_table_version(): error in db_query 0(12780) ERROR: <core> [db.c:436]: db_check_table_version(): querying version for table userblacklist 0(12780) ERROR: userblacklist [db_userblacklist.c:86]: userblacklist_db_init(): during table version check. 0(12780) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb5f0), called from db_cassandra: dbcassa_base.cpp: db_cassa_free_connection(203) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb318), called from core: db.c: db_do_close(356) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from core: db.c: db_do_init2(298) 0(12780) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module userblacklist (/usr/local/lib64/kamailio/modules/userblacklist.so) ERROR: error while initializing modules Any help would be highly appreciated. Regards, Sammy

2 2

route_if_exists crash / breakage
by Alex Balashov 12 May '16

12 May '16

Hi, I'm running 4.4.0:ea2033 and am seeing a crash when using route_if_exists() on a nonexistent route: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 512 -M 8 -u evaristesys -g'. Program terminated with signal 11, Segmentation fault. #0 0x0000000000456751 in do_action (h=0x7ffcce5bc4c0, a=0xf0f0f0f0, msg=0x7fba440abc80) at action.c:337 337 switch ((unsigned char)a->type){ gdb) print a $1 = (struct action *) 0xf0f0f0f0 (gdb) print a->type Cannot access memory at address 0xf0f0f108 The invocation looks like this: if(check_route_exists("AUX_RTPENGINE_OFFER_INBOUND")) route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); else { set_rtpengine_set("1"); rtpengine_offer("replace-origin replace-session-connection ICE=remove"); } #route[AUX_RTPENGINE_OFFER_INBOUND] { # xlog("L_INFO", 'action=R-AUX-RTPENGINE-OFFER-INBOUND'); #} Uncommenting the nonexistent route has another curious effect: script execution appears to break after route[AUX_RTPENGINE_OFFER_INBOUND] executes, and nothing further happens. Changing route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); to route(AUX_RTPENGINE_OFFER_INBOUND); resolves both issues (the crash and the script execution halting). But, of course, if the route is commented out - its normal state - then Kamailio won't start since it won't allow a nonexistent identifier to be passed to route(). -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

Kamailio - Manage several kamailio instances with Siremis
by José Seabra 12 May '16

12 May '16

Hi list, Anyone here knows if i can use one siremis instance to manage more than one Kamailio Instance? BR -- Cumprimentos José Seabra

1 0

Kamailio / issue with registration
by Ali Taher 12 May '16

12 May '16

Hello, I installed Homer with kamailio as capture agent in order to capture sip packets. The first test I did was using the default kamailio.cfg config located in homer-docker directory , where all Register and Invite packets were saved in the database and I was able to monitor them from web homer. After that , I tried to enable registration using the save(location) function in registrar module , yet I'm getting the below error in the log : 2016-05-11T18:48:36.443231+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f64702b1d98,767,0,192.168.26.3:3806,16): Bad file descriptor(9) 2016-05-11T18:48:36.443239+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: sl [../../forward.h:201]: msg_send_buffer(): udp_send failed 2016-05-11T18:48:36.443247+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK Can you please help with resolving this issue , noting that the registration works good if I remove the routing logic set for sipcapture. Any hint would be appreciated. Cheers, Ali Taher

1 1

Wiki update notice
by Mikko Lehto 12 May '16

12 May '16

Hello Some wiki work: 1. "What's new in 4.4" -content is no more present in devel version wiki page 2. based on recent mailing list discussion, I added note about github pull requests I think these make sense, please yell if not :) [1] http://kamailio.org/wiki/features/new-in-devel [2] http://kamailio.org/wiki/devel/github-contributions?rev=1456916916&do=diff -- Mikko

2 2

T_RELAY_TO distribute local proxy to a central one
by Sébastien Brice 12 May '16

12 May '16

Hi guys, i have a kamailio running on the wan (my central kamailio) and i have phones on a private network range 172.20.0.0/16 behind a NAT. i also run another kamailio locally to collect all sip phones on my network and this one use the DB of the remote central kamailio. the local kamailio does t_relay_to() the ip of the central kamailio route { if (uri==myself) { if (method=="REGISTER") { route(2); exit; }; route[2] { rewritehost("public_remote_kamailio"); if(!t_relay_to("udp:public_remote_kamailio:5060")) { sl_reply_error(); }; exit; } the bad thing is once the phones get registered with non-routable ip (like in the attached picture) trying to INVITE them directly by requesting public_kamailio_ip result in non-working situation. how should I rewrite my dst_domain when registring in a way INVITE triggers to the local kamailio. My local kamailio runs with the listen udp:private_local_kamailio advertise udp:public_local_kamailio I try to be explicit but my understanding is not perfect, it would be nice you could give me a hand. I put my local kamailio.cfg attached. thx

2 1

4.4 compiler warnings on debian jessie
by Juha Heinanen 12 May '16

12 May '16

debian jessie c compiler complains about these in kamailio 4.4: CC (gcc) [sip-proxy] mem/tlsf_malloc.o mem/tlsf_malloc.c: In function 'tlsf_malloc_init_pkg_manager': mem/tlsf_malloc.c:1353:16: warning: assignment from incompatible pointer type ma.xmalloc = tlsf_malloc; ^ mem/tlsf_malloc.c:1355:16: warning: assignment from incompatible pointer type ma.xrealloc = tlsf_realloc; ^ mem/tlsf_malloc.c:1358:16: warning: assignment from incompatible pointer type ma.xavailable = tlsf_available; ^ mem/tlsf_malloc.c: In function 'tlsf_malloc_init_shm_manager': mem/tlsf_malloc.c:1501:20: warning: assignment from incompatible pointer type ma.xmalloc_unsafe = tlsf_malloc; ^ it would be highly appreciated to get rid of them. -- juha

5 12

calling rpc command in cfg
by Yasin CANER 12 May '16

12 May '16

3 2

speeddial, owners
by Filippo Graziola 11 May '16

11 May '16

Hello everybody, I have a kamailio running on debian, and it works well with a little modification of the kamailio-advanced file. I would like to have some speed dial short numbers for all local users, because I used a random generator script and usernames and password are 25 digits (not only numbers) long. My problem is with the ownership of a every speeddial number. Example: SIP accounts: 5001,5002,5003 SIP domain: mydomain.com add a speeddial number: kamctl speeddial add 5002(a)mydomain.com 51(a)mydomain.com sip:5001@mydomain.com I can call from user 5002 to user 5001 typing 51 correctly. what I would like to do is that sd-id 51(a)mydomain.com could be called from every user registered at mydomain.com. I read the module documentation, kamctl help but I'm not able to find an answer. Could you please help me? Thank you in advance

2 2

kamailio does not see rtpengine module
by Dmitry 11 May '16

11 May '16

Hello when I compiled rtpengine for kamailio (in Kazoo)? there were no errors while compiling, but when restarting kamailio - I saw:kamailio: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found [root@ip-192-168-130-136 kamailio]# locate rtpengine.so/usr/lib64/kamailio/modules/rtpengine.so default.cfg:mpath="/usr/lib64/kamailio/modules/" Does it mean that something wrong with rtpengine compilation? (I compiled manually) Appreciate any help.

1 0

Skype-like service, RTP flows strangely
by Sébastien Brice 11 May '16

11 May '16

Hi guys i did the "running skype-like service in less than one hour" and got it running ok. Kamailio run remotely in a datacenter. but i am trying to get the communication working locally between two devices behind the same NAT. I know the #!define WITH_NAT shoud get the call working bidirectionnaly but that is not the case, I've got one way dial every time i'm attempting call ! running a network capture (please find attached to this email )reveal rtp does not flow correctly: the device A 192.168.3.33 send RTP straight to device B 192.168.3.98 ---> OK device B does not behave like that, rather to talk to A (192.168.3.33) it sends RTprotocol to the public_ip of my internet box (refered as public_ip red text in my .png capture) and the WAN leg of my box answers ICMP 242 Destination unreachable (port unreachable) So comes blind communicatios. But that is not supposely act like that, right ? RTP should be relayed by the rtpproxy running on the remote Kamailio and we should have: device A -----> remote_kamailio_ip ----> device B device B -----> remote_kamailio_ip ----> device A thx you for any help PS: see attached file

2 1

VersionUp for CentOS7's
by 中島宗治 11 May '16

11 May '16

Please version up for CentOS7's rpm. -- Muneharu Nakasima <mune(a)vasara.net> <vasaralamp(a)gmail.com>

2 1

Kamailio and Oracle database
by gmele 11 May '16

11 May '16

Hello, we are trying to connect our Kamailio SIP to an Oracle database. We tried both db_oracle and db_unixodbc modules to connect to the DB. With db_oracle, Kamailio core dumps when trying to access the active_watchers table (caused by the oracle CLOB?). Here is the kamailio logs: Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: <core> [db_query.c:131]: db_do_query_internal(): DB-QUERY: select presentity_uri,callid,to_tag ,from_tag,event from active_watchers where updated=:1 AND event=:2 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN PRESENTITY_URI TYPE 4 SIZE 129 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN CALLID TYPE 4 SIZE 256 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN TO_TAG TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN FROM_TAG TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN EVENT TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:378]: get_rows(): OCIStmtFetch2() returned 100 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23667]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 16 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23667]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 14 What is really strange is that this table is accessed several times before the crash, without any problem. The other point is that this table is absolutely empty... With db_unixodbc, all requests having dates check fail because because the date format is not understood by oracle. The date should be converted using the oracle to_date(date, format) funtion to work... Here are the Kamilio traces: Apr 29 14:10:38 newvm1 /usr/sbin/kamailio[17021]: ERROR: usrloc [dlist.c:183]: get_all_db_ucontacts(): query error Apr 29 14:10:38 newvm1 /usr/sbin/kamailio[17021]: ERROR: nathelper [nathelper.c:2089]: nh_timer(): failed to fetch contacts Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1. Query= select received,contact,socket,cflags,path,ruid from$ Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: db_unixodbc [connection.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=HY000:1:933:[Oracle][ODBC][Ora]ORA-00$ Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query I tried the option modparam("db_unixodbc", "use_escape_common", 1), but without success. Has anyone been able to make Kamailio 4.4.0 work with Oracle 11g, either with db_oracle or with db_unixodbc module? Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-and-Oracle-database-tp1479… Sent from the Users mailing list archive at Nabble.com.

3 28

Garbage when decoding uri
by Glenn O Larsen 11 May '16

11 May '16

During a re-invite, it looks I get a strange decode error if the Route header is to long, or is there something else I have missed... 23(23945) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <transport=tls;r2=on;lr=on;ftag=as03d8c4db;vsf=AAAAAEFVRVNSRxtbXFhFITUbBwcLEQBaQAFVDkVFUXI9cGhvbmU-> 23(23945) DEBUG: uac [replace.c:527]: restore_uri(): getting 'vsf' Route param 23(23945) DEBUG: uac [replace.c:535]: restore_uri(): route param is 'AAAAAEFVRVNSRxtbXFhFITUbBwcLEQBaQAFVDkVFUXI9cGhvbmU-' (len=52) 23(23945) DEBUG: uac [replace.c:609]: restore_uri(): decoded uris are: new=[sip:xxxxxxxxxx@u*5?%"9tu3{6dB @hone] old=[sip:xxxx.xxxx@10.0.0.1:5060] Kamailio 4.3.4 is used. -- Best regards Glenn Larsen

1 0

Kamailio Simple Config File Request
by Ali Dakroub~Vanrise Technical Support 10 May '16

10 May '16

Hello, I am a newbie in Kamailio configuration. I would be very grateful if you can provide me with a simple Kamailio.cfg file that achieves the below scenario: 1. Machines A and B are able to register on Kamailio Sip Server. 2. Machine A is able to call Machine B through Kamailio SIP server using x-lite. Noting that Machines A and B are on the same network. Thanks in advance. Regards, Ali

1 0

Setting source interface on relay
by Phil Lavin 10 May '16

10 May '16

Kamailio receives a fairly arbitrary out of dialog NOTIFY on an interface with a private IP address (eth1). It runs it through the following code which relays it to an IP/port combo pulled from a hash table. The IP that it is relaying to is a public address and should be addressed out of the interface with a public IP address (eth0). It is, however, sent on the private interface (eth1) and thus never reaches its target. route[PHONEREBOOT] { # Only handle check-sync (Polycom) NOTIFY if (!is_method("NOTIFY") || $hdr(Event)!="check-sync") return; # Look up ip:port combo in the agentloc hash table if ($sht(agentloc=>$rU)==$null) return; # Set SIP URI to point to the agent $ru = "sip:" + $rU + "@" + $sht(agentloc=>$rU); route(RELAY); exit; } Is there a way to explicitly set the source interface to relay on? Thanks Phil Lavin Telecoms Systems Manager CloudCall by SYNETY www.cloudcall.com<http://www.cloudcall.com/> T: +44 (0) 330 335 0000 / +1 617 982 1600 D: +44 (0) 116 424 4790 / +1 716 242 6604 SM: LinkedIn<https://uk.linkedin.com/pub/phil-lavin/25/422/750> READ OUR BLOG FOR SMARTER COMMUNICATIONS<http://t.sidekickopen03.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XX43…> Confidentiality: This e-mail transmission, including any attachments, is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you have received this transmission in error, or are not the named recipient(s), please notify the sender immediately by return e-mail and permanently delete this transmission, including any attachments. Security: This e-mail and any attachments are believed to be free from any virus but it is the responsibility of the recipient to ensure this is so. E-mail is not a 100% secure communication.

3 4

Kamailio v4.4.1 Released
by Daniel-Constantin Mierla 10 May '16

10 May '16

Hello, Kamailio SIP Server v4.4.1 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since release of v4.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.1. For more details about version 4.4.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Looking forward to meeting many of them at Kamailio World Conference 2016 next week in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

extension of auth module
by JBF 10 May '16

10 May '16

Hello, We would like to use some functionalities provided in ims_auth. But our deployment uses auth/auth_db (we have another server with credentials and we would like to avoid adding a diameter interface to it) In particular, we want to add an Authentication-Info header to the response Do we have to patch the auth module to achieve this, or is there some other way ? If we patch it, would you merge it to the master ? its important to us, so we can easily upgrade to the next versions of kamailio. -- View this message in context: http://sip-router.1086192.n5.nabble.com/extension-of-auth-module-tp148332.h… Sent from the Users mailing list archive at Nabble.com.

1 0

Planning the release of Kamailio v4.4.1
by Daniel-Constantin Mierla 10 May '16

10 May '16

Hello, I am considering to do a minor release from branch 4.4 next week, likely on Tue or Wed, May 10 or 11, 2016. If there are issues you are aware of and not reported yet to the tracker, do it as soon as possible to try to fix what we can by the date. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

WSS configuration sample needed
by Sunil More 09 May '16

09 May '16

Hello All, I am trying to connect Kamailio over WSS and the error I get is "tls_accept: client did not present a certificate" . It works fine for WS, however Please guide me to connect over WSS. the following are the logs i See.. DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... ay 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake done May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:718]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:411]: tls_accept(): TLS accept successful May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:418]: tls_accept(): tls_accept: new connection from sunilmore.in:49703 using TLSv1/SSLv3 AES256-SHA 256 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:421]: tls_accept(): tls_accept: local socket: sunilmore.in:10082 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:432]: tls_accept(): tls_accept: client did not present a certificate And the websocket closes. Please help. Thanking You, Sunil More Ph : 9503338275

2 5

sdp_remove_line_by_prefix
by Alberto Sagredo 09 May '16

09 May '16

Hi Im trying to remove my Asterisk s= line on SDP but when doing: sdp_remove_line_by_prefix("s=Asterisk"); Nothing happens. Loaded sdpops.so module as well. Is there any additional needed? BR

4 10

Getting sip.instance in INVITE
by Cibin Paul 09 May '16

09 May '16

Hi, How can I get sip.instance value in the sip header of INVITE? I do get this value during REGISTER Thanks & Regards Cibin

2 2

Third party registration help
by Code Developer 09 May '16

09 May '16

Hello, I'm after some help on third-party SIP registration in Kamailio - specifically, how to perform registration of user agents that don't register themselves, but only send out occasional messages. The UAs not registering themselves are devices on a GSM network that try to save band-width by not sending REGISTER requests for themselves. They only ever send out SIP messages when necessary, to which they expect a reply message from our processing server. (Note, details like external IP addresses have been changed for security.) The configuration is as follows. We have a SIP server running Kamailio (4.3, location db enabled, nathelper enabled) on a static external IP address (50.170.158.149), a processing server on an internal network registered with the SIP server (let's say with a User agent name of '1000'), and a monitoring device in another part of the country connected to the internet via a GSM network (let's say User agent name of '2000'). The device is effectively NAT'd on the GSM provider's private network. At some point the device (UA 2000) sends a SIP message to the processing server via Kamailio (to sip:1000@50.170.158.149). The processing server is registered, so the SIP server knows where it is and forwards the message accordingly. 200/OK replies propagate back successfully to the device. The message flow appear as below. Only relevant header fields are shown: Message received by the SIP server from remote device: ======== IP Src:60.250.149.198, Dst:50.170.158.149 UDP Src Port: 59177, Dst Port: 5060 -- Request-Line: MESSAGE sip:1000@50.170.158.149 Via: SIP/2.0/UDP 10.92.240.131:5060;branch=z9hG4bK... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== Message forwarded from SIP server, received by processing server as shown: ======== IP Src:50.170.158.149, Dst: 192.168.0.235 UDP Src Port:5060, Dst Port:5060 -- Request-Line: MESSAGE sip:1000@50.170.158.146:5060; ob SIP/2.0 Via: SIP/2.0/UDP 50.170.158.149;branch=z9hG... Via: SIP/2.0/UDP 10.92.240.131:5060;rport=59177;received=60.250.149.198;branch=z9hG... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== The processing server now needs to send back a reply message to the remote, unregistered device which is NATd on the GSM internal network. We can do this by sending back a reply to the external IP address and rport values found in the second Via header of the message received by the processing server (60.250.149.198, port 59177). The reply can only be sent from the SIP server or the GSM network's router will ignore it - it's only a temporary reply address/port open for a minute or so as it's effectively just a NAT route to the internet. In the first instance, when the processing server attempts to send back a message to the remote device via the SIP server, the SIP server replies with a 404 as the remote device hasn't registered and the SIP server has no idea where to send it. After looking at RFC3261 my conclusion was that we must get the processing server to perform a third-party registration on behalf of the remote device before we can send a reply message back to that device. My understanding is that to perform the third party registration I should send a REGISTER request from our processing server to the SIP server: 1) Set the 'From' field to the AOR of the UA performing the registration (which is the processing server, 1000). 2) Set the 'To' field to the AOR of the UA being registered (i.e. the remote device, 2000). 3) Add a 'Contact' header field set to the IP address and port of the remote device (from Via = 60.250.149.198 port 59177). The message flow for the register is as follows (only relevant header fields shown): Processing server sends a register request to the SIP Server: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.235:5060;rport;branch=z9hG... From: <sip:1000@50.170.158.149>;tag=e6e2d70... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> ======== SIP server responds to the processing server with a 401 unauthorized response containing the auth header field (nonce challenge): ======== WWW-Authenticate: Digest realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" ======== Processing server sends a revised register request to the SIP Server, augmented with the authorisation header for the supplied nonce: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 50.170.158.146:1096; rport;branch=z9hG4... From: <sip:1000@50.170.158.149>;tag=dae68a... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> Authorization: Digest username="1000", realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" response="d4d7ffba8..." ======== However, the SIP Server again responds with a 401 and won't perform the third-party registration. I haven't been able to determine why. I *think* the response hash the processing server is sending back is correct. I'm using PJSIP on our processing server to perform all SIP functionality. The processing server does obviously register itself, the difference being that the To and From fields are the same (<sip:1000@50.170.158.149>) and there is no Contact header. I'm at the point now of being unable to find any information or examples on how third-party registration requests should be performed - apart from RFC3261, which doesn't really describe the process very well. Can anyone tell me why the third-party registration would fail, despite replying with the authorisation header? Does Kamailio first need to be configured to allow third-party reg somehow? Alternatively, should I be attempting to send back SIP messages to unregistered devices in some other fashion? I hope someone can help or point me in the right direction. As an aside, when the device (UA2000) is plugged into a network via ethernet, it does then perform its own registration and everything works - messages go back and forth without issue.

2 1

LCR Module Question
by Alberto Sagredo 09 May '16

09 May '16

Hi I have it working but i have re-read documentation and do not see how to do what i need. I explain it :) Now i have only one LCR provider and i need to add a backup one. I do not know if its enough to add under same lcr_id or its better to add with different one and add several lcr_rule and lcr_rule_target Acordding to next_gw() function, what is better to be used? In my case. a primary providers is used and only a backup one for the momment. Thanks for your help Alberto

3 3

Media-Proxy & NAT
by cao tao 09 May '16

09 May '16

Hello: I use kamailio as my SIP signal proxy, and media-proxy as RTP relayer. It works well at many situations. But when it comes to two layer of NAT(a user behind two layer of NAT), I got a single way media(the sip signal layer was ok). I can’t hear the voice of user who behinds two layer of NAT, but he can hear the voice of mine. Most part of my kamailio config is default, and simply enable media-proxy by engage_media_proxy()： if (is_method("INVITE") && !has_totag()) { engage_media_proxy(); } And in my UAC/UAS side, I use PJSIP. Can anybody give me some help? Any suggestion will be appreciated!

2 1

User offline
by 王欣鑫 06 May '16

06 May '16

Hello, I use kamailio set up an IMS on a cloud server in a public network, and have alice and bob registered on the server using ImsDroid in a private network. When I use alice to call bob, I got "User offline" error. Below are .pcap file captured from 4060 port and pcscf.log/icscf.log/scscf.log and all config files(in kamailio.zip) 4060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148166/4060.pcap> pcscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/pcscf.log> icscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/icscf.log> scscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/scscf.log> kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148166/kamailio.zip> My kamailio version and debian operating system version is like below: root@iZ94lg22es7Z:~# kamailio -version version: kamailio 4.4.0 (x86_64/linux) d4f23c flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d4f23c compiled on 13:02:46 May 5 2016 with gcc 4.9.2 root@iZ94lg22es7Z:~# root@iZ94lg22es7Z:~# cat /etc/issue Debian GNU/Linux 8 \n \l root@iZ94lg22es7Z:~# Can anybody give me some help? Any suggestion will be appreciated! -- View this message in context: http://sip-router.1086192.n5.nabble.com/User-offline-tp148166.html Sent from the Users mailing list archive at Nabble.com.

2 14

Clients are busy
by kai.ohnacker＠cbc.de 06 May '16

06 May '16

Hello Community, in my scenario works multiple user with the kamailio server and uses different clients. They work in a mobile scenario and setup the clients and break the clients in a high period. If they break the clients, sometimes they don't send a bye, but they take off the cable connection during a call. If these devices will connected again and other clients try to call them, they are busy. That is my problem :). Does the Kamailio server provides any function to send a bye message if a clients don't register at a required time? Or is there another solution available? Thanks for help. Cheers, Kai

2 1

making rtpengine gives error on centos
by Yasin CANER 05 May '16

05 May '16

4 5

Kamailio - Sip_capture Module
by Ali Taher 05 May '16

05 May '16

Hello, We are currently using kamailio sip_capture module, and then we need to create a Wireshark file (pcap format)from fields in the database. As we noticed in sip_capture table there is msg fields that contains only the sip content. And in order to create our file we need to have the entire packet and not part of it. We are hopefully asking if you can give us a hand with that , beside the Homer project, what we actually wanted is to know if there is another module in kamailio that store the entire packet captured and not only the part of it. Regards, Ali Taher

2 3

Pause a Transaction for a short while
by Sirvan Paraste 05 May '16

05 May '16

Hello all, is there anyway to pause a transaction for a short while (1 second) ? I have implemented the push notification, when the callee get register I need a short time to be sure that the app is completely start and can do its duty. otherwise it missed the call. so I want to pause the transaction. I have tested async_sleep("1"); it worked. is there any better solution ? $var(hjoin) = 0; lock("$tU"); async_sleep("1");

2 1

ACK not forwarded by Proxy when LOCATION lookup
by Al S 05 May '16

05 May '16

Hi, I am trying to simulate Inbound calls from UAC to Kamailio to UAS. Kamailio performs location lookup within route[LOCATION] and updates URI with the address of UAS. However, it does not add a new Record-Route. Therefore, ACK message sent by UAC is not routed by the proxy: UAC ---> (send INVITE) ---> Kamailio ---> (send INVITE) ---> UASUAS ---> (send 180) ---> Kamailio ---> (send 180) ---> UACUAS ---> (send 200) ---> Kamailio ---> (send 200) ---> UAC UAC ---> (send ACK) ---> Kamailio ???? ACK message has "Route" header with the address of Proxy. Thanks,AI S

2 1

You must register first with a S-CSCF/Server error on LIR select next S-CSCF
by 王欣鑫 05 May '16

05 May '16

Hello, I'm using kamailio and OpenImsCore HSS module to set up an IMS enviroment(P/I/S-CSCF and HSS). All these four modules are set up in one cloud server in public network with different ports(4060/4061 5060 6060 3869 3870) to commonicate with each other. Now I have two users Alice and Bob, they are in private network. I can register Alice and Bob with ImsDroid on this IMS, however when I use Alice to call Bob, I can see "You must register first with a S-CSCF" error. As I found this was caused by kamailio.cfg in pcscf module, I comment out these lines below. Then I got another error "Server error on LIR select next S-CSCF". And I captured packets on port 5060, I can see several "Request-Line: OPTIONS sip:sbc-2.ng-voice.com SIP/2.0" and "Request-Line: OPTIONS sip:sbc-1.ng-voice.com SIP/2.0". #if (!pcscf_is_registered("location")) { # send_reply("403","Forbidden - You must register first with a S-CSCF"); # break; #} What's wrong with my IMS? Could any body give me some help? Thank you! Here are my IMS configure files. kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/kamailio.zip> Here are my hss configure files. hss_config.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/hss_config.zip> -- View this message in context: http://sip-router.1086192.n5.nabble.com/You-must-register-first-with-a-S-CS… Sent from the Users mailing list archive at Nabble.com.

1 0

Using websocket server for events
by Yuriy Gorlichenko 04 May '16

04 May '16

Hi is there is a valy to send websocket events (not only SIP) on remote subscribed client from config file? I just want make shure that i need to use redis for this staff insead of native websocket events

2 3

Using Kamailio as UAC on any PBX
by Jöran Vinzens 04 May '16

04 May '16

Hi All, we are trying to use the Kamailio as an UAC on a PBX. It should behave just like a normal SIP UAC (e.g. an SNOM phone). The reason why we want that, is connecting a service to any other PBX with Client features and security (DA). Our setup looks like: Kamailio UAC w/ register <-> Kamailio OBP/SBC <-> PBX The register is working fine. We insert the credentials into DB and trigger kamailio to reload the config (kamctl uac.reload). Then the register is sent out including the unauth (401) handling. After than we are registered on the PBX. The Register is sent though the OBP as configured in DB. We are trying to make calls: Calls from PBX to Kamailio are working fine but we struggle with calls from kamailio torwards PBX. We have the l_uuid in a variable and we try to get the PV ready for the invite. First we need to do is rewrite the "from" URI. We thought we can get a proper FROM URI with the uuid we have. But the uac_replace_from just sets the $fu as given and does not make any logic with UAC credentials. To handle the RURI, we trigger uac_reg_lookup with the uuid and it replaces the $ru with user-name and domain from the data stored in DB. In this case that makes no sense. The $rU should be what the user had dialed, the $rd should be the server given in uacreg DB table. The OBP given in DB is ignored completely. After we managed to rewrite everything in SIP as needed we try to catch the 407 as it's shown in the module documentation with failure route. Here we have not found any way to get the credentials for the uuid. Now we wonder if we have missed any global setting which sets everything right. If anybody has a similar setup and a working config, we appreciate any help we can get. our config looks like: ... loadmodule "uac.so" .... modparam("uac","auth_realm_avp","$avp(arealm)") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","restore_from_avp","$avp(original_uri_from)") modparam("uac","restore_to_avp","$avp(original_uri_to)") modparam("uac","credential","username:domain:password") modparam("uac", "reg_db_url","DBURI") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_contact_addr", "1.2.3.4:5060") ... route { xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru\n"); route("QUERY_UUID"); # <- sets $var(UUID) to l_uuid of the DB $var(request_user) = $rU; uac_reg_lookup("$var(UUID)", "$ru"); $var(request_from_user) = $rU; $rU = $var(request_user); uac_replace_from("$var(request_from_user)","sip:$var(request_from_user)@ 1.2.3.4"); t_on_failure("TRUNKAUTH"); xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru $avp(auser) $avp(apass)\n"); # avp "auser and apass are "<null>" if (!t_relay()) { xlog("L_ERR", "error to <$tu> from <$fu>\n"); sl_reply_error(); }; } failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { # $avp(auser) = "test"; <-should be set from the l_uuid profile anyhow # $avp(apass) = "test"; $var(request_user) = $rU; if (uac_auth()) { # We do never get here since uac_auth never returns true. $rU = $var(request_user); route("FORWARD_MESSAGE"); exit; } xlog("L_NOTICE", "DAMN!!!"); } } -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 1

rtpengine default set
by Alex Balashov 04 May '16

04 May '16

Hi, After upgrading to Kamailio v4.4, I receive the following error on boot: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: NOTICE: rtpengine [rtpengine.c:1788]: mod_init(): Default rtpp set 0 NOT found My rtpengine module config is: --- #!ifdef WITH_RTPENGINE #!subst "/RTPENGINE_SOCK/udp:127.0.0.1:5050/" modparam("rtpengine", "rtpengine_sock", "1 == RTPENGINE_SOCK") modparam("rtpengine", "rtpengine_disable_tout", 30) modparam("rtpengine", "rtpengine_tout_ms", 2000) modparam("rtpengine", "rtpengine_retr", 2) #!endif --- Furthermore, I have set_rtpengine_set("1") before every call to the rtpengine_* module function(s) in my route script. Subsequently to that, Kamailio starts normally and connects to rtpengine: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24975]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24974]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled [...] It might be something on my end, but my guess is that in v4.4 there is a fixup that runs beforehand and looks for a default set. Strictly speaking, I don't need an explicitly identified set, since I'm only using one rtpengine instance. However, I explicitly specify it because I'd like the code to be open in the future to selecting a given rtpengine relay based on some criteria. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

create packet
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

2 1

create PCAP file
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

1 0

Kamailio 5-0 - Executing event routes in embedded languages
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, as announced earlier this morning, most of the routing blocks can be written directly in an embedded language (right now supported being Lua and Python). The missing blocks are (most of) the event routes. The event route from tm for branch failure route is already supported, as the name has to be set via t_on_branch_failure(...). In the native configuration, event routes are executed if they are defined. Their names (as in event_route[name]) are not valid function names for the other programming languages to be used directly corresponding callback function name. I was thinking of the alternative, next are the ones I came up with, more suggestions are welcome. 1) add parameters for event route callback names, like: modparam("htable", "evrtcb_mod_init", "ksr_htable_mod_init") would define the name of the function for the alternative to event_route[htable:mod-init]. This has a benefit in speed, if the parameter is not defined, then there is no overhead in trying to execute a callback function in an embedded language. The drawback is that there is need to code a bit of C to add those parameters to the components executing event routes. 2) do a direct mapping using some translation rules, like: - prefix with ksr_ - replace ':' and '-' with underscore '_' So event_route[htable:mod-init] will have automatically the correspondent function ksr_htable_mod_init(). The drawback is that each time the even route should be called, a lookup on embedded language symbols table needs to be done to see if it exists, then execute it. There is still some C code to do, but less than adding parameters like for 1). 3) like 2), but instead of an automatic translation, keep a mapping table: struct mapping { str kname; str ename; } table[] = { str_init("htable:mod-init"), str_init("ksr_htable_mod_init"), } Adding a new event route will require adding a new record in the mapping table. The benefit is that the function name in embedded language can be selected to be whatever, with or without ties to the event_route name. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

A question about the code
by ngregory＠reprivata.com 04 May '16

04 May '16

3 4

Kamailio 5.0 - Routing blocks in Lua or Python
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, I want to announce that the framework (named kemi) which allows writing the routing blocks in an embedded language is already in place. Lua or Python can be use as alternative to the native scripting language to write the routing blocks. More details can be found at: - https://www.kamailio.org/wiki/devel/config-engines Example of scripts that offer more or less same features as etc/kamailio-basic.cfg are available in source tree at examples/kemi. For convenience of reading, here are the links to git repository: - the common file setting global and modules parameters as well as specifying the config engine: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in native language: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Lua: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Python: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… You can compare the last three above and see how different or similar they look for providing same SIP routing features using different languages. A lot of modules need to export now functions to embedded config interface, which is quite straightforward: https://www.kamailio.org/wiki/devel/config-engines#exporting_functions_to_k… The main missing part is dealing with event_route blocks -- I will send a separate email to discuss that. Interesting that some incipient performance tests to compare the interpreters show good results for using especially Lua but also Python: - https://www.kamailio.org/wiki/devel/config-engines#interpreters_performances Anyone willing to help can start exporting functions from modules to embedded interface or test some more complex configs in Lua or Python. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

t_on_branch_failure never called AND t_relay never fails in a, forked scenario
by Tristan Mahé 03 May '16

03 May '16

Hi fellow kamailio users, I'm playing actually with t_on_branch_failure, in a simultaneous forking scenario ( lookup("location") returns multiple contacts ). Any idea on how to get those "on_branch_failure" working ? Thanks in Advance ! * Test by putting failure_exec_mode: - at 1 => no branch failure being called, never. - at 0 => t_relay does not fail if a single branch answers the transaction, even if all other branches are failing. * Test by moving the t_on_branch_failure("manage_branch_failure"); line before or after the lookup("location") or on a _on_branch route to set it per branch. - The on_route branch are executed, but never the failure one. - tried also with failure_exec_mode = 0 or 1. * Using set_forward_no_connect, I can see in the logs a "Sending to branch X failed" but no failure route being executed. Without it, nothing in the logs indicating that the branch has timeout. * Config extract: # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # 3 by default # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "reparse_on_dns_failover", 0) # 1: Allow branches to have a per branch failure route # 0: t_relay should fail if a branch fail modparam("tm", "failure_exec_mode", 1) # Allow locally generated 408 to be forgotten in case a reply is received. #modparam("tm", "faked_reply_prio", 2000) # USER location service route[LOCATION] { t_on_branch_failure("manage_branch_failure"); if (!lookup("location")) { .... } # Let's get back after 2.5s if no 100/Trying received. t_set_fr(0, 2500); # set_forward_no_connect(); #t_on_branch("SET_BRANCH_FAILURE"); route(RELAY); exit; } route[RELAY] { .... if (!t_relay()) { xlog("L_WARN","$rm|$ci|$fu|$tu|RELAY_FAILED"); sl_reply_error(); } exit(); } # set failure route per branch branch_route[SET_BRANCH_FAILURE] { xlog("L_WARN","$rm|$ci|$fu|$tu|SETTING_BrANCH_FAILURE_ROUTE"); t_on_branch_failure("manage_branch_failure"); } # manage failure routing cases per branch event_route[tm:branch-failure:manage_branch_failure] { xlog("L_WARN","$rm|$ci|$fu|$tu[$avp(dest_to)]|MANAGE_BRANCH_FAILURE"); }

3 2

set_body_multipart and memory leak
by Sergey Basov 03 May '16

03 May '16

Hi, All. I have some problems with pkg_memory leak. After each call used private memory of the "udp receiver" process increases at 2500 bytes. On test environment I hav set it to 2Mbyte, but it is full after 700 calls. I need to modify SDP in invite and other packets. When I do ==================================================== if(has_body("application/sdp")) set_body_multipart(); if (msg_apply_changes()) { xlog("L_INFO", "ISUP 1 Changes Applied Succesfully"); } } record_route(); dlg_manage route(RELAY); ==================================================== Debug from kamailio: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:2627]: has_body_f(): content type is 196611 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<17>:[unique-boundary-1] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<44>:[multipart/mixed;boundary="unique-boundary-1"] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1766]: set_multibody_helper(): set flag FL_BODY_MULTIPART Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: ERROR: <core> [msg_translator.c:1691]: get_boundary(): Content-Type hdr has no params Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: WARNING: <core> [msg_translator.c:1954]: build_req_buf_from_sip_req(): check_boundaries error Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 251 (251) Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:0637304494@10.56.42.27:5070;user=phone> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK09e6.2654d2382f7c4e4fb8bb62fe85f62dfa.0>; state=6 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 236, <i> = <1>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: INFO: <script>: ISUP 1 Changes Applied Succesfully Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKsr-s7wTDLa0zUfsTql5zpl0zpl0z.lRD.zok.TLz.lSk2Fw1RY9PJYMxpdED.dED.aEG38LkpcsWhqMQr1v1bFAyJGCxrCoQdB71Mtceh52GpZKDpX3zRFAzVFJzhBLGpeJe.TczW7cDuT8GLs7O37vBbn6BbTfGhM2Gpl*>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=ffffffffffffffff Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:509]: parse_headers(): parse_headers: this is the second via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [46]; uri=[sip:0637304494@10.56.42.27:5070;user=phone] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:0637304494@10.56.42.27:5070;user=phone>#015#012] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=251 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 16

xmlrpc crash
by Juha Heinanen 03 May '16

03 May '16

i was sending some xmlrpc requests to kamailio and got crash. -- juha (gdb) where #0 0x00007f9c465da634 in __GI___libc_free (mem=0x17828c0) at malloc.c:2945 #1 0x00007f9c39e0b797 in collect_garbage () at xmlrpc.c:741 #2 0x00007f9c39e12669 in dispatch_rpc (msg=0x0, s1=0x17828b0 "ams", s2=0x7f9c46903628 <main_arena+8> "`\352w\001") at xmlrpc.c:2440 #3 0x000000000053c2c0 in do_action (h=0x7ffd17c562d0, a=0x7f9c46355c28, msg=0x7ffd17c563b0) at action.c:1060 #4 0x000000000053ad85 in run_actions (h=0x7f9c46903620 <main_arena>, a=0x17828b0, msg=0x7f9c46903628 <main_arena+8>) at action.c:1549 #5 0x00007f9c39e0d04b in em_receive_request (orig_msg=0x7f9c45cc43e8, new_buf=0x0, new_len=24651952) at xmlrpc.c:2303 #6 0x00007f9c39e14034 in process_xmlrpc (msg=0x7f9c45cc43e8) at xmlrpc.c:2385 #7 0x00000000004fdce3 in nonsip_msg_run_hooks (msg=0x7f9c45cc43e8) at nonsip_hooks.c:111 #8 0x0000000000549610 in receive_msg (buf=0x0, len=1171014632, rcv_info=0x7f9c3e4b0110) at receive.c:188 #9 0x00000000005cb97a in tcp_read_req (con=0x7f9c3e4b00f8, bytes_read=0x7ffd17c56e64, read_flags=0x7ffd17c56e68) at tcp_read.c:1418 #10 0x00000000005d0293 in handle_io (fm=0x7f9c3e4b00f8, events=10416, idx=-1) at tcp_read.c:1611 #11 0x00000000005d6bce in io_wait_loop_epoll (h=<optimized out>, t=<optimized out>, repeat=<optimized out>) at io_wait.h:1061 #12 tcp_receive_loop (unix_sock=1183856160) at tcp_read.c:1781 #13 0x00000000004d9df7 in tcp_init_children () at tcp_main.c:4788 #14 0x000000000050736f in main_loop () at main.c:1690 #15 0x000000000041cdbc in main (argc=0, argv=0x0) at main.c:2616

2 15

Kamailio Client-Server
by Shiv Patidar 02 May '16

02 May '16

hiii 1.We are writing our own SIP Client to test my Client i am using Kamailio server 2. So i am trying to map the identities used between Client and Server 3. If can possible can u please help me to the steps has to be follower to set identities common between Client and Server. 4. Or if u have some test code for client can u share the SIP message dump for test code so i can re-use the same for configuration my client code. Thank you

3 2

unixODBC failure/bug?
by Nagorny, Dimitry 02 May '16

02 May '16

Hi all, I am trying to use the unixODBC module to connect to MariaDB as a first step and then to test it to connect to MSSQL (please don't blame me on MSSQL...). I am using Kamailio 4.3.5 on CentOS7.2 fully patched. Here info about installed unixODBC: [root@r5dev-KAM1 ~]# yum list installed | grep unix unixODBC.x86_64 2.3.1-11.el7 @base unixODBC-devel.x86_64 2.3.1-11.el7 @base Which leads me to my possible problem that I can't start Kamailio with db_unixodbc.so because it is giving me this error: r5dev-KAM1 /usr/local/kamailio/sbin/kamailio[16307]: ERROR: <core> [db.c:203]: db_bind_mod(): Module db_unixODBC not found. Missing loadmodule? Followed by errors from modules depending on the DBURL are failing as well. Here my modules part: ... #!define DBURL "unixODBC://root:<passw>@localhost/MARIADB" ... # Module Path mpath="/usr/local/kamailio/lib64/kamailio/modules/" loadmodule "tls.so" # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio/etc/kamailio/tls.cfg") #loadmodule "db_mysql.so" loadmodule "db_unixodbc.so" ... I doublechecked that the module was compiled and is present: [root@r5dev-KAM1 ~]# ls -ll /usr/local/kamailio/lib64/kamailio/modules/ | grep unix -rwxr-xr-x. 1 root root 184451 28. Apr 08:56 db_unixodbc.so I tried to check the db.c and db_unixodbc.c but I am not familiar with the c-syntax. I somehow suspect that the error is related to Linux case sensitivity if db.c is trying to check if "unixODBC" is installed and then trying to match it with the "db_unixodbc.so" but I'm am not sure. Can someone please give further advice? Respectfully Dimitry Nagorny Trainee

2 5

SIP Notifications
by Safdar Khan 02 May '16

02 May '16

Hello, How i enable push notification in kamailio.How i proceed to configure the kamailio configuration file and which kamailio module should i look for? Thanks

3 4

dlg_bridge
by huseyin kalyoncu 29 Apr '16

29 Apr '16

Hello I am trying to implement click 2 call functionality on Kamailio with dlg_bridge. But It seems that Kamailio routes REFER messages to devices according to Contact field of location table instead of Received field. But if the the IP in the Contact field is private REFER messages are failing so does the dlg_bridge. I called dlg_bridge in the config like this: dlg_bridge("sip:$param1", "sip:$param2", "sip:IP_ADDRESS_OF_KAMAILIO:5060"); It rings the device in $param1 but after answering on that device Kamailio fails to send REFER if the Contact field for that device is Private IP. So it fails to initiate a call. How can I force Kamailio to use Received field for REFER messages instead of Contact field? Thanks Huseyin

2 11

kamailio stop to print log meesage
by Shiv Patidar 29 Apr '16

29 Apr '16

kamailio.log and syslog stop to print log message i have set debug 4 and also try with debug 2 and debug 3 and kamalio.log and syslog not print any log message what i need to do please help me

2 1

How to run MySQL queries in Kamailio
by Ahmad Fuad 28 Apr '16

28 Apr '16

Dear Support, Kindly find below my Database Table (kamailio.ext). Column names are "ext" and "ip" mysql> select * from ext -> ; +------+----------------+ | ext | ip | +------+----------------+ | 202 | 192.168.55.157 | | 203 | 192.168.55.158 | | 204 | 192.168.55.159 | +------+----------------+ 3 rows in set (0.00 sec) I want to run this type of queries in kamailio. I have tried following command but no luck so far. Although I have added following line in the configuration file as well. Thanks modparam("sqlops", "sqlcon", "cb=>mysql://root:123456@localhost/kamailio") sql_query("cb", "select ip from ext where ext=$rU", "ra"); sl_send_reply("302","$ra"); PLEASE HELP !!! when we see the traces using wireshark it shows internal error. Thanks 275.733635441 192.168.55.144 -> 192.168.55.157 SIP 431 Status: 302 Internal Server Error | -- Best Regards, Ahmad Fuad

2 1

kamailio-radius RPM for Centos
by Antonio Reale 28 Apr '16

28 Apr '16

Hi all, I see that the kamailio-radius rpm package is available only for Opensuse. Is there a particular reason that it is unavailable for Centos? Thanks. Regards. Antonio

2 10

fetch data from Mysql Using Kamailio
by Shahid Mehmood 28 Apr '16

28 Apr '16

Dear All, i am getting error when i trying to fetch data from mysql data base using kamailio. i have following code into kamailio.cfg file loadmodule "db_mysql.so" loadmodule "sqlops.so" ................................ modparam("sqlops","sqlcon","ca=>mysql://root:123456@localhost/kamailio") ............................... ............................... sql_query("ca", "select * from ext", "ra"); sql_result_free("ra"); *Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not availableApr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: sqlops [sqlops.c:276]: fixup_sql_query(): invalid connection [ca]Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:514Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:534* -- Shahid Mehmood

2 4

fetch data from Mysql Using Kamailio
by Shahid Mehmood 28 Apr '16

28 Apr '16

Dear All, i am getting error when i trying to fetch data from mysql data base using kamailio. i have following code into kamailio.cfg file loadmodule "db_mysql.so" loadmodule "sqlops.so" ................................ modparam("sqlops","sqlcon","ca=>mysql://root:123456@localhost/kamailio") ............................... ............................... sql_query("ca", "select * from ext", "ra"); sql_result_free("ra"); *Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not availableApr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: sqlops [sqlops.c:276]: fixup_sql_query(): invalid connection [ca]Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:514Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:534* -- Shahid Mehmood

2 1

Kamailio IMS_QOS
by José Seabra 28 Apr '16

28 Apr '16

Hello There, I'm trying using the ims_qos module with PCRF for QOS, but I'm facing some issues. When kamailio receives an initial invite (originator), it will execute the function - "Rx_AAR("ORIG_SESSION_AAR","orig","",-1)", but the PCRF is not contacted and i see the following messages: 5(12153) DEBUG: <script>: Diameter: Orig authorizing media via Rx 5(12153) DEBUG: ims_qos [mod.c:609]: w_rx_aar(): Looking for route block [ORIG_SESSION_AAR] 5(12153) DEBUG: ims_qos [mod.c:621]: w_rx_aar(): Rx AAR called 5(12153) DEBUG: ims_qos [mod.c:1345]: create_return_code(): Creating return code of [-2] for aar_return_code 5(12153) DEBUG: ims_qos [mod.c:1354]: create_return_code(): created AVP successfully : [aar_return_code] 5(12153) DEBUG: ims_qos [mod.c:627]: w_rx_aar(): Can't do AAR for call session in request. Why kamailio says that I Can't do AAR for call session in request? I have looked into source code and I found that the condition that is blocking the communication with PCRF is: //We don't ever do AAR on request for calling scenario... if (msg->first_line.type != SIP_REPLY) { LM_DBG("Can't do AAR for call session in request\n"); return result; } Anyone can help me understand why it is happening? why it can't proceed if isn't a reply? Thank you for your support BR -- José Seabra

4 15

Call http service inside config , Push notification TSILO
by Sirvan Paraste 27 Apr '16

27 Apr '16

Hello All, I am new to kamailio, but I have setup a server that is ok for call functionalities, I have write an android app it I wnat to add push notification ability to the app when it is clised or unregisterd to the sip server. I have done all push notification part using GCM service and a third party server name Uniqush-push that is connected to GCM to handle push notification, anyway the app and push server part all work well now, The problem is this: kamilio send a http request like this : curl http://127.0.0.1:9898/push -d service=myserviceName -d subscriber=6000 -d msg="Hello World" *given 6000 as a kamilio user,* I ask a nice guy and he replied me by using the TSILO as following link: http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… thanks to him it made the process more easier I think. But I dont know how I can congigured kamailio to call my server as I said above in order to send push. please help me by example because I am new, I searched a lot but I get confused, I will make a tutorial for other people to help them. thanks all

2 2

A: Ongoing process of selecting a new website theme
by Daniel-Constantin Mierla 27 Apr '16

27 Apr '16

Hello, following the last developers' meeting on IRC, we are going to update the website at kamailio.org to use a responsive theme, in order to make it more friendly for mobile browsing. Until one is selected, you may see various forms of website design for short time, as variants are enabled to see how it looks with the real content. All this should not take too long, in 1-2 days the new theme should be in place. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 1

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