- sr-users - kamailio.org

Caller not receiving RTP feed
by Benjamin Trent 23 Dec '13

23 Dec '13

Hey all, I have kamailio set up behind a nat(port restricted, with firewall rules to allow sip transactions and allowing rtpproxy packet handling if needed) on Amazon EC2. I can register and calls complete, however, the Caller(the one initiating the transaction) receives no rtp media feed. I am running with NAT enabled on kamailio and have rtpproxy installed listening on the public IP. Kamailio and the rtpproxy are communicating(I have verified via the kamailio debug logs). If I make a call between the exact same voip machines directly via local IP on the same Nat(skipping kamailio), the calls complete and they both receive feeds. Both the Caller(party making the call) and the Callee(party receiving the call) are behind a Port Restricted Nat. This is a folder containing the debug output for one of these calls and the kamailio.cfg settings https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sha… Quick FYI, the Caller Display Name and the Callee SIP UserName are the same string. However, in my understanding about sip, the display name means pretty much nothing and is just a human readable string that does not effect packet transport. If I am wrong and should test with a different display name, let me know. Thank you for the assistance, ben

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Kamailio 4.1, Debian Squeeze, RTPProxy 1.2.1 Segfault
by Jr Richardson 20 Dec '13

20 Dec '13

Hi All, root@sip-router3-ve206:/etc/kamailio# kamailio -V version: kamailio 4.1.0 (x86_64/linux) 350d2e Doing some testing, can't seem to get rtpproxy to not segfault. I've loaded version from deb-squeeze pkg, from source http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz and from git:// sippy.git.sourceforge.net/gitroot/sippy/rtpproxy (which should be the latest) but all versions segfault as soon as a call sets up. I've configured each versions control socket for both udp or unix, kamailio starts and sees the rtpproxy fine with no errors, but when a call hits, rtpproxy segfaults. I've run rtpproxy in the forground with degug and get this response (IP's washed x.x.x): ------------------ DBUG:handle_command: received command "7505_12 USIEc0,101 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76 x.x.x.76 23388 as33dd7c98;1" INFO:handle_command: new session 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76, tag as33dd7c98;1 requested, type strong Segmentation fault ------------------ root@sip-router3-ve206:/etc/kamailio# more /etc/default/rtpproxy # Defaults for rtpproxy # The control socket. CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: #CONTROL_SOCK="udp:127.0.0.1:7722" # Additional options that are passed to the daemon. EXTRA_OPTS="-l x.x.x.20" kamailio.cfg: loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") inside route[] add_path_received(); rtpproxy_manage("cwei"); record_route(); Any guidance on further identifying the issue? Thanks. JR -- JR Richardson Engineering for the Masses

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Kamailio v4.0.5 Released
by Daniel-Constantin Mierla 20 Dec '13

20 Dec '13

Hello, Kamailio SIP Server v4.0.5 stable release is out. This is a maintenance release of the stable branch 4.0, that includes fixes since release of v4.0.4. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.5. For more details about version 4.0.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/12/kamailio-v4-0-5-released/ Note that 4.0.x is previous stable release series, the latest one is now 4.1.x, with v4.1.0 released about two weeks ago. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Caller Not Receiving RTP Feed
by Benjamin Trent 20 Dec '13

20 Dec '13

Hey all, **I apologize if this is a duplicate, I received a bounce back on my first email. I have kamailio set up behind a nat(port restricted, with firewall rules to allow sip transactions and allowing rtpproxy packet handling if needed) on Amazon EC2. I can register and calls complete, however, the Caller(the one initiating the transaction) receives no rtp media feed. I am running with NAT enabled on kamailio and have rtpproxy installed listening on the public IP. Kamailio and the rtpproxy are communicating(I have verified via the kamailio debug logs). If I make a call between the exact same voip machines directly via local IP on the same Nat(skipping kamailio), the calls complete and they both receive feeds. Both the Caller(party making the call) and the Callee(party receiving the call) are behind a Port Restricted Nat. This is a folder containing the debug output for one of these calls and the kamailio.cfg settings https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sha… Quick FYI, the Caller Display Name and the Callee SIP UserName are the same string. However, in my understanding about sip, the display name means pretty much nothing and is just a human readable string that does not effect packet transport. If I am wrong and should test with a different display name, let me know. Thank you for the assistance, ben

2 2

ack not transfer kamailio-3.3.4
by dolphin 20 Dec '13

20 Dec '13

hello, I’m trying to build two kamailio-3.3.4 server like this: ua1 ----> kamailio1(k1)-----> kamailio2(k2) -----> ua2 ua1 (5978003 ) is register k1 ua2 (5001005) is register k2 When ua1 callout to ua2 sip process through k1 and k2 The problem is k2 server ACK not transfer to ua2 and k2 has received ack signals from u1 through k1 I ngrep sip process for k2 and k2 /var/log/debug see the attached File ngrep_k2server.txt, debug_k2 1. First ACK uri have ua2 number sip:5001005@xxx.xxx.xxx.xxx(k2):5003;ob , is correct !! 2. Second ack not include 5001005 , again loop itself ; Route: <sip:xxx.xxx.xxx.xxx(k2):5003;lr;did=763.941> loose_route Next hop: 'sip:xxx.xxx.xxx.xxx(k2):5003;lr;did=763.941' is loose router. so next hop Not found in usrloc db 5001005 user and not transfer to ua2 How to fix this problem, any suggestions ?

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how to detect un-REGISTER
by Henry Fernandes 20 Dec '13

20 Dec '13

Is there a Kamailio module/function to detect if a REGISTER request is for unregistering? -H

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Kamailo - Problems with RTP streaming.
by Wingsravi R 20 Dec '13

20 Dec '13

Dear All, I am running a kamailio 4.0.3 server, it was working fine with the features like- audio calling, video calling , SMS (but presence is not working). But from last two days I’m facing problems with RTP packets streaming between SIP clients (IMSDroid clients). Session establishment between two clients works well but without audio and video streaming. And after sometime again it works well. What is this wierd behaviour? Following attachments are Audio calling cases of working and not working (ngrep based), And i have found some errors in Syslog (attached) and my config file for any reference. What could be the problem? How can i solve the issue? Anybody please help me. Regards,

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planning release of v4.0.5
by Daniel-Constantin Mierla 20 Dec '13

20 Dec '13

Hello, I am considering to release a new version out of branch 4.0 (previous stable branch) before Christmas holidays. Depending on the available time, it could be this Friday or next week on Monday. If there are patches to be backported, do them as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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ACC issue on 4.0 trunk when doing serial fork
by Kelvin Chua 20 Dec '13

20 Dec '13

this is the 2nd branch, the first branch is sent to 100.200.30.40 $rd is set properly according to xlog (10.20.30.40). tcpdump proves packets are sent to 10.20.30.40 however, after failing, ACC logs dst_domain as the previous branch's $rd (100.200.30.40), not the current one. is this the expected behavior? seems incorrect. modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;callrate=$avp(callrate)") failure_route[SERIAL] { if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if ( isbflagset(FLB_CANCEL) ) { xlog("cancel bflag set!\n"); exit; } else if ( !t_next_contacts() ) { xlog("serial fork done!\n"); t_reply("404", "Not here or busy"); exit; } else { setflag(FLT_ACC); setflag(FLT_ACCMISSED); xlog("failure_route new branch - $ru\n"); t_on_failure("SERIAL"); $avp(callrate) = ""; if ( $sel(ruri.params["rtmr"]) != $null ) { t_set_fr($sel(ruri.params["rtmr"])*1000,$sel(ruri.params["dtmr"])*1000); $avp(ABSOLUTE) = $sel(ruri.params["tmr"]); } if ( $sel(ruri.params["callrate"]) != $null ) $avp(callrate) = $sel(ruri.params["callrate"]); xlog("failure_route sending INVITE to $rd\n"); t_relay(); } } ERROR: <script>: failure_route sending INVITE to 10.20.30.40 DEBUG: <script>: new branch [1] to sip:14X88X72X2X@10.20.30.40:5060 ;dtmr=1;rtmr=20;tmr=60;callrate=0.007600000000 DEBUG: siputils [checks.c:103]: has_totag(): no totag DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid DEBUG: <core> [parser/sdp/sdp_helpr_funcs.c:565]: extract_mediaip(): located IP address [192.168.5.191] in `o=' field DEBUG: <core> [parser/sdp/sdp_helpr_funcs.c:565]: extract_mediaip(): located IP address [192.168.5.191] in `c=' field DEBUG: rtpproxy [rtpproxy.c:2630]: force_rtp_proxy(): proxy reply: 10370 2X8.1X1.3X.2X#012 DEBUG: siputils [checks.c:103]: has_totag(): no totag DEBUG: <core> [msg_translator.c:461]: clen_builder(): clen_builder: content-length: 172 (172) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(1X5.2X2.1X7.1X0, 192.168.5.191, 0) DEBUG: <core> [parser/sdp/sdp.c:793]: free_sdp(): _sdp = 0x7f53c9a5e580 DEBUG: <core> [parser/sdp/sdp.c:795]: free_sdp(): sdp = 0x7f53cb87d3e0 DEBUG: <core> [parser/sdp/sdp.c:797]: free_sdp(): session = 0x7f53cb87d480 DEBUG: tm [t_reply.c:1710]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=-1 DEBUG: tm [t_reply.c:1211]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=408 DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f53c1511148, callback type 128, id 0 entered DEBUG: acc [acc_logic.c:539]: tmcb_func(): acc callback called for t(0x7f53c1511148) event type 128, reply code 408 NOTICE: acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1387019197;method=INVITE;from_tag=wyudf;to_tag=;call_id=cjfbdfydrjielti@X340saucy;code=408;reason=Request Timeout;src_user=kelvin;src_domain=2X8.1X1.3X.2X;src_ip=1X5.2X2.1X7.1X0;dst_ouser=14X88X72X2X;dst_user=14X88X72X2X;dst_domain=100.200.30.40;callrate=0.007600000000 Kelvin Chua

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memory debug
by Victor V. Kustov 20 Dec '13

20 Dec '13

Hi, all debug=2 memdbg=0 memlog=0 mem_summary=0 i saw it in kamcmd cfg.get... but anyway q_malloc messages in log. where i make mistake? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.10.24-calculate GNU/Linux

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Kamailio v3.3.6 Released
by Daniel-Constantin Mierla 19 Dec '13

19 Dec '13

Hello, Kamailio SIP Server v3.3.6 stable release is out. This is a maintenance release of the previous stable branch, 3.3, that includes fixes since release of v3.3.5. There is no change to database schema or configuration language structure that you have to do on installations of previous v3.3.x. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.6 or a newer stable release. For more details about version 3.3.6 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/12/kamailio-v3-3-6-released/ Be aware that this is intended to be the last version released from branch 3.3. Note that latest stable series is now 4.1.x, with v4.1.0 released about two weeks ago. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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timers
by Kelvin Chua 19 Dec '13

19 Dec '13

just a quick question, somebody out there might have an idea: currently, we can easily control the ff: Time from INVITE to 200 OK ------ fr_inv_timer Time from INVITE to 100 TRYING ------ fr_timer what i hope to achieve is: Time from Session Progress (180/183) to 200 OK Time from INVITE to Session Progress (180/183) is there any hack to do this? Kelvin Chua

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failure in accessing r-uri params
by Juha Heinanen 19 Dec '13

19 Dec '13

i started to wonder, why my config does not work when there is ';user=phone' param in r-uri. in order to investigate, i added these to the config: xlog("L_INFO", "r-uri = '$ru'\n"); xlog("L_INFO", "r-uri params = '$(ru{uri.params})'\n"); xlog("L_INFO", "r-uri params = '$sel(ruri.params)'\n"); and made two tests. in the first, r-uri param is ;user=phone: Dec 19 07:08:10 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri = 'sip:+35862345670@test.tutpro.com;user=phone' Dec 19 07:08:10 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri params = '' Dec 19 07:08:10 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri params = '<null>' and in the second r-uri param is ;param=value: Dec 19 07:09:59 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri = 'sip:+35862345670@test.tutpro.com;param=value' Dec 19 07:09:59 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri params = 'param=value' Dec 19 07:09:59 rautu /usr/sbin/sip-proxy[5107]: INFO: r-uri params = 'param=value' note the difference. in the first, neither uri transformation nor ruri select works, but they work in the second. i checked from rfc3261 and both request uris are syntactically valid and in both cases there is a valid r-uri parameter. my current hack is to use tobody.params, but would like to get rid of it. -- juha

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planning last release for 3.3 branch
by Daniel-Constantin Mierla 19 Dec '13

19 Dec '13

Hello, the last two stable branches are now 4.0 and 4.1. I am considering packaging the latest branch 3.3 as release 3.3.6 and mark the end of releasing from that branch. Patches can still go there if developers find time for them, but the focus on release will go on 4.0 and 4.1. Hopefully I will find some time tomorrow afternoon for it. If anyone has patches to backport, do it before noon GMT tomorrow, or if there is something that should go there and is no time for it now, reply here with description and eventually propose a new date (or when you think what you want to do can be finished). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

broken ims modules links in online doc
by Andreas Granig 19 Dec '13

19 Dec '13

Hi, Seems like two ims module links (linked from http://kamailio.org/docs/modules/4.1.x/) are broken: http://kamailio.org/docs/modules/4.1.x/modules/ims_registrar_pcscf.html http://kamailio.org/docs/modules/4.1.x/modules/ims_usrloc_scscf.html The same for http://kamailio.org/docs/modules/devel/, as far as I can see. Thanks for checking! Andreas

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call limit by trunk/user
by Oliver Roth 18 Dec '13

18 Dec '13

Hi all I need a solution to limit the amount of concurrent calls by trunk or user/subscriber. We have the following situation: Kamailio loadbalancer ==> 3 kamailio gateway with routing/business logic Kamailio is version 3.3 for all 4 systems. Loadbalancer has only the absolute minimum config to act as loadbalancert, the whole business logic is on gateway side (db connections, carrierroute, ...) Do you have a suggestion how to limit calls/connections? Has this to be done on the loadbalancer? Or is there a solution to do this on the gateways - so every gateway knows the currently opened connections? Thanks for helping Oli

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Kamailio Admin Book Update?
by JR Richardson 18 Dec '13

18 Dec '13

Hi Daniel, What is the status and possible release date for your admin book. The last update is it will be released as an e-Book only, that's ok with me. Thanks. JR JR Richardson - - - - - - - - - - - Engineering for the Masses

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I-cscf module not reading correctly UserAuthenticationAnswer
by Gennaro Sarnataro 18 Dec '13

18 Dec '13

Hello everyone, I'm using kamailio modules as Interrogating-cscf and Serving-cscf. Registering an user i perform correctly an UAR and the HSS answers correctly with an UAA giving user'c server-capabilities. Even if i have a success result code ( as you can see in the attached wireshark dump) the configuration code bring me in the failure branch. route[REG_UAR_REPLY] > { > xlog("L_INFO", "FB: received uaa"); > xlog("L_INFO", "<<<<<<<<<<<<<<<< [$avp(s:uaa_return_code)]"); > switch ($avp(s:uaa_return_code)){ > case 1: #success > xlog("L_INFO", "FB:success"); > if (I_scscf_select("0")){ > t_on_failure("register_failure"); > t_on_reply("register_reply"); > #now relay to appropriate SCSCF > if (!t_relay()) { > t_reply("500", "Error forwarding to > SCSCF"); > } > } else {#select failed > I_scscf_drop(); > t_reply("500", "Server error on SCSCF Select > (UAR)"); > } > break; > case -1: #failure > xlog("L_INFO", "failure"); > xlog("L_ERR", "UAR failure - error response sent from > module\n"); > break; > case -2: #error > xlog("L_INFO", "error"); > xlog("L_ERR", "UAR error - sending error response now\n"); > t_reply("500", "UAR failed"); > break; > default: > xlog("L_ERR", "Unknown return code from UAR, value is > [$avp(s:uaa_return_code)]\n"); > t_reply("500", "Unknown response code from UAR"); > break; > } > } as you can see from my log i think tre's some problem when reading UserAuthenticationAnswer return code. Dec 17 12:31:40 localhost /usr/sbin/kamailio[28053]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=325 a=24 n=I_scscf_drop > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28053]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=327 a=26 > n=I_perform_user_authorization_request > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28053]: INFO: ims_icscf > [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : > [uaa_return_code] > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: ims_icscf > [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: ims_icscf > [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed > finding avp > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: ims_icscf > [cxdx_uar.c:211]: async_cdp_uar_callback(): Empty capability list returning > 600 > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: ims_icscf > [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : > [uaa_return_code] > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=336 a=26 n=xlog > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: <script>: FB: > received uaa > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=337 a=26 n=xlog > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: <script>: > <<<<<<<<<<<<<<<< [18446744073709551615] > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=366 a=21 n=switch > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=354 a=26 n=xlog > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: INFO: <script>: > failure > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=355 a=26 n=xlog > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: <script>: UAR > failure - error response sent from module > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: ERROR: *** cfgtrace: > c=[/home/voiceeng/kamailio/i-cscf/kamailio.cfg] l=356 a=2 n=exit > Dec 17 12:31:40 localhost /usr/sbin/kamailio[28045]: WARNING: tm > [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release > transaction I hope you will clarify this matter... thanks in advance for your help -- Gennaro Sarnataro

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Re: [SR-Users] memory
by Keith 17 Dec '13

17 Dec '13

Hi, Yeah sorry, I had to restart it as it's a production box. I have the log file which I can show you? Cheers Keith

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Re: [SR-Users] Memory
by Keith 17 Dec '13

17 Dec '13

Hi, Here are the details: Version: 4.02 *kamcmd pkg.stats:* { entry: 0 pid: 15738 rank: 0 used: 1521632 free: 2504528 real_used: 1689776 } { entry: 1 pid: 15745 rank: 1 used: 1530112 free: 2495248 real_used: 1699056 } { entry: 2 pid: 15746 rank: 2 used: 1529920 free: 2496240 real_used: 1698064 } { entry: 3 pid: 15747 rank: 3 used: 1529920 free: 2496240 real_used: 1698064 } { entry: 4 pid: 15748 rank: 4 used: 1529920 free: 2496240 real_used: 1698064 } { entry: 5 pid: 15749 rank: 5 used: 1530192 free: 2494816 real_used: 1699488 } { entry: 6 pid: 15751 rank: 6 used: 1530208 free: 2494768 real_used: 1699536 } { entry: 7 pid: 15753 rank: 7 used: 1530192 free: 2494848 real_used: 1699456 } { entry: 8 pid: 15754 rank: 8 used: 1530192 free: 2494848 real_used: 1699456 } { entry: 9 pid: 15756 rank: 9 used: 1555456 free: 2458096 real_used: 1736208 } { entry: 10 pid: 15757 rank: 10 used: 1557328 free: 2455264 real_used: 1739040 } { entry: 11 pid: 15758 rank: 11 used: 1557216 free: 2455536 real_used: 1738768 } { entry: 12 pid: 15759 rank: 12 used: 1555952 free: 2457632 real_used: 1736672 } { entry: 13 pid: 15760 rank: -1 used: 2891024 free: 2496064 real_used: 3029088 } { entry: 14 pid: 15761 rank: -1 used: 2879152 free: 2496096 real_used: 3017216 } { entry: 15 pid: 0 rank: 0 used: 1539584 free: 2486512 real_used: 1707792 } { entry: 16 pid: 15776 rank: -2 used: 1532704 free: 2493424 real_used: 1700880 } { entry: 17 pid: 15778 rank: -1 used: 1532224 free: 2493872 real_used: 1700432 } { entry: 18 pid: 15779 rank: -1 used: 1532032 free: 2494096 real_used: 1700208 } { entry: 19 pid: 15782 rank: 13 used: 1610320 free: 2415744 real_used: 1778560 } { entry: 20 pid: 15784 rank: 14 used: 1610320 free: 2415744 real_used: 1778560 } { entry: 21 pid: 15786 rank: 15 used: 1610320 free: 2415744 real_used: 1778560 } { entry: 22 pid: 15788 rank: 16 used: 1610320 free: 2415744 real_used: 1778560 } { entry: 23 pid: 15790 rank: -4 used: 1603056 free: 2423008 real_used: 1771296 } *kamcmd stats* core:bad_URIs_rcvd = 0 core:bad_msg_hdr = 0 core:drop_replies = 0 core:drop_requests = 50 core:err_replies = 0 core:err_requests = 0 core:fwd_replies = 0 core:fwd_requests = 352 core:rcv_replies = 5181 core:rcv_requests = 5522 core:unsupported_methods = 0 dialog:active_dialogs = 34 dialog:early_dialogs = 0 dialog:expired_dialogs = 0 dialog:failed_dialogs = 60 dialog:processed_dialogs = 272 dns:failed_dns_request = 0 mysql:driver_errors = 0 registrar:accepted_regs = 103 registrar:default_expire = 3600 registrar:default_expires_range = 0 registrar:max_contacts = 0 registrar:max_expires = 3600 registrar:rejected_regs = 0 shmem:fragments = 1633 shmem:free_size = 64059008 shmem:max_used_size = 3755584 shmem:real_used_size = 3049856 shmem:total_size = 67108864 shmem:used_size = 2911792 siptrace:traced_replies = 0 siptrace:traced_requests = 0 sl:1xx_replies = 0 sl:200_replies = 616 sl:202_replies = 0 sl:2xx_replies = 0 sl:300_replies = 0 sl:301_replies = 0 sl:302_replies = 0 sl:3xx_replies = 0 sl:400_replies = 0 sl:401_replies = 81 sl:403_replies = 0 sl:404_replies = 0 sl:407_replies = 54 sl:408_replies = 0 sl:483_replies = 0 sl:4xx_replies = 0 sl:500_replies = 0 sl:5xx_replies = 0 sl:6xx_replies = 0 sl:failures = 0 sl:received_ACKs = 50 sl:sent_err_replies = 0 sl:sent_replies = 751 sl:xxx_replies = 0 sst:expired_sst = 0 tcp:con_reset = 0 tcp:con_timeout = 0 tcp:connect_failed = 0 tcp:connect_success = 0 tcp:current_opened_connections = 0 tcp:current_write_queue_size = 0 tcp:established = 0 tcp:local_reject = 0 tcp:passive_open = 0 tcp:send_timeout = 0 tcp:sendq_full = 0 tmx:2xx_transactions = 4543 tmx:3xx_transactions = 0 tmx:4xx_transactions = 795 tmx:5xx_transactions = 110 tmx:6xx_transactions = 0 tmx:UAC_transactions = 2365 tmx:UAS_transactions = 5449 tmx:active_transactions = 1 tmx:inuse_transactions = 8 tmx:local_replies = 1317 usrloc:location-contacts = 35 usrloc:location-expires = 2 usrloc:location-users = 35 usrloc:registered_users = 35 Thanks!

2 1

Calling end_media_session twice
by Teddy Rabary 17 Dec '13

17 Dec '13

Hi all, I see this discussion about mediaproxy and it's the same what happens to me http://opensips.org/pipermail/users/2010-August/013997.html Indeed in my kamailio.cfg, I call twice end_media_session on "bye" and on "cancel" route[CANCEL] { if (is_method("CANCEL")) { end_media_session(); if (t_check_trans()) t_relay(); exit; } } and route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { end_media_session(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(s:can_uri) = $ru; } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { ... I thought both are necessary, so can I make something like this at route[WITHINDLG] ? ... if (is_method("BYE|CANCEL")) { end_media_session(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(s:can_uri) = $ru; } ... and then remove end_media_session at route[CANCEL] ? Regards.

2 1

Memory
by Keith 16 Dec '13

16 Dec '13

Hi, Not sure if this is a bug or config but getting this error message: insert_subst_lump_after(): ERROR: insert_new_lump_after: out of memory Any ideas? Keith

2 1

S-CSCF strange execution effects
by Flavio Battimo 16 Dec '13

16 Dec '13

Hello, i'm encountering in some strange behaviours while working with S-CSCF version: kamailio 4.1.0 (x86_64/linux) 350d2e Making some tests, in particular with Ro interface for online charging, the script execution doesn't follow normal flow (checking it by adding some xlog call). Looking at the following code script, i have made some modifications: 1) Modify the P-Asserted-Identity so as to add angle brackets to the value. 2) Add user=phone parameter to uri if it doesn't exists 3) Enabled RO interface for accounting The amazing thing is that if kamailio is executed with this script the whole flow terminates without any notice just after record_route() call. Making some attempts, removing record_route() call flow continues until Ro_CCR method, where same behaviour appears (call flow is unpredictable terminated and no diameter CCR request is made) Which kind of operations can lead to this strange behaviour If is needed i can attach some logs in debug level mode Thanks in advance ###################################################################### # Originating, Intial Requests ###################################################################### route[orig] { xlog("L_DBG","Enter orig route\n"); # modifica di P-Asserted-Identity subst('/P\-Asserted\-Identity:[ ]*([\+0-9a-z@\.\-\:]*)/P-Asserted-Identity: <\1>\r\n/ig'); if ($ru =~ ".*;user=phone.*" ){ xlog("L_DBG","Parameter user=phone already exists"); } else { xlog("L_DBG","Adding parameter user=phone"); $ru = $ru + ";user=phone"; } xlog("L_DBG","$rm $ru"); set_dlg_profile("orig"); # we MAYBE need something like this to check if a user is barred # if (S_originating_barred()){ # sl_send_reply("403","Forbidden - Originating Public Identity barred"); # exit; # } xlog("L_DBG","Checking AVP mo custom\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_DBG","Adding AVP mo custom parameter\n"); $avp(RR_CUSTOM_USER_AVP)="mo"; xlog("L_DBG","call record_route\n"); record_route(); } #!ifdef WITH_RO # before we allow call - lets check credit xlog("L_DBG","Checking if RO must be called\n"); if (is_method("INVITE")) { xlog("L_DBG","Sending initial CCR Request for call\n"); Ro_CCR("CHARGING_CCR_REPLY", "orig", "SCUR", "", "30"); } #!else xlog("L_DBG","calling route[finalize_orig]"); route(Finalize_Orig); #!endif } -- Flavio Battimo flavio.battimo(a)gmail.com fbattimo(a)meetecho.com skype: flaviobattimo

3 2

500 I'm terribly sorry error
by Helena Garcia-Nieto 16 Dec '13

16 Dec '13

Hello, Thanks in advanced for the help. I am almost new with kamailio and still struggling through silly problems so please forgive me if the solution is so obvious. I have a network like Xlitle -- Kamailio -- GW The GW is more or less out of my reach for changing the behaivour. As devices I have xlitle Kamailio is on version 4.0.2 I've changed only few things from the default config file. Add mysql support, auth, userlocdb, pstngw. For this part, gw routing , I've defined gw ip and port inside the PSTN definition like: #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" iskratel.gw_ip = "10.XX.XX.XX" iskratel.gw_port = "5060" #!endif I route the calls with: route(ISKRATEL); And defined a routing function route[ISKRATEL] { #!ifdef WITH_PSTN # check if ISKRATEL GW IP is defined if (strempty($sel(cfg_get.iskratel.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but iskratel.gw_ip not defined\n"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.iskratel.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.iskratel.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.iskratel.gw_ip) + ":" + $sel(cfg_get.iskratel.gw_port); } # Add profix to ISKRATEL: A99901 subst_uri('/^sip:(.*)/sip:A99901\1/i'); # add A99901 route(RELAY); exit; #!endif return; } I can see that the code goes all the way to the route(relay) but all I can see is the 500 I'm terribly sorry, server error occurred (7/SL) and a second response 500 I'm terribly sorry, server error occurred (7/TM) I've tried defining the IP as the given pstn.gw_ip and route(pstn) without changing anything on the pst default routing but the response of the server is the same. I cannot see any special error on the logs. I have the exact same config for this part, in another test server and the calls go to the gw without reporting any error. I'll appreciate any help from you! Thanks in advanced for your time Helena

3 8

Trying to understand the interal workings of Dispatcher and timers
by Grant Bagdasarian 16 Dec '13

16 Dec '13

Hello, I'm trying to understand how the dispatcher module works and how the timers in the TM module influence the behavior. My architecture looks like this: UA1 ---- SBC1 ---- Kamailio ----- SBC2 ---- FreeSWITCH App Server(x2) What I got so far: 1) UA1 sends an INVITE to SBC1, which bridges the call between the UA1 and Kamailio 2) INVITE is received by Kamailio. 3) Kamailio looks in its dispatcher table to dispatch the INVITE to its destination 4) Kamailio rewrites the $du value to that of the SBC2 5) Kamailio rewrites the request-uri and sends the INVITE to the destination in $du. 6) SBC2 is a FreeSWITCH instance which bridges the call (does not answer). It only responds back with a 100 TRYING 7) SBC2 tries to contact the endpoint (FS App Server 1) in the request-uri, but the destination does not respond because it is unavailable. 8) Kamailio keeps waiting for a final reply. 9) The fr_inv_timer is set to 5000 on startup 10) Once the timer hits, Kamailio executes the route set in t_failure_route 11) The failure route looks like this: if(t_is_expired()) { ... } if (t_is_canceled()) { ... } if(t_check_status("500") || t_branch_timeout() || !t_branch_replied()) { .... } 12) Kamailio enters the third if block in which the next domain is selected and the $du is rewritten again to that of SBC2 13) A new INVITE is generated and sent to its destination (FS App Server 2) through SBC2. Nothing is sent back to SBC1 yet, because Kamailio handles the failover here. Is this correct? Are there any other timers which are fired which I missed, or does the fr_inv_timer have nothing to do with this? If Kamailio is supposed to load balance between FS App Server 1 and 2 using the Round-Robin algorithm, does it in case of an unreachable first destination take the next destination in the route set? Regards, Grant

2 1

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