- sr-users - kamailio.org

releasing v4.0.2
by Daniel-Constantin Mierla 12 Jun '13

12 Jun '13

Hello, I am considering to release kamailio v4.0.2 next week on Wednesday, June 12. Developers that committed fixes on master branch and should be backported, have to do it before. Overall, everyone should keep an eye and be sure their demanded fixes are backported. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 4

Need help in setting up Presence
by SamyGo 12 Jun '13

12 Jun '13

Hi List, I've been trying to make the Presence thing work with kamailio but the very basic presence doesn't seem to work. I've tried multiple modules and different how-tos for running successful presence aware configuration but seems something is missing. The ultimate goal is to give user online/busy/offline status updaes to Subscribers. My Kamailio version is: kamailio 4.0.1 (i386/linux) 55f7de I've loaded the following modules. loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo" loadmodule "pua.so" #loadmodule "sca.so" loadmodule "pua_dialoginfo.so" loadmodule "pua_usrloc.so" loadmodule "pua_reginfo" The Presence route contains this: route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; xlog("L_INFO", "[$fU@$si:$sp]{$rm} In Presence Route \n"); #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } Then I've my x-lite phone (which is known to work with Presence in Asterisk) tries to register and subscribe to its own extensions. Nothing happens. The trace from sipgrep is attached. Please help me in making presence work. Thanks, Sammy

4 10

Re: [SR-Users] Missed calls notification supporting
by sipatse＠in.gr 11 Jun '13

11 Jun '13

3 2

How to iterate comma separated string
by Mino Haluz 11 Jun '13

11 Jun '13

Hi, I have to iterate string like 1,2,3,4 in kamailio. How could you do this? I know we have s.select,index,separator but I dont know how to go through all elements. Thanks, Mino

3 2

Kamailio/Asterisk combination + hashed passwords?
by Daniel Pocock 11 Jun '13

11 Jun '13

I was just looking over: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb A couple of things I noticed: - Kamailio is using a column sippasswd which is not hashed. Asterisk doesn't use that column at all. Is there any reason this can't be done with the H(A1) and H(A1b) columns? The INSERT example shows a non-encrypted password. - Is it all considered valid for Kamailio 4 and Asterisk 11? (maybe a disclaimer could be added at the top) - The Asterisk columns `md5secret' and `secret' are left empty so that Asterisk won't challenge. I believe there are other ways of doing this: for example, telling Kamailio to be the registrar and forcing Asterisk to use outbound proxy mode. I managed to make this work against repro - Asterisk no longer receives any REGISTER messages, but all INVITEs go through Asterisk, so the double-challenge problem only arises for INVITEs. Maybe Asterisk can be told that Kamailio's source IP:port is `trusted' and doesn't need to be challenged - is anybody aware of such an option in Asterisk?

5 11

Modify the Request Line
by Klaus Darilion 10 Jun '13

10 Jun '13

Hi! Is there some method to modify the request line, eg. subst/replace ...? AFAIK the textops functions only work starting with the first header line. thanks Klaus

2 2

is using Gerrit for Kamailio a good idea?
by Victor Seva 10 Jun '13

10 Jun '13

Hello fellow Kamailio users and developers, I would like to throw the idea of use Gerrit[0] for code contributions reviews. This is not a replacement for the bugtracker, Gerrit will help developers at the task of review and merge contributors patches. I'm going to use mainly [1] to try to explain what Gerrit is: "Gerrit is a web-based tool that is used for code review. Its main features are the side-by-side difference viewing and inline commenting which makes code reviews quick and simple task. It is used together with Git version control system. Gerrit allows authorized contributors to submit changes to Git repository, after reviews are done. Contributors can get their code reviewed with a little effort, and get their changes quickly through the system." It has a nice interface to review the different patch versions of the change, you can comment any line of code easily and the change has to be approved to be merged. I would like to know your opinion on this subject and your experience using it if you have it. Cheers, Victor [0] http://gerrit-documentation.googlecode.com/svn/Documentation/2.6/intro-quic… [1] http://qt-project.org/wiki/Gerrit-Introduction

3 2

Problems with Siremis Register Page
by Alexander Albert 10 Jun '13

10 Jun '13

Hi there, i hope you can help me, i am having trouble with the register page of siremis, strange thing is, i can open the register page, but only after i have logged my self in, if not, it redirects me directly to the login page.. how can i setup the register page to be accessible without the need of logging in ? Also i am getting the info after trying to register, that "Public registration is not enabled!" but i have set siremis/modules/ser/config/common.Main.php to $cfg_siremis_public_registrations = true; What is the problem, please help. Greetings, Alex

2 1

Re: [SR-Users] Problem with ACK
by phillman25 10 Jun '13

10 Jun '13

Hello Daniel Thanks for your response, can you explain to me how to apply this latest branch? Thanks again! Phillip Hello, ok, apparently there was an issue introduced by addition of outbound handling in rr module affecting strict routing. It was fixed in trunk by Peter Dunkley but not backported. I just did it: - http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=6eb6d92… Can you try with latest branch 4.0? It will be part of 4.0.2 which is planned for mid of next week. Cheers, Daniel

3 3

Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)
by Thomas Martin 10 Jun '13

10 Jun '13

Hello, being new to Kamailio, I have been closely following Daniel's tutorial "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database", hoping to be able to setup a basic SIP server with voicemail-boxes attached to the the accounts. (Eventually, UACs that live behind either port-restricted or symmetric NATs should be supported (i.e. experience two way audio), so, I guess, eventually "WITH_NAT" needs to be defined and rtpproxy needs to be installed as described in the "Run your own Skype-like service in less than one hour" tutorial... ?) Anyway, I am encountering a couple of difficulties in an early stage and hope for some insights form the experts: 1) Even though launching Kamailio with "/etc/init.d/kamailio start" seems to succeed ... root@jm1:~# /etc/init.d/kamailio start Starting Kamailio: loading modules under /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 178.254.20.156:5060 tcp: 127.0.0.1:5060 tcp: 178.254.20.156:5060 Aliases: tcp: rv1192.1blu.de:5060 tcp: localhost:5060 udp: rv1192.1blu.de:5060 udp: localhost:5060 *: jingle-me.org:* kamailio started. root@jm1:~# ... SIP Clients (I have tried several) cannot register and receive the response: 401 Unauthorised. cat /var/log/syslog yields: Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr [rr_mod.c:159]: outbound module not available Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: usrloc [hslot.c:53]: locks array size 512 Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376 Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376 Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2200]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) I suspect that the "INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled" hints to the problem, but have no clue how to fix that. 2) Kamailio does not start on boot-up, even though "/etc/init.d/kamailio start (stop/restart/status)" work(s), and /etc/default/kamailio looks like this: RUN_KAMAILIO=yes USER=kamailio GROUP=kamailio SHM_MEMORY=64 PKG_MEMORY=4 SSD_SUID=no DUMP_CORE=no Right after a re-boot, root@jm1:~# chkconfig kamailio yields: "kamailio on", however, root@jm1:~#/etc/init.d/kamailio status yields: "Status of Kamailio: kamailio is not running." at this time. Any hints are appreciated! Best regards, -Thomas

4 5

Missed calls notification supporting
by sipatse＠in.gr 10 Jun '13

10 Jun '13

2 1

Strict routing configuration
by julian arsanches 10 Jun '13

10 Jun '13

If anyone can help me on this one thanks before hand, i have a setup on kamilio and everything has been working fine, i need to interface with a huge carrier and they don't like the way kamailio sends the invites so i need to implement stric routing . now when kamailio sends an invite it goes like +0.000140 Kamailio:5060 -> CarrierIP:5060 INVITE sip:XXXXXX@Kamailio IP SIP/2.0. Record-Route: <sip:Kamailio IP;lr=on;ftag=as4640f260<sip:54.236.192.19;lr=on;ftag=as4640f260> >. Via: SIP/2.0/UDP Kamailio IP:5060;branch=z9hG4bK9156.eeb6fbf7.0. Via: SIP/2.0/UDP Kamailio IP:5060;branch=z9hG4bK7aeceb75;rport=5060. Max-Forwards: 16. To: sip:XXXXXX@KAMAILIO IP(a)54.236.192.19<sip:85630573108863792@54.236.192.19@54.236.192.19> . and they wanted to receive it like this INVITE sip:XXXXXX@CARRIER IP;user=phone SIP/2.0. Via: SIP/2.0/UDP KAMAILIO IP:5062; From: <sip:XXXXXX@KAMAILIOIP:5060;user=phone>;tag=514594693-3792808143-4087363760-57672641. To: <sip:XXXXXXXX@CARRIERIP;user=phone>. Call-ID: 8517ac46cfa811e2b038a0f3c1037003(a)204.2.202.12. CSeq: 1 INVITE. I dont know how to achieve this in kamailio, i am working fine with several carriers using loose routing here is my config any help is greatly apreciated debug=4 log_stderror=yes #!else #debug=2 #log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes children=6 auto_aliases=no #listen=udp:KAMAILIO IP listen=udp:KAMAILIO IP:5060 advertise EXTERNALIP:5060 port=5060 enable_tls=no #tcp_connection_lifetime=3605 #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #path="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" oadmodule "db_mysql.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "acc.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "pv.so" loadmodule "dispatcher.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "siputils.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "avpops.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif loadmodule "xlog.so" #mysqlcdr modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_url", "mysql://some mysql server/openser") modparam("acc", "db_extra", "some credentials" ) #loadmodule "carrierroute" modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.lst") modparam("rr", "enable_full_lr", 1) !ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif modparam("xlog", "buf_size", 8192) modparam("xlog", "force_color", 1) #carrierroute #modparam("carrierroute", "config_file", "/etc/kamailio/carrierroute.conf") #end carrierroute modparam("pv", "shvset", "ruta=s:$ru") request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } # if(method=="BYE") { #Account BYE transactions setflag(2); }; #CANCEL processing if (method=="CANCEL") { setflag(2); if (t_check_trans()) t_relay(); exit; }; # if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; ##CANCEL processing if (method=="CANCEL") { setflag(3); if (t_check_trans()) t_relay(); exit; }; if (!t_relay()) { sl_reply_error(); } exit; } if (is_method("INVITE")) { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); record_route(); } #removed 1 from string #(ai{s.substr,4,0}); xlog("before strip,< time [$Tf] > Aip= $src_ip, r-uri($ru\n"); #MYvariables $var(x) = $ru; $var(y)=$(var(x){s.substr,4,3}); $var(z)=$(var(y){s.int}); $var(t)=$(var(x){s.substr,7,0}); xlog("dsroute $var(z)fullruri($ru)\n"); xlog(">>>>>>>>>>>>>>>>>la header TO $var(t)----------fullruri($ru)<<<<<<<<<<<<<<<<<<\n"); xlog("all---$Tf,$ct,$cs,<$cs>,callid-$ci,$re,$du,$rt,$ru,$si,$sp,$ua,$rm,$rs,$fn,$fd,$fu,$fU "); strip(3); xlog("after strip time [$Tf] method ($rm) via el code ($var(z)-- ($var(y))\n"); ds_select_dst("$var(z)", "4");#carrer dynamic #ds_select_dst("103", "4");#to change static carrier xlog("print el destination y el $var(z) ($du)>>($td)>>> ($ru)\n"); xlog("errors---$err.info,$err.rcode,$err.rreason,$rc,$rc,$rc,$rr,--end errors "); #remove_hf("To") ; #$insert_hf("To: sip:$var(t)@$rd\r\n", "From"); # trying to change to header #if (!forward()){; #if forwarding # if(!t_relay()){; if (!t_relay_to_udp()) {#proxing sl_reply_error(); exit; }; }

2 1

Register VoIP phones with Kamailio server
by Ravindra Gowda 07 Jun '13

07 Jun '13

Dear All, I was successful in integrating kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version, by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Both the servers are running parallely without any problems. And i choosen siremis-4.0.0 to get the control panel of kamailio server and i got its GUI (By following below link): http://kb.asipto.com/siremis:install32x:main But when i tried to register my VoIP phones with the Kamailio server, I am getting the status code reply as *'401 Unauthorized* (0 bindings)'. What could be the wrong? How can i solve it? And how can i retrieve my Users (are which i had added to MySQL database) in SIP server (kamailio) GUI? Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Engineer - Systems and Products*. *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

Re: [SR-Users] Msilo dumping offline messages
by sipatse＠in.gr 07 Jun '13

07 Jun '13

2 1

Re: [SR-Users] Problem with ACK
by phillman25 07 Jun '13

07 Jun '13

Hello Daniel/Stoyan Thanks for your reply, here is a full sip trace from the first INVITE message to the last acknowledge which is sent to PGW. 192.168.10.189 ==> 81.21.38.34 INVITE sip:94294294@81.21.38.34 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1692903116 1692903116 IN IP4 192.168.10.189 s=Asterisk PBX 1.8.21.0 c=IN IP4 192.168.10.189 t=0 0 m=audio 12584 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 81.21.38.34 ==> 81.21.38.5 INVITE sip:3005A94294294@81.21.38.5 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 16 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 INVITE sip:7000A94294294@81.21.38.34:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 15 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.34==>81.21.38.55 INVITE sip:94294294@81.21.38.55 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 14 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.55 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 242 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.55 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.55 t=0 0 m=audio 49156 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 81.21.38.34 ==> 81.21.38.5 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.34 ==> 192.168.10.189 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes # U 192.168.10.189:5060 -> 81.21.38.34:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport* Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,< sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 70* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* 81.21.38.34:5060 -> 81.21.38.5:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060* Route: <sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 16* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* Regards Phillip

2 1

memcache problem - reporting protocol error
by Dragos Oancea 06 Jun '13

06 Jun '13

Hello We are having issues with the memcached module (Kamailio 4.0.0) . All of a sudden we are getting this kind of messages in the logs, and the values for the requested keys are not retrieved. ALERT: memcached [memcached.c:189]: ALERT: memcached: mcm_fetch_cmd():1305: memcache(4) protocol error: STORED#015#012VALUE 29515_HA1 0 32#015#012b52299247f16c6adfa4c8cc53b34e74a#015#012END#015#012STORED#015#012 [...] ALERT: memcached [memcached.c:189]: ALERT: memcached: mcm_fetch_cmd():1176: memcache(4) protocol error: server sent data for key not in request I suspect that at some point there was a network problem between the memcached server and Kamailio that triggered this abnormal behavior and I also suspect that it is connected with the (old) libmemcache library (libmemcache-1.4.0.rc2) . This is fixed only with a Kamailio restart. Has anyone experience this ? Is there a fix ? I noticed that the Kamailio from the trunk compiles the memcached module with the more new libmemcached library from here: https://launchpad.net/libmemcached/+download . Maybe this is one of the reasons libmemcache is going to be dropped ? Thank you very much. Regards, Dragos

2 2

Re: [SR-Users] Problem with ACK
by phillman25 06 Jun '13

06 Jun '13

Hello Daniel/Stoyan Thanks for your reply, here is a full sip trace from the first INVITE message to the last acknowledge which is sent to PGW. 192.168.10.189 ==> 81.21.38.34 INVITE sip:94294294@81.21.38.34 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1692903116 1692903116 IN IP4 192.168.10.189 s=Asterisk PBX 1.8.21.0 c=IN IP4 192.168.10.189 t=0 0 m=audio 12584 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 81.21.38.34 ==> 192.168.10.189 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 81.21.38.34 ==> 81.21.38.5 INVITE sip:3005A94294294@81.21.38.5 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 16 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Content-Length: 0 81.21.38.5 ==> 81.21.38.34 INVITE sip:7000A94294294@81.21.38.34:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 15 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.34 ==> 81.21.38.5 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 81.21.38.34==>81.21.38.55 INVITE sip:94294294@81.21.38.55 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 14 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.55 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 242 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.55 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.55 t=0 0 m=audio 49156 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 81.21.38.34 ==> 81.21.38.5 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.34 ==> 192.168.10.189 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes # U 192.168.10.189:5060 -> 81.21.38.34:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport* Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,< sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 70* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* 81.21.38.34:5060 -> 81.21.38.5:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060* Route: <sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 16* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* Message: 3 Date: Thu, 06 Jun 2013 16:51:42 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Problem with ACK Message-ID: <51B0A1FE.6040406(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" On 6/6/13 4:34 PM, Stoyan Mihaylov wrote: > We had some similar problems. But what was the actual problem? At least in the two ACKs provided below, loose routing handling with looks correct. Is something that Asterisk doesn't like? Cheers, Daniel > Our configuration is: > SIP client 1 <-> Kamailio <-> Asterisk <->Kamailio<->SIP client 2 > My solution was to check $td and $si and if they are same as Kamailio, > to forward call to Asterisk. > Because I planed to use more then 1 Asterisk, I keep in variable which > one to use. > > > > On Thu, Jun 6, 2013 at 5:28 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > the incoming ACK has the top Route with lr parameter, meaning is > loose routing. By that, the proxy removes the top route header, > preserves the R-URI and sends to the URI in the next Route header. > > From what I can see in the Route stack, it seems a spiral back to > the proxy because ip 81.21.38.34 is two times there. > > If you can't sort it out, send the full SIP trace taken on the > proxy from the initial INVITE to the ACK. Then we can see how > Record-Route headers are set and the signaling flow. > > Cheers, > Daniel > > On 6/6/13 3:30 PM, phillman25 wrote: >> Dear list further to the above problem i observed the following: >> >> ACK message coming from PABX1: >> >> U +0.001877 192.168.10.189:5060 <http://192.168.10.189:5060> -> >> 81.21.38.34:5060 <http://81.21.38.34:5060> >> ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55> >> SIP/2.0* >> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport* >> Route: >> <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,< sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,< sip:81.21.38.34;lr=on;ftag=as181922af>* >> Max-Forwards: 70* >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af* >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551- 13c4-50022-1c3096-5cc2673d-1c3096* >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>>* >> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* >> <mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*> >> CSeq: 102 ACK* >> User-Agent: FPBX-2.8.1(1.8.21.0)* >> Content-Length: 0* >> >> >> >> ACK message sent to PGW from Kamailio1 >> >> U +0.001254 81.21.38.34:5060 <http://81.21.38.34:5060> -> >> 81.21.38.5:5060 <http://81.21.38.5:5060> >> ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55> >> SIP/2.0* >> Via: SIP/2.0/UDP >> 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0* >> Via: SIP/2.0/UDP >> 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060* >> Route: <sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,< sip:81.21.38.34;lr=on;ftag=as181922af>* >> Max-Forwards: 16* >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af* >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551- 13c4-50022-1c3096-5cc2673d-1c3096* >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>>* >> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* >> <mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*> >> CSeq: 102 ACK* >> User-Agent: FPBX-2.8.1(1.8.21.0)* >> Content-Length: 0* >> >> >> >> >> Shouldn't the ACK message to the PGW have the header ACK >> sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6 and the >> Route: <sip:81.21.38.34;lr=on;ftag=as181922af>* ??? >> >> >> >> >> Your help is much appreciated!! >> >> Phillip >> >> >> >> On Thu, Jun 6, 2013 at 12:26 PM, phillman25 <phillman25(a)gmail.com >> <mailto:phillman25@gmail.com>> wrote: >> >> Dear List >> >> I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an >> issue for the below scenario: >> >> PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2 >> >> >> I understand that this is a hairpin scenario but was working >> normally on v 3.3. >> >> Checking in the syslog i see: >> ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via >> found in reply >> >> Checking the sip trace i see that when calling from PABX1 to >> PABX2. After PABX2 answers and the the 200 OK is eventually >> sent to PABX1 , PABX1 answers with ACK but seems like its >> not sent back to PABX2 as a result PABX resends a 200 OK and >> the cycle continues until PABX2 sends a BYE message. Please >> see below the ACK received from PABX1: >> >> ACK sip:94294294@81.21.38.55 >> <mailto:sip%3A94294294@81.21.38.55> SIP/2.0 >> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK6bffe37c;rport >> Route: >> <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1;did=e36.c471>,< sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-26eb>,< sip:81.21.38.34;lr=on;ftag=as1cd4f8f1> >> Max-Forwards: 70 >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as1cd4f8f1 >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d94f08-37261551- 13c4-50022-1c1e67-87fe958-1c1e67 >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>> >> Call-ID: 03042a717e27a87e759f7f4879e70377@192.168.10.189:5060 >> <http://03042a717e27a87e759f7f4879e70377@192.168.10.189:5060> >> CSeq: 102 ACK >> User-Agent: FPBX-2.8.1(1.8.21.0) >> Content-Length: 0 >> >> >> Is there an issue with the above ACK message? Is there any >> way to solve this issue quickly perhaps by disabling loose route? >> I have observed that this issue occurs only when hairpinned. >> >> >> Thanking you in advance! >> >> Phillip >> >> >>

1 0

Problem with ACK
by phillman25 06 Jun '13

06 Jun '13

Dear List I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an issue for the below scenario: PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2 I understand that this is a hairpin scenario but was working normally on v 3.3. Checking in the syslog i see: ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via found in reply Checking the sip trace i see that when calling from PABX1 to PABX2. After PABX2 answers and the the 200 OK is eventually sent to PABX1 , PABX1 answers with ACK but seems like its not sent back to PABX2 as a result PABX resends a 200 OK and the cycle continues until PABX2 sends a BYE message. Please see below the ACK received from PABX1: ACK sip:94294294@81.21.38.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK6bffe37c;rport Route: <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1;did=e36.c471>,< sip:94294294@81.21.38.5 ;pgw-call=call-26eb>,<sip:81.21.38.34;lr=on;ftag=as1cd4f8f1> Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as1cd4f8f1 To: <sip:94294294@81.21.38.34 >;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67 Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 03042a717e27a87e759f7f4879e70377@192.168.10.189:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.21.0) Content-Length: 0 Is there an issue with the above ACK message? Is there any way to solve this issue quickly perhaps by disabling loose route? I have observed that this issue occurs only when hairpinned. Thanking you in advance! Phillip

3 5

No ACK from Cisco
by Victor V. Kustov 06 Jun '13

06 Jun '13

Hi all! I've got a problem with Kamailio<->Cisco<->PSTN. Called from PSTN: 16:20:14.328786 IP (tos 0x80, ttl 255, id 0, offset 0, flags [none], proto UDP (17), length 1166) 172.16.16.3.58446 > 172.16.17.8.sip: SIP, length: 1138 INVITE sip:599674@172.16.17.8:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Remote-Party-ID: <sip:595311@172.16.16.3>;party=calling;screen=no;privacy=off From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:599674@172.16.17.8> Date: Thu, 06 Jun 2013 04:25:44 GMT Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4251252826-3449229794-2149122083-881571867 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1370492744 Contact: <sip:595311@172.16.16.3:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 279 v=0 o=CiscoSystemsSIP-GW-UserAgent 6723 8551 IN IP4 172.16.16.3 s=SIP Call c=IN IP4 172.16.16.3 t=0 0 m=audio 18550 RTP/AVP 8 18 101 c=IN IP4 172.16.16.3 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 16:20:14.329130 IP (tos 0x10, ttl 64, id 17361, offset 0, flags [none], proto UDP (17), length 327, bad cksum 0 (->bc99)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 299 SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:599674@172.16.17.8> Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.335619 IP (tos 0x10, ttl 64, id 17363, offset 0, flags [none], proto UDP (17), length 359, bad cksum 0 (->bc77)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 331 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:599674@172.16.17.8> Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.362576 IP (tos 0x10, ttl 64, id 17365, offset 0, flags [none], proto UDP (17), length 623, bad cksum 0 (->bb6d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 595 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: <sip:172.16.17.8;lr=on;did=558.b04> From: <sip:0074832595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:0074832599674@172.16.17.8>;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Contact: <sip:0074832599674@10.120.0.18:32225;user=phone> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 16:20:23.621104 IP (tos 0x10, ttl 64, id 17380, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba50)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: <sip:172.16.17.8;lr=on;did=558.b04> From: <sip:0074832595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:0074832599674@172.16.17.8>;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Contact: <sip:0074832599674@10.120.0.18:32225;user=phone> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 266 v=0 o=0074832599674 8002 8000 IN IP4 10.120.0.18 s=SIP Call c=IN IP4 10.120.0.18 t=0 0 m=audio 39206 RTP/AVP 8 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 16:20:24.126589 IP (tos 0x10, ttl 64, id 17383, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba4d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:25.135006 IP (tos 0x10, ttl 64, id 17389, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba47)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:27.144412 IP (tos 0x10, ttl 64, id 17395, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba41)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK And no ACK from Cisco. Is it Cisco config problem? P.S. no sip-ua configuration, dial-peer voice 10002 voip description ** xxx ** preference 1 destination-pattern 5T voice-class codec 1 session protocol sipv2 session target ipv4:xxx session transport udp I had no idea about no ACK. Maybe OK from Kamailio incorrect? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.4-calculate GNU/Linux

2 5

New developer: Camille Oudot
by Daniel-Constantin Mierla 06 Jun '13

06 Jun '13

Hello, I want to announce that a new person got developer GIT write access to repository: Camille Oudot - from Orange, France. Currently, he is developing mainly on Kamailio IMS modules, several patches being submitted lately by him. His git commit id is: coudot My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

New developer: Torrey Searle
by Daniel-Constantin Mierla 06 Jun '13

06 Jun '13

Hello, I want to announce that a new person got developer GIT write access to repository: Camille Oudot - from Orange, France. Currently, he is developing mainly on Kamailio IMS modules, several patches being submitted lately by him. His git commit id is: coudot My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

Re: [SR-Users] cnxcc developer module pulse concept
by Carlos Ruiz Díaz 06 Jun '13

06 Jun '13

Hi Eduardo, please take into account that there might be telephony concepts mistranslated from Spanish to English. In case you find something else that seems confusing, please let me know. Regards, Carlos On Thu, Jun 6, 2013 at 7:56 AM, Eduardo Lejarreta <lejarreta.e(a)sarenet.es>wrote: > Good evening Carlos.**** > > ** ** > > I’ve found an example on git repository where you have additional > information.**** > > ** ** > > For a credit of 50, a discount per second of 0,5, a initial_pulse of 30 > and final_pulse of 6 ….**** > > ** ** > > # if only one call is established, that call should last 1m, 36s**** > > ** ** > > Now It makes sense to me.**** > > ** ** > > Thanks and regards.**** > > -- **** > > Eduardo Lejarreta.**** > > ** ** > > *De:* Eduardo Lejarreta [mailto:lejarreta.e@sarenet.es] > *Enviado el:* jueves, 06 de junio de 2013 11:40 > *Para:* 'sr-users(a)lists.sip-router.org' > *Asunto:* cnxcc developer module pulse concept**** > > ** ** > > Good morning Carlos.**** > > ** ** > > On cnxcc_set_max_credit() function I don’t understand the concept “pulse” > (sure it’s something easy but I’m a not aware about this matter)**** > > ** ** > > Could you please explain it in an example?**** > > ** ** > > Thanks and best regards.**** > > -- **** > > Eduardo Lejarreta.**** > > ** ** > > _______________________________________________ > 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 > > -- Carlos http://caruizdiaz.com +595981146623

1 0

cnxcc developer module pulse concept
by Eduardo Lejarreta 06 Jun '13

06 Jun '13

Good evening Carlos. I've found an example on git repository where you have additional information. For a credit of 50, a discount per second of 0,5, a initial_pulse of 30 and final_pulse of 6 .. # if only one call is established, that call should last 1m, 36s Now It makes sense to me. Thanks and regards. -- Eduardo Lejarreta. De: Eduardo Lejarreta [mailto:lejarreta.e@sarenet.es] Enviado el: jueves, 06 de junio de 2013 11:40 Para: 'sr-users(a)lists.sip-router.org' Asunto: cnxcc developer module pulse concept Good morning Carlos. On cnxcc_set_max_credit() function I don't understand the concept "pulse" (sure it's something easy but I'm a not aware about this matter) Could you please explain it in an example? Thanks and best regards. -- Eduardo Lejarreta.

1 0

Microseconds not milliseconds in cfgutils usleep
by David K 06 Jun '13

06 Jun '13

Hello, According to this page : http://kamailio.org/docs/modules/stable/modules/cfgutils.html#idp15252400 usleep will sleep for the number of milli-seconds specified. I am familiar with Linux's usleep which accepts milliseconds and therefore had a look at the code in Kamailio, it looks to me that your usleep accepts micro-seconds ( 1/1000000 of a second ) and not milli seconds. Can anyone confirm this? If so, maybe update the doc. Thanks, David

2 1

cnxcc developer module pulse concept
by Eduardo Lejarreta 06 Jun '13

06 Jun '13

Good morning Carlos. On cnxcc_set_max_credit() function I don't understand the concept "pulse" (sure it's something easy but I'm a not aware about this matter) Could you please explain it in an example? Thanks and best regards. -- Eduardo Lejarreta.

1 0

if (t_check_status("486|408"))
by hiro 06 Jun '13

06 Jun '13

i'm trying to use the example kamailio.cfg to route to voicemail server on busy or decline. Only thing I did was adding decline code to t_check_status("486|408"), enabling the preprocessor variable for voicemail and changing the voicemail host and port to my voicemail server. No requests arrive on my voicemail server and the dial tone keeps ringing even if phone is busy and when I decline on the receiving phone there's a new INVITE sent to the phone directly afterwards. Am I doing something wrong here?

3 15

Msilo dumping offline messages
by sipatse＠in.gr 05 Jun '13

05 Jun '13

2 1

dialog ka-timer/ka_interval results in flood op OPTIONS
by Daniel Tryba 05 Jun '13

05 Jun '13

Trying to use the dialog module to implement calllimits I'm running into a problem when trying to use keep-alives to detect dead dialogs. Limits work fine without keepalives. But as soon as either dlg_set_property("ka-dst"); or dlg_set_property("ka-src"); are used, after the ka_interval timeout kamailio will start to send OPTIONS as fast as possible. There is no difference in behavior for inbound of outbound calls. Older threads on this seem to end without solution. Increasing shmem/pkgmem makes no difference. System is running on Debian/stable with deb http://deb.kamailio.org/kamailio wheezy main packages. # kamailio -V version: kamailio 4.0.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:13:24 Mar 29 2013 with gcc 4.7.2 modparam("dialog","dlg_flag",4) modparam("dialog","hash_size",4096) modparam("dialog","enable_stats",1) modparam("dialog","db_url",DBURL) modparam("dialog","db_mode", 2) modparam("dialog","db_update_period", 15) modparam("dialog","profiles_with_value","maxchannels") modparam("dialog","profiles_no_value", "all") modparam("dialog","dlg_match_mode", 1) modparam("dialog","send_bye", 1) modparam("dialog","default_timeout", 43200) modparam("dialog","ka_timer", 15) modparam("dialog","ka_interval", 30) dlm_manage() uses as: route[DIALOG] { if (is_method("CANCEL|ACK") || (has_totag() && is_method("INVITE| BYE"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag()) { if($au==$null) { route(MAXCALLSINBOUND); } else { route(MAXCALLSOUTBOUND); } dlg_manage(); } } route[MAXCALLSOUTBOUND] { if(method=="INVITE") { if($au && $avp(src_maxchannels)>0) { $var(maxchannels) = 0; get_profile_size("maxchannels", "$au", "$var(maxchannels)"); if($var(maxchannels)>=$avp(src_maxchannels)) { send_reply("403","Call limit reached"); exit; } set_dlg_profile("maxchannels","$au"); dlg_set_property("ka-src"); get_profile_size("maxchannels", "$au", "$var(maxchannels)"); } } return; } MAXCALLSINBOUND is nearly the same but uses $rU instead of $au (and routes to voicemail/redirects if applicable). Resulting in network traffic: U +6.719318 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.6baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000041 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.7baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000028 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.8baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000026 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.9baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . [6412 more OPTIONS before hanging up or phone reboots] syslog output (another call since turning debug on will choke the machine in disk IO to syslog), it keeps repeating stuff like this: 14:19:56 DEBUG: topoh [topoh_mod.c:345]: the COOKIE is [xxx] 14:19:56 DEBUG: dialog [dlg_req_within.c:387]: keep-alive sent to caller 14:19:56 DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x7fb4bfe3d2e0 with 1 -> 2 14:19:56 DEBUG: dialog [dlg_hash.c:588]: ref dlg 0x7fb4bfe3d2e0 with 1 -> 3 14:19:56 DEBUG: dialog [dlg_hash.c:590]: dialog id=7871 found on entry 2710 14:19:56 DEBUG: dialog [dlg_req_within.c:366]: sending BYE to caller 14:19:56 DEBUG: tm [uac.c:243]: DEBUG:tm:t_uac: next_hop=<sip: +31880100781@10.0.34.226:27819> 14:19:56 DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 6504 14:19:56 DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 14:19:56 DEBUG: <core> [parser/msg_parser.c:625]: method: <OPTIONS> 14:19:56 DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip: +31880100781@10.0.34.226:27819> 14:19:56 DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 14:19:56 DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK8691.df660fd1.0>; state=16 14:19:56 DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 14:19:56 DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 14:19:56 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 14:19:56 DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=a2b76e7ff411361o0 14:19:56 DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 14:19:56 DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [54]; uri=[sip:+31880100781@10.0.32.42] 14:19:56 DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [sip: +31880100781@109. 235.32.42] 14:19:56 DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <101> <OPTIONS> 14:19:56 DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 14:19:56 DEBUG: <core> [parser/msg_parser.c:106]: found end of header 14:19:56 DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=as23ba5e76 14:19:56 DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 3

Re: [SR-Users] kamcmd vs kamctl
by DanB 05 Jun '13

05 Jun '13

Hey Ovidiu, Thanks for the tip. Will have a look. Have a good one! DanB

1 0

Asterisk 11.4.0+Kamailio 4.0.1 Integration
by Ravindra Gowda 04 Jun '13

04 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But after all, when i tried to start Kmailio server with /etc/init.d/kamailio start command , i am getting the following errors: asterisk-kamailio@asterisk- kamailio:/$ sudo /etc/init.d/kamailio start [sudo] password for asterisk-kamailio: /etc/init.d/kamailio: 1: /etc/init.d/kamailio: sed: not found /etc/init.d/kamailio: 118: /etc/init.d/kamailio: arithmetic expression: expecting primary: "" what could be the wrong, and how can i solve it? Please share your valuable suggestions on this. Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Engineer - Systems and Products*. *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

Re: [SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Michael Leuker 04 Jun '13

04 Jun '13

Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described: Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy: chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" <sip:xxxxxxxx@198.23.139.21>;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy I doubt that I can make the necessary modifications even with your hints, but would be willing to pay for your (or anybody's) time solving the matter in a way that (a) leaves the FreePBX installation untouched as far as possible and (b) is easy to apply to subsequent versions. Ideally, it would be a slight modification of Daniel's excellent tutorial at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, allowing to create extensions in FreePBX. Anybody willing to give it a shot? Just let me know what you would charge and we can talk about it.

2 5

Re: [SR-Users] kamcmd vs kamctl
by DanB 04 Jun '13

04 Jun '13

Hey Daniel, Thanks for the answer. It makes more sense now. Have a good one! DanB

2 1

Re: [SR-Users] [Spce-user] FYI: Package update(s) in release 2.8
by Andrew Pogrebennyk 04 Jun '13

04 Jun '13

Dear 2.8 users, this update includes the fixes for kamailio-lb config (multiple sockets-related): - added record-route param to store socket; review logic for relaying in-dialog requests (fallback to default send socket if it's not defined explicitly); fixed socket selection for ACK after 4xx and replies to in-dialog requests to callee; - also file descriptors limit has been lifted for sems. The upgrade is highly recommended if you run the 2.8 version. On 06/04/2013 12:16 PM, Sipwise Repository Tracker wrote: > One or more updates have been released in NGCP release 2.8: > > * ngcp-templates-ce-asterisk was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-cdr-exporter was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-check-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-cleanup-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-iaxmodem was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-kamailio was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-lsb was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mediator was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mysql was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-nginx was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-odbc was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-reminder was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-sems was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-system was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-vmnotify was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-asterisk was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-callingcard was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-cdr-exporter was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-check-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-cleanup-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-glusterfs was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-ha was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-heartbeat2 was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-iaxmodem was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-kamailio was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-lsb was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mediator was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-monit was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-monitoring-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mysql was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-nginx was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-odbc was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-pushd was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-redis was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-reminder was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-sems was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-system was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-vmnotify was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-voisniff-ng was updated from version 2.8.6 to version 2.8.7 > > To apply the latest update(s) execute: > > apt-get update && apt-get upgrade && \ > ngcp-update-db-schema && ngcp-update-cfg-schema && \ > ngcpcfg apply > > If you encounter any problems please report a bug at > the spce-user(a)lists.sipwise.com mailing list. > > Further documentation is available at: > > http://www.sipwise.com/products/spce/documentation/ > > This is an automated mail by the Continuous Delivery platform at Sipwise.com > > _______________________________________________ > Spce-user mailing list > Spce-user(a)lists.sipwise.com > http://lists.sipwise.com/listinfo/spce-user >

2 1

Kamailio and Siremis
by Isaac A. McDonald 04 Jun '13

04 Jun '13

Hello All, I'm a bit confused as to how Siremis interacts with Kamailio. From what I understand, the dial-plan is done in the kamailio.cfg with regex statements and requires a restart of the Kamailio process to take effect. How then does Siremis change the dial-plan in Kamailio without writing re-writing the config? I'm assuming Siremis is merely writing to a MySQL database and some additional code is required in the kamailio.cfg in order to be able to use the dial-plan functionality of Siremis? What must I change in the kamailio.cfg to be able to use Siremis for dial-plan modification?

2 1

Kamailio Postgres error - unknown field type: 2
by Jayesh Nambiar 04 Jun '13

04 Jun '13

Hi, I am using Kamailio with Postgres and I get this error while using the load_credentials parameter in the auth_db module. Basically I am loading a column whose datatype is BIGINT or int8 in postgres. Kamailio shows up with this error in syslog when authorization is done: NOTICE: <core> [db_ut.c:485]: unknown field type: 2, setting value to null And it sets the value of AVP as null. When I change the datatype to int4 and put a smaller value it works. I googled a little and figured that kamailio supports bigint, but I still get this error. I am using Kamailio 4.0.1 to test. Any pointers or directions that I should look at?? Would really appreciate the response. Thanks, --- Jayesh

2 1

Handling forking in kamailio
by Ram Anji 04 Jun '13

04 Jun '13

Hi, I want to handle the responses for forked Requests in kamailio. Please suggest me that where i can handle this in configuration. Here is my requirement. If Clinet A is registered with kamailio server on pc 1 and pc 2. If Client B calls A, Server is sending INVITE to A for both PC 1 and PC 2. As per current implementation Server closing the INVITE session if any one(pc1/pc2) responds with 486. But i want to handle the 486 response, and wait till other one(pc1/pc2) also responds instead of closing INVITE session. Regards, Raman

2 1

Re: [SR-Users] is_uri_host_local() maybe not considering IP:PORT
by DanB 03 Jun '13

03 Jun '13

Klaus, Thanks for the tip. I have been using (uri==myself) up to know but I was trying to move towards a more multidomain configuration, was kinda hoping to get rid of hardcoded stuff in particular servers. Daniel, Can the fact that port in ruri makes host part not longer matching as own domain be considered a bug or it is the accepted working mechanism? The "register_myself" parameter made me think that it should match host part at least (port is not important for me), but in my case port in ruri makes the whole host not matching. Hope I have clearly expressed my thoughts ;). Ta, DanB

2 3

is_uri_host_local() maybe not considering IP:PORT
by DanB 03 Jun '13

03 Jun '13

Guys, I was wondering if I am doing something wrong or domain module not considering properly local ips within is_uri_host_local(). Although I have domain loaded with register_myself on 1, the request going to my IP is simply not matching local domain (listening on 127.0.0.1 port 5070). Bellow the trace of such request: """ # U 2013/05/30 12:35:56.436485 10.10.10.21:5060 -> 127.0.0.1:5070 OPTIONS sip:127.0.0.1:5070 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Max-Forwards: 70. Content-Length: 0. User-Agent: iClass4-EP 4.0.0. . # U 2013/05/30 12:35:56.440410 127.0.0.1:5070 -> 10.10.10.21:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>;tag=46a6e639fa023622ac1ba4fea686e961.d61e. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Server: iClass4-AP 4.0.0. Content-Length: 0. """ My script looks something like bellow, so Address Incomplete should never be reached: """ #!define LISTEN_IP 127.0.0.1 #!define LISTEN_PORT 5070 ... listen=LISTEN_IP port=LISTEN_PORT auto_aliases=yes ... # ----- domain params ----- modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) ... if (is_method("OPTIONS") && is_uri_host_local()) { options_reply(); exit; } ... if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } """ I am running on git master with the test patch Daniel did few days back for me. Thanks in advance for any tip! DanB

3 3

Asterisk 11.4.0 + Kamailio 4.0.1 integration
by Ravindra Gowda 03 Jun '13

03 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But after all, when i tried to start Kmailio server with /etc/init.d/kamailio start command , i am getting the following errors: asterisk-kamailio@asterisk-kamailio:/$ sudo /etc/init.d/kamailio start [sudo] password for asterisk-kamailio: /etc/init.d/kamailio: 1: /etc/init.d/kamailio: sed: not found /etc/init.d/kamailio: 118: /etc/init.d/kamailio: arithmetic expression: expecting primary: "" what could be the wrong, and how can i solve it? Please share your valuable suggestions on this. Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Engineer - Systems and Products*. *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

Faild to add Subscription via WebRTC2SIP websocket GW
by Raj Roy Ghandhi 03 Jun '13

03 Jun '13

Hi Friends, I am using Kamailio 4.0 SIP Proxy GW + WebRTC2SIP 2 web socket GW to maintain the presence status of SIPMP5 clients. When I subscriber for presence statue of contact I get the following error and no record get inserted into the "active_wathcers" table. 5(7435) WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found 5(7435) ERROR: tm [t_msgbuilder.c:1367]: assemble_via: via building failed 5(7435) ERROR: tm [t_msgbuilder.c:1540]: build_uac_req(): Error while assembling Via 5(7435) ERROR: tm [uac.c:338]: t_uac: Error while building message 5(7435) ERROR: presence [notify.c:1591]: in function tmb.t_request_within 5(7435) ERROR: presence [subscribe.c:1613]: send Notify not successful 5(7435) ERROR: presence [subscribe.c:2129]: in function handle_expired_record 5(7435) ERROR: presence [subscribe.c:464]: Failed to delete subscription from memory Here is the SIPML5 console log for that perticular area State machine: tsip_dialog_generic_Any_2_InProgress_X_oSubscribe tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 session event = type = connecting - description = Connecting... test_api.html:51 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=b01221cac0ed2edd8409481b9ce2d72c.ac8c Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Content-Length: 0 Proxy-Authenticate: Digest realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",stale=FALSE Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Proxy-Authorization: Digest username="1004",realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",uri="sip:1005@sip.mysip.com ",response="223e983b8e2de2b04dbcd2ac9fc0b17d",algorithm=MD5 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 202 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_Connected_X_2xx tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=NOTIFY sip:1004@192.168.1.55:20124;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 User-Agent: kamailio (4.0.1 (x86_64/linux)) Max-Forwards: 70 Event: presence Subscription-State: active;expires=200 tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_Any_2_Any_X_iNotify tsk_utils.js?svn=20:110 session event = type = i_ao_request - description = OK test_api.html:51 session event = type = connected - description = Connected test_api.html:51 SEND: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:1004@df7jal23ls0d.invalid;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 tsk_utils.js?svn=20:110 session event = type = i_notify - description = Incoming NOTIFY test_api.html:51 NOTIFY content = test_api.html:72 NOTIFY content-type = null Please guide me on this. Best Regards, Roy.

2 1

Does Kamailio check for CSeq, Call-ID when unregister ?
by Khoa Pham 03 Jun '13

03 Jun '13

Hi, When I want to unregister, I have 2 ways 1. Append ;expires=0 in Contact field 2. Add another Expires : 0 header with CSeq, Call-ID not the same with when i register Does Kamailio check for the saem Cseq, Call-ID to allow unregistration ? -- Khoa Pham HCMC University of Science www.fantageek.com

3 2

Asterisk 12.04 LTS + kamailio 4.0.1 Integration problem.
by Ravindra Gowda 03 Jun '13

03 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But i am unable to accomplish it, i just ended with some mysql errors. I am very new to this aspect. So can anyone suggest me any other standard reference procedure or any tutorials to do this. PS: my host Architecture: i686 Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

{voipfreak.net}
by Laszlo 02 Jun '13

02 Jun '13

http://guardianwarriorfoundation.com/zffhwbdc/tadou.hdoho

1 0

[6/2/2013 12:59:07 PM]
by Laszlo 02 Jun '13

02 Jun '13

http://helptocats.ru/zwwnzzty/zldopj.wjfxdwp

1 0

Re: [SR-Users] SIP Trunks
by Keith Hubner 01 Jun '13

01 Jun '13

Hi, Authentication for users is working fine for me, I add a user to the db and the phone can register. I am looking to connect a SIP trunk but there doesn't seem to be any authentication? What I want is to allow users to connect from any IP but only allow SIP trunks to connect from certain IPs and authenticate. Thanks, Keith

2 1

How to make BLF work
by SamyGo 31 May '13

31 May '13

Hi list, I've been trying to make my Yealink phone to give BLF indications but I haven't been able to achieve this successfully yet so I need some expert advise here. My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights. The same phone registering to Kamailio sends SUBSCRIBE and nothing happens. The Presence Handling is enabled in configurations and yet no dialog-info+xml NOTIFY is received. Please help. Best Regards, Sammy Go.

4 12

pua: lock release
by Halina Nowak 31 May '13

31 May '13

missing lock release: --- a/modules/pua/hash.c Mon May 27 16:42:23 2013 +0200 +++ b/modules/pua/hash.c Fri May 31 16:36:01 2013 +0200 @@ -339,7 +339,10 @@ if (temp_dialog) delete_htable(temp_dialog, hash_code); else + { + lock_release(&HashT->p_records[hash_code].lock); return -1; + } insert_htable(dialog, hash_code);

2 1

Why does Kamailio return many Contacts in Contact header in REGISTER response ?
by Khoa Pham 30 May '13

30 May '13

When I make REGISTER request to server Kamailio. Kamailio sometimes return me REGISTER 200 OK response with many Contacts in Contact header field Contact: <sip:user1@1.1.1.1:58492 ;transport=TLS;ob>;expires=29;received="sip:1.1.1.1:58492;transport=TLS", <sip:user1@3.3.3.3:58520;transport=TLS;ob>;expires=244, <sip:user1@1.1.1.1:58529 ;transport=TLS;ob>;expires=284;received="sip:1.1.1.1:58529;transport=TLS", <sip:user1@3.3.3.3:58548;transport=TLS;ob>;expires=329, <sip:user1@3.3.3.3:58562;transport=TLS;ob>;expires=393, <sip:user1@1.1.1.1:58571 ;transport=TLS;ob>;expires=483;received="sip:1.1.1.1:58571;transport=TLS", <sip:user1@2.2.2.2:58588;transport=TLS;ob>;expires=538, <sip:user1@1.1.1.1:58600 ;transport=TLS;ob>;expires=587;received="sip:1.1.1.1:58600;transport=TLS", <sip:user1@2.2.2.2:58611;transport=TLS;ob>;expires=630, <sip:user1@1.1.1.1:58624 ;transport=TLS;ob>;expires=670;received="sip:1.1.1.1:58624;transport=TLS", <sip:user1@2.2.2.2:58632;transport=TLS;ob>;expires=706, <sip:user1@1.1.1.1:58650 ;transport=TLS;ob>;expires=826;received="sip:1.1.1.1:58650;transport=TLS", <sip:user1@2.2.2.2:58661 ;transport=TLS;ob>;expires=900;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00007dtrf0a4c>";reg-id=1 Why is that ? -- Khoa Pham HCMC University of Science www.fantageek.com

5 9

change cli in unconditional call Forward
by Barry Flanagan 30 May '13

30 May '13

Hi I have the same issue with the CLI on unconditional call forwarding being set to the original caller's number as in this post from March 2012 http://lists.sipwise.com/pipermail/spce-user/2012-March/001149.html In that post it suggests that version 2.6 might have a fix. I am on 2.8 and have tried various settings but can't see a way to make the From: header contain the subscriber's CLI. Is this still not possible without a custom tweak? -Barry Flanagan

1 0

Re: [SR-Users] git branch moved?
by Gertjan Wolzak 30 May '13

30 May '13

Never mind... Problem on my end.... not between keyboard and chair. Rgds, Gertjan From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Gertjan Wolzak Sent: donderdag 30 mei 2013 11:39 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] git branch moved? Gents, Trying to install Kamailio 4.0 from GIT, but it looks like only the master is still there... Something gone wrong ?? root@sip:/usr/local/src/kamailio-4.0/kamailio# git status # On branch master nothing to commit (working directory clean) root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout -b 4.0 origin/4.0 fatal: git checkout: updating paths is incompatible with switching branches. Did you intend to checkout 'origin/4.0' which can not be resolved as commit? root@sip:/usr/local/src/kamailio-4.0/kamailio# git branch * master root@sip:/usr/local/src/kamailio-4.0/kamailio# git show show show-branch root@sip:/usr/local/src/kamailio-4.0/kamailio# git show show show-branch root@sip:/usr/local/src/kamailio-4.0/kamailio# git show-branch [master] documentation: Rebuild all modified READMEs root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout HEAD master origin/HEAD origin/master root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout Regards, Gertjan Wolzak

1 0

git branch moved?
by Gertjan Wolzak 30 May '13

30 May '13

Gents, Trying to install Kamailio 4.0 from GIT, but it looks like only the master is still there... Something gone wrong ?? root@sip:/usr/local/src/kamailio-4.0/kamailio# git status # On branch master nothing to commit (working directory clean) root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout -b 4.0 origin/4.0 fatal: git checkout: updating paths is incompatible with switching branches. Did you intend to checkout 'origin/4.0' which can not be resolved as commit? root@sip:/usr/local/src/kamailio-4.0/kamailio# git branch * master root@sip:/usr/local/src/kamailio-4.0/kamailio# git show show show-branch root@sip:/usr/local/src/kamailio-4.0/kamailio# git show show show-branch root@sip:/usr/local/src/kamailio-4.0/kamailio# git show-branch [master] documentation: Rebuild all modified READMEs root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout HEAD master origin/HEAD origin/master root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout Regards, Gertjan Wolzak

1 0

I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 30 May '13

30 May '13

Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 106 to 107 I get the followng error : 1. in the asterisk console: " Unresolvable destination (478/SL)" 2. in the kamailio log: May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Any idea about the cause of this problem? ps:kamailio and asterisk are running in the same machine. ps:attachment is Asterisk CLI log Best Regards, zhengyw 郑友闻医疗终端部东软熙康健康有限公司沈阳市浑南新区新秀街2号，A1楼216 Postcode：110179 Tel：+86 24 83662278 Mobile：15242493836 Email : zhengyw(a)neusoft.com http://www.neusoft.com --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

4 14

outbound loose routing question
by אורן אברהם 29 May '13

29 May '13

Dear list I'm trying to route using the loose_route() method for an ACK message of an INVITE. the ACK message contains a Route header field with the outbound proxy address (which is the server processing this message - aka "myself") and a flow token to the UAC. but the routing is failed because the process_outbound() function in loose.c deliberatly ignores my flow-token because of the condition in loose.c:522 which is: else if (!ip_addr_cmp(&rcv->src_ip, &_m->rcv.src_ip || rcv->src_port != _m->rcv.src_port). I don't understand why i cannot use this flow token to send the ACK back to the UAC... Thanks in advance...

2 2

"active_watchers" table does not get filled.
by Raj Roy Ghandhi 29 May '13

29 May '13

Hi Friends, I am using Kamailio 4 with SIPML5 HTML client. When I subscriber for presence I see that data get filled into the "presentity" table. But no into the "active_watchers". When I use QuteCom SIP client The "active_watchers" table get filled with the contacts in the QuteCom. Can we fix this for SIPML HTML client ? Best Regards, Roy.

1 1

Presence Module with DBOnly Mode
by Raj Roy Ghandhi 29 May '13

29 May '13

Hi Freinds, Has anyone done configuring the presence module only with DB. In my requirement, I am going to have multiple servers and share the same DB for presence info and enable Publish/Unpublish and Subscribe/UnSubscribe for all the clients who will be connecting to different Kamailio servers. Also expecting the NOTIFY to happen without any issues. I tried to get this done with modparam("presence", "subs_db_mode", 3) But failed. Please guide me for this. Best Regards, Roy.

1 0

Dispatcher ds_select_dst on branches
by DanB 29 May '13

29 May '13

Hey Guys, I was wondering if I could get some guidance on my ideas. I would like to use dispatcher ds_select_dst on ruri in a branch. Is this scenario feasible or out of internals (winning branch maybe) impossible? The idea behind my scenario is to be able to dispatch separately branches received from redirect server. Thanks in advance for any tip! DanB

3 5

Kamailio 4 + Asterisk 11: Local Extensions busy / congested
by Michael Leuker 29 May '13

29 May '13

Hi everybody, I've been following Daniel's excellent tutorial at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and managed to get registrations and outbound dialing to run. The problem is that whenever I try to call a local extension (directly or via trunk), CLI reports == Everyone is busy/congested at this time (1:0/0/1) and plays the according VM message. "sip show peers" shows the extension as online / the registration correctly passed along by Kamailio: 1001/1001 198.23.139.21 D A 5060 OK (1 ms) 198.23.139.21 is the server IP and used for both Kamailio (5060) and Asterisk (5080). Asterisk only allows for UDP, Kamailio for for UDP and TCP connections. The server is connected directly to the internet and the only NAT is on the client side. I have disabled NAT support both in Asterisk and Kamailio because the clients support ICE and can connect using TURN to the Asterisk echo-test just fine. Enabling NAT support (with and without rtpproxy) in Kamailio doesn't solve the above problem (and doesn't seem to be able to traverse all NAT situations in any case). To be clear: The Asterisk configuration itself is working flawlessly. I don't encounter the problem at all if I connect directly to the Asterisk server, either in its original, standalone configuration or on port 5080 (after disabling the ACL allowing only the Kamailio IP and re-enabling the password). Please let me know if you need specific logs (I wouldn't want to needlessly clutter the original description with useless information) to take a closer look at the problem. If this is a known issue, a friendly pointer would be much appreciated as well! I've searched for hours and hours and never seemed to find anything helpful. Have a great day and with Best Regards, Michael

2 2

Re: [SR-Users] Adding Contacts info from branches in 302 redirect not as expected
by DanB 29 May '13

29 May '13

Daniel, Good catch, I do not recall it but it could be I have used in first case (when it worked) normal clone, and it second just the dept=1. I have just rechecked now and it works checking out directly your commit, so sorry for additional noise. Thanks and have a good one! DanB

1 0

MySQL storage engine
by Lefteris Zafiris 29 May '13

29 May '13

Hi everybody, I noticed (after checking the files in utils/kamctl/mysql) that kamailio uses the MyISAM storage engine for its database tables in mysql, is there some particular reason for this? Isn't the read/write concurency that InnoDB provides desirable? And on a side note, enabling 'low_priority_updates' ( http://dev.mysql.com/doc/refman/5.0/en/server-options.html#option_mysqld_lo…) that prioritizes reads over writes in my mysql config can lead to any potential issues with kamailio? Anybody tested this before? Regards, Lefteris Zafiris

3 4

Re: [SR-Users] Dispatcher ds_select_dst on branches
by DanB 29 May '13

29 May '13

Hey Daniel, Thanks for the answer. Now if I think deeper the AVPs should be also overwritten by sequential ds_select_dst so I guess it can end up nasty (was kinda planning to use two dispatchers on the same transaction-one to reach redirect server and second to process the answer in failure route). Will anyway redesign my ideas ;). Have a good one! DanB

2 1

Re: [SR-Users] Adding Contacts info from branches in 302 redirect not as expected
by DanB 29 May '13

29 May '13

Daniel, The command which failed for me it was checking out your commit in the git sources: Here is how I run it: """ root@IECDev:/usr/local/src/kamailio/kamailio.git# git checkout 74bf08ef30cf18fb609c2c067584f1515ede23da fatal: reference is not a tree: 74bf08ef30cf18fb609c2c067584f1515ede23da root@IECDev:/usr/local/src/kamailio/kamailio.git# git branch * master """ DanB

2 1

misc_radius false error
by Victor V. Kustov 28 May '13

28 May '13

Hello! Radius pairs has two similar types: integer and ipaddr. generate_avps In modules/misc_radius/functions.c, function generate_avps() we see that attribute checked for PW_TYPE_INTEGER (integer) type. That incorrect: when rad attribute has PW_TYPE_IPADDR (for example Framed-IP-Addr) then we raise error, but we must processing ipaddr same as integer. As result misc_radius cant sent queries with PW_TYPE_IPADDR at all: error "attribute ## not integer" raised. Need correct this same way: @@ -142,7 +196,8 @@ static void generate_avps(struct attr *attrs, VALUE_PAIR* received) do { \ if ((_len) != 0) { \ if ((_len) == -1) { \ - if (_attrs[_attr].t != PW_TYPE_INTEGER) { \ + if (_attrs[_attr].t != PW_TYPE_INTEGER) \ + if (_attrs[_attr].t != PW_TYPE_IPADDR) { \ LM_ERR("attribute %d is not of type integer\n", \ _attrs[_attr].v); \ goto error; \ Daniel, what about my previous patch? May be resend with this corrections? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 6

Re: [SR-Users] Kamailio + Siremis Outbound route ---- Re Sent Due to bad English
by Daniel-Constantin Mierla 28 May '13

28 May '13

Hello, On 5/27/13 1:03 AM, Tony Turner wrote: > > (re-sent due to bad English) > > Hi, > > I have 3 registered test users, how can I configure Siremis to do the > trunk to freeswitch using LCR or Carrierroute rather than using the > code below. I am keen to be able to setup Inbound + Outbond trunks via > Siremis. Do you know if there is a manual for Siremis or a how to / > step by step? > > if($rU =~"^01") { > $ru = "sip:" + $rU + "@__FREESWITCHIP__"; > route(RELAY); > exit; > } > Siremis is a web interface for the database use for kamailio, not for kamailio configuration file. For using LCR or carrierroute, just read the documentation of these modules and add to your configuration file. LCR should be rather simple to setup, besides loading it and setting module parameters, you have to use two new functions in the routing logic of config file. Carrier route should be pretty much the same, just that I haven't checked it lately. > Currently with the above code if a user phones one of the other > extensions it tries to route out to the PSTN network rather than the > extension, is that because I have put the above code in the wrong > place in the config so it never gets to the code to route to the > extension? (routing to PSTN is fine) > > Or do I need an if else statement wrap checking if local user, please > can you give me some idea of the code ... > If you don't have numbering rules that you can sort out if it is local subscriber or not, the best is to check with subscriber table (see uri_db or sqlops modules, both of them can help for this). Cheers, Daniel > > Thanks > > Tony > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 20 May 2013 16:19 > *To:* tony.turner(a)nodemax.com <mailto:tony.turner@nodemax.com>; > Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] Kamailio + Siremis Outbound route > > Hello, > > if you want to send all calls that arrive to kamailio having the > prefix 01 to freeswitch: > > if($rU =~"^01") { > $ru = "sip:" + $rU + "@__FREESWITCHIP__"; > route(RELAY); > exit; > } > > Be sure calls are authenticated at that point and, if needed, the call > is not actually coming from freeswitch. > > Cheers, > Daniel > > On 5/20/13 11:33 AM, Tony Turner wrote: > > Hi > > Version Kamailio v4.0 + Siremis installed on Debian Wheezy via > apt-get install > > I want to use Kamailio as a proxy edge register to our network. > > I have installed Kamailio and freeswitch. > > I can register on Kamailio but I can't route a call from my sip > client from Kamailio to freeswitch and out to PSTN > > Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway > --- Carriers > > If I register direct on Freeswitch I can route out to PSTN but I > don't understand Kamailio routing. > > Can someone let me how I route say from SIP client registered on > Kamailio to prefix 01% which goes out to Freeswitch > > Many Thanks > > Tony > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 > *http://asipto.com/u/katu * > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

Re: [SR-Users] Adding Contacts info from branches in 302 redirect not as expected
by DanB 28 May '13

28 May '13

Hey Daniel, Many thanks for so fast reaction. As I always say, opensource beats proprietary in more than one way ;). The changes you did work smooth for me, the new redirect comes with proper enclosure characters for uri in branches: """ # U 2013/05/28 10:44:53.201933 127.0.0.1:5070 -> 10.10.10.21:5060 SIP/2.0 302 iClass4-AP Redirect. Call-ID: ab9c9470915850c7ec7c8922dc4eed95@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. From: "dan" <sip:dan@iec.itsyscom.com>;tag=e65ed733. To: <sip:dan@iec.itsyscom.com>;tag=46a6e639fa023622ac1ba4fea686e961.8db5. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bKf5a1.88785e1b21984da781a03be04b99e6b9.0. Via: SIP/2.0/UDP 10.10.10.154:5060;rport=5060;branch=z9hG4bK-313932-636d0be5350bb5038fc59a6d30111b9d. Contact: <sip:dan@10.10.10.141:3072;line=113rlmlo;rcv=sip:10.10.10.141:3072>;q=1, <sip:dan@10.10.10.154:5060;transport=udp;registering_acc=iec_itsyscom_com;rcv=sip:10.10.10.154:5060>. Server: iClass4-AP 4.0.0. Content-Length: 0. """ Have a good one! DanB PS: Checking out specific commit fails for me after Juha's merge, could be I am doing something wrong though.

2 1

Via header, branch parameter and syn_branch
by Richard Brady 28 May '13

28 May '13

Hi folks The syn_branch global parameter results in the use of a "synonym" branch parameter in the Via header for statelessly forwarded requests as a performance optimisation. This was originally done by setting branch=0 which, while not strictly compliant with 3261 (8.1.1.7 and 16.6 item 8), would not cause any problems with a downstream proxy or UA compliant to either 3261 or 2543 since it does not contain the cookie and is therefore self-evidently not guaranteed to be unique. Then a patch was added (commit ebb3b08) to change it from branch=0 to branch=z9hG4bKcydzigwkX which includes the cookie, thus declaring itself to be unique when it clearly is not. This takes the optimisation too far as it deliberately misleads downstream proxies with regard to the uniqueness of the parameter and breaks the mechanism in 17.2.3 for identifying unreliable (non-unique) branch parameters for transaction matching which should then fall back to header inspection but does not. The result is that unrelated requests can be identified as duplicates of each other. Also note 3261 section 16.11 on Stateless Proxy behaviour: "The requirement for unique branch IDs across space and time applies to stateless proxies as well." Has the commit above introduced a bug and should it be reverted? And should the next major release have a default of syn_branch=0? Since with syn_branch=1 the branch=0 version has been known to cause interop issues in the past (see below) and I can confirm the branch=z9hG4bKcydzigwkX version causes interop issues in the present. Regards, Richard See also these related threads: [SR-Users] Via header branch parameter in ACK message not unique http://lists.sip-router.org/pipermail/sr-users/2011-December/071207.html [SR-Users] "branch" tag in the "Via" header of the ACK message for the re-INVITE http://lists.kamailio.org/pipermail/sr-users/2012-November/075560.html [Serusers] branch parameter http://lists.sip-router.org/pipermail/sr-users/2007-February/060147.html [SR-Users] about syn_branch http://lists.sip-router.org/pipermail/sr-users/2011-March/067732.html [SR-Users] Via branch parameter in end2end ACK http://lists.sip-router.org/pipermail/sr-users/2011-May/068627.html [SR-Users] Broken Via/reply-matching for natping OPTION http://lists.sip-router.org/pipermail/sr-users/2011-April/068340.html [Serusers] ACK message Via field branch=0 problem http://lists.sip-router.org/pipermail/sr-users/2004-July/037731.html

2 3

kamailio 4.0.1 websocket configuration error
by Rupayan Dutta 28 May '13

28 May '13

Hi, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's rong with it? please help. Rupayan Dutta

1 0

pv_trans:312 - substr out of range
by Uri Shacked 28 May '13

28 May '13

Hi, can anyony tell me when will i get the following error : "substr out of range". i saw it has something to do with the pv_trans.c file and i saw it there. but, is it related to "shvset" or to "substr" command BR, Uri

2 2

How does save(location) choose the subscriber
by Mino Haluz 28 May '13

28 May '13

Hi, if I do save(location) when receiving REGISTER, what is the header which indicates the subscriber for which it will be registered ? $fu?$au?$tu? Thanks, Mino

3 2

TLS
by Moacir Ferreira 28 May '13

28 May '13

I would appreciate some help on the following questions I have: - If I use TLS mutual authentication, do I still need a subscriber password or the TLS successful mutual session setup will assume that the client is "trusted" so it can register what it is asking to register? - For large deployments, can I issue a single certificate and install it on all my telephone sets making them "trusted" to me or I need one certificate per telephone/subscriber? - Anyway, can you share your "good practices" advises for large deployment? - Finally, do you know any free softphone that implements mutual TLS authentication? Thanks, Moacir

3 5

topoch and xHTTP
by Julia 28 May '13

28 May '13

Hello, I am testing the xHTTP module on kamailio-3.3 with topology hiding (topoch) and I get parsing error for each received HTTP message: /usr/local/sbin/kamailio[2937]: ERROR: <core> [parser/parse_from.c:60]: ERROR:parse_from_header: bad msg or missing FROM header /usr/local/sbin/kamailio[2937]: ERROR: topoh [topoh_mod.c:215]: cannot parse FROM header Is it configuration issue? Thanks, Julia.

2 5

question
by Dave Paxton 27 May '13

27 May '13

I have a login with iptel. I am trying to set up jitsi or linphone with someone in argentina. The server sees us both on but we cannot get the thing to work. Each program shows we are online but there is no way to connect. I must be doing something stupid. Thanks, Dave Paxton 6408 Colonial Drive Boise, ID 83709 208-570-9755 dpaxton(a)me.com Skype: dpaxton

4 4

Business directory reminder
by Daniel-Constantin Mierla 27 May '13

27 May '13

Hello, a quick reminder to our old and new community members that companies and individuals offering services for or products based on Kamailio can get listed on project's business directory. There are few basic rules that apply, you can read more at: - http://www.kamailio.org/w/business-directory/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

Adding Contacts info from branches in 302 redirect not as expected
by DanB 27 May '13

27 May '13

Hey Guys, I was wondering if someone can help me with identifying my issue (which I suspect it as a bug). I am doing some redirect tests between 2 servers, both running same Kamailio version - 4.0.0 out of debian packages. """ root@IECDev:/home/dan# kamailio -V version: kamailio 4.0.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:13:24 Mar 29 2013 with gcc 4.7.2 """ On redirect server I have location stored, doing lookup and right after using "sl_send_reply("302", "iClass4-AP Redirect");" Up to information stored in the Contact of redirect request everything works smooth, the problem which I get is with the information in the Contact of the redirect: """ # U 2013/05/27 16:55:39.171237 127.0.0.1:5070 -> 10.10.10.21:5060 SIP/2.0 302 iClass4 AP Redirect. Call-ID: 1886b401c36753c96bb85a0f4ad7ef88@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. From: "dan" <sip:dan@iec.itsyscom.com>;tag=9f9fd3ed. To: <sip:dan@iec.itsyscom.com>;tag=46a6e639fa023622ac1ba4fea686e961.affb. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK0947.8bc60857.0. Via: SIP/2.0/UDP 10.10.10.154:5060;rport=5060;branch=z9hG4bK-363833-3efe2f43332be9a60ebd1e1aad124280. Contact: <sip:dan@10.10.10.141:3072;line=qhjb2bk0;rcv=sip:10.10.10.141:3072>;q=1, sip:dan@10.10.10.154:5060;transport=udp;registering_acc=iec_itsyscom_com. Server: iClass4-AP 4.0.0. Content-Length: 0. """ It appears that the information of the first contact is correctly encapsulated but the second (out of first branch) not. Due to this in the server capturing the redirect I will only have first contact correctly written in the ruri and second one partially. Example of loging this: """ May 27 16:55:39 IECDev /usr/sbin/kamailio[3556]: ERROR: <script>: Extracted destination set: Contact: <sip:dan@10.10.10.154:5060>;q=0.01, <sip:dan@10.10.10.141:3072;line=qhjb2bk0;rcv=sip:10.10.10.141:3072>;q=1, ru: sip:dan@iec.itsyscom.com, du: sip:127.0.0.1:5070 May 27 16:55:39 IECDev /usr/sbin/kamailio[3556]: ERROR: <script>: Branches: <sip:dan@10.10.10.154:5060>;q=0.01, <sip:dan@10.10.10.141:3072;line=qhjb2bk0;rcv=sip:10.10.10.141:3072>;q=1 """. I would suspect the redirect function not properly inserting the content of each contact uri within it's own <> boundaries but I would appreciate a confirmation on this from someone more experienced with Kamailio's code. Thanks in advance for any tip! DanB

2 1

SCTP support of MultiHoming...
by Jignesh Gandhi 27 May '13

27 May '13

Hello, I read in the documentation that Kamailio SCTP supports multi homing. Does anyone know how I can configure this? Thanks in advance, --Jignesh Gandhi

2 2

PRESENCE entries duplication
by Raj Roy Ghandhi 27 May '13

27 May '13

Hi Friends, I am trying to use the Kamalio 4 and SIPML5 and so for it is working fine. But when I subscribe for presentity of another then I receive all the entries of that user in the presentity table in mysql db. It seem that it I receive the entries that are not expired. Is there a way to get the latest status of the user or just avoid duplicate entries in the presentity table. Please advice. Best Regards, Roy.

2 1

kamailio 4.0.1 websocket support
by Rupayan Dutta 27 May '13

27 May '13

Dear All, I have build and installed kamailio 4.0.1 from source(tarball release) in Cent Os 5.8 in i386 machine.I can start kamailio but when i want to connect kamailio using SipMl5 client which uses sip over websocket the client can't connect with the server.I am attaching my generated kamailio.cfg file with this mail.Is there any additional configuration needed for websocket support in kamailio 4.0.1?I have located the example websocket.cfg file in kamailio source but totally confused what modification to be done in the end of the file(e.g Main SIP request routing logic blocks). Note: I moified *modules.lst and include db_mysql auth auth_db sl nathelper tm xhttp tls msrp websocket modules *

2 1

Re: [SR-Users] Kamailio + Siremis Outbound route ---- Re Sent Due to bad English
by Tony Turner 26 May '13

26 May '13

(re-sent due to bad English) Hi, I have 3 registered test users, how can I configure Siremis to do the trunk to freeswitch using LCR or Carrierroute rather than using the code below. I am keen to be able to setup Inbound + Outbond trunks via Siremis. Do you know if there is a manual for Siremis or a how to / step by step? if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Currently with the above code if a user phones one of the other extensions it tries to route out to the PSTN network rather than the extension, is that because I have put the above code in the wrong place in the config so it never gets to the code to route to the extension? (routing to PSTN is fine) Or do I need an if else statement wrap checking if local user, please can you give me some idea of the code ... Thanks Tony From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 20 May 2013 16:19 To: tony.turner(a)nodemax.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + Siremis Outbound route Hello, if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch: if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch. Cheers, Daniel On 5/20/13 11:33 AM, Tony Turner wrote: Hi Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install I want to use Kamailio as a proxy edge register to our network. I have installed Kamailio and freeswitch. I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing. Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch Many Thanks Tony _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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Re: [SR-Users] Kamailio + Siremis Outbound route
by Daniel-Constantin Mierla 26 May '13

26 May '13

Hello, if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch: if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch. Cheers, Daniel On 5/20/13 11:33 AM, Tony Turner wrote: > > Hi > > Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get > install > > I want to use Kamailio as a proxy edge register to our network. > > I have installed Kamailio and freeswitch. > > I can register on Kamailio but I can't route a call from my sip client > from Kamailio to freeswitch and out to PSTN > > Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- > Carriers > > If I register direct on Freeswitch I can route out to PSTN but I don't > understand Kamailio routing. > > Can someone let me how I route say from SIP client registered on > Kamailio to prefix 01% which goes out to Freeswitch > > Many Thanks > > Tony > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

LCR Module - Rule_id
by Eloy Coto Pereiro 24 May '13

24 May '13

Hi, It Is possible get the rule_id after execute next_gw() function? Cheers

2 4

Calls remain stuck in state 1
by Giany 24 May '13

24 May '13

Hello, We are using dipatcher to limit the concurrent number of calls, problem is that from time to time calls remain stuck in state 1 and it breaks our concurrent limits..I was not able to make a kamailio log with high debug as it happens randomly. Attached is a tcpdump flow: Conv.| Time | serverA | Provider | | | | RemoteEnd | 112 |938.355 | INVITE SDP (g729 g711U GSM X-NSERTPType-100 te...hone-eventRT) | |(5060) ------------------> (5050) | | 112 |938.357 | 100 Trying| | | | |(5050) ------------------> (5060) | | 113 |938.420 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5050) ------------------> (5060) | | 113 |938.422 | 100 trying -- your call is important to us | | |(5060) ------------------> (5050) | | 113 |938.422 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |938.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |939.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |941.906 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |945.907 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 112 |948.358 | CANCEL | | |SIP Request | |(5060) <-------------------------------------- (61016) | 112 |948.359 | CANCEL | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.359 | 200 canceling | |SIP Status | |(5060) --------------------------------------> (61016) | 112 |948.359 | 487 Request Terminated | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | 200 OK | | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | ACK | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.360 | 487 Request Terminated | |SIP Status | |(5060) --------------------------------------> (61016) | 113 |948.360 | CANCEL | | |SIP Request | |(5050) ------------------> (5060) | | 113 |948.360 | 200 canceling | |SIP Status | |(5060) ------------------> (5050) | | 112 |948.365 | ACK | | |SIP Request | |(5060) <-------------------------------------- (61016) | dialog:: hash=136:689416016 state:: 1 ref_count:: 3 timestart:: 0 timeout:: 0 callid:: 745eed805cad6b9a3e1727d169cf3461@serverA:5050 from_uri:: sip:fromnumber@serverA:5050 from_tag:: as53f6bee4 caller_contact:: sip:fromnumber@serverA:5050 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:serverA:5060 callee_bind_addr:: to_uri:: sip:internalnr@serverA:5060 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: As you see the remoteEnd does not answer at all to this request (due to network issue most likely) and the provider sends a CANCEL after approx 3 seconds. >From what I see the INVITE that is sent from asterisk towards kamailio remains stuck (938.420). We are using Kamailio 3.1.6. Any idea what could be the reason for this? Thank you.

2 1

"SER Getting Started" - what's the copyright and license?
by Olle E. Johansson 24 May '13

24 May '13

http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf I'm looking for the authors of this document to find out what license it is published under. It would be nice to be able to use it and update it instead of starting a new "getting started" from scratch. Please contact me if you get this message! Sorry for cross-posting. /O

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Re: [SR-Users] Simulating kamailio config flows
by Daniel-Constantin Mierla 24 May '13

24 May '13

On 4/22/13 5:30 PM, Juha Heinanen wrote: > Andreas Granig writes: > >> Let me know if this is stupid and/or a complete overkill, but what about >> introducing some kind of dummy mode, where you'd pipe a message into >> kamailio via stdin, and get the resulting message out on stdout (e.g. in >> ngrep style with ip information as first line, plus the content >> following), plus a dump of internals (e.g. vars, avps) on stderr as >> they're being assigned, and once the message is processed, kamailio >> would just shut down again? > in addition to the message, you would need to be able to tell which > ip addr it is coming from and which ip/port in kamailio it is going to. > also, on the output side, kamailio would need to tell which > proto/ip/port it would use to send the message out. Details about going out can be printed on onsend_route. In this route block, one can execute drop and nothing is sent to the wire. It can be a config started with a special define specified with -A parameter. Enabling debugger module with cfg trace should give valuable information about what has been executed from config. From what Andreas suggesting, printing the value of variables as they are assigned is missing, probably can be added by hooking in the interpreter when doing the assignment operation. Grouping all above under some global/command line parameter can be useful to make it easier to do a dry run. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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add_path() and save()
by אורן אברהם 23 May '13

23 May '13

Dear list. when using registrar module with path support does the following code work: add_path(); save(); or maybe the path is added only for outgouing messages and the save() won't be affected ?

2 3

nat_keepalive() and fix_contact() are incompatible
by David 23 May '13

23 May '13

Hello, So I want to setup a Kamailio SIP Proxy( version 4 ) that will do NAT signalling handling. With the help of nat_traversal and path modules. Attached is my kamailio.cfg file. My problem is I want to do keep alive on users who are behind NAT, so I called nat_keepalive();. The script will work fine, but as soon as I enable either add_contact_alias() or fix_contact(), the keep alive doesn't activate. I modified nat_traversal.c to identify the issue and found that it is comparing the contact header from the request and the reply to find the expiration. Trouble is, the request contact header was taken from before I called fix_contact() and the reply header has the contact from after the fix_contact() call. So my traces show that on the return trip the callback is called, I bet the stack trace looks like this : 1. Some method in tm 2. __tm_reply_in() defined at nat_traversal.c:1358 3. get_register_expire called from nat_traversal.c:1377 and defined at nat_traversal.c:878 4. STR_MATCH_STR called from nat_traversal.c:937 and defined at nat_traversal.c:77 STR_MATCH_STR returns false because the headers are different. I added else { LM_ERR("failed to match because request %s is not reply %s", contact->uri.s, r_contact->uri.s ) ; } to the code and see : May 21 12:18:52 kamailio-dev-nat /usr/sbin/kamailio[30163]: ERROR: nat_traversal [nat_traversal.c:946]: failed to match because request sip:username@mytld#015#012#015#012 is not reply sip:username@myip:59080>;expires=60;received="sip:64.18.189.196:5060"#015#012Server: CenseredUA 5.2-rc1#015#012Content-Length: 0#015#012#015 So as you can see the original Contact header has the server's domain name and the response has the IP address that set by fix_contact(). So I think that nat_keepalive() should not be comparing the Contact from the original request but using the contact header from the modified request. Any ideas what to do next ? Thanks, David

3 5

Radius and FLT_ACCFAILED
by Victor V. Kustov 23 May '13

23 May '13

Hello! I do radius accounting. When I imitate call fail, BYE/CANCEL messages are loose, account stop packets not sending. I see FLT_ACCFAILED flag, can I set it for acc_radius module? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.2-calculate GNU/Linux

2 1

auth module Readme on website has missing ToC
by Andrew Pogrebennyk 23 May '13

23 May '13

Hi, please check the link http://kamailio.org/docs/modules/stable/modules/auth.html - the Table of Contents is missing, there's only examples list. Andrew

1 0

Re: [SR-Users] Contents of sr-users Digest, Vol 96, Issue 68
by Jignesh Gandhi 23 May '13

23 May '13

Thanks for the quick reply. 1. Re: SCTP question (Daniel-Constantin Mierla) What I meant is if the SCTP association goes down from the Invite, 200 ok w/sdp to BYE. Should there be not an error at t_relay() level , since the association does not exists ? Currently I don't get any error when I do if (!t_relay()) { xlog("L_INFO", "T_Relay return code is $retcode\n"); sl_reply_error(); } I have set up $du with the IP:port of the host where I received the Invite and ACK to 200 ok w/sdp. I have shut down the other side , so I know that SCTP is not up. Thanks, --Jignesh -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, May 20, 2013 2:25 PM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 96, Issue 68 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: SCTP question (Daniel-Constantin Mierla) 2. Re: [SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour (Daniel-Constantin Mierla) 3. Re: Kamailio + Siremis Outbound route (Daniel-Constantin Mierla) 4. SCTP support of MultiHoming... (Jignesh Gandhi) 5. Re: SCTP support of MultiHoming... (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Mon, 20 May 2013 16:56:55 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Jignesh Gandhi <jigpgandhi(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] SCTP question Message-ID: <519A39B7.3050503(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, On 5/20/13 6:12 AM, Jignesh Gandhi wrote: > Hello, > > I recently started using SCTP relay feature of Kamailio and am > receiving SCTP-SIP and relaying it UDP-SIP and vice versa. > > What happens , if an SCTP association is broken from the distant end > in the middle of the call, is there a way to re transmit the SCTP > message via another route ? By middle of the call, do you mean in between the INVITE and the BYE? I am not that familiar with SCTP, but probably a new one is created with the BYE. The same should be happening with tcp. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

websocket support for kamailio 4.0.1
by Rupayan Dutta 23 May '13

23 May '13

I have installed kamailio 4.0.1 in centOs 5.8 and trying to connect it using SipML 5 client (supplied by Doubango telecomm) using SIP over websocket.But the client can't connect to the server.Is there any special configuration needed for websocket support in kamailio 4.0.1? please help.

2 1

Kamailio 4.0.1 websocket support
by Rupayan Dutta 23 May '13

23 May '13

I have installed kamailio 4.0.1 in centOs 5.8 and trying to connect it using SipML 5 client (supplied by Doubango telecomm) using SIP over websocket.But the client can't connect to the server.Is there any special configuration needed for websocket support in kamailio 4.0.1? please help

2 1

memcached and usrloc
by Victor V. Kustov 23 May '13

23 May '13

Hello! Can I use memcached for usrloc db? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.2-calculate GNU/Linux

5 10

[SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
by johnc 23 May '13

23 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I updated this config to work with version 4. See below. I have calls working over TLS between jitsi sip clients registered to the same proxy of either one of two proxies I have built using this updated config. I can make calls to/from clients over TLS registered to the openrcs.com proxy from both of these proxies. I can't make calls between the two proxies configured with the config below. Each domain has commercial SSL certs. I have rtpproxy configured and working. I would be very grateful if somebody would check the config and see if I have made a mistake. Many thanks. I will post the final working config so it may be of help to others. Regards, John #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_MULTIDOMAIN # # Kamailio (OpenSER) SIP Server v4.0 - By Daniel-Constantin Mierla # http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour # updated for version 4.0 by John Cahill johnc<AT>aktivix.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=3 log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching #modparam("domain", "db_mode", 1) JC commented # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@somedomain.com") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); setflag(FLT_NATS); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRmf9DAAoJELy1jPQ1KER71LUP+wZGvLn3wQuEVpLQIzZ4U7YR UD7l6OpLQEEp349zmWgyNoOWmcZrRcBYhgKHZde+9ZKIDFTMiWUxqPJJrmcDUTi1 ybq98z9Es9KpT6d5ytv/AMG2DNpM6zRgIVTLXqXfnyzAqdbGZJUhIiLwXiLHBaxy P/JBJCmri4qeahIr/hNJOcmD3pUi8KMS2AE2cxvgl4Y4X1bwEPxDH374Z55/ka3A IJlSGTwFUFjW/Ujc5ih4+TYyRgO4khoCtnzQaSQk8VVNEGGkcuIXV3j9JOOQ+yOT B7TdsbeCu5pBZZe8Qy/aIcTRonvtpexrUXI5jSzfxRaWFYjw01KINBRIprNTaUQ0 w+c+MQWz8gOLI3hcw4/9if9fkJMbmUD7deaS6N5Ovu647TLMbhgB3hDI+77+s5Cs EPJNeW8P1iSPJVuASrx9jqfDNeOKvPOxvd6CemCBHngoSW3NAsuospmfDPo+8QFA MbtJQXJ4p2b6g6CrP1FVx1q2cdPbvC8ZFUqxkfPj8AUtAg5vtKvjU+PPnbtDDwbi RxF0KGpCnixVtc3AkTBdX9JcP+4Pc6RvprNlCEsi1AfaK6yzSlCZeblC2OjZ/2oa PkSaWQeBwLQKMi90QVhoSJTI7Z4O6T03j09V9Wul7qkZorcHkZlzi+W4tBqyxL5R 8prN+zlBCZ3nZN5nFosn =39pZ -----END PGP SIGNATURE-----

2 5

Websockets and outbound config file
by אורן אברהם 23 May '13

23 May '13

Dear list Does someone has a working config file for a registrar + outbound edge proxy on the same kamailio server instance that works ? (like the /examples/websocket.cfg but updated to use outbound & path) thanks...

1 0

[OT] Crocodile Network & WebRTC Launch broadcast
by Peter Dunkley 22 May '13

22 May '13

Hello, Crocodile is having a launch event this evening for our new SIP network and SDK (making use of Kamailio with SIP over WebSocket and outbound). The launch will be broadcast live from 18:30 BST. Anyone interested can view the broadcast at http://lnx.so/croc (the link currently points to our blog but will go live properly shortly before the broadcast). Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Re: [SR-Users] I need you help-----about Kamailio 3.3.x andAsterisk 10.7.0 Realtime Integration
by Gertjan Wolzak 21 May '13

21 May '13

Hello, I looked at the asterisk config, looks like you have not configured that. You need to add a dialplan in the extensions config and trunks in the sip confiig. If you want to make the dialplan realtime as well, you should add the dialplan to the db. Success. Gertjan -----Original message----- From: zhengyw zhengyw(a)neusoft.com Date: Mon, 20 May 2013 04:07:33 +0200 To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x andAsterisk 10.7.0 Realtime Integration > Hi Barry: > This issue have not been resolved after following by your method that > modifed the video1_sipregs table struct,attachment is table info and > asterisk log. > Can you help me with this problem? thank you very much! > > best > zhengyw > *asterisk log* asterisk_log.txt > <http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt> > *dbinfo.txt* dbinfo.txt > <http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt> > > > > -- > View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-… > 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

2 1

db_url bug?
by Ján Hrnko 21 May '13

21 May '13

Hi, I have a problem when using uri_db module. When I leave everything on default and not specify modparam "uri_db", "db_url", then it will not connect to the database. In the var/log/syslog I have seen that it tries to connect to database okmaialio instead of kamailio. Is it bug, or did I misconfig something? I left the kamctlrc file intact and everything to default... can you confirm this please? Thank you!

2 1

remove_body() leaves header Content-Type: application/sdp ?
by Konstantin M. 21 May '13

21 May '13

Hello, I'm using a very ugly code to test remove_body(): request_route { remove_body(); $rd = "192.168.254.85"; t_relay(); } Original INVITE (with SDP): --------------------------------- U 192.168.254.102:5060 -> 192.168.254.104:5060 INVITE sip:123@192.168.254.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 269. . v=0. o=root 1763381182 1763381182 IN IP4 192.168.254.102. s=Asterisk PBX 1.8.7.1. c=IN IP4 192.168.254.102. t=0 0. m=audio 16162 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. --------------------------------- Modified INVITE (after remove_body()): --------------------------------- U 192.168.254.104:5060 -> 192.168.254.85:5060 INVITE sip:123@192.168.254.85 SIP/2.0. Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK6c5c.6160e017.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. *Content-Type: application/sdp.* Content-Length: 0. . --------------------------------- So, remove_body() is actualy removes SDP from request but leaves Content-Type header. Why ?

2 1

SCTP question
by Jignesh Gandhi 20 May '13

20 May '13

Hello, I recently started using SCTP relay feature of Kamailio and am receiving SCTP-SIP and relaying it UDP-SIP and vice versa. What happens , if an SCTP association is broken from the distant end in the middle of the call, is there a way to re transmit the SCTP message via another route ? Thanks, --Jignesh Gandhi

2 1

Re: [SR-Users] Msilo configuration‏
by sipatse＠in.gr 20 May '13

20 May '13

2 1

Kamailio + Siremis Outbound route
by Tony Turner 20 May '13

20 May '13

Hi Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install I want to use Kamailio as a proxy edge register to our network. I have installed Kamailio and freeswitch. I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing. Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch Many Thanks Tony

1 0

Kamailio 3.1.x and FreeSWITCH 1.0.6+ for Media Services and SBC Author: Daniel-Constantin Mierla
by Carlton Thompson 18 May '13

18 May '13

http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc#dokuwi… Hello I was trying to use your tutorial on the above page. I am trying to do so using PGQL vs MYSQL. I changed all the mysql parts to pgsql however when i try to start kamailio it fails. I am building the system on alpine linux and attached my kamailio.cfg and kamctlrc. i am able to setup using the default config making necessary changes but when i use yours it fails. Any help would be appreciated. -- Regards, Carlton Thompson, MS Doctoral Student College of Engineering University of Florida Gainesville, FL 32611

1 0

Can I use Kamailio as B2BUA
by Kamal Palei 18 May '13

18 May '13

Dear Kamailio experts I have a typical use case where I want Kamailio to behave as a B2BUA. What I mean here is (assume Kamailio is using TCP for SIP call establishment) 1. For each call it should create a separate TCP connection with next proxy in path. 2. When call ends, it should close that connection. If that call is active for 10hrs, then connection should stay alive till 10hrs. Please somebody help. Thanks kamal

3 8

New developer: Victor Seva
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, I want to announce that a new person got developer GIT write access to repository: Victor Seva. He helped pushing back Kamailio in official Debian packages and lately has submitted many patches to various modules, among upcoming ones is some work on newly added per module debug level (see the logs from yesterday's IRC meeting for more about this feature). His git commit id is: vseva My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

IMS In Kamailio 4.0.0 - what is it?
by Olle E. Johansson 17 May '13

17 May '13

Jason Penton, Richard Good and Carsten Bock has written an excellent article about the IMS extensions in Kamailio 4.0.0. What it is, what you can do with it and what they're currently working on. Go read it on our web site now to learn more about this work: http://www.kamailio.org/w/2013/05/ims-kamailio/ Have a great weekend! /O

2 1

Radius - new style AVP's (misc_radius)
by Victor V. Kustov 17 May '13

17 May '13

Good day! Now I test new style AVP's for radius response storing. Today additional radius responses return in special radpair: SIP-AVP - name, string "email:sr-users@lists.sip-router.org session-timeout#161 nexthopip:h323/0001111@myvoip-gate.kamailio.org" - value. I don't know why developers chose this way. But it's addition kamailio's feature, so let it be. But if radius server responded with standard pairs? Sending Access-Accept of id 60 to 172.16.X.X port 59736 Session-Timeout = 4261674 next-hop-ip = "SIP/00111222333444@cisco-out" SIP-AVP = "email:sr-users@lists.sip-router.org session-timeout#161 next-hop-ip:h323/0001111@myvoip-gate.kamailio.org" session-protocol = "SIP" No chance to get full radius response at this moment. Only parse SIP-AVP pair. This mean that when you wanna integrate K. in existing infrastructure (in my case - billing), u need work with K. and radius. But why, if you already had worked voip devices? I think more flexible solution ADD in misc_radius function to get radpairs as AVP's. See that i got: misc_radius [functions.c:159]: AVP 'Session-Timeout'/0='null'/4261674 has been added misc_radius [functions.c:143]: AVP 'next-hop-ip'/0='SIP/005555777888@cisco-out'/0 has been added misc_radius [functions.c:143]: AVP 'session-protocol'/0='SIP'/0 has been added After radius response i get: $avp(Session-Timeout)=4261674 (integer) $avp(next-hop-ip)="SIP/005555777888@cisco-out" (string) $avp(session-protocol)="SIP" (string) I think its usable and more flexible. I see 3 way to integrate it: 1. additionally with existing radius_load_caller_avps/radius_load_callee_avps (i make same for test) 2. new function in same module, like radius_authorize_request (caller,callee). 3. replace existing style. it's bad for backward compability, of cause. Or it stay my local patch :) If it really needed and usable i may made any (1..3) way with doc and examples :) diff --git a/modules/misc_radius/functions.c b/modules/misc_radius/functions.c index 2a42024..f156939 100644 --- a/modules/misc_radius/functions.c +++ b/modules/misc_radius/functions.c @@ -107,6 +107,60 @@ error: return -1; } +static void generate_avps_rad(VALUE_PAIR* received) +{ + int_str name, val; + unsigned short flags; + VALUE_PAIR *vp; + + vp = received; + + for( ; vp ; vp=vp->next) { + flags = AVP_NAME_STR; + switch(vp->type) + { + case PW_TYPE_STRING: + flags |= AVP_VAL_STR; + name.s.len = strlen(vp->name); + val.s.len = strlen(vp->strvalue); + name.s.s = vp->name; + val.s.s = vp->strvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + case PW_TYPE_INTEGER: + name.s.len = strlen(vp->name); + name.s.s = vp->name; + val.n = vp->lvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + default: + LM_ERR("skip attribute type %d (non-string)", vp->type); + continue; + } + return; + } +} + /* Generate AVPs from Radius reply items */ static void generate_avps(struct attr *attrs, VALUE_PAIR* received) @@ -212,6 +266,7 @@ int radius_load_caller_avps(struct sip_msg* _m, char* _caller, char* _s2) LM_DBG("success\n"); rc_avpair_free(send); generate_avps(caller_attrs, received); + generate_avps_rad(received); rc_avpair_free(received); return 1; } else { -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

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Msilo configuration
by andreas tseiko 17 May '13

17 May '13

Dear All, We would like to store SIP text Messages when the destination Subscriber is Offline.We have insert into kamailio cfg file the configuration lines below.Unfortunately storing messages is unsuccessful.Any ideas of what missing or what could be wrong? Best regards. ************************************************************************************************************loadmodule "msilo.so" #!ifdef WITH_MSILOmodparam("msilo","db_url","mysql://[% kamailio.proxy.dbrwuser %]:[% kamailio.proxy.dbrwpw %]@[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar@xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU@xxxxxxx.local")modparam("msilo","contact_hdr","Contact: <sip:registrar@xx.xx.xx.xx:5062>;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” #########################################################################Store messages to offline Subs########################################################################route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay();} failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };}***********************************************************************************************

1 0

Kamailio and PJSIP Issue
by Ali Jawad 17 May '13

17 May '13

Hi I did setup kamailio to handle websocket, and I am testing using pjsip, my problem is that I can logon using the pjsip client but SIP does not register, pjsip displays a SIP connection timeout error. My xlite client can logon just fine though. The log from kamailio can be seen below http://pastebin.com/XrsQM8Cf Please advice

1 0

[OT] announcement: openrcs.com - free SIP service
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, following several requests during Kamailio World, but also in the direction of federating open source and local communities via SIP (as a result of FOSDEM panel and discussions), a SIP service was built up to be offered for free on best effort manner - the hardware and bandwidth are offered by asipto.com, the service is using the domain openrcs.com. Main goal is to offer a free option for community members (but not restricted to) to create and use SIP accounts. The plan is to enable many (common) new features to allow people to play with, helping also to test kamailio. There is a rather basic (and alpha stage) web portal based on wordpress+buddypress, good enough for the moment to create an account and tune some options - it is available at: - https://www.openrcs.com Along with voice/video calling and instant messaging, the service has enabled features such as simple presence, msrp relay, xcap server and websocket (web portal embeds a bare sip app built using jssip library). Other features will be enable over the time, but have in mind **it is a not a telephony service**, thus no such specific services like emergency, pstn interconnect, etc. Hope it will be useful for many people - who ever wants to try it, head to https://www.openrcs.com ! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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config
by johnc 16 May '13

16 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Daniel, I nearly forgot, can you post your openrcs config ? Many thanks, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRlOvSAAoJELy1jPQ1KER7juoP/RmguFzHtfzdyr+RydqtE/xp TDfyzQCOh/iciln+pXANMtgajwyyHa83504HP05uhXnk3RLvLEoze/8t7/KkwSdR mksANtDJ/cR4Q9LI9iXVplIUORIR98qGl0/C+LODSU5Hsd3/FS5V0v0B+d1gEZPV na4UPlu/FGgCbRfUa/eFVfo/gE38fU8TNbtZqwggSYmmXrKdEUlzqN8cxCeFg0fT LVygThcLhtam1r4NQWpXOjzHl+qDrVNnIwuDng0W1kg3OLvt2GMS2l0FsLglZcMf xr6CcVQT+BvKUKGgubF1PfjcMuVhNGXgLH4dcXEbNs5Z41pDL0ap7nnz/SXiZQxa qHNEgFV9cJQJtPGK0fTuIPVV4YFA7iaMoXlLJbvwI5YwZUG9W3WnytDBuLUcXnSG 0E8Js9KHUHj975BQMaMUFDohLKL1kFv+wiRKbxT0QDVsEtMTMugi3vkLGRgskrNA e798aVReyh+8gB1hcm9bEUCt7Pc/aQBWYiLM40JqdlVS2Y28X4jQ8BYtSVROLGQa Ws0/nspwKWIXmC5DKF1AJNlBI1HSdW78PO1R2hc5uCatjcidthrvOmCwwhC3B/yA 9rPD0ECquOkRzLw7IHIFzM9pAruRDlKVdjhRaoN4hMuN57e/nqKuASPpMMXbj45L 4rSCPsSV9/A7kCLd9miV =88yR -----END PGP SIGNATURE-----

2 1

Delay in "100 Trying" reply message
by Robert R 16 May '13

16 May '13

Hi, There is a big delay in our Kamailio server when repling "100 Trying" after incoming INVITE messages (more than 500 ms). It seems my perl script (and database query) is executed before "100 Trying" being send out. I was wondering if sending "100 Trying" is configurable so I can send it right after INVITE and before executing my perl script. request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); if (is_method("INVITE")) { perl_exec("test"); } t_check_trans(); route(SIPOUT); route(PRESENCE); route(REGISTRAR); .. route(RELAY); } (morthen Thanks R

3 4

IRC Development Meeting
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, it's probably the time to have a IRC development meeting to set the targets for the next major release as well as discuss what is needed these days around the project. I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin time). I created a page to collect the topics, only few from my list being added for now: * http://www.kamailio.org/wiki/devel/irc-meetings/2013a Feel free to add topics there. Also, I would be available on Tuesday at the same time if proves to be more convenient for more people. If none of these dates suit for majority, we can look for other alternatives -- just propose and we can start a poll. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 3

event_route
by Bruno Bresciani 16 May '13

16 May '13

Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards

2 6

redirecting irc channel #sip-router to #kamailio
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, already suggested on the irc channels, I plan to redirect #sip-router to #kamailio to group in a single place and have consistency with the name of the project. That means making #sip-router an invite-only channel and automatically redirecting those that want to join to #kamailio. Hope is ok with everyone, invite-only flag can be removed later. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Trevor Burt 16 May '13

16 May '13

Dear All, using the following link:- http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I have successfully integrated Asterisk and Kamailio on the same box for testing, but am now facing the problem of getting Freepbx to use the same MySQL database tables. My Kamailio and Asterisk install uses the following tables:- sipusers sipregs voicemail voicemail_data voicemail_messages However Freepbx uses the following tables. devices users Any idea on how I fully integrate Freepbx with my Asterisk and Kamailio configuration using Mysql? My goal is to have a Kamailio LB to multiple Asterisk/Freepbx servers sharing the same extension and voicemail databases. I will also probably integrate a2billing and share the a2billing database between the Asterisk/Freepbx servers to create a fully scalable solution. Any advice most gratefully received.

3 2

SER-GettingStarted… update?
by David Wilson 16 May '13

16 May '13

Hi All, One of the most thorough documents I've found for an introduction to Kamailio is the SER-GettingStarted.pdf, but it is now quite dated. Is there an updated version available, or in the works? Cheers, Dave.

1 0

[PATCH] Memory corruption using s.subst transformation
by Juha Heinanen 16 May '13

16 May '13

Daniel wrote: please do the patch to store the new value in _tr_buffer and attach it to mailing list or bug tracker. I haven't looked at code yet, but sounds like there is indeed an issue. I will review the patch and apply it. What was the conclusion about this? Was the patch ever submitted and applied? If not, is the bug still there or has is been fixed by some other means? I'm asking because this looked like a quite serious issue. -- Juha

2 4

Kamailio crash
by kiran bhosale 16 May '13

16 May '13

Hi we have developed the custom module where in we have created shared memory area which is not used by any other module or process and we locked it using locks as recommended. but because of locks the kamailio is getting crashed every now and then . is it really important to protect the shared memory area with locks. if yes then how to avoid the crash ?? so to locate the crash we built kamailio wit h MM_DBG , and other GCC debug flags . but we are getting error "/No symbol table info available/" . how to locate the bugs. the kamailio is built wit h 1 MB of shared memory on MIPS based VOIP gateway which is having * MB of flash and 16 MB or memory (RAM) . thanks in advance .

2 3

kamailio as a proxy
by Giuseppe Carella 16 May '13

16 May '13

Hi all, I'm trying to use kamailio as a proxy between webrtc clients and an openIMSCore. For this reason I'm currently using the dispatcher module to distribute all the requests coming from the clients to the IMS core, and enabled the websocket module for getting the SIP messages in. SO far it works fine for the registration, but it doesn't for the INVITEs. I noticed that kamailio receives the message but it doesn't deliver it to the clients. Please find attached the log and the configuration file that I used. I'm not an expert with the configuration, so any help or feedback would be very appreciated. Thank you in advance. Best regards, Giuseppe Carella

2 1

ERROR messages in DEBUG level
by Alex Solt 16 May '13

16 May '13

Hi, After changing log level from INFO to DEBUG level, I am seeing bunch of ERROR messages which are unknown to me: May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=498 a=26 n=is_methodMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=765 a=26 n=mf_process_maxfwd_headerMay 15 10:26:44 DEBUG: maxfwd [mf_funcs.c:85]: value = 68May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=775 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=27 n=sanity_checkMay 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0May 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:104]: found end of headerMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=snl_0156137698_NSN_CLIENTMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29May 15 10:26:44 DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=508 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=976 a=3 n=returnMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=511 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=833 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=780 a=25 n=has_totagMay 15 10:26:44 DEBUG: siputils [checks.c:106]: totag foundMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=830 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=loose_route The error message are complaining about simple statements in the config file: if (is_method("OPTIONS")) { sl_send_reply("200", "Ok"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } I appreciate your help. Thanks,AS

2 1

home pbx server experience
by u 15 May '13

15 May '13

I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and "scalability". But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage("c") in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u

5 4

log reply message number
by Robert R 15 May '13

15 May '13

Hi, I am trying to add a log when receiving 1xx or 2xx reply message: # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","reply message: $T_reply_code ci:$ci"); route(NATMANAGE); } } Adding the above line does not log the reply message number (180 or 183 or 200, ...). Thanks, R

3 3

Re: [SR-Users] home pbx server experience
by u 15 May '13

15 May '13

Should UAC work with Sipgate and freevoipdeal (dellmont)? Thanks for your answers everyone. 2013/5/15 u <ueberwachungsstaat(a)googlemail.com>: > Should UAC work with Sipgate and freevoipdeal (dellmont)? > > Thanks for your answers everyone. > > 2013/5/15 Stefan Sayer <stefan.sayer(a)googlemail.com>: >> >> if you don't get it all (auth + registrations) working with kamailo, >> you might try sems (iptel.org/sems) with sbc module (auth of outgoing >> calls) and for registrations either reg_agent (cfg file) or >> db_reg_agent (accounts from db) modules. >> >> load_plugins=uac_auth;registrar_client;reg_agent;sbc;xmlrpc2di >> application=sbc >> >> >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_sbc.html >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_reg_agent.html >> http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html >> >> BR >> Stefan >> >> o Edson - Lists on 05/15/2013 06:40 AM: >>> Hi... >>> >>> Use UAC module to manage registrations and play a little with the >>> config (INVITE section) to forward output calls correctly. >>> --- >>> Edson. >>> >>> Em 14/05/2013 11:27, u escreveu: >>>> I would like to share my experience with kamailio and other home pbx >>>> servers. >>>> >>>> Kamailio on my kirkwood home router for my 6 SIP users is perhaps >>>> overkill: I don't really need mysql and "scalability". But at last I >>>> finally managed to make calling between registered users work stable. >>>> My voip clients only work in all NAT scenarios if I work around some >>>> bugs: to use csipsimple on android I had to change rtpproxy_manage() >>>> to rtpproxy_manage("c") in kamailio's default config, so that problems >>>> with conflicting c: entries in the SDP go away. >>>> >>>> I propose kamailio could ship with a special example >>>> kamailio-compatible.cfg that doesn't try to be RFC compliant, but >>>> compatible to the most common voip clients. Right now the only thing I >>>> would change for this is the option for rtpproxy_manage, but I'm sure >>>> others will know more common quirks that could safely be enabled to >>>> increase compatibility. I think this compatibility idea is what yate >>>> sticks to for their defaults. In freeswitch you also have to do it all >>>> manually, and it's much more work to figure things out in their >>>> enormous config files. >>>> >>>> The other SIP proxies I had tried before kamailio officially fit all >>>> my requirements, including support for multihomed dynamic IPs, but >>>> contrary to their claims it didn't work. >>>> Yate was easy to set up, but the default dialplan is more confusing >>>> than powerful and after having made everything work I realised yate >>>> was clogging my CPU and RAM and after some time always randomly >>>> stopped working. This is with only 2 users connected! It also wasn't >>>> possible to fix NAT sdp while leaving the codecs section in the SDP >>>> alone at the same time. I tried to debug the code, but the C++ was so >>>> complex that I had to give up. >>>> Freeswitch was much more difficult to setup, a multihomed setup with >>>> dynamic IP was super buggy and it also didn't help that the >>>> unintuitive configuration is all in complex unreadable XML >>>> configuration files. >>>> >>>> Kamailio and rtpproxy don't officially support dynamic IP address, but >>>> I can just restart both each time my DSL provider forces me to a new >>>> IP address. This happens automatically in the night and is no big >>>> hassle really. The most simple, least-featureful solution works best >>>> it seems. >>>> >>>> Now the last problem I have with kamailio: I don't know how to connect >>>> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). >>>> I would like a simple way to do this, preferably without other >>>> features that always seem to complicate the matters. Is there >>>> something more lightweight and simple than asterisk, freeswitch and >>>> yate, that people use successfully for this task together with >>>> kamailio and rtpproxy? >>>> >>>> u >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> . >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

HELP! I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end.
by Future Lian 15 May '13

15 May '13

I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end. in CRIT.log of siremis/log, I get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all.

2 1

Kamailio 3.3.x is crashing frequently...
by Krishna Kurapati 15 May '13

15 May '13

Hi, When the Server is running it is crashing occasionally. (gdb) bac #0 0x000000000043359e in ?? () #1 0x0000000000000000 in ?? () (gdb) file /usr/local/sbin/kamailio Reading symbols from /usr/local/sbin/kamailio...done. (gdb) bac #0 free_lump_list (lump_list=<value optimized out>) at data_lump.c:504 #1 del_nonshm_lump (lump_list=<value optimized out>) at data_lump.c:661 #2 0x00007f571ff6cd9c in ?? () #3 0x00000000008db480 in mem_pool () #4 0x00007f571ff6c7e1 in ?? () #5 0x000000000139086e in ?? () #6 0x00000039294546e1 in ?? () #7 0x000000000140c7d0 in ?? () #8 0x00007f561f7acb80 in ?? () #9 0x00007f561f3fa6e0 in ?? () #10 0x00007f561f3fa6e0 in ?? () #11 0x0000000000000001 in ?? () #12 0x0000000000000000 in ?? () What could be going wrong here? Krish Kura

2 1

kamailio 4.0.1 and nokia e72 udp
by hiro 14 May '13

14 May '13

using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. tcplog, rtp should be running, but no voice: ... 00:02:13.829682 IP p5795BC1A.dip0.t-ipconnect.de.49172 > koln-5d81d2a9.pool.mediaWays.net.49184: UDP, length 73 00:02:13.847063 IP koln-5d81d2a9.pool.mediaWays.net.49184 > p5795BC1A.dip0.t-ipconnect.de.49152: UDP, length 73 ... /usr/sbin/rtpproxy -l 93.129.210.169 -s unix:/var/run/rtpproxy/rtpproxy.sock -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid ... DBUG:handle_command: received command "Uc100,8,0,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 192.168.1.56 40000 o5r7rmu5p1hc7vci0591;1" INFO:handle_command: new session _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB, tag o5r7rmu5p1hc7vci0591;1 requested, type strong INFO:handle_command: new session on a port 49184 created, tag o5r7rmu5p1hc7vci0591;1 INFO:handle_command: pre-filling caller's address with 192.168.1.56:40000 DBUG:doreply: sending reply "49184 93.129.210.169 " DBUG:handle_command: received command "Lc100,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 87.149.188.26 49172 o5r7rmu5p1hc7vci0591;1 5qt1pav3 INFO:handle_command: lookup on ports 49184/49196, session timer restarted INFO:handle_command: pre-filling callee's address with 87.149.188.26:49172 DBUG:doreply: sending reply "49196 93.129.210.169 " INFO:rxmit_packets: callee's address filled in: 87.149.188.26:49152 (RTP) DBUG:handle_command: received command "D _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 5qt1pav3dsafun5p47o30tj2od1urm6v o5r7rmu5p1hc7vci0591" INFO:handle_delete: forcefully deleting session 1 on ports 49184/49196 INFO:remove_session: RTP stats: 1 in from callee, 3465 in from caller, 3466 relayed, 0 dropped INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session: session on ports 49184/49196 is cleaned up DBUG:doreply: sending reply "0 what do you guys think? it only occured with kamailio+rtpproxy so far. any insight about why the e72 sends such packet? it doesn't happen when i use sip over tcp. is my analysis right, is it a bug in rtpproxy? greetings hiro

4 9

Kamailio for Debian blog?
by Daniel Pocock 14 May '13

14 May '13

I'd like to write a brief blog about the status of WebRTC in Debian, with a focus on SIP I understand Kamailio 4.0.1 is already in unstable, is that recommended for potential WebSocket users? Has anybody else written any quickstart blog about WebRTC with that particular version, possibly with examples that are consistent with the Debian usage? For client side, SIPml5 is packaged, and I've had discussions with the JsSIP guys about packaging. For TURN, has anybody tried the TURN server project from Google code? It appears more advanced than the existing two TURN servers in Debian (e.g. it has database-backed authentication) http://code.google.com/p/rfc5766-turn-server/ and there is a package in progress: http://mentors.debian.net/package/rfc5766-turn-server

6 11

Iptel SIP used with Bria iPhone App
by JE 14 May '13

14 May '13

I'm using an iPhone Bria app and have input Iptel SIP information in the settings, however, when I try to dial a land line I get a "Forbidden (403)" error message. What could be wrong?

2 1

4.0 forking behaviour
by Alex Balashov 14 May '13

14 May '13

Hello, Has something changed about default forking behaviour in >= 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 9

Re: [SR-Users] confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b
by Future Lian 14 May '13

14 May '13

I can't install siremis 3.0 for kaimailio after install kamailio and begin to install siremis in CRIT.log of siremis/log, i get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. 2013/5/14 <sr-users-request(a)lists.sip-router.org> > Mailing list subscription confirmation notice for mailing list > sr-users > > We have received a request from 218.79.223.111 for subscription of > your email address, "lianhc(a)gmail.com", to the > sr-users(a)lists.sip-router.org mailing list. To confirm that you want > to be added to this mailing list, simply reply to this message, > keeping the Subject: header intact. Or visit this web page: > > > http://lists.sip-router.org/cgi-bin/mailman/confirm/sr-users/cd4884b9a2ea4a… > > > Or include the following line -- and only the following line -- in a > message to sr-users-request(a)lists.sip-router.org: > > confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b > > Note that simply sending a `reply' to this message should work from > most mail readers, since that usually leaves the Subject: line in the > right form (additional "Re:" text in the Subject: is okay). > > If you do not wish to be subscribed to this list, please simply > disregard this message. If you think you are being maliciously > subscribed to the list, or have any other questions, send them to > sr-users-owner(a)lists.sip-router.org. > -- Future Lian

2 1

Does Kamailio support PRACK ?
by Alex Solt 13 May '13

13 May '13

Hi, Does kamailio support PRACK method ? Any configuration change is needed?It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE --> Kamailio --> proxy the packet to BB send 180 Ringing --> Kamailio -> proxy the packet to AA send PRAK (increase Cseq) --> Kamailio --> proxy the packet to BB send 200 OK --> Kamailio -> proxy the packet to AKamailio re-send 180 Ringing to A Thanks,AS

3 4

how to install kamailio with websocket function and msrp function on ubuntu？
by 李启明 13 May '13

13 May '13

hi, i am chinese,i am not good at kamailio.i want to install kamailio with websocket function and msrp function on ubuntu,could you give me a help？

3 3

Doing automatic unregister when a WEBSOCKET connection is closed.
by אורן אברהם 13 May '13

13 May '13

Dear List. I'm trying to issue an automatic unregister() (of a registration) when a websocket connection is closed. it is useful when the browser is closed without issuing REGISTER with expires=0. i've tried implementing this using the event_route[websocket:closed] but i don't know how to find the registration uri from the $si:$sp. (that is needed for the unregister() function from the registrar module) thanks for all replies in advance....

5 15

Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 13 May '13

13 May '13

Hello: I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0…: 1. Could you find any mistake in my configuration file? 2. I also wonder how could I be sure of "Be sure you update the listen IP and port as well if Asterisk is running on the same system with Kamailio" and "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio". thank you very much PS：attachment is kamailio、asterisk's congifure file. Best Regards zhengyw ----- Original Message ----- From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration > Hello, > > I am not that familiar to troubleshoot asterisk configuration files, but > from logs I could see the resulting URI is: > > INVITEsip:106@(null) SIP/2.0 > > That is wrong, meaning something incorrect is done when setting it in > asterisk. Maybe someone else can help more with asterisk. > > Cheers, > Daniel > > On 5/13/13 4:02 AM, zhengyw wrote: >> hello daniel: >> thank you very much! but I can't find the problem in the asterisk. >> attachment is asterisk's configure file, kamailio's configure file and >> data.can you help with this problem? >> >> ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 >> >> >> Best Regards, >> zhengyw kamailio.cfg >> <http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg> >> sip.conf <http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf> >> extconfig.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf> >> extensions.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf> >> db_result.txt >> <http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt> >> >> >> >> -- >> View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-… >> 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 > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 > * http://asipto.com/u/katu * > > > _______________________________________________ > 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 --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

1 0

branche_route support
by Bruno Bresciani 13 May '13

13 May '13

Hi all, currently I have been used kamailio and rtpproxy to listen on two different network (bridge mode), and the register of sip users are saved on different locations ("aliases" and "location") according the interface where register is received. When there is a call to a user registered, I use the registered() function of module registrar on my route[ ] block, to discover the network where sip user is connected and use the correct parameters in rtpproxy functions. >From now, I need that my plataform has forking call support and when the same sip user is registered on both networks, I have to use the branch_route[ ] block to process every branch separately because is necessary different rtpproxy parameters. I am with difficult in branch_route to set the correct rtpproxy parameters because I can't find a way to know the network that the branch will be send. Could anybody tell me if exist a form to solve my problem? Best Regards

2 3

SIP notify to UAC, triggered from DB or web
by Attila Megyeri 13 May '13

13 May '13

Hello, I would like to send a SIP NOTIFY message to my SIP client(s), initiated by an event on the server side. //OFF Asterisk has a similar functionality, for instance ("sip notify <phone vendor>-reboot"), to request a reboot of the phone device from the server side. There is no need for an associated SUBSCRIBE request in such a case. ON// Our requirement is, that if a specific event happens on the server side, I want to send an event to the client. My first idea is to use SIP NOTIFY, but perhaps a SIP MESSAGE could do the job as well. The client is registered over TLS, if necessary it could even subscribe for the event. What would you recommend, using the websockets? Or creating a module from the scratch? Thank you in advance, Attila

2 2

(no subject)
by 李启明 13 May '13

13 May '13

hi, i am from china.i have install a kamailio server with wss support ,and i set 4443 as the port(set 8080 as the ws port);the result of "netstat -tnl" is: "Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 127.0.0.1:53 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN tcp 0 0 my_ip:4443 0.0.0.0:* LISTEN tcp 0 0 my_ip:5060 0.0.0.0:* LISTEN tcp 0 0 my_ip:5061 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN tcp 0 0 my_ip:8080 0.0.0.0:* LISTEN" but,when i use ws,it's ok;when i use wss,it come up a 106 error!why?

2 3

i386 vs. amd64
by Juha Heinanen 11 May '13

11 May '13

debian wheezy multiarch support allows installation of i386 packages to amd64 host provided that there is i386 version of libraries used by the package. is there any advantage of amd64 kamailio versus i386 one? how does kamailio benefit from 64 bit architecture or does it not? if not, the on wheezy amd64 packages would not be needed. this is what e.g. google has done with chrome browser packages. i have noticed that amd64 version of kamailio requires quite a lot more memory than i386 version, which it not necessarily a good thing. -- juha

3 2

Is RTP proxy compatilble with SER
by Priyaranjan Nayak 11 May '13

11 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Presence and watcher_domain
by andpe 10 May '13

10 May '13

Hi I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones (trying to run :-)). The phones are seen as being registered in the PRESENCE, REGINFO work unless properly. But I have a question: in the table active_watchers watcher_domain field is the IP address of the phone and not the domain address. Is this normal? Applies only to telephones using REGINFO. eg Jitsi has a valid entry. Andy

3 2

Is RTP proxy compatible with SER
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Cleaning registrations and presentity table after restart
by אורן אברהם 10 May '13

10 May '13

Dear list I want to remove all raws from the presentity table after restart of kamailio. how can i achieve this ? (preferably without deleting the table manualy)

2 1

Serial forking
by Vitaliy Aleksandrov 10 May '13

10 May '13

Hello. As I understand t_load_contacts() / t_next_contacts() can help only in case when branches have different "q" values. The only way I see is to get all contacts via reg_fetch_contacts(), save the list to an AVP and then iterate through the list in the way described in tm module docs. Does anybody know a better way of how serial forking can be implemented if all contacts have equal "q" values ?

2 2

I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 10 May '13

10 May '13

Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error : 1. in the asterisk console: " Unresolvable destination (478/SL)" 2. in the kamailio log: May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Any idea about the cause of this problem? ps:kamailio and asterisk are running in the same machine. Best Regards, zhengyw 郑友闻医疗终端部东软熙康健康有限公司沈阳市浑南新区新秀街2号，A1楼216 Postcode：110179 Tel：+86 24 83662278 Mobile：15242493836 Email : zhengyw(a)neusoft.com http://www.neusoft.com --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

2 1

How to configure Kamailio to use different source port for different call
by Kamal Palei 10 May '13

10 May '13

Dear All I have a Kamalio proxy, and number of phones are connected to it. When all the phones make call, Kamailio forwards00 request to main proxy from same source port. Is there any configuration option in Kamailio.cfg for changing source port for each call. Basically one dedicated connection per session. Kindly let me know if it is there. Thanks Kamal

2 1

Need kamailio-3.1.5 configuration
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, After successful install of kamailio-3.1.5 I am trying to start kamailio,but it throws following error loading modules under /usr/local/lib/kamailio/ modules_k/:/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_s/ 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 Listening on udp: 192.168.90.210:5035 udp: 192.168.90.210:5036 udp: 127.0.0.1:5036 tcp: 192.168.90.210:5035 tcp: 192.168.90.210:5036 tcp: 127.0.0.1:5036 Aliases: tcp: localhost:5036 udp: localhost:5036 0(2580) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2582) INFO: usrloc [hslot.c:53]: locks array size 512 0(2582) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2582) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 Please suggest what can be done ? -- Thanks Priyaranjan

2 1

Siremis v4.0.0 Released
by Elena-Ramona Modroiu 09 May '13

09 May '13

Hi, a new release of Siremis is out - v4.0.0 - web management interface that is compatible out of the box with Kamailio v4.0.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ Regards, Ramona

2 2

Do we need rtpproxy for all kind of NAT?
by John Chen 08 May '13

08 May '13

Hi, I can configure kamailio + rtpproxy to enable calling between user behind the NAT. Thanks for the example config, it is very easy to did it. Do we need rtprpoxy for all kind of NAT? From what i have read, only symmetric NAT that hard to be traversal-ed. While other type of NAT can be traversaled using STUN. But my device that behind openwrt router can't work without rtpproxy. Is it expected behaviour? Or i'm doing something wrong in the config? Thanks

4 6

Call routing
by mark＠brightvoip.co.uk 08 May '13

08 May '13

Hi all, Posted a similar query a few weeks ago, without much interest - any advice appreciated. I have two sites and will send calls between them. I have Kamailio at each site which will route the calls out/in. There are multiple distinct network routes between the sites, accessible via different IP addresses. Each Kamailio has multiple IP's, one for each route. The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable. When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%). I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability. My main headache is that RTP must follow the same route as SIP for each call. After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route). Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio. Any comments or alternate solutions/suggestions would be of interest. Many thanks, Mark

2 1

Problem with Realtime Kamailio-Asterisk integration
by aaaa aaaa 08 May '13

08 May '13

Hello, I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error: [Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'. Kamailio and Asterisk are running in the same machine. Any idea about the cause of this problem? Best Regards, Theo

5 6

sip peering
by johnc 07 May '13

07 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have built a kamailio based phone system along the lines of the tutorial below. I have updated it for version 4.0 and it is running successfully on Debian 7.0. I have set my system up with a Gandi SSL cert rather than a self signed one. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I want to federate/peer with another proxy running Kamailio with TLS. What do I need to do to make this work? Thanks. Regards, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRiDBUAAoJELy1jPQ1KER7sBoP/1G7I5rlDWcy2eC5GaN7aIa6 xjtKSB8/2cHIyws9d5Zb1mJFKbKgRPJRlSODxUD4rqLdUwq3oIP8buVBAp4rx7UT Xg8POBwRTPqccF7+AxelhkplnWsh11rVhfCT/2L0FLgKSelGGCaAl5SNI5rwj9HO On9ACQeCarpP4UbyZK15iSkreMe4eMfErQgqhfyJwjmlIoEYlBiT4QrUPR3I0MbA jWJq2IWl2Jz/ZeDe2Z7vX6vDY6iidYL+kfFqyXOzoajw+/HTpzok79pGwPSMWZVx 2UHximuxyzJKqGXH1G1Ai4sRrWurM/02jH5iaddb3K2k/fGI8NxvDVgS4p7hyUTA JTmXetcYhTFstgQlnaD0U+FO268uPtGbVQjzuxnYy/bwHRnRV8sxKsjihpRTJrUY R6/QU2xMuaKigJn2FTzSodLMeHtSifUc3Od/9THoh4r6RlQOwMMxBGH+xSdTIXOo wmzLqly3sFPEsSp/30ncs33xT+P7SvMGzu+XHx16UZHCvrFhG3ME10a4C06dn53H mOvXPQ/xm0a47uytp859lbhG3PiyuiF5OkBmuaxndKon2w+HisEZ6+FqteYL3Xjq K3a2z8aoDGFzkYgqqyh5Fpn/hGzd7/E8KkI6Ldrffo2QY3uQx8CWnwXZnYsqFzXy dFS7DTGFXWeKZSAwDO+I =Iezv -----END PGP SIGNATURE-----

2 2

Max concurrent TCP connection that Kamailio support ?
by Khoa Pham 07 May '13

07 May '13

Hi, What is the maximum number of concurrent TCP connection that Kamailio can handle ? I see the max_tcp_connections (which is set to 2048), is that the answer ? How to test for this ? Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

Question about relaying
by Leo Brown 07 May '13

07 May '13

Hi My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers. MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch | |---> Our internal routing switch The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other. When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be). What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way. Leo

3 7

Presence and Cisco 79xx phones BLF
by Andrzej 07 May '13

07 May '13

Is it possible to run on phones Cisco 79xx series BLFs? So to show the status of the shared lines. Phones registered with Kamailio. Andy

3 2

rfc: restructuring source code tree
by Daniel-Constantin Mierla 07 May '13

07 May '13

Hello, among the topics discussed just before the last major release series (4.0.x) was one about restructuring the source code tree. It started mainly as a proposal to move source code belonging to core in a dedicated folder, but there could be more variants. It's time start discussing if we do it, and if yes, how. Here is what I could collect so far. a) no change, keep like it is b) move source code of core in a dedicated folder named 'core' c) move all source code in a folder named src, with following sub-structure src/core - for core src/lib - for internal libraries src/modules - for module Feel free to propose other variants or express your opinion regarding the proposed options. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

missing AVP OriginatingRequest in LIR
by Khue Nguyen Minh 07 May '13

07 May '13

Hi all, I use ims_icscf in Kamailio to send LIR message to HSS. But, In LIR message that I received in HSS haven't OriginatingRequest AVP. How I can insert it into LIR message? Thanks & Best regards, Khue Nguyen.

1 0

max number of pipes for pipelimit module?
by Julia 07 May '13

07 May '13

Hello, Is there a limit on the max number of pipes for pipelimit module? Thanks, Julia

2 2

Publish Failed
by Raj Roy Ghandhi 07 May '13

07 May '13

Hi, I installed and configured the latest Kamalio and all the features works except PRESENCE Also it works with NAT clients with RTPProxy. When I register with Jitsi or QuteCom I get 5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found 6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1001@sip.abc.com via sip:1001@124.43.201.156:5060 on behalf of sip:1000@sip.abc.com for event presence 6(10517) ERROR: presence [publish.c:388]: No E-Tag and no body found ERROR: presence [publish.c:388]: No E-Tag and no body found Please advice me how to fix this. Best Regards, Roy.

3 3

How to use Kamailio as load balancer
by Khoa Pham 07 May '13

07 May '13

Hi, As someone asked in http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html about TCP overload, Daniel answerd that "if you have some lengthily operations that have to be applied to your sip traffic and you get congestion because of that, the solution is to use a kamailio as load balancer in front of a farm of proxy doing the actual processing" How to use Kamailio as load balancer ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 2

xcap-diff support in xcap_server module?
by Juha Heinanen 04 May '13

04 May '13

does xcap_sever module support xcap-diff, i.e., is it able to publish xcap-event when some modification of a document happens? -- juha

2 2

Handling of CANCEL in case of no matching INVITE
by Vassilis Radis 03 May '13

03 May '13

Hello all, In the documentation of the t_relay_cancel() (TM module) there is an example that reads: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } What bothers me is the phrase #do the same as or INVITEs, because in RFC( http://tools.ietf.org/html/rfc3261#section-16.10 ) it says: If a response context is not found, the element does not have any knowledge of the request to apply the CANCEL to. It MUST *statelessly* forward the CANCEL request (it may have statelessly forwarded the associated request previously). So aren't we supposed to immediately statelessly forward the CANCEL if t_relay_cancel() did not find the INVITE transaction, instead of doing the same as INVITEs, which could be t_relay() (not stateless)?. Like below: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, forward(); } Am I correct or am i missing something? Thank you in advance, Bill

4 7

service uri not found in rls-services document
by Juha Heinanen 03 May '13

03 May '13

my presence server tries to handle subscribe to sip:jh+presence@test.tutpro.com. that service uri is stored in rls-services document of user sip:jh@test.tutpro.com, but for some reason k does not find the service uri. here is some debug: May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: Handling SUBSCRIBE <sip:jh+presence@test.tutpro.com> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:155]: searching document for user sip:jh@test.tutpro.com May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:240]: rls_services document:#012<?xml version='1.0' encoding='UTF-8'?>#012<rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service><service uri="sip%3Ajh%2Bdialog%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_dialog_rls%22%5D</resource-list><packages><package>dialog</package></packages></service></rls-services> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:253]: service uri sip:jh+presence@test.tutpro.com not found in rl document for user sip:jh@test.tutpro.com any idea why the service uri is not found even when it appears to be in the document: <service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service> could the escaped chars explain the failure or the fact that resource-list is an uri? -- juha

2 8

extra_id_pv parameter
by Bruno Bresciani 03 May '13

03 May '13

Hi All, I need forking a call to multiple destinations in paralel on different network segments, requiring different rtpproxy parameters. Reading the rtpproxy module documentation I discovered that is possible by setting the "b" parameter on rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and rtpproxy_destroy() functions, but I'm with some doubts... I can't understanding how set the extra_id_pv parameter correctly to use when the "b" parameter is used. The "b" parameter is supported on rtpproxy version 1.2.1? Best Regards

2 3

How to decrypt SIP message
by Khoa Pham 03 May '13

03 May '13

Hi, Currently, I'm using TLS and it works fine. But eventually, TLS is just used to agree upon a secret key. And both client and Kamailio use that secret key to encrypt message. In my solution, I have a fixed secret key. And client encrypt SIP message with that secret key. What code should I insert in Kamalio to decrypt it? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 4

Re: [SR-Users] service uri not found in rls-services document
by Peter Dunkley 02 May '13

02 May '13

Last I looked (around February) there were some things in Kamailio presence that aren't in OpenSIPS. For example, the separate notifier processes in both presence and rls (I believe one may have been ported across now by Saul - but not both). There are also extensions relating to the built-in XCAP server (particularly in presence_xml and rls) that are in Kamailio and not OpenSIPS. I know that there are some specific RLS fixes and features in Kamailio that only work when you use the built-in XCAP server. I also made some enhancements to the way pidf-manipulation is handled in Kamailio that I do not think have been ported to OpenSIPS. Basically, while OpenSIPS presence may be more advanced in some ways I am quite sure it is less developed in some others. Having spent a lot of time over the course of around 18 months doing work in Kamailio presence I would strongly resist any attempts to junk what is currently there because I know that I need a lot of the stuff that is in the Kamailio implementation. I don't currently have the time to develop it further at the moment either. I think the "junk them and replace with opensips" approach would be a disaster. Regards, Peter > looks like kamailio presence and especially rls is lagging behind in > development as compared to opensips. it properly handles the escapes > that i had trouble with, support external references in presence > rules, xcal-diff, etc. > > what should we do about kamailio presence modules? junk them and > replace with opensips ones or implement the missing stuff ourselves? > i personally don't have resources for the latter. > > -- juha > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 02 May '13

02 May '13

Thanks for reply, I checked log messages at callee (IPv6 client - Linphone) and result was little surprising for me. This IPv6 client as Callee didnt receive INVITE messages (I thought that yes because of Wireshark at SIP server). So it is my bad somewhere in my local network. But it is little interesting, that in other direction (from IPv6 caller to IPv4 callee) all Reply messages are received at IPv6 client and RTP/UDP packets go in both directions. I checked Iptables and Firewall at all nodes, but nothing (they all are in the same network) and I dont know what filteres messages to IPv6 callee client. btw. the proxy server in the middle is SIP Protocol gateway, which translates messages between IPv4 and IPv6 and packets at this node are OK. thanks for help Jakub Hrabovsky

1 0

Topoh module also masking Via?
by Grant Bagdasarian 01 May '13

01 May '13

Hello, The description of this module says it hides the SIP routing headers, does this also include the Via headers? Grant

3 5

rtpproxy virtual or physical machine
by Grant Bagdasarian 01 May '13

01 May '13

Hello, I understand that machines processing RTP are recommended to be physical machines. Does the same apply for machines just proxy'ing RTP? Also, after how many RTP session would you start noticing issues with RTP when using rtpproxy on a virtual VMware machine? Regard, Grant

2 2

gruu and dead-lock in registrar module
by Andreas Granig 01 May '13

01 May '13

Hi, We're hitting an issue in a deployment where all udp receivers are sitting in FUTEX_WAIT caused by save() -> lock_udomain() and seem to have deadlocked themselves every couple of days. Looking at the code, enable_gruu in registrar is active by default, and in lookup there is a code path /* temp-gruu lookup */ res = ul.get_urecord_by_ruid(_d, ahash, &inst, &r, &ptr); but no lock_udomain is obtained. However, when the execution falls through to the "done:" marker, it does ul.unlock_udomain(_d, &aor); without having called ul.lock_udomain first. 1.) Could someone please review this part? Looks a bit suspicious, although I don't know what implicitly happens in this case. If it were a semaphore and you decrease it to -1 by decrementing it without prior increment, it's essentially causing a dead-lock, but the current locking implementation might work completely different. 2.) Since I have no clue how gruu is supposed to work in detail, and since in our config we don't explicitly handle gruu (no lookup in loose-route, but gruu is enabled by default in registrar and we don't explicitly turned it off), I'm not even sure if we ever hit this code path. I only see that the ruid column in the location table is filled, but in order to get to this part, the ";gr" flag needs to be set in the R-URI for a lookup(), which I don't know whether that happened somehow in some call flows (we only log $ru, which I don't think logs these parameters, right?). Some input is highly appreciated! Andreas

2 4

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus dispatcher
by Aldo Antignano 01 May '13

01 May '13

> Sorry, yeah. you should probably ignore that part :-) The same dispatcher >deals with client connections from the "public" network as well as inbound Ok thanks, it's clear now. :) >> In the REGFWD route you send all the registrations to ALL the Asterisks? >> If so, why? > By sending the registration to all the available Asterisk servers then they > all know where each SIP client can be located. If a call comes in from the > PSTN or from a SIP client Kamailio can just be sent it to any of the * > servers and it will find the registered client. This will work even with > multiple load balancers/dispatchers using DNS SRV. In this way should be an option to keep registration in Kamailio only, let Asterisk do its specific job, and configure Asterisk to accept all calls coming from Kamailio as authorized calls by its IP? > In this scenario Kamailio is mainly acting as a load balancer, topology > hider and security/anti-flood front-end to create a scalable and redundant > Asterisk PBX system. Really depends what you are after. These are exactly my goals and was trying to understand which is the best way to achieve this. Any suggestion is welcome, thanks. Anty ------------------------------ Message: 3 Date: Tue, 30 Apr 2013 23:02:20 +0100 From: Barry Flanagan <barry(a)flanagan.ie> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus Message-ID: <CALJb54_1FGmGV4HfmR1zHgcAGqMsSPu6kriTv72W3adyHF3dQw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

info: monthly mailing list reminder disabled
by Daniel-Constantin Mierla 01 May '13

01 May '13

Hello, I disabled the monthly mailing list membership reminder, being rather annoying when subscribed to lots of mailing lists. The option can be enabled on per subscriber basis, if anyone wants it, then it can be set in personal options at: - http://lists.sip-router.org/cgi-bin/mailman/options/sr-users/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

dlg_end_dlg
by Dragos Oancea 01 May '13

01 May '13

Hi everyone, I believe I've run into a problem when using dlg_end_dlg (with mi_datagram ) called from an external perl script with perl_exec() . Kamailio just locks after a while (few seconds) after disconnecting a call. Some parts of my debug log can be found here: http://pastebin.com/ANaTEifC The problem was observed while performing a stress test. Is it a good idea to call dlg_end_dlg() like that or I can kill an established dialog from the routing script in some other way ? Regards, Dragos

2 5

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus
by Aldo Antignano 30 Apr '13

30 Apr '13

Thank you very much for your reply Barry. Can you explain us yours ip 10.5.75. and 10.5.76. ? Seems your asterisks have two interfaces, one private and one public? In my infrastructure I have: SIP client -> Kamailio ---> Asterisk 01 ---> Asterisk 02 ... ---> Asterisk N Only Kamailio has a public IP address. In the REGFWD route you send all the registrations to ALL the Asterisks? If so, why? I'm trying to achieve something like: SIP clientA ---> REG ---> Kamailio Is Asterisk01 alive? Yes --> REG ---> Asterisk 01 NO --> Choose next one ---> Asterisk 02 ... ---> Asterisk N ---> Asterisk 01 SIP clientB ---> REG ---> Kamailio Is Asterisl02 alive? Yes --> REG ---> Asterisk 02 NO --> Choose next one ---> Asterisk 03 ... ---> Asterisk N and so on, with a round robin algorithm. When Sip ClientA want to call Sip ClientB, it asks Kamailio, that know that SIP Client B is registered on Asterisk 02 and so send the invite to it. I'm going to work over this in the next few days and let you know the result, if positive I'll write a new complete guide on this topic with your valuable suggestions. Any suggestion is welcome BR, Anty ------------------------------------------------------------------------------------------------------ >> I have read and applied the excellent guide found on: >> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb >> >> Now I have added to Kamailio the HA/Load Balancer support, with the >> "dispatcher" module. >> This way I have 1 Kamailio and 2 Asterisk machines. >> >> How can I change the routing logic of the sections route[REGFWD] | >> route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the >> guide above the asterisk binded ip address is cabled in the kamailio config >> code) I have done that. Relevant route entries below. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } # for each Asterisk server in setid=2 (External) we send a registration on behalf of the user. sql_xquery("ca", "SELECT SUBSTRING_INDEX(destination,':',-1) AS port, SUBSTRING_INDEX(SUBSTRING(destination,5),':',1) AS address FROM dispatcher WHERE setid = 2", "ra"); $var(i) = 0; while($xavp(ra[$var(i)]) != $null) { $var(rip) = $xavp(ra[$var(i)]=>address); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $xavp(ra[$var(i)]=>port); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(i) = $var(i) + 1; } sql_result_free("ra"); } # Test if coming from Asterisk. We check the dispatcher "ds_is_from_list()" function to see if this is one of our Asterisk IPs route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } } # Send to Asterisk route[TOASTERISK] { # If call comes in to the .75 iface, we need to send it to the .75 iface of Asterisk as well. # otherwise we send to the .76 iface. We do this by calling different dispatcher sets. This is # because Asterisk needs to use NAT on the .76. (public) interface but not on the .75. if($td=~"10.5.75.") { $var(setid) = 4; xlog("SCRIPT: Call to 10.5.75. ip - using set $var(setid) \n"); } else { $var(setid) = 2; xlog("SCRIPT: Call from $fn to 10.5.76. ip - using set $var(setid) \n"); } # round robin dispatching on set determined above if(!ds_select_dst($var(setid), "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Hope this helps. -Barry Flanagan

2 1

Kamailio 3.x and Asterisk Realtime Integration plus Dispatcher module
by Aldo Antignano 30 Apr '13

30 Apr '13

I have read and applied the excellent guide found on: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb Now I have added to Kamailio the HA/Load Balancer support, with the "dispatcher" module. This way I have 1 Kamailio and 2 Asterisk machines. How can I change the routing logic of the sections route[REGFWD] | route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the guide above the asterisk binded ip address is cabled in the kamailio config code) Thanks, Anty

2 1

[SR-users] Radius AAA
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi again! Need tips to some tricks with radius. 1. when receive sip message with REGISTER method - initiate auth with radius. (Authentification) - radius_www_authorize in auth_radius.so 2. when receive INVITE message - ask radius 'bout permissions and parse answer: get permission and next hop IP. (Authorization) ???? misc_radius ???? 3. when call a) before call send Acct-Status-Type = Start with attributes. b) after call send Acct-Status-Type = Stop with attributes. (Accounting) - acc_radius.so 1,3 - clear, I understand how it works. 2 - ? Kamailio must send request to radius server, get answer and route call to server from "next-hop-IP" field. Can I make it? -- С уважением, Виктор В. Кустов ООО "Брянск Связь-ТВ" +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

siptrace + flatstore
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi all loadmodule "siptrace.so" modparam("siptrace", "db_url", "flatstore:/var/log/siptrace") call sip_trace(); nothing in /var/log/siptrace, no errors in logs. whats wrong? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

var vs avp...
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi! whats difference $var() and $avp()? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

7 24

riud access from route block
by Juha Heinanen 30 Apr '13

30 Apr '13

when i call t_relay() on a tcp contact, branch route is executed. in that branch route i call t_on_branch_failure. now if the tcp connection to the contact is broken, for example, due to uas crash, the branch failure route is never executed and thus i cannot get access to $T_reply_ruid. there is $branch(name) pv that gives access to other branched, but not the "main branch". is there any way to get access to ruid of the main branch when t_relay() fails when called from route block? for example, if there is only one contact (the $ru one), which results in t_relay() failure, is there a way to find out the ruid of this contact? -- juha

1 1

kamailio 3.1.4
by Ryan d'Eon 30 Apr '13

30 Apr '13

Hello, I'm trying to support a kamailio install, and they are having a crashing issue. I'm not looking for detailed help on the problem, I just noticed one thing I'd like to check up on here: kamailio 3.1.4 (the version they are running) isn't available for download anymore, though quite a few other old versions are. Are there known bugs? Meaning, should I be worried about this, and go back with the recommendation to upgrade? If so, which is the ideal upgrade path? 3.1.5? 3.1.6? Thanks for any help. Ryan

2 1

!!! about private emails
by Daniel-Constantin Mierla 29 Apr '13

29 Apr '13

Hello, I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. There are several common sense as well as technical reasons. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. There are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else or this application is missing what so ever tutorial giving the solution you need or simply asking for full config of complex requirements, is not helping, making people ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, nobody here owe you anything. Therefore, if anyone sent me a private message asking about kamailio or continuing a discussion started on mailing lists, then the messages have to be resent on mailing list if one wants an answer. A quick look in my unfiltered folder revealed hundreds of messages during the past year, soon many will be deleted anyhow. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24016, 2013 * http://asipto.com/u/katu *

2 1

Rewrite RURI
by Grant Bagdasarian 29 Apr '13

29 Apr '13

Hello, Is it possible to have Kamailio rewrite the Request URI of the INVITE message but sent the INVITE to another address first? For example - Kamailio (10.0.0.1) receives an INVITE - Kamailio rewrites the RURI to 10.0.0.3 - Kamailio sends the INVITE to 10.0.0.2 - The application at 10.0.0.2 sends the INVITE to 10.0.0.3 Regards, Grant

5 5

Re: [SR-Users] Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

Hi, Sincere apologies, not sure how this happened - please ignore, it was not me who sent it :/ Regards, Charles On 2 April 2013 08:21, Charles Chance <charles.chance(a)sipcentric.com> wrote: > http://www.toyosoken.co.jp/ivtduw.php > > > _______________________________________________ > 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 > -- *Charles Chance* Managing Director t. 0121 285 4400 m. 07932 063 891 -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

1 0

Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

http://www.toyosoken.co.jp/ivtduw.php

1 0

siptrace source ip causes martians
by Thilo Bangert 29 Apr '13

29 Apr '13

Hi, i have a kamailio instance, which is connected to two subnets (A and B) and exchanges sip messages between them. I have setup the siptrace module to send hep tracing messages of all sip messages to a host in subnet B. This works really well for sip messages arriving on the IP of subnet B - for those arriving on subnet A however it doesnt work. The hep messages have the interface address of subnet A set as source address. Since the machine receiving the HEP messages also is connected to both subnet A and B, these messages are (correctly) rejected as martians and never reach the sipcapture module on that host. this is on kamailio 3.3.4. Is there a way to force the source IP of outgoing HEP messages in the siptrace module? thanks kind regards Thilo

2 4

kamaillio as SBC
by Vu Ngo thanh 29 Apr '13

29 Apr '13

I am interested in testing Kamaillio acting as an SBC. The SBC as Outbound proxy server, the IPPhone can register to SIP Server via SBC and make a call via SBC. the topology is bellow The IP Phone A, The IP Phone B --------- register ------- FreeSwitch (SBC-Outbound Proxy) -------- SIP Server Could you please help me how to install and configure FreeSwitch as SBC-Outbound Proxy ? Thanks so much

3 2

Montoring/Reporting Solution - Commercial/Open Source
by Ali Jawad 26 Apr '13

26 Apr '13

Hi I am in need of a solution that does provide me with advanced graphing and report for Kamailio, I.e. ASR/ACD over day/month/destination etc. with tresholds and alerts. Does anyone know about a solution that is compatible with Kamailio and does this ? Thanks

4 5

Issue with app_perl - return value, logging and append_branch
by Mark Boyce 26 Apr '13

26 Apr '13

Hi All I may well be me missing something obvious but I have got three odd ones whilst trying to test app_perl module which I'd appreciate some assistance with. Setup -------- Kamailio version : 4.0.0 (x86_64/linux) from Peter Dunkley's YUM Repository Perl version : v5.10.1 OS version : CentOS release 6.4 Kamailio.cfg highlights; loadmodule "app_perl.so" modparam("app_perl", "filename", "/tmp/test.pl") modparam("app_perl", "modpath", "/usr/lib64/kamailio/perl/") # ** would someone like to add the app_ to 5.1 and 5.2 of the manuals please request_route { . . route(PERL); . . } route[PERL] { if ($rU=~"^124" && method=="INVITE") { xlog("L_INFO","perl: match $rm $ru from:$fu ($si:$sp) to:$tu rpid:$re\n"); if (perl_exec("testrouting")) { xlog("L_INFO","perl: true exit\n"); } else { xlog("L_INFO","perl: false exit\n"); } } } /tmp/test.pl has the following in it ... #!/usr/bin/perl use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; return 1; } There is no valid route for 124@ and the above works exactly as I'd expect; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- Kamailio log gets "perl: true exit" <- Proxy responds with "404 Not Found" -> Phone sends ACK ------------------------------------------ First problem is that if I swap the "return 1" in test.pl for "return 0" what I see is; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- I Hang Up Phone --> Phone send CANCEL --> Phone send CANCEL --> Phone send CANCEL In this second test once perl_exec is called all routing seems to stop. There's nothing from the xlog to say if perl_exec returned true or false. -------------------------------------- The second problem I have is when I add "Kamailio::log(L_INFO," line it never arrives in any of the logs use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; Kamailio::log(L_INFO, "PERL : testrouting\n"); return 1; } -------------------------------------- The third one is that I cannot get append_branch to do anything. If I add to the perl script the lines; my $m = shift; $m->rewrite_ruri("sip:anotheruser\@ourserver"); The call gets diverted as expected. However if I use append_branch; $m->append_branch("sip:anotheruser\@ourserver"); It has no effect at all. -------------------------------------- That's as far as I have experimented with the module so I can't say if everything else is working as expected :-) Anyone have any thoughts? Cheers Mark

2 1

Problem with PUA
by andpe 26 Apr '13

26 Apr '13

Hi I am new to Kamailio. I have a configuration with existing phones. I call between them. However, I have a problem with the configuration of PRESENCE. I guess I'm doing something wrong. Can someone send me a sample setup working. I looked for a long time information, but I could not fix my problem. In the logs I get the information: 6 (19259) ERROR: pua [send_publish.c: 578]: New PUBLISH and no body found-invalid request 6 (19259) ERROR: pua_usrloc [ul_publish.c: 326]: while sending publish [cycle # 12, if constant make sure server lives] Version: Server :: kamailio (4.0.0 (i386/linux)) Build :: mi_core.c compiled on Apr 4 2013 11:20:52 with gcc 4.4.5 Flags :: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT :: 1d009c regards Andy

2 1

RPM packages for v4.0.1
by Daniel-Constantin Mierla 26 Apr '13

26 Apr '13

Hello, rpm packages for v4.0.1 are no available on opensuse build service - more details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

misc_radius - too many extras
by Victor V. Kustov 26 Apr '13

26 Apr '13

Hi all I've got same error: misc_radius [extra.c:80]: too many extras -> please increase the internal buffer How I get increase buffer in config? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 1

RLS Module - Getting NOTIFY for offline users
by Attila Megyeri 25 Apr '13

25 Apr '13

Dear Users, We use Kamailio as our SIP proxy and Presence server, leveraging XCAP and RLS functionality. After issuing SUBSCRIBE rls@<domain> I would like to get presence information for the contacts in the RLS list not only when they are online, but also if they are offline. Is this possible? Currently we are getting NOTIFY only for those contacts that are online (presence open) Thank you in advance, Attila

2 2

Kamailio v4.0.1 Released
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, Kamailio SIP Server v4.0.1 stable release is out. This is a maintenance release of the latest stable branch, 4.0, that includes fixes since release of v4.0.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.0.1. For more details about version 4.0.1 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/04/kamailio-v4-0-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 25 Apr '13

25 Apr '13

Hi, 1.) Here INVITE message sent to IPv6 SIP Client, but he doesnt respond: INVITE sip:jano@[2001:4118:300:122:bca7:31ac:ccab:d6f6];line=e7b07b8cb7a33ee SIP/2.0 Record-Route: <sip:[2001:4118:300:122:444A:3FFF:FECC:61A0];lr=on> Record-Route: <sip:[2001:4118:300:122:A00:27FF:FE60:83C];r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.207;r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.46;lr=on;nat=yes> Via: SIP/2.0/UDP [2001:4118:300:122:444A:3FFF:FECC:61A0];branch=z9hG4bK1c7c.bf9123f4.0 Via: SIP/2.0/UDP [2001:4118:300:122:A00:27FF:FE60:83C];branch=z9hG4bK1c7c.76b1542.0 Via: SIP/2.0/UDP 158.193.139.46;rport=5060;branch=z9hG4bK1c7c.af9123f4.0 Via: SIP/2.0/UDP 10.0.2.15:5060;received=158.193.139.205;rport=55754;branch=z9hG4bK1417756364 From: <sip:jakub@ip02.sip.uniza.sk>;tag=632445346 To: <sip:jano@ip02.sip.uniza.sk> Call-ID: 267806174 CSeq: 21 INVITE Contact: <sip:jakub@158.193.139.205:55754> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 67 User-Agent: Linphone/3.3.2 (eXosip2/3.3.0) Subject: Phone call Content-Length: 470 P-hint: outbound v=0 o=jakub 123456 654321 IN IP6 2001:4118:300:122:a00:27ff:fe60:83c s=A conversation c=IN IP6 2001:4118:300:122:a00:27ff:fe60:83c t=0 0 m=audio 51652 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000/1 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=on a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=nortpproxy:yes 2.) I get here cap files of SIPv64 server and Bridge as Enclosure. Here i get my hosts file for wireshark for clear view of SIP entities: 158.193.139.46 SIPserver_IP4 158.193.139.207 Bridge_IP4 2001:4118:300:122:444a:3fff:fecc:61a0 SIPserver_IP6 2001:4118:300:122:a00:27ff:fe60:83c Bridge_IP6 158.193.139.205 WinPC_IP4 2001:4118:300:122:bca7:31ac:ccab:d6f6 IP6_Client 10.0.2.15 IP4_Klient

2 1

Kamailio dialog termination
by Grant Bagdasarian 25 Apr '13

25 Apr '13

Hello, Using the dialog module. According to the documentation a dialog is automatically destroyed when a BYE is received. In case of no BYE, the dialog lifetime is controlled via the default timeout. I'm assuming the dialog module only creates a dialog when a 200 OK is received on the INVITE. So if a CANCEL request is sent for this INVITE or the server answers with anything but a 200 OK, no dialog is created and maintained by the dialog module? Also, when the dialog is terminated, is it guaranteed the dialog entry is removed from memory at all times, but there might be a chance the dialog remains stored in the database table because of some error? I'm using the dialog module to control the number of calls allowed to a certain destination, but it sometimes does occur that an entry in the database is not removed, but I'm not sure if the dialog is also removed from memory when this happens. I hope someone could clear this out for me. Thanks, Grant

3 5

★ SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List, Phillip left a message for you
by Badoo 25 Apr '13

25 Apr '13

You can instantly reply using our message exchange system. Check your message http://eu1.badoo.com/0138644786/in/y9a..eJmaMo/?lang_id=3&g=57&m=61&mid=517… Some other people in the area: If clicking the links in this message does not work, copy and paste them into the address bar of your browser. You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=sr-users%40lists.s… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at Media Village, 131 - 151 Great Titchfield Street, London, W1W 5BB.

1 0

Planing to release v4.0.1
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, I am considering releasing v4.0.1 next week on Thursday. There are some issues under discussions on mailing list, they will be sorted by that time, but if you are aware of others, please report them on dev mailing list or tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

www.kamailio.org rebooted for maintenance
by Henning Westerholt 24 Apr '13

24 Apr '13

Hello, I've needed to reboot www.kamailio.org two times for maintenance (kernel and grub boot loader update) a few minutes ago. The machine which its services is now up again, please let me know if you find some issues because of the change. Thanks and regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications Tel. +49 721 91374-4483, Fax. +49 721 91374-2734 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Robert Hoffmann, Andreas Hofmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen, Martin Witt, Christian Würst. Aufsichtsratsvorsitzender: Michael Scheeren. Amtsgericht Montabaur / HRB 6484

1 0

BYE dialog timeout bad syntax
by Julia 24 Apr '13

24 Apr '13

Hello, The BYE, generated by Kamailio in case of dialog timeout, is malformed: BYE <sip:anonymous@192.168.225.2:50601> sip:anonymous@192.168.225.2:50601 SIP/2.0 Via: SIP/2.0/UDP 172.16.60.113;branch=z9hG4bK77b6.941684b6.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-JfymiMenCtp4urS5CXSnCHJnCXJZiHvRCHYJb6ggNPN9NVDL3 -TWhdW0UvOfN1KPTReRiHy1CHmcs6mZiHS* To: <sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1> sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1 From: <sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587> sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587 CSeq: 465332801 BYE Call-ID: <mailto:1aebb42c-01e3a8c0-000007c2@192.168.225.2> 1aebb42c-01e3a8c0-000007c2(a)192.168.225.2 Content-Length: 0 User-Agent: kamailio (3.3.2 (x86_64/linux)) Max-Forwards: 69 The bad syntax is located in the "To" header field. According to the RFC 3261 section 25.1, the grammar of the To header field is as follow: To = ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param ) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI / SIPS-URI / absoluteURI to-param = tag-param / generic-param The RFC 3261 section 20.10 explains that: Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". As the "addr-spec" <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2 contains a semicolon, it means that the "name-addr" form must be used. The "To" header field should be encoded as follow: To: < <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2>;tag=01e3a8c0-c1c2-t-1 BR, Julia

2 1

Fwd: save accdb records with milisec in timestamp field
by Uri Shacked 24 Apr '13

24 Apr '13

Hi, can anyone tell me if it is possible to save the time and date using accdb (mysql) with milisec as well? the field on the DB is set as TIMESTAMP. BR, Uri

3 5

Re: [SR-Users] Kamailio 4.0 IMS
by Jason Penton 23 Apr '13

23 Apr '13

Thanks Carsten. We have also started some other docs with Richard and Olle. Olle have you managed to publish the doc Rich and I shared with you - at least this is a nice overview of the modules, etc? I know there are still some module docs outstanding from me which I will get to ASAP - apologies for that! Cheers Jason On Tue, Apr 23, 2013 at 11:40 AM, Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>wrote: > Thank you Carsten very much**** > > ** ** > > Palo73**** > > ** ** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Carsten Bock > *Sent:* Tuesday, April 23, 2013 10:08 AM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* andreea.ancuta.corici(a)fokus.fraunhofer.de; Dragos Vingarzan > *Subject:* Re: [SR-Users] Kamailio 4.0 IMS**** > > ** ** > > Hi,**** > > i was approached by several people after Kamailio World 2013, that our > installation guide is not available anymore.**** > > The "old" version was at some point quite outdated, that is the reason why > i disabled the page. However, due to the great demand i will update the IMS > on Kamailio how-to later this week and publish the updated version (and i > will move it to the Kamailio Wiki so other points can be easily added by > different people).**** > > IMS rocks!**** > > ** ** > > Kind regards, > Carsten**** > > ** ** > > ** ** > > 2013/4/23 Pavel Segeč <pavel.segec(a)fri.uniza.sk>**** > > Sir, > > Thank you for your answer. Aggree all you wrote and I'm aware of that. > Just I'm saying helps us..we use, popularize and love Kamailio project > however we need some help ...that's all... Kamilio 4.0 IMS is a quite new > thing and we are starting to use it during official student classes and > labs, we are gaining experiences and we share them,we reports bugs and so > on, we do not hide our knowledge > > palo73**** > > > > > > > > > _______________________________________________ > 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**** > > > > > -- **** > > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/**** > > ** ** > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________**** > > ** ** > > Tuto spravu preveril ESET NOD32 Antivirus.**** > > ** ** > > http://www.eset.sk**** > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > _______________________________________________ > 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 > > -- Jason Penton Senior Manager: Applications and Services Smile Communications Johannesburg, South Africa Phone: +27 83 283 7000 This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 1

Simulating kamailio config flows
by Andreas Granig 23 Apr '13

23 Apr '13

Hi, I'd like to put a topic up for discussion to test kamailio config logics. The standard way of doing so is to start kamailio as usual and write sipp or sipsak scenarios to perform automated tests. This can get quite complex pretty quickly, as you always have to take care of covering the full transaction on the client side, where in most of the cases you really just want to know which IP/port kamailio would send the message to, and how some headers look like after processing. Also, it doesn't really allow to inspect your variables and avps etc. on the fly other than logging them to syslog and parsing it from there. Let me know if this is stupid and/or a complete overkill, but what about introducing some kind of dummy mode, where you'd pipe a message into kamailio via stdin, and get the resulting message out on stdout (e.g. in ngrep style with ip information as first line, plus the content following), plus a dump of internals (e.g. vars, avps) on stderr as they're being assigned, and once the message is processed, kamailio would just shut down again? There are quite some things to consider, like not doing any timer handling in tm, but would this be something anyone would be interesting to discuss in more detail? If I'm completely on a wrong track, what are suggested alternatives to testing full sip call flows other than running automated UAs against kamailio? Andreas

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Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 23 Apr '13

23 Apr '13

Hi, Once in a while it captures outbound SIP Invite but it captures always 100 trying, 180 and 200 OK and some times it captures inbound SIP INVITE as well. Thanks Ram

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