- 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

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