sr-users June 2013

sr-users@lists.kamailio.org

94 participants
170 discussions

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

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

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sr-users June 2013