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37455 discussions

[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

I am afraid that rating all calls the same regardless of destination will not work unless you define some destination to match all your traffic. The rating engine must find a destination prefix from which it derives the price. Read the RATING.txt to understand the logic behind it. Adrian >> I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination...

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Re: [Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

Yes You need to add at list one customer entry, associated profiles and finally the rates linked with the profile which contain the actual prices for each destination. You have example rating files that you can modify. Adrian On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote: > That's not a problem. I am currently able to successfully match > destinations by looking for anything that starts with a '1'. I have > a destination setup called "Billable" which applies to any > destination beginning with 1. I also have destinations setup for > 1800, 1877, etc to filter out the 800 numbers. > > What I cannot figure out is how to link a rate to a destination. Do > I have to setup a profile and customer as well? Or should I be able > to price calls using only destination and rates? > > On Thu, Jul 31, 2008 at 8:42 AM, Adrian Georgescu <ag@ag- > projects.com> wrote: > I am afraid that rating all calls the same regardless of destination > will not work unless you define some destination to match all your > traffic. > The rating engine must find a destination prefix from which it derives > the price. Read the RATING.txt to understand the logic behind it. > Adrian > >> > I'm attempting to understand how call rating works within CDRTool. > I am > able to link destinations to the calls, but unable to generate > prices. I'm > using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What > I'm > trying to setup seems simple: I have one set rate of $.08/min for any > call > beginning with a 1 (I'll exclude 800 numbers later). This rate should > be > applied globally with no differentiation between caller party or > destination... > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

Hi Neill, The outgoing audio works fine. The audio that is not passed is from PSTN -> Phone. The Snom does have symmetrical RTP set. I did have this working about a week ago then it stopped working! How does asterisk deal with NAT, as if I connect the phone directly to Asterisk everything works perfectly! Both Asterisk & Kamailio are on publically accessible IP addresses. Thank you for your help so far. Ross From: neill.wilkinson(a)btinternet.comTo: users(a)lists.kamailio.orgDate: Thu, 31 Jul 2008 13:32:05 +0100Subject: Re: [Kamailio-Users] NAT Problems Which direction is the one way audio in – from the Phone to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s the audio returning to the phone from the PSTN you’re missing? Do you have symmetrical RTP set in the Line settings (under NAT) on the SNOM? Is Kamailio inside or outside the NAT? Neill...;o) Neill WilkinsonPrincipal Consultant Aeonvista Ltd - opening up new ideas From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross BeerSent: 31 July 2008 12:50To: users(a)lists.kamailio.orgSubject: [Kamailio-Users] NAT Problems I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= if (!mf_process_maxfwd_header("10")) {sl_send_reply("483","Too Many Hops");exit;};if (msg:len >= 2048 ) {sl_send_reply("513", "Message too big");exit;};# NAT detectionforce_rport();if(nat_uac_test("3")){fix_nated_contact();}if (!method=="REGISTER"){record_route();}if (loose_route()) {append_hf("P-hint: rr-enforced\r\n"); route(1);};if (!uri==myself) {append_hf("P-hint: outbound\r\n"); route(1);};if (uri==myself) {if (method=="REGISTER") {if (!www_authorize("", "subscriber")) {www_challenge("", "0");exit;};if (isflagset(5)) {# set branch flag -- when someone will call this user# the INVITE will have branch flag 6 set after lookup("location")setbflag(6); };fix_nated_contact();fix_nated_register();consume_credentials();save("location");exit;};if (!lookup("location")) {sl_send_reply("404", "Not Found");exit;};append_hf("P-hint: usrloc applied\r\n"); };if(method=="MESSAGE"){if (!lookup("location")){sl_send_reply("404", "User Offline");exit;}route(4);}if (method=="REGISTER"){save("location");} route(1);} route[1] {if (subst_uri('/(sip:.*);nat=yes/\1/')){setbflag(6);};if (isflagset(5)||isbflagset(6)) {route(3);}t_on_reply("1");if (!t_relay()) {sl_reply_error();};exit;} route[3] {if (is_method("BYE")) {unforce_rtp_proxy();} else if (is_method("INVITE")){fix_nated_sdp("3"); };if (isflagset(5)){search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');}t_on_reply("1");}route[4]{if (!t_relay()) {sl_reply_error();};exit;} failure_route[2] {if (isbflagset(6) || isflagset(5)) {unforce_rtp_proxy();}}onreply_route[1] {fix_nated_contact();if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) {fix_nated_sdp("2");force_rport(); }search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');} ==================== SIP PACKETS ===========================INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asteriskMax-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxxs=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY ////////////////////SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 Find out how to make Messenger your very own TV! Try it Now! _________________________________________________________________ Invite your Facebook friends to chat on Messenger http://clk.atdmt.com/UKM/go/101719649/direct/01/

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[Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection force_rport(); if(nat_uac_test("3")) { fix_nated_contact(); } if (!method=="REGISTER") { record_route(); } if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # the INVITE will have branch flag 6 set after lookup("location") setbflag(6); }; fix_nated_contact(); fix_nated_register(); consume_credentials(); save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; if(method=="MESSAGE") { if (!lookup("location")) { sl_send_reply("404", "User Offline"); exit; } route(4); } if (method=="REGISTER") { save("location"); } route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { fix_nated_sdp("3"); }; if (isflagset(5)) { search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } route[4] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[2] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { fix_nated_contact(); if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) { fix_nated_sdp("2"); force_rport(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } ==================== SIP PACKETS =========================== INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asterisk Max-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxx s=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY //////////////////// SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID *** CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508 v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP *** t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 _________________________________________________________________ Find the best and worst places on the planet http://clk.atdmt.com/UKM/go/101719807/direct/01/

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[Kamailio-Users] Kamailio and Snom phones
by Jason Penton 31 Jul '08

31 Jul '08

Hi * not sure if someone has encountered this before but I have noticed a problem with Snom phones registering with Kamailio. It seems that the Snom phones register with a contact port of 2049. However, when Kamailio does the usrloc lookup and the corresponding t_relay, a packet dump shows me that Kamailio is trying to send the invite to port 5060. Can anyone shed some light? Kind regards Jason

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[Kamailio-Users] Error in database module
by Schumann Sebastian 31 Jul '08

31 Jul '08

Dear all I use OpenSER with the perl extension to publish presence states for non-SIMPLE/PUBLISH capable devices using an outbound proxy. My idea is to send all SIP messages to an outbound-proxy, that forwards them to the respective registrar/SIP proxy. The outbound proxy uses SIP messages (INVITE, BYE, CANCEL...) as triggers to call a perl script that publishes the presence state to a presence agent using PUA_MI. Currently, registrar, presence agent and the publishing functionality are hosted on one machine. But I think that should not have any impact on the whole idea... All what I wanted to do works fine so far, except that after some time, I get MySQL errors. The Perl script does not use the database, only from OpenSER script, actions are called. I posted below some of the error messages I receive and after, the part of my config performing database queries. Does anyone see a obvious mistake in there? I tried OpenSER lastest branch and current trunk, both produce the error after some time. Thanks for your help! Sebastian Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:07:30 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:07:30 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:pua:db_update: while querying db table pua Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:xcap_client:query_xcap_update: in sql query Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_delete: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:update_db_subs: deleting expired information from database Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_watchers_clean: querying database for expired messages Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Jul 30 11:09:10 [4329] ERROR:mysql:db_mysql_query: error while submitting query Jul 30 11:09:10 [4329] ERROR:presence:msg_presentity_clean: querying database for expired messages 203 if (uri==myself) { 204 if (method=="REGISTER") { 205 # Uncomment this if you want to use digest authentication 206 if (!www_authorize("IP", "subscriber")) { 207 www_challenge("IP", "0"); 208 exit; 209 }; 210 211 # query if dialog exists for registering URI 212 avp_delete("*"); 213 if (avp_db_query("SELECT from_uri, 214 to_uri 215 FROM dialog 216 WHERE from_uri='$tu' or to_uri='$tu' or from_uri='$tu:5060' or to_uri='$tu:5060'")) { 217 #xlog("L_INFO", ">>> Dialog active: caller=$avp(i:1) callee=$avp(i:2)\n"); 218 } else { 219 #xlog("L_INFO", ">>> No Dialog active: avp_db_query() returned no records\n"); 220 }; 221 xlog("L_INFO", ">>>>>>>>> REGISTER received ($tu)\n"); 222 if(!is_avp_set("$avp(i:1)")) { # no call active for registering to_uri user 223 xlog("L_INFO", ">>>>>>>>> PUBLISHED\n"); 224 perl_exec("publish"); 225 } else { 226 xlog("L_INFO", ">>>>>>>>> NOT PUBLISHED, call active\n"); 227 } 228 save("location"); 229 exit; 230 } 231 232 if(is_method("SUBSCRIBE|PUBLISH")) { 233 if(is_method("PUBLISH")) { 234 if($si == "IP") { 235 if (!t_newtran()) { 236 sl_reply_error(); 237 exit; 238 }; 239 handle_publish(); 240 t_release(); 241 } else { 242 sl_send_reply("488","Not Acceptable Here"); 243 } 244 } else if(is_method("SUBSCRIBE")) { 245 if (!proxy_authorize("", "subscriber")) { 246 proxy_challenge("", "1"); 247 exit; 248 }; 249 handle_subscribe(); 250 t_release(); 251 }; 252 exit; 253 }; 252 exit; 253 }; 254 255 lookup("aliases"); 256 257 # native SIP destinations are handled using our USRLOC DB 258 if (!lookup("location")) { 259 sl_send_reply("404", "Not Found"); 260 exit; 261 } else { 262 append_hf("P-hint: usrloc applied\r\n"); 263 if(method=="INVITE") { 264 # Store Dialog in db 265 setflag(5); 266 # Session timer 267 setflag(6); 268 if (sstCheckMin("1")) { 269 xlog("L_ERR", "422 Session Timer Too Small reply sent.\n"); 270 exit; 271 } 272 xlog("L_INFO", ">>>>>>>>> INVITE received ($fu busy)\n"); 273 perl_exec("publish"); 274 } 275 }; 276 };

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[Kamailio-Users] FYI: A1 InnovationDays
by Daniel-Constantin Mierla 31 Jul '08

31 Jul '08

Maybe some developer has a brilliant idea involving openser and looks for some support. Also, maybe you will find there ideas that will help you in the future, so you can comment and vote for it. Cheers, Daniel :: Final Call for the International Developer Challenge A1 InnovationDays :: Since almost 4 weeks, the international developer challenge A1 InnovationDays attracted more than 30 ideas from creative minds. The challenge is open to anyone who addresses the grand theme "Open Communication by Open Standards" and can come up with ideas for cool services built on IP-based communication (SIP & IMS). Ideas can be posted on the platform www.a1innovations.at/en/static/a1innovationdays until August 15, 2008. All entries are open for voting and discussing by all users. On August 15, the 10 highest ranked ideas will be short-listed and judged by a panel of recognized experts as well as representatives from the sponsors. End of August, the best teams behind the ideas will be invited to a prototyping event near Vienna in order to realize their ideas and demonstrate their potential. The winning team can choose among several awards, such as a week of powder snow skiing in Austria. After that, the sponsor (mobikom Austria) is keen to bring your idea to the market. All submissions must be based on Open Source Software. : More Information : For all details about this challenge, go to the FAQs: http://www.a1innovations.at/en/static/faq For registration, go directly to: http://www.a1innovations.at/registrierung : Important dates ahead : August 1, 2008: Two prizes for the best interim ideas will be awarded August 15, 2008: End of idea entries (CET 24:00) August 18, 2008: Announcement of the five selected teams for the prototyping event August 29 - September 1, 2008: Prototyping event at the monastery Und (near Vienna: http://www.und.at/) Other notes about the event: http://blog.tmcnet.com/blog/tom-keating/voip/a1-innovationdays-contest-look… -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Detecting Two Phones Behind Same NAT
by Ross Beer 31 Jul '08

31 Jul '08

Hi, Is there a way to detect two phones behind the same NAT device? Thanks in advance, Ross _________________________________________________________________ The John Lewis Clearance - save up to 50% with FREE delivery http://clk.atdmt.com/UKM/go/101719806/direct/01/

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[OpenSER-Users] Firewall
by CSB 31 Jul '08

31 Jul '08

I have an error which is driving me crazy: Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:core:udp_send: sendto(sock,0x81aaed8,825,0,0xb61216f0,16): Operation not permitted(1) Jul 24 16:58:34 beta /sbin/openser[5446]: ERROR:tm:msg_send: udp_send failed This is firewall related because when I turn iptables off the problem goes away. But although all ACCEPT and DENY messages are logged by the firewall, there is no corresponding message logged. When a call is made between two UACs and the far end attempts to answer the call, the call is not answered successfully. The dialogue is shown below. If I turn the firewall off, start a call and then turn the firewall on the call continues successfully. However the next call is not successful. We use Mediaproxy and I suspect that is trying to do some communication that is being blocked by the firewall but that is only a suspicion. The dialogue for this call is shown at the bottom. When comparing these two dialogues with the firewall turned on: - when the callee answers, OpenSER receives a 200 OK but doesn't immediately pass that on to the caller - after a number of 200 messages from the callee, OpenSER sends the 200 to the caller but the Contact is the private IP address whereas when the firewall is turned off it is the public IP address I have also included the iptables config. Note that some lines are commented out due to the testing I'm doing and IP addresses have been changed. Any advice on the changes required would be appreciated. Thanks **************************** SIP dialogue with firewall on U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2222 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK31ee9ba27B17140D;rport=5060;received=58.28.001.0 01..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790239 1216790239 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35982 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 192.168.1.124:5065ACK sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=566CA8D1-4C8E0458>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK4b4d0d4aF803AD55..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK4b4d0d4aF803AD55..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..CSeq: 1 ACK..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK07da.73971d95.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..From: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bK31ee9ba27B1714 0D..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 298748 298748 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16424 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKc0cc.a198b237.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=566CA8D1-4C8E0458;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-947b0ac7..From: <sip:44556644@domain.com;user=phone>;tag=e07209dc8d8de14fi5..To: "CSB" <sip:44556648@domain.com>;tag=566CA8D1-4C8E0458..CSeq: 101 BYE..Call-ID: 7f806604-ea80e337-e14b8216@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=e07209dc8d8de14fi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************************** SIP dialogue with Firewall off U 58.28.001.001:5060 -> 147.202.001.001:5060INVITE sip:44556644@domain.com:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 251....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 192.168.1.102..t=0 0..m=audio 2224 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca52e47AD4AD366;rport=5060;received=58.28.001.00 1..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Server: OpenSER (1.3.2-notls (i386/linux))..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065INVITE sip:44556644@192.168.1.124:5065 SIP/2.0..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>..CSeq: 1 INVITE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 69..Content-Type: application/sdp..Content-Length: 253....v=0..o=- 1216790431 1216790431 IN IP4 192.168.1.102..s=Polycom IP Phone..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 8 18 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000.. U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 100 Trying..To: <sip:44556644@domain.com;user=phone>..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 180 Ringing..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5060 -> 147.202.001.001:5060ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 70..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065ACK sip:44556644@58.28.001.001:5065 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=3D7BF99A-F3B3ACE5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.2..Via: SIP/2.0/UDP 192.168.1.102;received=58.28.001.001;branch=z9hG4bK9bb7064fBA56088E..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..CSeq: 1 ACK..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Max-Forwards: 69..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKf294.1cda4646.0..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@192.168.1.124:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 208..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 192.168.1.124..t=0 0..m=audio 16426 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 147.202.001.001:5060 -> 58.28.001.001:5060SIP/2.0 200 OK..To: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..From: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.1.102;rport=5060;received=58.28.001.001;branch=z9hG4bKca52e47AD4AD36 6..Record-Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..Contact: <sip:44556644@58.28.001.001:5065>..Server: Linksys/SPA962-5.1.18(SC)..Content-Length: 209..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces..Content-Type: application/sdp....v=0..o=- 317989 317989 IN IP4 192.168.1.124..s=-..c=IN IP4 147.202.001.001..t=0 0..m=audio 35984 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 69..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 58.28.001.001:5060 -> 147.202.001.001:5060SIP/2.0 200 OK..Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bKac76.c88550f.0..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@192.168.1.102>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... U 58.28.001.001:5065 -> 147.202.001.001:5060BYE sip:44556648@58.28.001.001:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.124:5065;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..CSeq: 101 BYE..Max-Forwards: 70..Route: <sip:147.202.001.001:5060;nat=yes;ftag=3D7BF99A-F3B3ACE5;lr=on>..User-Agent: Linksys/SPA962-5.1.18(SC)..Content-Length: 0.... U 147.202.001.001:5060 -> 58.28.001.001:5065SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.124:5065;received=58.28.001.001;branch=z9hG4bK-6829d146..From: <sip:44556644@domain.com;user=phone>;tag=9f9da889431cd5afi5..To: "CSB" <sip:44556648@domain.com>;tag=3D7BF99A-F3B3ACE5..CSeq: 101 BYE..Call-ID: 26e8c161-7e674928-90cba56b@192.168.1.102..Contact: <sip:44556648@58.28.001.001:5060>..Record-Route: <sip:147.202.001.001;lr=on;ftag=9f9da889431cd5afi5>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037..Content-Length: 0.... ************************** iptables config $IPTABLES -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A OUTPUT -m state --state ESTABLISHED,RELATED -j ACCEPT $IPTABLES -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT # # Rule 0 (tun0,tun1,tun2) # echo "Rule 0 (tun0,tun1,tun2)" # # # $IPTABLES -N In_RULE_0 $IPTABLES -A INPUT -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A INPUT -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun0 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun1 -m state --state NEW -j In_RULE_0 $IPTABLES -A FORWARD -i tun2 -m state --state NEW -j In_RULE_0 $IPTABLES -A In_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A In_RULE_0 -j ACCEPT $IPTABLES -N Out_RULE_0 $IPTABLES -A OUTPUT -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A OUTPUT -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun0 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun1 -m state --state NEW -j Out_RULE_0 $IPTABLES -A FORWARD -o tun2 -m state --state NEW -j Out_RULE_0 $IPTABLES -A Out_RULE_0 -j LOG --log-level warning --log-prefix "RULE 0 -- ACCEPT " $IPTABLES -A Out_RULE_0 -j ACCEPT $IPTABLES -A Out_RULE_0 -j ACCEPT # # Rule 1 (lo) # echo "Rule 1 (lo)" # # # $IPTABLES -N In_RULE_1 $IPTABLES -A INPUT -i lo -p icmp -m icmp --icmp-type any -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j In_RULE_1 #$IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j In_RULE_1 $IPTABLES -A INPUT -i lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_1 $IPTABLES -A In_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A In_RULE_1 -j ACCEPT $IPTABLES -N Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p icmp -m icmp --icmp-type any -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p tcp -m tcp -m multiport --dports 25060,8008,25,443,80,22,3306,5060 -m state --state NEW -j Out_RULE_1 #$IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 5060,1813 -m state --state NEW -j Out_RULE_1 $IPTABLES -A OUTPUT -o lo -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_1 $IPTABLES -A Out_RULE_1 -j LOG --log-level warning --log-prefix "RULE 1 -- ACCEPT " $IPTABLES -A Out_RULE_1 -j ACCEPT # # Rule 2 (eth0) # echo "Rule 2 (eth0)" # # # $IPTABLES -N Out_RULE_2 $IPTABLES -A OUTPUT -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A FORWARD -o eth0 -d 10.8.1.1 -m state --state NEW -j Out_RULE_2 $IPTABLES -A Out_RULE_2 -j LOG --log-level warning --log-prefix "RULE 2 -- ACCEPT " $IPTABLES -A Out_RULE_2 -j ACCEPT # # Rule 3 (eth0) # echo "Rule 3 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E561900.0 $IPTABLES -A INPUT -i eth0 -d 147.202.001.001 -m state --state NEW -j Cid485B0E561900.0 $IPTABLES -N Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E561900.1 $IPTABLES -A Cid485B0E561900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E561900.1 $IPTABLES -N In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 203.89.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 58.28.001.001 -j In_RULE_3 $IPTABLES -A Cid485B0E561900.1 -s 64.38.001.001 -j In_RULE_3 $IPTABLES -A In_RULE_3 -j LOG --log-level warning --log-prefix "RULE 3 -- ACCEPT " $IPTABLES -A In_RULE_3 -j ACCEPT # echo "Rule 4 (eth0)" # # ping test for nagios # $IPTABLES -N Cid485B0E6A1900.0 $IPTABLES -A OUTPUT -o eth0 -s 147.202.001.001 -m state --state NEW -j Cid485B0E6A1900.0 $IPTABLES -N Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p icmp -m icmp --icmp-type any -j Cid485B0E6A1900.1 $IPTABLES -A Cid485B0E6A1900.0 -p tcp -m tcp -m multiport --dports 25,3306 -j Cid485B0E6A1900.1 $IPTABLES -N Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 203.89.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 58.28.001.001 -j Out_RULE_4 $IPTABLES -A Cid485B0E6A1900.1 -d 64.38.001.001 -j Out_RULE_4 $IPTABLES -A Out_RULE_4 -j LOG --log-level warning --log-prefix "RULE 4 -- ACCEPT " $IPTABLES -A Out_RULE_4 -j ACCEPT # # Rule 5 (eth0) # echo "Rule 5 (eth0)" # # # $IPTABLES -N In_RULE_5 $IPTABLES -A INPUT -i eth0 -p tcp -m tcp -m multiport -d 147.202.001.001 --dports 5060,22,443,80,53,25060,8008 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 10000:20000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -d 147.202.001.001 --dport 35000:36000 -m state --state NEW -j In_RULE_5 #$IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport -d 147.202.001.001 --dports 1194,5065,5060,53,10000:20000,35000:36000 -m state --state NEW -j In_RULE_5 $IPTABLES -A INPUT -i eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j In_RULE_5 $IPTABLES -A In_RULE_5 -j LOG --log-level warning --log-prefix "RULE 5 -- ACCEPT " $IPTABLES -A In_RULE_5 -j ACCEPT # # Rule 6 (eth0) # echo "Rule 6 (eth0)" # # # $IPTABLES -N Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p tcp -m tcp -m multiport -s 147.202.001.001 --dports 22,53,80,443,5060,8008,25060,25 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 10000:20000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -s 147.202.001.001 --dport 35000:36000 -m state --state NEW -j Out_RULE_6 #$IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport -s 147.202.001.001 --dports 53,1194,5060,5065,1813,123,10000:20000,35000:36000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A OUTPUT -o eth0 -p udp -m udp -m multiport --dports 1:55000 -m state --state NEW -j Out_RULE_6 $IPTABLES -A Out_RULE_6 -j LOG --log-level warning --log-prefix "RULE 6 -- ACCEPT " $IPTABLES -A Out_RULE_6 -j ACCEPT # # Rule 7 (global) # echo "Rule 7 (global)" # # # $IPTABLES -N RULE_7 $IPTABLES -A OUTPUT -j RULE_7 $IPTABLES -A INPUT -j RULE_7 $IPTABLES -A FORWARD -j RULE_7 $IPTABLES -A RULE_7 -j LOG --log-level warning --log-prefix "RULE 7 -- DENY " $IPTABLES -A RULE_7 -j DROP # #

2 1

[Kamailio-Users] wildcard in avp_db_load with db_schema
by Andreas Granig 31 Jul '08

31 Jul '08

Hi, For loading all AVPs from "normal" avp tables, I can use something like this: avp_db_load("$fu/domain", "*/domain_preferences"); But when I want to use a db schema to load AVPs belonging for example to a call_id for some sort of dialog tracking, it fails to start with the following error message: "ERROR:avpops:parse_avp_db: inconsistent usage of DB scheme without complet specification of AVP name" My config looks like that: modparam("avpops","db_scheme", "my_schema:table=some_table;uuid_col=call_id;value_col=value") ... avp_db_load("$ci", "*/$my_schema"); If I put something like "$avp(i:10)" instead of the "*", it works, but then I just get this specific AVP instead of all of them. Any idea if this is supposed to work, and if so, how? Thanks, Andreas

2 2

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