[SR-Users] ACK not sending to the invite to header but to the contact.
anfecora
anfecora at gmail.com
Tue Oct 29 17:05:21 CET 2013
Thank you , yes it is behind a private IP, and I am using advertise option,
i was able to solve my issue using
modparam("rr", "enable_double_rr", 0)
and then resplying ack o a loose_route() now it seems to work fine.
thanks
Andres Collazos,
On Tue, Oct 29, 2013 at 5:34 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:
> Hello,
>
> the config file is not complete, you don't pass the parameters. The ngrep
> trace that kamailio is forwarding the invite from the private IP but the
> route header is having the public IP.
>
> You have to set debug=3 in your config and send the log messages here to
> see what is executed.
>
> As a guess hit - if kamailio is listening on a private ip, being behind a
> port forwarding nat firewall, you may consider:
>
> listen=udp:privateip:5060 advertise publicip:5060
>
> See Core Cookbook from the kamailio.org wiki for more details about the
> above parameter.
>
> Cheers,
> Daniel
>
> On 10/24/13 12:01 AM, anfecora wrote:
>
> Hi all, can anyone help me to find out what is wrong with my setup, i have
> an asterisk behind a kamailio, kamailio is proxying all packages to the
> outside.
>
> when the call is bridge it gets disconnected after a few seconds, it
> seems that our voip carrier is sending a bye because we didn't answer to
> their 200 ok propperly, but as the trace shows we did only that kamailio is
> answering to the contact header ip not the ip that is sending the ok.
>
> any help is apreciated .
>
> thanks.
>
> my setup
>
> 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
>
> };
>
>
> if (method=="CANCEL") {
> if (t_check_trans()) t_relay();
>
> exit;
> };
>
>
>
> if (loose_route()) {
>
>
> t_relay();
> exit;
> }
>
>
> if (is_method("INVITE")) {
>
>
> record_route();
>
> }
> f (!t_relay_to_udp("3.1.1.1", "5060")) {
> sl_reply_error();
> exit;
> };
> exit
> };
>
> here is a trace to a call made to a hotel.
> i had changed the real ips for obvious reasons.
> thanks.
>
>
> asterisk ip 1.1.1.1
> kamailio internal 1.1.1.2
> kamailio external 2.0.0.1
> Voip Carrier 3.1.1.1
> voip contact ip 3.1.1.2
>
>
>
> U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060
> INVITE sip:23276341079 at 2.0.0.1 SIP/2.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport.
> Max-Forwards: 70.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Date: Wed, 23 Oct 2013 21:26:46 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH.
> Supported: replaces, timer.
> Privacy: off.
> P-Asserted-Identity: sip:+19812457865 at 1.1.1.1.
> Cisco-Guid: 25655507-3591552378-379709
> Content-Type: application/sdp.
> Content-Length: 333.
> .
> v=0.
> o=root 519803789 519803789 IN IP4 1.1.1.1.
> s=Asterisk PBX 1.8.15-cert2.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 49926 RTP/AVP 0 18 3 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Server: kamailio (4.0.4 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060
> INVITE sip:76890723276341079 at 3.1.1.1:5060 SIP/2.0.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Max-Forwards: 16.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.1>.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Date: Wed, 23 Oct 2013 21:26:46 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH.
> Supported: replaces, timer.
> Privacy: off.
> P-Asserted-Identity: sip:+19812457865 at 1.1.1.1.
> Cisco-Guid: 25655507-3591552378-379709
> Content-Type: application/sdp.
> Content-Length: 333.
> .
> v=0.
> o=root 519803789 519803789 IN IP4 1.1.1.1.
> s=Asterisk PBX 1.8.15-cert2.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 49926 RTP/AVP 0 18 3 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.1>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Server: gProxy (1.8.3 (i386/Linux)).
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP]
> .....insert into acc
> (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode
> ) values ('INVITE','as4bc322e9','3591552407-393967','
> 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060','200','OK','2013-10-23
> 17:26:16','sip:+19812457865 at 1.1.1.1','sip:23276341079 at 2.0.0.1
> ','+19812457865','1.1.1.1','sip:76890723276341079 at 3.1.1.1:5060','
> sip:23276341079 at 2.0.0.1','OUT')
>
> U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 70.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 16.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.2>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 70.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 16.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.2>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 70.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 16.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.2>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060
> SIP/2.0 200 OK.
> Session-Expires: 3600;refresher=uas.
> Require: timer.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060.
> Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 INVITE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Call-Info: <sip:3.1.1.2>
> ;method="NOTIFY;Event=telephone-event;Duration=1000".
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Length: 202.
> .
> v=0.
> o=MSXB 4755 8544 IN IP4 3.1.1.2.
> s=sip call.
> c=IN IP4 204.15.40.111.
> t=0 0.
> m=audio 33408 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
> U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 70.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:23276341079 at 2.0.0.1>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060
> ACK sip:76890723276341079 at 3.1.1.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,
> <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
> Max-Forwards: 16.
> From: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> To: <sip:76890723276341079 at 3.1.1.2>;tag=3591552407-393967.
> Contact: <sip:+19812457865 at 1.1.1.1:5060>.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060
> BYE sip:+19812457865 at 1.1.1.1:5060 SIP/2.0.
> Max-Forwards: 69.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: "+19812457865" <sip:+19812457865 at 1.1.1.1>;tag=as4bc322e9.
> From: <sip:76890723276341079 at 3.1.1.1>;tag=3591552407-393967.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 2 BYE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
> Via: SIP/2.0/UDP 3.1.1.2:5060
> ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Content-Length: 0.
> .
>
>
> U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060
> BYE sip:+19812457865 at 1.1.1.1:5060 SIP/2.0.
> Max-Forwards: 16.
> Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
> To: "+19812457865" <sip:176822213 at 1.1.1.1>;tag=as4bc322e9.
> From: 0 <sip:079 at 3.1.1.1>;tag=3591552407-393967.
> Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0 at 1.1.1.1:5060.
> CSeq: 2 BYE.
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE.
> Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0.
> Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
> Via: SIP/2.0/UDP 3.1.1.2:5060
> ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
> Contact: <sip:76890723276341079 at 3.1.1.2:5060>.
> Content-Length: 0.
> .
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Trainings - Berlin, Nov 25-28
> - more details about Kamailio trainings at http://www.asipto.com -
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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