<div dir="ltr">Hello. I try to provide call scheme:<br><br>internal client -> asterisk -> Kamailio -> provider -> external endpoint call<br><br>when I make call I see this:<br><br>asterisk kamailio provider<br>
invite --> invite --> <br> <-- 407<br> ACK --> <br> invite w/Auth --><br> <-- 100 <-- 100<br>
<-- 180 <-- 180<div> <-- 183 <-- 183</div><div> <-- 200 <-- 200</div><div> ACK --> ACK --></div><div><br>My problem with last ACK, that I send to provider. Provider ignores it, and sends me some OK packets. As resultI can notend session ( answer to BYE 481 - transaction does not exists). I think it is wrong ACK but can not undrtand where I do mistake.<br>
<br>Please help me to find it: <br><br>My invite (with Auth creditans):<br><br><div>IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 1606</div><div>E...]. .@..R</div><div>...6........N0TINVITE sip:12345678900@my.provider.ip:5060 SIP/2.0</div>
<div>Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on></div><div>Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1</div><div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600</div>
<div>Max-Forwards: 70</div><div>From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div><div>To: <sip:12345678900@my.provider.ip:5068></div><div>Contact:<provider_username@my.external.ip:5068></div>
<div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 12.5.0</div><div>Date: Wed, 27 Aug 2014 22:02:58 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Content-Type: application/sdp</div><div>Content-Length: 544</div><div>Proxy-Authorization: Digest username="provider_username", realm="my.provider.ip", nonce="U/5Wv1P+VZNjFBLf6fwPizgd6iLto5St", uri="sip:12345678900@my.provider.ip:5060", qop=auth, nc=00000001, cnonce="2888860875", response="9f23110471fe9ff751cd55466e70ded2", algorithm=MD5</div>
<div><br></div><div>v=0</div><div>o=root 1370647246 1370647246 IN IP4 12.34.56.78</div><div>s=Asterisk PBX 12.5.0</div><div>c=IN IP4 12.34.56.78</div><div>t=0 0</div><div>a=ice-lite</div><div>m=audio 30296 RTP/AVP 0 3 8 101</div>
<div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=maxptime:150</div><div>a=sendrecv</div>
<div>a=rtcp:30297</div><div>a=ice-ufrag:p5k92ynl</div><div>a=ice-pwd:FIOYKt96NlBfEqKsQipUuadUev1g</div><div>a=candidate:vV3V06Tv</div><br><br><br>Provider trying<br><br><br><div>IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 500</div>
<div>E.........PX6...</div><div>..........ySIP/2.0 100 trying -- your call is important to us</div><div>Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1;rport=5068;received=12.34.56.78</div>
<div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600</div><div>From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div><div>To: <sip:12345678900@my.provider.ip:5068></div>
<div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div><div>CSeq: 102 INVITE</div><div>Server: kamailio (4.1.2 (x86_64/linux))</div><div>Content-Length: 0<br>
<br><br><br><br>provider ringing<br><br><br><br><br><div>IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1098</div><div>E..f......M.6...</div><div>........RV.SIP/2.0 180 Ringing</div><div>Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1</div>
<div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600</div><div>Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1></div><div>Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on></div>
<div>From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div><div>To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH</div><div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div>
<div>CSeq: 102 INVITE</div><div>Contact: <sip:12345678900@67.192.253.160:5060;transport=udp></div><div>User-Agent: Plivo</div><div>Accept: application/sdp</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div>
<div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div><div>
Content-Length: 0</div><div>Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip>;party=calling;privacy=off;screen=no</div></div><br><br><br>provider seesion in progress<br><br><br><br><div>IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1887</div>
<div>E..... ...,.6...</div><div>........g.DSIP/2.0 183 Session Progress</div><div>Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1</div><div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600</div>
<div>Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1></div><div>Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on></div><div>From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div>
<div>To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH</div><div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div><div>CSeq: 102 INVITE</div>
<div>Contact: <sip:12345678900@67.192.253.160:5060;transport=udp></div><div>User-Agent: Plivo</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div>Supported: timer, precondition, path, replaces</div>
<div>Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div>
<div>Content-Length: 742</div><div>Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip>;party=calling;privacy=off;screen=no</div><div><br></div><div>v=0</div><div>o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160</div>
<div>s=FreeSWITCH</div><div>c=IN IP4 67.192.253.160</div><div>t=0 0</div><div>a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD</div><div>m=audio 27180 RTP/AVP 0 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=ssrc:326362635 cnam<br><br><br><br></div></div><div><br></div><div>provider OK<br><br><br><br><div>IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 2026</div>
<div>E..... ...,.6...</div><div>...........SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1</div><div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600</div>
<div>Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1></div><div>Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on></div><div>Fл2rom: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div>
<div>To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH</div><div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div><div>CSeq: 102 INVITE</div>
<div>Contact: <sip:12345678900@67.192.253.160:5060;transport=udp></div><div>User-Agent: Plivo</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div>SupлЛ</div>
<div>o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160</div><div>s=FreeSWITCH</div><div>c=л2IN IP4 67.192.253.160</div><div>t=0 0</div><div>a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD</div><div>m=audio 27180 RTP/AVP 0<br>
<br><br><br><br>my ACK<br><br><br><br><br><div>IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 614</div><div>E...]...@...</div><div>...6........n.hACK sip:12345678900@my.provider.ip:5060;transport=udp SIP/2.0</div>
<div>Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.4250e4d315c4aa6697b6d7f70e861b62.0</div><div>Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK4d28fc11;rport=50600</div><div>Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1></div>
<div>Max-Forwards: 70</div><div>From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50</div><div>To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH</div><div>Contact:<provider_username@my.external.ip:5068></div>
<div>Call-ID: <a href="http://2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600">2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600</a></div><div>CSeq: 102 ACK</div><div>User-Agent: Asterisk PBX 12.5.0</div><div>Content-Length: 0<br>
<br><br><br>So after this ACK provider still sends me 200 OK and my server still sends ACK....<br><br>tags and call-id always one.<br><br><br>Thanks </div></div></div></div>